diff --git a/marked/Rel-11/26_series/26071/997233d405f0d4b89ddeb7683e047f66_img.jpg b/marked/Rel-11/26_series/26071/997233d405f0d4b89ddeb7683e047f66_img.jpg new file mode 100644 index 0000000000000000000000000000000000000000..3f4c87d79f455a241616781180bdff2f8b18e7fc --- /dev/null +++ b/marked/Rel-11/26_series/26071/997233d405f0d4b89ddeb7683e047f66_img.jpg @@ -0,0 +1,3 @@ +version https://git-lfs.github.com/spec/v1 +oid sha256:25d84eb1a1f4077bcf045b46d821c8168e3384285a5e63b95ee7d192a3af0f04 +size 151166 diff --git a/marked/Rel-11/26_series/26071/raw.md b/marked/Rel-11/26_series/26071/raw.md new file mode 100644 index 0000000000000000000000000000000000000000..efd6f17066fcafcda2544480d5b089fd23e7f814 --- /dev/null +++ b/marked/Rel-11/26_series/26071/raw.md @@ -0,0 +1,267 @@ + + + + + + +# --- Contents + +| | | +|----------------------------------------------------------------------------|----| +| Foreword ..... | 4 | +| 1 Scope..... | 5 | +| 2 References..... | 5 | +| 3 Definitions and abbreviations ..... | 6 | +| 3.1 Abbreviations ..... | 6 | +| 4 General ..... | 6 | +| 5 Adaptive Multi-Rate speech codec transcoding functions..... | 8 | +| 6 Adaptive Multi-Rate speech codec ANSI C-code ..... | 8 | +| 7 Adaptive Multi-Rate speech codec test vectors ..... | 8 | +| 8 Adaptive Multi-Rate speech codec source controlled rate operation ..... | 9 | +| 9 Adaptive Multi-Rate speech codec voice activity detection..... | 9 | +| 10 Adaptive Multi-Rate speech codec comfort noise insertion ..... | 10 | +| 11 Adaptive Multi-Rate speech codec error concealment of lost frames ..... | 10 | +| 12 Adaptive Multi-Rate speech codec frame structure ..... | 10 | +| 13 Adaptive Multi-Rate speech codec interface to RAN..... | 10 | +| 14 Adaptive Multi-Rate speech codec performance characterisation..... | 11 | +| Annex A (informative): Change history..... | 12 | + +# --- Foreword + +This Technical Specification has been produced by the 3rd Generation Partnership Project (3GPP). + +The contents of the present document are subject to continuing work within the TSG and may change following formal TSG approval. Should the TSG modify the contents of the present document, it will be re-released by the TSG with an identifying change of release date and an increase in version number as follows: + +Version x.y.z + +where: + +- x the first digit: + - 1 presented to TSG for information; + - 2 presented to TSG for approval; + - 3 or greater indicates TSG approved document under change control. +- y the second digit is incremented for all changes of substance, i.e. technical enhancements, corrections, updates, etc. +- z the third digit is incremented when editorial only changes have been incorporated in the document. + +# --- 1 Scope + +The present document is an introduction to the speech processing parts of the narrowband telephony speech service employing the Adaptive Multi-Rate (AMR) speech coder. A general overview of the speech processing functions is given, with reference to the documents where each function is specified in detail. + +# --- 2 References + +The following documents contain provisions which, through reference in this text, constitute provisions of the present document. + +- References are either specific (identified by date of publication, edition number, version number, etc.) or non-specific. +- For a specific reference, subsequent revisions do not apply. +- For a non-specific reference, the latest version applies. In the case of a reference to a 3GPP document (including a GSM document), a non-specific reference implicitly refers to the latest version of that document *in the same Release as the present document*. + +- [1] GSM 03.50: "Digital cellular telecommunications system (Phase 2); Transmission planning aspects of the speech service in the GSM Public Land Mobile Network (PLMN) system". +- [2] 3GPP TS 26.090: "Transcoding functions". +- [3] 3GPP TS 26.073: "Adaptive Multi-Rate (AMR); ANSI C source code". +- [4] 3GPP TS 26.074: "Adaptive Multi-Rate (AMR); Test sequences". +- [5] 3GPP TS 26.093: "Source Controlled Rate operation". +- [6] 3GPP TS 26.094: "AMR Speech Codec; Voice Activity Detector". +- [7] 3GPP TS 26.092: "Mandatory Speech Codec speech processing functions; AMR Speech Codec; Comfort noise aspects". +- [8] 3GPP TS 26.091: "Mandatory Speech Codec speech processing functions; AMR Speech Codec; Error concealment of lost frames". +- [9] 3GPP TS 26.101: "Frame Structure". +- [10] 3GPP TS 26.102: "AMR Speech Codec; Interface to Iu and Uu". +- [11] 3GPP TS 26.901: "AMR wideband speech codec feasibility study report". +- [12] ITU-T Recommendation G.711: "Pulse code modulation (PCM) of voice frequencies". +- [13] ITU-T Recommendation H.324: "Terminal for low bit-rate multimedia communication". + +# --- 3 Definitions and abbreviations + +## 3.1 Abbreviations + +For the purposes of this TS, the following abbreviations apply: + +| | | +|-------|-------------------------------------------------------------------------------------------------| +| ACELP | Algebraic Code Excited Linear Prediction | +| AMR | Adaptive Multi-Rate | +| BFI | Bad Frame Indication | +| CHD | Channel Decoder | +| CHE | Channel Encoder | +| GSM | Global System for Mobile communications | +| ITU-T | International Telecommunication Union – Telecommunication standardisation sector (former CCITT) | +| PCM | Pulse Code Modulation | +| PLMN | Public Land Mobile Network | +| PSTN | Public Switched Telephone Network | +| RX | Receive | +| SCR | Source Controlled Rate | +| SPD | SPeech Decoder | +| SPE | SPeech Encoder | +| TC | Transcoder | +| TX | Transmit | +| UE | User Equipment (terminal) | + +# --- 4 General + +The AMR speech coder consists of the multi-rate speech coder, a source controlled rate scheme including a voice activity detector and a comfort noise generation system, and an error concealment mechanism to combat the effects of transmission errors and lost packets. + +The multi-rate speech coder is a single integrated speech codec with eight source rates from 4.75 kbit/s to 12.2 kbit/s, and a low rate background noise encoding mode. The speech coder is capable of switching its bit-rate every 20 ms speech frame upon command. + +A reference configuration where the various speech processing functions are identified is given in Figure 1. In this figure, the relevant specifications for each function are also indicated. + +In Figure 1, the audio parts including analogue to digital and digital to analogue conversion are included, to show the complete speech path between the audio input/output in the User Equipment (UE) and the digital interface of the network. The detailed specification of the audio parts is not within the scope of the present document. These aspects are only considered to the extent that the performance of the audio parts affect the performance of the speech transcoder. + +![Figure 1: Overview of audio processing functions. The diagram is split into TRANSMIT SIDE and RECEIVE SIDE. The TRANSMIT SIDE shows the flow from BSS side only (GSM 06.60.AMR) and MS side only (GSM 03.50) through various processing blocks like LPF, A/D, 8bit/A-law to 13-bit uniform, and into the Speech Encoder (GSM 06.60.AMR). It also includes Voice Activity Detector (VAD) (GSM 06.82.AMR), Comfort Noise TX Functions (GSM 06.62.AMR), and DTX Control and Operation (GSM 06.81.AMR) blocks, resulting in outputs for SP flag and Info. bits. The RECEIVE SIDE shows the reverse process, starting from Info. bits, BFI, SID, and TAF inputs into the DTX Control and Operation block, followed by Speech frame substitution, Speech Decoder (GSM 06.60.AMR), and Comfort Noise RX Functions (GSM 06.62.AMR), leading to the final output through D/A and LPF blocks (GSM 03.50) to the BSS side only (GSM 06.60.AMR).](997233d405f0d4b89ddeb7683e047f66_img.jpg) + +Figure 1: Overview of audio processing functions. The diagram is split into TRANSMIT SIDE and RECEIVE SIDE. The TRANSMIT SIDE shows the flow from BSS side only (GSM 06.60.AMR) and MS side only (GSM 03.50) through various processing blocks like LPF, A/D, 8bit/A-law to 13-bit uniform, and into the Speech Encoder (GSM 06.60.AMR). It also includes Voice Activity Detector (VAD) (GSM 06.82.AMR), Comfort Noise TX Functions (GSM 06.62.AMR), and DTX Control and Operation (GSM 06.81.AMR) blocks, resulting in outputs for SP flag and Info. bits. The RECEIVE SIDE shows the reverse process, starting from Info. bits, BFI, SID, and TAF inputs into the DTX Control and Operation block, followed by Speech frame substitution, Speech Decoder (GSM 06.60.AMR), and Comfort Noise RX Functions (GSM 06.62.AMR), leading to the final output through D/A and LPF blocks (GSM 03.50) to the BSS side only (GSM 06.60.AMR). + +**Figure 1: Overview of audio processing functions** + +- 1) 8-bit A-law or $\mu$ -law PCM (ITU-T Recommendation G.711 [12] ), 8 000 samples/s; +- 2) 13-bit uniform PCM, 8 000 samples/s; +- 3) Voice Activity Detector (VAD) flag; +- 4) Encoded speech frame, 50 frames/s, number of bits/frame depending on the AMR codec mode; +- 5) Silence Descriptor (SID) frame; +- 6) TX\_TYPE, 2 bits, indicates whether information bits are available and if they are speech or SID information; +- 7) Information bits delivered to the 3G AN; +- 8) Information bits received from the 3G AN; +- 9) RX\_TYPE, the type of frame received quantized into three bits. + +# 5 Adaptive Multi-Rate speech codec transcoding functions + +The adaptive multi-rate speech codec is described in [2]. The technical content is identical to that of 3GPP TS 26.090. + +As shown in Figure 1, the speech encoder takes its input as a 13-bit uniform Pulse Code Modulated (PCM) signal either from the audio part of the UE or on the network side, from the Public Switched Telephone Network (PSTN) via an 8-bit A-law or $\mu$ -law to 13-bit uniform PCM conversion. The encoded speech at the output of the speech encoder is packetized and delivered to the network interface. In the receive direction, the inverse operations take place. + +The detailed mapping between input blocks of 160 speech samples in 13-bit uniform PCM format to encoded blocks (in which the number of bits depends on the presently used codec mode) and from these to output blocks of 160 reconstructed speech samples is described in [2]. The coding scheme is Multi-Rate Algebraic Code Excited Linear Prediction. The bit-rates of the source codec are listed in Table 1. + +An AMR speech codec capable UE shall support all source rates listed in Table 1. + +**Table 1: Source codec bit-rates for the AMR codec.** + +| Codec mode | Source codec bit-rate | +|------------|--------------------------| +| AMR_12.20 | 12,20 kbit/s (GSM EFR) | +| AMR_10.20 | 10,20 kbit/s | +| AMR_7.95 | 7,95 kbit/s | +| AMR_7.40 | 7,40 kbit/s (IS-641) | +| AMR_6.70 | 6,70 kbit/s (PDC-EFR) | +| AMR_5.90 | 5,90 kbit/s | +| AMR_5.15 | 5,15 kbit/s | +| AMR_4.75 | 4,75 kbit/s | +| AMR_SID | 1,80 kbit/s (see note 1) | + +NOTE 1: Assuming SID frames are continuously transmitted + +NOTE 2: GSM-EFR is the 3GPP TS 26.090 Enhanced Full Rate Speech Codec (also identical to the TTA TDMA-US1 Enhanced speech codec) + +NOTE 3: IS-641 is the TTA/EIA IS-641 TDMA Enhanced Full Rate Speech Codec + +NOTE 4: PDC-EFR is the ARIB 6.7 kbit/s Enhanced Full Rate Speech Codec + +# 6 Adaptive Multi-Rate speech codec ANSI C-code + +The ANSI-C code of the speech codec, VAD and CNG system are described in [3]. The ANSI C-code is mandatory. The ANSI C-code is identical to that of 3GPP TS 26.073 [3]. + +# 7 Adaptive Multi-Rate speech codec test vectors + +A set of digital test sequences is specified in [4], thus enabling the verification of compliance, i.e. bit-exactness, to a high degree of confidence. The test vectors are identical to those of 3GPP TS 26.074 [4]. + +The test sequences are defined separately for: + +- The speech codec described in [2], +- The VAD described in [6], +- The CN generation described in [7]. + +The adaptive multi-rate speech transcoder, VAD, SCR system and comfort noise parts of the audio processing functions (see Figure 1) are defined in bit exact arithmetic. Consequently, they shall react on a given input sequence always with + +the corresponding bit exact output sequence, provided that the internal state variables are also always exactly in the same state at the beginning of the test. + +The input test sequences provided shall force the corresponding output test sequences, provided that the tested modules are in their home-state when starting. + +The modules may be set into their home states by provoking the appropriate homing-functions. + +NOTE: This is normally done during reset (initialisation of the codec). + +Special inband signalling frames (encoder-homing-frame and decoder-homing-frame) described in [2] have been defined to provoke these homing-functions also in remotely placed modules. + +At the end of the first received homing frame, the audio functions that are defined in a bit exact way shall go into their predefined home states. The output corresponding to the first homing frame is dependent on the codec state when the frame was received. Any consecutive homing frames shall produce corresponding homing frames at the output. + +# --- 8 Adaptive Multi-Rate speech codec source controlled rate operation + +The source controlled rate operation of the adaptive multi-rate speech codec is defined in [5]. + +During a normal telephone conversation, the participants alternate so that, on the average, each direction of transmission is occupied about 50 % of the time. Source controlled rate (SCR) is a mode of operation where the speech encoder encodes speech frames containing only background noise with a lower bit-rate than normally used for encoding speech. A network may adapt its transmission scheme to take advantage of the varying bit-rate. This may be done for the following two purposes: + +- 1) In the UE, battery life will be prolonged or a smaller battery could be used for a given operational duration. +- 2) The average required bit-rate is reduced, leading to a more efficient transmission with decreased load and hence increased capacity. + +The following functions are required for the source controlled rate operation: + +- a Voice Activity Detector (VAD) on the TX side; +- evaluation of the background acoustic noise on the TX side, in order to transmit characteristic parameters to the RX side; +- generation of comfort noise on the RX side during periods when no normal speech frames are received. + +The transmission of comfort noise information to the RX side is achieved by means of a Silence Descriptor (SID) frame, which is sent at regular intervals. + +# --- 9 Adaptive Multi-Rate speech codec voice activity detection + +The adaptive multi-rate VAD function is described in [6]. + +The input to the VAD is the input speech itself together with a set of parameters computed by the adaptive multi-rate speech encoder. The VAD uses this information to decide whether each 20 ms speech coder frame contains speech or not. + +The VAD algorithm is described in [6], and the corresponding C code is defined in [3]. The verification of compliance to [6], is achieved by use of digital test sequences applied to the same interface as the test sequences for the speech codec. + +# --- 10 Adaptive Multi-Rate speech codec comfort noise insertion + +The adaptive multi-rate comfort noise insertion function is described in [7]. + +When speech is absent, the synthesis in the speech decoder is different from the case when normal speech frames are received. The synthesis of an artificial noise based on the received non-speech parameters is termed comfort noise generation. + +The comfort noise generation process is as follows: + +- the evaluation of the acoustic background noise in the transmitter; +- the noise parameter encoding (SID frames) and decoding, and +- the generation of comfort noise in the receiver. + +The comfort noise processes and the algorithm for updating the noise parameters during speech pauses are defined in detail in [7], and the corresponding C code is defined in [3]. The comfort noise mechanism is based on the adaptive multi-rate speech codec defined in [2]. + +# --- 11 Adaptive Multi-Rate speech codec error concealment of lost frames + +The adaptive multi-rate speech codec error concealment of lost frames is described in [8]. + +Frames may be lost due to transmission errors or frame stealing in a wireless environment. Actions which shall be taken in these cases, both for lost speech frames and for lost SID frames are described in [8]. Error concealment actions shall be used also in the case of lost speech packets in the transport network. The methods described in [8] may with some modifications be used as a basis for such actions. + +In order to mask the effect of isolated lost frames, the speech decoder shall be informed and the error concealment actions shall be initiated, whereby a set of predicted parameters are used in the speech synthesis. Insertion of speech signal independent silence frames is not allowed. For several subsequent lost frames, a muting technique shall be used to indicate to the listener that transmission has been interrupted. + +# --- 12 Adaptive Multi-Rate speech codec frame structure + +The adaptive multi-rate speech frame structure is described in [9]. The output interface format from the encoder and input interface format to the decoder is divided into two parts; the core speech data part, which is the speech coded bits, and the other part is an additional data part with mode information. + +The interface format described in [9] is termed AMR interface format 1 (AMR IF1). + +Annex A of [9] describes an octet aligned frame format which shall be used in applications requiring octet alignment, such as for ITU-T Recommendation H.324 [13]. This format is termed AMR interface format 2 (AMR IF2). + +# --- 13 Adaptive Multi-Rate speech codec interface to RAN + +The adaptive multi-rate speech service interface to RAN is described in [10]. + +# --- 14 Adaptive Multi-Rate speech codec performance characterisation + +The adaptive multi-rate speech channel performance characterisation is described in [11]. + +# Annex A (informative): Change history + +| Document history | | | | | | | | | +|-------------------------|-----------------|-------------------------------------------|-----------|------------|--------------------------------------|------------|------------|--| +| V.0.1.0 | March 1999 | First Draft | | | | | | | +| V.0.1.1 | April 1999 | References changed | | | | | | | +| V.1.0.0 | April 22, 1999 | Editorial changes | | | | | | | +| V.2.0.0 | June 15, 1999 | Minor Editorial changes | | | | | | | +| V.3.0.0 | June 22, 1999 | Approved at 3GPP TSG SA#4 Plenary meeting | | | | | | | +| V.3.0.1 | August 22, 1999 | Reformatted 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0000000000000000000000000000000000000000..e495bb4e57c910ff79c72f2130514e287c2970a3 --- /dev/null +++ b/marked/Rel-11/26_series/26077/raw.md @@ -0,0 +1,1748 @@ + + + + + + +# Contents + +| | | +|---------------------------------------------------------------------------------------------------------------------------------------------------------------------------------|-----------| +| Foreword ..... | 6 | +| 1 Scope..... | 7 | +| 2 References..... | 7 | +| 3 Definitions and abbreviations ..... | 8 | +| 3.1 Definitions..... | 8 | +| 3.2 Abbreviations ..... | 8 | +| 4 Description of Noise Suppression applied to AMR..... | 8 | +| 4.1 Applicability of Noise Suppression to Basic Services..... | 8 | +| 5 Requirements to be assessed by Objective Means..... | 8 | +| 5.1 Bit Exactness of the Speech Encoder..... | 8 | +| 5.2 Bit Exactness of the Speech Decoder ..... | 9 | +| 5.3 Impact on Speech Path Delay..... | 9 | +| 5.4 Impact on Channel Activity ..... | 10 | +| 6 Requirements to be assessed by subjective tests..... | 10 | +| 6.1 Impact on Speech Quality ..... | 10 | +| 6.1.1 Initial Convergence Time..... | 10 | +| 6.1.2 No Degradation in Clean Speech ..... | 10 | +| 6.1.3 No degradation of Speech and no Undesirable Effects in Residual Noise in Conditions with Background Noise ( residual noise = background noise after AMR/NS ) ..... | 10 | +| 6.1.4 Quality Impact compared to AMR ..... | 11 | +| 7 Performance Objectives assessed by Objective Measures..... | 11 | +| 7.1 Impact on Active Speech Level ..... | 11 | +| 7.2 Objective Speech Quality Measures ..... | 11 | +| 8 Interaction with supplementary services..... | 12 | +| 8.1 General ..... | 12 | +| 8.2 Explicit Call Transfer (ECT)..... | 12 | +| 8.3 Call wait/Call hold. .... | 12 | +| 8.4 Multiparty..... | 12 | +| 8.5 Service Announcements..... | 12 | +| 9 Interaction with Alternate and Followed by services..... | 12 | +| 10 Interaction with other speech services ..... | 12 | +| 11 Interaction with DTMF and other signalling tones..... | 13 | +| 12 Interaction with Lawful Intercept ..... | 13 | +| 13 Interaction with TFO..... | 13 | +| Annex A (informative): Method for generating Objective Performance Measures ..... | 14 | +| A.1 Notations ..... | 14 | +| A.2 Test material..... | 15 | +| A.3 Objective measures for characterization of NS algorithm effect ..... | 15 | + +| | | | +|-----------------------------|------------------------------------------------------------------------------------------------------------------------------------------------|-----------| +| Annex B (normative): | Methodology for Measuring Subjective SNR Improvement for CCR Experiments ..... | 20 | +| Annex C (normative): | Test Plan for Checking Conformance to Requirements ..... | 21 | +| C.1 | Introduction ..... | 21 | +| C.2 | Document Structure ..... | 21 | +| C.3 | References, Conventions, and Contacts ..... | 22 | +| C.4 | Key Acronyms ..... | 22 | +| C.4.1 | Contact Names ..... | 22 | +| C.5 | Roles and Responsibilities ..... | 23 | +| C.6 | Information relevant to all Experiments ..... | 23 | +| C.6.1 | General Technical Notes ..... | 23 | +| C.6.2 | Codec Adaptation and Error Conditions ..... | 23 | +| C.6.3 | Speech Material ..... | 23 | +| C.6.3.1 | Availability of Pre-recorded Speech Material ..... | 24 | +| C.6.3.2 | Recording Your Own Speech Databases ..... | 24 | +| C.6.3.3 | Format for Single Sentence Speech Samples ..... | 24 | +| C.6.3.4 | Format for Short Speech Samples ..... | 25 | +| C.6.3.5 | Format for Long Speech Samples ..... | 25 | +| C.6.3.6 | Processing of the Speech Files ..... | 25 | +| C.6.4 | Listening Environment ..... | 26 | +| C.6.5 | Experimental Procedure ..... | 26 | +| C.6.6 | Preliminary Conditions ..... | 27 | +| C.6.7 | Reference Conditions ..... | 27 | +| C.6.8 | Noise Material ..... | 27 | +| C.7 | Experiment 1: Degradation in Clean Speech (Pair Comparison Test) ..... | 27 | +| C.7.1 | Introduction ..... | 27 | +| C.7.2 | Test Factors and Conditions ..... | 28 | +| C.7.3 | Preliminary Conditions ..... | 29 | +| C.7.4 | Speech Material ..... | 29 | +| C.7.5 | Experimental Design ..... | 29 | +| C.7.6 | Processing ..... | 29 | +| C.7.7 | Randomizations ..... | 29 | +| C.7.8 | Duration of the PC Experiment ..... | 30 | +| C.7.9 | Votes Per Condition ..... | 30 | +| C.7.10 | Test Procedure ..... | 30 | +| C.7.11 | Opinion Scale ..... | 30 | +| C.7.12 | Statistical Analysis ..... | 30 | +| C.7.13 | Test Conditions for Experiment 1 ..... | 31 | +| C.8 | Experiments 2a, 2b & 2c: No degradation of Speech and no Undesirable Effects in Residual Noise in Conditions with Background Noise (ACR) ..... | 32 | +| C.8.1 | Introduction ..... | 32 | +| C.8.2 | Test Factors and Conditions ..... | 32 | +| C.8.3 | Preliminary Conditions ..... | 33 | +| C.8.4 | Speech Material ..... | 34 | +| C.8.5 | Experimental Design ..... | 34 | +| C.8.6 | Processing ..... | 34 | +| C.8.7 | Randomizations ..... | 34 | +| C.8.8 | Duration of the ACR Experiments 2a, 2b, and 2c ..... | 34 | +| C.8.9 | Votes Per Condition ..... | 35 | +| C.8.10 | Test Procedure ..... | 35 | +| C.8.11 | Opinion Scale ..... | 35 | +| C.8.12 | Test Conditions for Experiments 2a, 2b and 2c ..... | 36 | +| C.8.13 | Statistical Analysis ..... | 36 | +| C.9 | Experiments 3a & 3b: Performances in Background Noise Conditions (Mod-CCR) ..... | 37 | +| C.9.1 | Introduction ..... | 37 | + +| | | | +|------------------------|--------------------------------------------------------------------------------------------------------------|----| +| C.9.2 | Test Factors and Conditions..... | 37 | +| C.9.3 | Preliminary Conditions ..... | 39 | +| C.9.4 | Speech Material..... | 39 | +| C.9.5 | Experimental Design ..... | 39 | +| C.9.6 | Processing..... | 40 | +| C.9.7 | Randomizations..... | 40 | +| C.9.8 | Duration of the CCR Experiments 3a and 3b..... | 40 | +| C.9.9 | Votes Per Condition ..... | 40 | +| C.9.10 | Test Procedure..... | 40 | +| C.9.11 | Opinion Scale ..... | 40 | +| C.9.12 | Test Conditions for Experiments 3a and 3b..... | 41 | +| C.9.13 | Statistical Analysis ..... | 41 | +| C.10 | Experiments 4: Influence of Input Level, Voice Activity Detection and Discontinuous Transmission (CCR) ..... | 42 | +| C.10.1 | Introduction ..... | 42 | +| C.10.2 | Test Factors and Conditions..... | 42 | +| C.10.3 | Preliminary Conditions ..... | 43 | +| C.10.4 | Speech Material..... | 44 | +| C.10.5 | Experimental Design ..... | 44 | +| C.10.6 | Processing..... | 44 | +| C.10.7 | Randomizations..... | 44 | +| C.10.8 | Duration of the Experiment..... | 44 | +| C.10.9 | Votes Per Condition ..... | 44 | +| C.10.10 | Test Procedure..... | 45 | +| C.10.11 | Opinion Scale ..... | 45 | +| C.10.12 | Test Conditions for Experiment 4 ..... | 45 | +| C.10.13 | Statistical Analysis ..... | 46 | +| C.11 | Instructions to subjects and data collection..... | 46 | +| C.11.1 | Example Instructions for Experiment 1 ..... | 47 | +| C.11.2 | Example Modified ACR Instructions for Experiment 2 ..... | 47 | +| C.11.3 | Example Instructions for Experiment 3 and 4..... | 48 | +| C.12 | Processing Tables..... | 48 | +| C.13 | Presentation Orders ..... | 49 | +| Annex D (informative): | Change history..... | 50 | + +# --- Foreword + +This Technical Specification has been produced by the 3rd Generation Partnership Project (3GPP). + +The contents of the present document are subject to continuing work within the TSG and may change following formal TSG approval. Should the TSG modify the contents of the present document, it will be re-released by the TSG with an identifying change of release date and an increase in version number as follows: + +Version x.y.z + +where: + +- x the first digit: + - 1 presented to TSG for information; + - 2 presented to TSG for approval; + - 3 or greater indicates TSG approved document under change control. +- y the second digit is incremented for all changes of substance, i.e. technical enhancements, corrections, updates, etc. +- z the third digit is incremented when editorial only changes have been incorporated in the document. + +# --- 1 Scope + +The present document specifies recommended minimum performance requirements for noise suppression algorithms intended for application in conjunction with the AMR speech encoder. This specification is for guidance purposes. Noise Suppression is intended to enhance the speech signal corrupted by acoustic noise at the input to the AMR speech encoder. + +The use of this recommended minimum performance requirements specification is not mandatory except for those solutions intended to be endorsed by SMG11. + +It is the intention of SMG11 to perform analysis and validation of any AMR noise suppression solution which is voluntarily brought to the attention of SMG11 in the future, using the requirements set out in this specification to facilitate such an analysis. In order for SMG11 to endorse such a solution, SMG11 must confirm that all the recommended minimum performance requirements are met. + +# --- 2 References + +The following documents contain provisions which, through reference in this text, constitute provisions of the present document. + +- References are either specific (identified by date of publication, edition number, version number, etc.) or non-specific. + - For a specific reference, subsequent revisions do not apply. + - For a non-specific reference, the latest version applies. In the case of a reference to a 3GPP document (including a GSM document), a non-specific reference implicitly refers to the latest version of that document *in the same Release as the present document*. +- [1] ITU-T Recommendation I.130 (1988): "General modelling methods - Method for the characterisation of telecommunications services supported by an ISDN and network capabilities of an ISDN". +- [2] 3GPP TR 01.04 (ETR 350): "Abbreviations and acronyms". +- [3] 3GPP TS 06.71: "Adaptive Multi-Rate (AMR); Speech processing functions; General description". +- [4] 3GPP TS 06.77: "Minimum Performance Requirements for Noise Suppressor Application to the AMR Speech Encoder". +- [5] 3GPP TS 06.73: "Adaptive Multi Rate (AMR) speech; ANSI-C code for the AMR speech codec". +- [6] 3GPP TS 03.50: "Transmission planning aspects of the speech service in the GSM Public Land Mobile Network (PLMN) system". +- [7] ITU-T Recommendation P.56: "Objective measurement of active speech level". +- [8] ITU-T Recommendation P.800: "Methods for subjective determination of transmission quality". +- [9] 3GPP TR 06.75: "Performance Characterization of the GSM Adaptive Multi-Rate (AMR) speech codec". +- [10] 3GPP TS 06.90: "Adaptive Multi-Rate (AMR) speech transcoding". +- [11] 3GPP TS 06.91: "Substitution and muting of lost frames for Adaptive Multi Rate (AMR) speech traffic channels". +- [12] 3GPP TS 06.92: "Comfort noise aspects for Adaptive Multi-Rate (AMR) speech traffic channels". +- [13] 3GPP TS 06.94: "Voice Activity Detection (VAD) for Adaptive Multi-Rate (AMR) speech traffic channels; General description". + +# --- 3 Definitions and abbreviations + +3GPP TR 01.04 [2] provides a list of abbreviations and acronyms used in GSM specifications. For the purposes of the present document, the following definitions and abbreviations also apply: + +## 3.1 Definitions + +None + +## 3.2 Abbreviations + +| | | +|--------|------------------------------------------------------------------------| +| AMR | Adaptive Multi-Rate | +| AMR/NS | Combination of the AMR speech codec and the Noise Suppression function | +| NS | Noise Suppression | + +# --- 4 Description of Noise Suppression applied to AMR + +Noise Suppression for the AMR codec is a feature designed to enhance speech quality in a range of environments where there is significant (acoustic) background noise. The noise suppression function is a pre-processing module that is used to improve the signal to noise ratio of a speech signal prior to voice coding. In so doing it may use functions and/or data from the AMR speech encoding function. This specification defines recommended minimum performance requirements for such a function when it is implemented in the mobile station (operating on the uplink speech signal). + +The AMR Speech decoder should not be altered by the Noise Suppression function. + +It shall be possible to disable the operation of the noise suppression algorithm using signalling when commanded by the network. + +## 4.1 Applicability of Noise Suppression to Basic Services. + +This feature shall be applicable (as an option) to all speech calls where the narrowband AMR codec is utilised. Provision of the feature in AMR-capable mobile stations is a manufacturer dependent option. The network shall be able to enable or disable this noise suppression function both at call set-up and in call. Signalling between network and mobile to allow this control has been provided. + +# --- 5 Requirements to be assessed by Objective Means + +## 5.1 Bit Exactness of the Speech Encoder + +The Noise Suppression shall be implemented as a separate pre-processing module prior to the speech encoding. The functionality and all internal states, tables and variables of the speech encoder shall remain unaltered by the Noise Suppression function. + +The Noise Suppression should be implemented as a stand-alone pre-processing module operating on the 160 samples input speech buffer to the speech encoder according to Figure 1. + +![Figure 1: Noise Suppression implementation block diagram. A linear flow starts with 'Input speech buffer' entering a block labeled 'NS'. The output is 'NS processed speech buffer', which enters a 'Speech Encoder' block. The final output is 'Coded bit stream'.](b3baf3a29b67c7425d2562ddbc52f0cc_img.jpg) + +``` + +graph LR + A[Input speech buffer] --> B[NS] + B --> C[NS processed speech buffer] + C --> D[Speech Encoder] + D --> E[Coded bit stream] + +``` + +Figure 1: Noise Suppression implementation block diagram. A linear flow starts with 'Input speech buffer' entering a block labeled 'NS'. The output is 'NS processed speech buffer', which enters a 'Speech Encoder' block. The final output is 'Coded bit stream'. + +**Figure 1: Noise Suppression implementation** + +Alternatively, for implementation in conjunction with the bit-exact fixed point C reference code [3GPP TS 06.73 [5]] the NS module may operate on the pre-processed input speech buffer "old\_speech[L\_TOTAL]" in the structure "cod\_amrState" in the AMR C code [3GPP TS 06.73] after the pre-processing module (sample down-scaling and input high pass filtering) of the speech encoder. The bit-integrity of the speech encoder for this implementation shall be verified according to Figure 2 where the signals at Test Points 1 and 2 shall be identical for any input signal and the Reference Encoder is the part of [3GPP TS 06.73] after the pre-processing module. Note: implementation in conjunction with the AMR floating point C code is for further study. + +![Figure 2: Verification of AMR speech encoder bit-exactness for embedded NS implementations. The diagram shows an input speech buffer going into a 'Pre-processing (scaling and filtering)' block. The output 'old_speech [L_TOTAL]' enters an 'NS' block. The output 'NS processed old_speech [L_TOTAL]' splits into two paths: one goes to 'Speech Encoder (rest of)' leading to 'Coded bit stream' at 'Test Point 1'; the other goes to a 'Store' block, then to a 'Reference Encoder' leading to 'Coded bit stream' at 'Test Point 2'.](dbe553cf16dd14073b89a8263a428664_img.jpg) + +``` + +graph LR + In[Input speech buffer] --> Pre[Pre-processing +scaling and filtering] + Pre --> Old["old_speech [L_TOTAL]"] + Old --> NS[NS] + NS --> NSProc["NS processed +old_speech [L_TOTAL]"] + NSProc --> SE[Speech Encoder +rest of] + SE --> TP1((Test Point 1)) + TP1 --> Out1[Coded bit stream] + NSProc --> Store[Store] + Store --> Ref[Reference Encoder] + Ref --> TP2((Test Point 2)) + TP2 --> Out2[Coded bit stream] + +``` + +Figure 2: Verification of AMR speech encoder bit-exactness for embedded NS implementations. The diagram shows an input speech buffer going into a 'Pre-processing (scaling and filtering)' block. The output 'old\_speech [L\_TOTAL]' enters an 'NS' block. The output 'NS processed old\_speech [L\_TOTAL]' splits into two paths: one goes to 'Speech Encoder (rest of)' leading to 'Coded bit stream' at 'Test Point 1'; the other goes to a 'Store' block, then to a 'Reference Encoder' leading to 'Coded bit stream' at 'Test Point 2'. + +**Figure 2: Verification of AMR speech encoder bit-exactness for embedded NS implementations** + +## 5.2 Bit Exactness of the Speech Decoder + +The AMR speech decoder shall remain unaltered by the Noise Suppression function. + +## 5.3 Impact on Speech Path Delay + +The one way algorithmic delay due to the activation of AMR noise suppression shall be no more than 5ms in excess of the delay inserted by the AMR speech codec. In the handsfree case, this delay is part of the 39ms delay specified in 3GPP TS 03.50 [6]. + +The total additional delay (comprising of algorithmic and processing delays) shall not exceed 10ms. The processing delay is calculated using the following formula with $E*S*P$ set to 50. + +$$\text{delay(proc)} = \text{WMOPS} * 20 / (E * S * P)$$ + +where $\text{WMOPS}$ = complexity in weighted operations per second evaluated through the theoretical worst case. (Direct means of measurement of total delay is for further study.). + +## 5.4 Impact on Channel Activity + +The AMR speech codec with noise suppression activated should not significantly increase channel activity when used in conjunction with DTX. + +Channel activity increase will be measured thanks to the Voice Activity factor (VAF), defined as follows. + +Let $x$ be the VAF measured by the AMR VAD as an averaged value on all clean speech signals + +Let $y$ be the VAF measured by the AMR VAD without AMR NS active as an averaged value on all clean speech + noise signals (where the applicable clean speech signal is the speech signal used in the measure of $x$ ). + +Let $w$ be the VAF measured by the AMR VAD with AMR NS active as an averaged value on all clean speech + noise signals (where the applicable clean speech signal is the speech signal used in the measure of $x$ ). $w$ is required to be not significantly more than the maximum of $y$ and $x$ . Any case where $w$ is greater than $y$ should be further investigated. + +These requirements shall apply to both standardized AMR VADs. ( $w, x, y$ ) are determined using one or both VADs, and, if both are used, the requirements are checked relatively to each AMR VAD independently. + +The definition of upper limits on VAF increase and attendant confidence intervals are for further study. + +# --- 6 Requirements to be assessed by subjective tests + +## 6.1 Impact on Speech Quality + +The following performance requirements are stated under the assumption that the noise suppressor is tested as an integral part of the AMR speech codec with the speech codec operating at the rates defined within the test plan (Annex C). The performance requirements must be met for all these stated speech codec rates. + +### 6.1.1 Initial Convergence Time + +The initial convergence time shall be a maximum of $T$ seconds with $T$ equal to 2s. The definition of this time interval shall be understood strictly in accordance with its means of use in subjective listening experiments. Its use shall be defined by a process whereby the first $T$ seconds of each sample processed through the AMR speech codec with and without noise suppression active, is deleted before presentation to listeners. It is assumed that this process does not reduce intelligibility, or introduce clipping or similar effects into the resultant speech plus noise material. + +### 6.1.2 No Degradation in Clean Speech + +The noise suppression function must not have a statistically significant distorting effect on clean speech, in comparison with the performance of the AMR codec without noise suppression applied. This requirement also applies when VAD/DTX is active. + +The requirement is checked with the use of a paired comparison test where the requirement is met if AMR/NS is preferred or equal to AMR within the 95 % confidence interval. + +### 6.1.3 No degradation of Speech and no Undesirable Effects in Residual Noise in Conditions with Background Noise (*residual noise = background noise after AMR/NS*) + +The noise suppression function must not introduce any degradation of speech and no undesirable effects in the residual noise, when there is (acoustic) background noise in the speech signal. This requirement also applies when VAD/DTX is active. + +The requirement is checked with the use of a modified ACR test with specific instructions where the requirement is met if AMR/NS is better than or equal to AMR within the 95 % confidence interval in all conditions. + +### 6.1.4 Quality Impact compared to AMR + +The AMR speech codec with noise suppression activated must produce an output in noisy speech which is preferred amongst test listeners with statistical significance, compared to the case where noise suppression is not used. This requirement also applies when VAD/DTX is active. + +The requirement is checked with the use of a CCR test where the requirement is met if AMR/NS is preferred to AMR within the 95 % confidence interval in at least 4 of the 6 conditions tested. Preference or equality within the 95 % confidence interval is required for the remaining conditions. + +Additionally, it is required that the subjective SNR improvement as measured by the methodology [Annex B] (where the measure is conducted on the associated CCR tests [Annex C]) meets the following requirements: + +- (a) In at least 2 of the 6 conditions tested the SNR improvement shall not be less than 6dB within the 95% confidence interval. +- (b) In at least 2 of the remaining 4 conditions the SNR improvement shall not be less than 4dB within the 95% confidence interval. + +# 7 Performance Objectives assessed by Objective Measures + +## 7.1 Impact on Active Speech Level + +The AMR speech codec with noise suppression activated must not significantly alter the active speech level. + +The requirement is checked with the use of a ITU-T Recommendation P.56 [7] speech level meter (the use of which remains for further study). Let $x$ be the averaged level of the clean speech material for one experiment and let $y$ be the averaged level of the processed material with AMR NS activated for the same experiment. The requirement is met if the absolute difference between $x$ and $y$ is less than 2 dB for all experiments. *The processed material should not be normalised to the nominal speech level before the listening tests.* + +*Note that this requirement does not preclude the use of active level control.* + +## 7.2 Objective Speech Quality Measures + +The objective measures of noise power level reduction (NPLR) and signal-to-noise ratio improvement (SNRI) defined in Annex 1 are to be used to characterise the performance of the AMR/NS solution. Objectives are defined for these measures in the following table. These measures will be used to provide additional information only and are not to be considered to be requirements. + +C source code is attached to this specification which shall be used to undertake these measurements. + +| Objective quality measure/test condition | Performance objective | +|-----------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------|-----------------------| +| NPLR
Assessment: To be evaluated using a predefined set of material (as used in the AMR/NS Selection Phase) comprising speech mixed with stationary car noise in the SNR conditions of 6 dB and 15 dB, following otherwise the guidelines set forth in [Annex 1]. | -7 dB or lower | +| SNRI
Assessment: To be evaluated using a predefined set of material (as used in the AMR/NS Selection Phase) comprising speech mixed with stationary car noise in the SNR conditions of 6 dB and 15 dB, following otherwise the guidelines set forth in [Annex 1]. | 6 dB or higher | + +# --- 8 Interaction with supplementary services + +## 8.1 General + +This clause defines requirements regarding the interactions between GSM supplementary services and the Noise Suppression Feature. + +The application of Noise Suppression shall not interfere with the provision or invocation of any supplementary services. + +## 8.2 Explicit Call Transfer (ECT) + +No adverse interaction. If the new party is a mobile station with support for the Noise Suppression feature, the noise suppression feature shall be invoked. + +## 8.3 Call wait/Call hold. + +No interaction. + +## 8.4 Multiparty + +No interaction. + +## 8.5 Service Announcements + +No interaction. + +# --- 9 Interaction with Alternate and Followed by services + +There shall be no impact on data transmission due the Noise Suppression Feature + +# --- 10 Interaction with other speech services + +There is no requirement for Noise Suppression in ASCI services. + +# --- 11 Interaction with DTMF and other signalling tones + +DTMF and other signalling tones transmission performance during the application of Noise Suppression shall be no worse than the case where Noise Suppression is turned off. + +# --- 12 Interaction with Lawful Intercept + +In the case where lawful intercept is required in a call where Noise Suppression is activated, the Noise Suppression shall not cause any degradation in the speech quality received by the A and B parties. + +# --- 13 Interaction with TFO + +No interaction. + +# --- Annex A (informative): Method for generating Objective Performance Measures + +This annex presents an objective methodology for characterising the performance of noise suppression (NS) methods. Two objective measures are specified to be used for characterising NS solutions complying with the AMR/NS specification. + +## A.1 Notations + +The following notations are used in this document: + +- The operator AMR( $\cdot$ ) corresponds to applying the AMR speech encoder and decoder on the input. +- The operator NR( $\cdot$ ) corresponds to applying the NS algorithm, and the AMR speech encoder and decoder on the input. +- The clean speech signals are referred to as $s_i$ , $i = 1$ to $I$ . +- The noise signals are referred to as $n_j$ , $j = 1$ to $J$ . +- The noisy speech test signals are referred to as $d_{ij} = \beta_{ij}(\text{SNR}) n_j + s_i$ , $i = 1$ to $I$ , $j = 1$ to $J$ , where $d_{ij}$ is built by adding $s_i$ and $n_j$ with a pre-specified SNR as presented below. +- The processed signal are referred to as $y_{ij} = \text{NR}(d_{ij})$ . +- The reference signal in the calculations shall be either the noisy speech test signal $d_{ij}$ itself or $d_{ij}$ processed by the AMR speech codec without NS processing. The latter signal will be referred to as $c_{ij} = \text{AMR}(d_{ij})$ , $i = 1$ to $I$ , $j = 1$ to $J$ . The relevant reference signal will be indicated in the formulation of each objective measure below. +- The notation $\text{Log}(\cdot)$ indicates the decimal logarithm. +- $\beta_{ij}(\text{SNR})$ is the scaling factor to be applied to the background noise signal $n_j$ in order to have a ratio SNR (in dB) between the clean speech signal $s_i$ and $n_j$ . The scaling of the input speech and noise signals is to be carried according to the following procedure: + - 1) The clean speech material is scaled to a desired dBov level with the ITU-T Recommendation P.56 [7] speech voltmeter, one file at a time, each file including a sequence of one to four utterances from one speaker. + - 2) A silence period of 2 s is inserted in the beginning of each of the resulting files to make up augmented clean speech files. + - 3) Within each noise type and level, a noise sequence is selected for every speech utterance file, each with the same length as the corresponding speech files, and each noise sequence is stored in a separate file. + - 4) Each of the noise sequences is scaled to a dBov level leading to the SNR condition corresponding to the $\beta_{ij}(\text{SNR})$ value in each of the test cases by applying the RMS level based scaling according to the P.56 [7] recommendation. + - The determination of which frames contain active speech is to be carried out with reference to the ITU-T Recommendation P.56 [7] active speech level measurement and is related to the classification of the frames into the presented speech power classes which is explained below. + +## A.2 Test material + +The test material should manifest at least the following extent: + +- Clean speech utterance sequences: 6 utterances from 4 speakers - 2 male and 2 female - totalling 24 utterances +- Noise sequences: + - car interior noise, 120 km/h, fairly constant power level + - street noise, slowly varying power level + +Special care should be taken to ensure that the original samples fulfil the following requirements: + +- the clean speech signals are of a relatively constant average (within sample, where 'sample' refers to a file containing one or more utterances) power level +- the noise signals are of a short-time stationary nature with no rapid changes in the power level and no speech-like components + +The test signals should cover the following background noise and SNR conditions: + +- car noise at 3 dB, 6 dB, 9 dB, 12 dB and 15 dB +- street noise at 6 dB, 9 dB, 12 dB, 15 dB and 18 dB + +A feasible subset of these conditions giving a practically useful indication of the achieved performance would be: + +- car noise at 6 dB and 15 dB +- street noise at 9 dB and 18 dB + +The samples should be digitally filtered before NS and speech coding processing by the MSIN filter to become representative of a real cellular system frequency response. + +## A.3 Objective measures for characterization of NS algorithm effect + +**Assessment of SNR improvement level:** The SNR improvement measure, *SNRI*, measures the SNR improvement achieved by the NS algorithm. SNR improvement is calculated separately in three groups of frames that represent power gated constituents of active speech signal. Hence, the *SNRI* measure is calculated separately in frames of high, medium and low power. These categories are used to characterise the effect of the NS processing on speech, allowing to distinguish the effect on strong, medium and weak speech. In addition to calculating the SNR improvement separately on the three categories, they are used to form an aggregate measure. A frame length of 80 samples is used since it has been found the most efficient to describe changes in the signal caused by NS processing. + +The calculation is here presented for the high power speech class: + +For each background noise condition j + +For each speaker i + +Construct a noisy input signal $d_{ij}$ as follows: + +$$d_{ij}(n) = \beta_{ij} n_j(n) + s_i(n)$$ + +where $\beta_{ij}$ depends on the SNR condition according to the procedure described above + +$$c_{ij} = \text{AMR}(d_{ij})$$ + +$$y_{ij} = \text{NR}(d_{ij})$$ + +$$\text{SNROUT\_h}_{ij} = \frac{\xi + \frac{1}{K_{\text{sph}}} \sum_{l=1}^{K_{\text{sph}}} \sum_{n=k_{\text{sph},l}}^{k_{\text{sph},l+1}-1} y_{ij}^2(n)}{\xi + \frac{1}{K_{\text{nse}}} \sum_{m=1}^{K_{\text{nse}}} \sum_{p=k_{\text{nse},m}}^{k_{\text{nse},m+1}-1} y_{ij}^2(p)} - 1$$ + +$$\text{SNRin\_h}_{ij} = \frac{\xi + \frac{1}{K_{sph}} \sum_{l=1}^{K_{sph}} \sum_{n=k_{sph,l} \cdot 80}^{k_{sph,l} \cdot 80+79} c_{ij}^2(n)}{\xi + \frac{1}{K_{nse}} \sum_{m=1}^{K_{nse}} \sum_{p=k_{nse,m} \cdot 80}^{k_{nse,m} \cdot 80+79} c_{ij}^2(p)} - 1$$ + +$$\text{SNRI\_h}_{ij} = \begin{cases} 0 & ; \text{ SNROUT\_h}_{ij} \leq \xi \vee \text{SNRin\_h}_{ij} \leq \xi \\ 10 \cdot [\text{Log}(\text{SNROUT\_h}_{ij}) - \text{Log}(\text{SNRin\_h}_{ij})] & ; \text{ else} \end{cases} \quad (1)$$ + +where $k_{sph}$ and $K_{sph}$ are the index and the total number of frames containing speech of a high power + +$k_{nse}$ and $K_{nse}$ are the corresponding index and total number of noise only frames + +$\xi > 0$ is a constant that should be set at $10^{-5}$ + +SNRI\_mij correspondingly for medium power frames + +SNRI\_lj correspondingly for low power frames + +$$\text{SNRI}_{ij} = \frac{1}{K_{sph} + K_{spm} + K_{spl}} (K_{sph} \cdot \text{SNRI\_h}_{ij} + K_{spm} \text{SNRI\_m}_{ij} + K_{spl} \text{SNRI\_l}_{ij}) \quad (2)$$ + +$$\text{SNRI}_j = \frac{1}{I} \sum_{i=1}^I \text{SNRI}_{ij} \quad (3)$$ + +$$\text{SNRI} = \frac{1}{J} \sum_{j=1}^J \text{SNRI}_j \quad (4)$$ + +In addition, measures for the SNR improvement in the high, medium and low power speech classes (SNRI\_h, SNRI\_m, SNRI\_l, respectively) shall be recorded based on the following formulae: + +$$\text{SNRI\_h} = \frac{1}{J} \sum_{j=1}^J \text{SNRI\_h}_j = \frac{1}{J} \sum_{j=1}^J \frac{1}{I} \sum_{i=1}^I \text{SNRI\_h}_{ij} \quad (5)$$ + +$$\text{SNRI\_m} = \frac{1}{J} \sum_{j=1}^J \text{SNRI\_m}_j = \frac{1}{J} \sum_{j=1}^J \frac{1}{I} \sum_{i=1}^I \text{SNRI\_m}_{ij} \quad (6)$$ + +$$\text{SNRI\_l} = \frac{1}{J} \sum_{j=1}^J \text{SNRI\_l}_j = \frac{1}{J} \sum_{j=1}^J \frac{1}{I} \sum_{i=1}^I \text{SNRI\_l}_{ij} \quad (7)$$ + +It is, in addition, informative to record separately the noise type specific SNR improvement measures, namely, SNRI\_hj, SNRI\_lj, SNRI\_mj and SNRIj for each j. + +To determine which frames belong to high, medium and low power classes of active speech and which present pauses in the speech activity (noise only), the active speech level (in dB) sp\_lvl of the noise free speech $s_i(n)$ is first determined according to the ITU-T Recommendation P.56. [7] Thereafter, the frames are classified into the four classes as follows . + +Let us first define four number sequences: $\{k_{sph}\}$ , $\{k_{spm}\}$ , $\{k_{spl}\}$ , $\{k_{nse}\}$ . All four sequences are initialized to an empty sequence: + +$$\begin{aligned} +\{k_{sph}\}_0 &= \emptyset \\ +\{k_{spm}\}_0 &= \emptyset \\ +\{k_{spl}\}_0 &= \emptyset \\ +\{k_{nse}\}_0 &= \emptyset +\end{aligned} \tag{8}$$ + +Then, the frame power is calculated in each signal frame $k$ : + +$$sp\_pow(k) = 10 \log \left[ \max \left\{ \varepsilon, \frac{\sum_{n=k \cdot 80}^{k \cdot 80+79} (s_i(n))^2}{80} \right\} \right] \tag{9}$$ + +We shall then classify each frame according to the frame power as follows: + +$$\begin{aligned} +&\text{if } sp\_pow(k) \geq sp\_lvl + th\_h \\ +&\quad \{k_{sph}\}_{length(k_{sph})+1} = \left\{ \{k_{sph}\}_{length(k_{sph})}, k \right\} \\ +&\text{else if } sp\_pow(k) \geq sp\_lvl + th\_m \\ +&\quad \{k_{spm}\}_{length(k_{spm})+1} = \left\{ \{k_{spm}\}_{length(k_{spm})}, k \right\} \\ +&\text{else if } sp\_pow(k) \geq sp\_lvl + th\_l \\ +&\quad \{k_{spl}\}_{length(k_{spl})+1} = \left\{ \{k_{spl}\}_{length(k_{spl})}, k \right\} \\ +&\text{else if } sp\_lvl + th\_nl \leq sp\_pow(k) < sp\_lvl + th\_nh \\ +&\quad \{k_{nse}\}_{length(k_{nse})+1} = \left\{ \{k_{nse}\}_{length(k_{nse})}, k \right\} +\end{aligned} \tag{10}$$ + +where $\varepsilon > 0$ is a constant whose value shall be such that in the dB scale, it shall be below $sp\_lvl + th\_nl$ ; a value of $10^{-7}$ should be used if $sp\_lvl = -26$ dBov and $th\_nl = -34$ dB, as proposed below + +$th\_h$ , $th\_m$ , $th\_l$ are pre-determined lower threshold power levels for classifying the speech frames to the high, medium, and low power classes, correspondingly. In the following, these threshold values are called *power class threshold values* + +$length(k)$ is a function returning the length of the number sequence $\{k\}$ + +The following notes on the formulation of the frame classification are made: + +- The lower bound for the power of the noise-only class of frames is motivated by a desire to restrict the analysis to noise frames that are among or close the speech activity, hence excluding long pauses from the analysis. This makes the analysis concentrate increasingly on the effects encountered during speech activity. +- In poor SNR conditions, the noise power level may occur to be higher than the lower bound of some of the speech power classes. However, even in this case, the information of the effect on the low power portions of speech may be informative. Another way of formulating the measure might be to make the power thresholds dependent on the noise level. This would, however, restrict the comparability of the SNR improvement figures of the different classes over experiments with different background noise content. + +The scaling for the clean speech material should be determined optimally so that the dynamics of the 16 bit arithmetic system is efficiently used but no waveform clipping is produced. Typically, a normalisation to the active speech level of -26 dBov is preferable. In such a case, the following values should be used for the power class thresholds: + +$$\begin{aligned} +\text{th\_h} &= -1 \text{ dB} \\ +\text{th\_m} &= -10 \text{ dB} \\ +\text{th\_l} &= -16 \text{ dB} \\ +\text{th\_nh} &= -19 \text{ dB} \\ +\text{th\_nl} &= -34 \text{ dB} +\end{aligned} \tag{8}$$ + +**Assessment of noise power level reduction.** The noise power level reduction **NPLR** measure relates to the capability of the NS method to attenuate the background noise level. + +The **NPLR** measure is calculated as follows: + +For each background noise condition j + +For each speaker i + +Construct a noisy input signal $d_{ij}$ as follows: + +$$d_{ij}(n) = \beta_j n_j(n) + s_i(n)$$ + +where $\beta_j$ depends on the SNR condition according to the procedure described above + +$$c_{ij} = \text{AMR}(d_{ij})$$ + +$$y_{ij} = \text{NR}(d_{ij})$$ + +$$\begin{aligned} +NPLR_{ij} = 10 \cdot \left\{ \text{Log} \left[ \xi + \frac{1}{K_{nse}} \sum_{m=1}^{K_{nse}} \sum_{n=k_{nse,m} \cdot 80}^{k_{nse,m} \cdot 80+79} y_{ij}^2(n) \right] \right. \\ +\left. - \text{Log} \left[ \xi + \frac{1}{K_{nse}} \sum_{l=1}^{K_{nse}} \sum_{p=k_{nse,l} \cdot 80}^{k_{nse,l} \cdot 80+79} c_{ij}^2(p) \right] \right\}, +\end{aligned} \tag{12}$$ + +where $\xi > 0$ is a constant that should be set at $10^{-5}$ ; + +$k_{nse}$ and $K_{nse}$ are the corresponding index and total number of noise only frames + +$$NPLR_j = \frac{1}{I} \sum_{i=1}^I NPLR_{ij} \tag{13}$$ + +$$NPLR = \frac{1}{J} \sum_{j=1}^J NPLR_j \tag{14}$$ + +Furthermore, it is informative to record separately the noise type specific NPLR measures, or $NPLR_j$ , for each j. + +**Comparison of SNRI and NPLR.** A comparison of the **SNRI** and **NPLR** measures can be used to acquire an indication of possible speech distortion produced by the tested NS method. If the **NPLR** parameter assumes clearly higher absolute values than **SNRI**, it can be expected that the NS candidate causes distortion to speech. This relation, however, should always be verified through a comparison with subjective test results. + +# Annex B (normative): Methodology for Measuring Subjective SNR Improvement for CCR Experiments + +The purpose of experiment 3 is to evaluate the performances of the NS algorithm in background noise conditions with two different bit-rates (5.9 kbps and 12.2 kbps). For these experiments three types of noise have been selected: car noise, street noise and babble noise. For each type of noise two different nominal SNR levels have been set: + +| Noise type | SNR [dB] | +|------------|----------| +| Car | 6, 15 | +| Street | 9, 18 | +| Babble | 9, 18 | + +For each sub-experiment and for each type of noise three ideal NS reference conditions will be processed. The exception is that for the higher SNRs (15dB for car noise and 18 dB for street, babble noise) only 2 ideal noise reference conditions will be tested (+3, +6dB): + +| Ideal SNR improvement | +|-----------------------| +| SNR sub-exp. +3 dB | +| SNR sub-exp. +6 dB | +| SNR sub-exp. +9dB | + +Each ideal NS will be compared during the sub-experiment with the speech+noise signals mixed at the nominal SNR levels. This leads to a total number of CCR reference results of 5 per sub-experiment corresponding to 3 (2 for the higher SNRs) SNR improvement levels. By connecting adjacent point by straight lines we will obtain a graph giving a correspondence between CCR scores and perceived SNR improvement (cf. figure B.1). + +Finally the perceived SNR improvement for an AMR-NS candidate is obtained using the CCR vs SNR graph as illustrated in figure B.1. + +![Figure B.1: Example of CCR versus SNR improvement graph. The graph plots SNR improvement [dB] on the y-axis (values +3, +6, +9) against CCR scale on the x-axis (values 3, 2, 1, 0). Two lines are shown: a solid line for 'Ideal NS score' (O) and a dashed line for 'AMR-NS candidate score' (*). Data points with error bars are plotted on both lines. A legend indicates that the solid line represents a '4 dB Estimate w/ estimated S.D.' and the dashed line represents a 'Candidate Data Point w/ Measured S.D.'.](a28fca9a7503d40707ef5273befe1be4_img.jpg) + +The graph shows the relationship between CCR scale (x-axis, decreasing from 3 to 0) and SNR improvement [dB] (y-axis, increasing from +3 to +9). Two data series are plotted: + + +- Ideal NS score (O):** Represented by a solid line. It passes through points approximately at (2.2, +3), (1.8, +6), and (1.2, +9). +- AMR-NS candidate score (\*):** Represented by a dashed line. It passes through points approximately at (2.1, +3), (1.7, +6), and (1.1, +9). + + Error bars are shown for each data point. A legend box on the right indicates: + + +- The solid line represents a "4 dB Estimate w/ estimated S.D." +- The dashed line represents a "Candidate Data Point w/ Measured S.D." + + A label "Perceived SNR improvement" with an arrow points to the y-axis. + +Figure B.1: Example of CCR versus SNR improvement graph. The graph plots SNR improvement [dB] on the y-axis (values +3, +6, +9) against CCR scale on the x-axis (values 3, 2, 1, 0). Two lines are shown: a solid line for 'Ideal NS score' (O) and a dashed line for 'AMR-NS candidate score' (\*). Data points with error bars are plotted on both lines. A legend indicates that the solid line represents a '4 dB Estimate w/ estimated S.D.' and the dashed line represents a 'Candidate Data Point w/ Measured S.D.'. + +**Figure B.1. Example of CCR versus SNR improvement graph** +**O: ideal NS score, \*:AMR-NS candidate score** + +# --- Annex C (normative): Test Plan for Checking Conformance to Requirements --- + +## C.1 Introduction + +The present document contains the complete set of subjective test experiments for the testing of the speech performance of Noise Suppression solutions for application to AMR. The purpose of the tests is to check for compliance to the recommended minimum performance requirements [1]. + +The AMR-NS Selection Tests are split into 4 main Experiments and 7 Sub-Experiments listed in the following table. + +| Exp. No. | Title | No. of Sub-Exp. | +|----------|-----------------------------------------------------------------------------------------------------------------|-----------------| +| 1 | Degradation in Clean Speech (PC) | 1 | +| 2 | No degradation of Speech and no Undesirable Effects in Residual Noise in Conditions with Background Noise (ACR) | 3 | +| 3 | Performances in Background Noise Conditions (Mod-CCR) | 2 | +| 4 | Influence of Input Level, Voice Activity Detection and Discontinuous Transmission (Mod-CCR) | 1 | +| | Total Number of Sub-Experiments: | 7 | + +## --- C.2 Document Structure + +The main body of the document starts at clause 4, and is arranged as follows: + +| | | | +|---------------|----------------------------------------------|-----------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------| +| Clause 4: | References, Conventions, and Contacts | References to specification documents, lists of abbreviations, and contact names for the different areas of the document | +| Clause 5: | Roles and Responsibilities | Identification of roles and allocation of Responsibilities. | +| Clause 6: | Information Relevant to all Experiments | Information relevant to all experiments. | +| Clauses 7-10: | Test Plans | Individual test plans. Information already covered in clause 6 is not repeated in the individual plans. Note that the processing tables for the experiments are collated in Annex B, and the randomizations (where required) in Annex C | +| Annex A: | Instructions to Subjects and Data Collection | For the Modified CCR, Pair Comparison, Modified ACR. | +| Annex B: | Processing Tables | Processing Tables for all experiments. These map which speech samples are to be processed through which conditions. | +| Annex C: | Presentation Orders | Randomized presentation orders for experiments. | + +## C.3 References, Conventions, and Contacts + +- [1] 3GPP TS 06.77 Minimum Performance Requirements for Noise Suppressor Application to the AMR Speech Encoder (latest version) +- [2] TBD Processing Function for the GSM AMR Noise Suppressor Selection Tests (Proposal - re-use selection phase document) +- [3] ITU-T Com 12 Handbook on Telephonometry +- [4] ITU-T Rec. P.800 Methods for subjective determination of transmission quality +- [4] 3GPP TS 06.71 Adaptive Multi-Rate Speech Codec; General Description +- [5] 3GPP TS 06.73 Adaptive Multi-Rate Speech Codec; ANSI C-Code +- [6] 3GPP TR 06.75 [9] Performance Characterization of the GSM Adaptive Multi-Rate Speech Codec +- [7] 3GPP TS 06.90 [10] Adaptive Multi-Rate Speech Codec; Transcoding Functions +- [8] 3GPP TS 06.91 [11] Adaptive Multi-Rate Speech Codec; Error Concealment of Lost Frames +- [9] 3GPP TS 06.92 [12] Adaptive Multi-Rate Speech Codec; Source Controlled Rate Adaptation +- [10] 3GPP TS 06.94 [13] Adaptive Multi-Rate Speech Codec; Voice Activity Detector + +## C.4 Key Acronyms + +| | | +|--------|--------------------------------------------------------------| +| ACR | Absolute Category Rating | +| AMR | Adaptive Multi-Rate Speech Codec for the GSM System | +| AMR-NS | Noise Suppressor for the AMR Speech Codec | +| BER | Bit Error Rate | +| C/I | Carrier to Interference Ratio | +| DCR | Degradation Category Rating | +| DECi | Dynamic Error Condition #i for Dynamic C/I conditions | +| ECx | Error Condition for static C/I conditions with C/I = x dB | +| EFR | GSM Enhanced Full Rate speech codec | +| EP | Error Pattern | +| FR | GSM Full Rate channel or existing GSM Full Rate speech codec | +| HR | GSM Half Rate channel or existing GSM Half Rate speech codec | +| MNRU | Modulated Noise Reference Unit | +| MOS | Mean Opinion Score | +| S/N | Signal to Noise Ratio | + +### C.4.1 Contact Names + +The following persons should be contacted for questions related to the test plan. + +| Clause | Contact Person/Email | Organization | Address | Telephone/Fax | +|---------------|-----------------------------------------------------------|-----------------------|-----------------------------------------------------------------------------|------------------------------------------------------------| +| Overall | | | | | +| Experiments 1 | Dominique Pascal/
dominique.pascal@rd.francetelecom.fr | France
Télécom R&D | 2 Av. Pierre Marzin
Technopole Anticipa
22307 Lannion Cedex
France | Tel : + 33 2 96 05 15
78
Fax : + 33 2 96 05 13
16 | +| Experiments 2 | Anders Eriksson/
anders.eriksson@era-t.ericsson.se | Ericsson | | | +| Experiments 3 | Steve Aftelak/
Stephen.Aftelak@motorola.com | Motorola | | | +| Experiment 4 | Steve Aftelak/
Stephen.Aftelak@motorola.com | Motorola | | | + +## --- C.5 Roles and Responsibilities + +It is the sole responsibility of the proponent of a noise suppression solution to ensure that the testing is conducted properly according to this test plan. It is strongly recommended that third party subjective testing laboratories be instructed to perform the tests according to this plan. + +It is additionally required that the test material is processed in accordance with the processing functions document [2]. + +Each experiment should be conducted in at least 2 languages. The proponent of the noise suppression solution is at liberty to choose the languages to be used, but it is recommended that a reasonable range of languages be incorporated, across the full set of experiments. + +## --- C.6 Information relevant to all Experiments + +### C.6.1 General Technical Notes + +Any and all deviations from the specifications contained in this document and the Processing Functions document [2] must be documented and submitted to SMG11/S4 along with the experimental results. + +### C.6.2 Codec Adaptation and Error Conditions + +The philosophy of the AMR system is that it is capable of dynamically altering the ratio of speech and channel coding to maximize speech performance as channel conditions change. Each of the combinations of speech and channel coding rates is known as a mode. + +However, for the purpose of the AMR Noise Suppressor tests, only fixed mode operation will be considered. + +### C.6.3 Speech Material + +All AMR-NS Experiments are subjective listening experiments using pre-recorded speech passed through the candidate algorithms and simulated impairment conditions prior to use in the experiments. Three types of speech sample are used in these experiments: + +Single sentence samples, 4 seconds in length + +Short samples; sentence pairs, 8 seconds in length. + +Long samples; sentence quadruplets, 16 seconds in length. + +The experiment investigating the equivalence of the candidate Noise Suppressor algorithms to the AMR algorithm without noise suppression in a quiet environment (PC experiment 1) will use the single sentence stimuli. The experiments investigating the possible introduction of artefacts and clipping by the candidate Noise Suppressor algorithms (ACR experiments 2a, 2b & 2c) will use the long 16-second samples. Experiment 2 includes conditions investigating level dependency, VAD and DTX. All other experiments will use the short 8-second samples. + +For all original speech samples a 2s header will be added to accommodate the Initial Convergence Time of the Noise Suppressor algorithms. For all experiments this header should be removed at the end of the processing prior to being used in subjective listening tests. + +Information for constructing these sentences is provided in the remainder of this clause. + +Unless stated otherwise in the individual plans, each source speech file will contain unique speech material (i.e. none of the sentences used in any given sample should be used in any other sample for the same, or any other talker within any sub experiment). + +Pre-recorded source speech material may possibly be purchased as described in Clause 6.3.1. Preferably, the test house should provide its own source speech material. The guidelines contained in Clause 6.3.2 should be followed. + +To avoid noise contrast effects, any silence gaps and/or pauses added to the speech files to pad them out into the specified formats for the source speech samples described in clauses 6.3.3, 6.3.4 and 6.3.5, should not be pure digital silence. Padding out should be done by adding the ambient noise present during the recording of the speech material between the sentences. + +The information in clauses 6.3.3, 6.3.4 and 6.3.5 should be used in the preparation of the material that the talkers will utter, as well as how the recorded material should be constructed. + +#### C.6.3.1 Availability of Pre-recorded Speech Material + +A "Multi-lingual Speech Database for telephonometry 1994", on 4 CD-ROM disks, was available from NTT-AT, No.7 Hakuei Building, 2-4-15 Naka-machi, Musashino-shi, 180 Japan (phone: +81 422 37 0823, fax: +81 422 60 4806). + +In this database, the speech samples consist of pairs of short sentences with a total length of 8-10 seconds. Each sentence lasts approximately 2 to 3 seconds. Four male and four female native speakers are assigned to each of the 21 languages and 96 speech samples are available for each language. The sampling rate is 16 kHz. Active speech level (as defined in ITU-T Rec. P.56) of every speech sample is adjusted to -26dBovl. + +Each CD consists of two different areas: audio and data. Speech samples in the audio area are digitized by 44.1 kHz and 16 bits word length linear PCM and can be played back by a commercial CD player. All speech samples in the data area are recorded in standardized format in 16-bit, 2's complement, low-byte first (little endian) format and can be retrieved by an ordinary PC-DOS system and CD-ROM reader. + +#### C.6.3.2 Recording Your Own Speech Databases + +All speech recordings should be made in acoustical and electrical environments complying with the requirements given in Annex B.1.1 of ITU-T Rec. P.800. + +The recommended method is to record the speech with a linear microphone and a low-noise amplifier with flat frequency response, digitize the speech, and then flat filter and level equalize. To achieve optimum SNR, the microphone should be positioned 15 to 20 cm from the talker's lips. A windscreen should be used if breath puffs from the talker are noticed. + +The recordings should be made directly into a computer (A/D) or via a high quality recording system such as a DAT. + +#### C.6.3.3 Format for Single Sentence Speech Samples + +Each source speech file will contain one sentence and will last nominally 4s. All source speech files within an experiment will be exactly the same length. This enhances the ability to recognize processing problems. An approximate 0.5 seconds period of silence precedes the sentence, and a similar period of silence follows the sentence. The speech files are organized as in the example shown in Figure 6.3.1. The sentences will be simple meaningful sentences as described in Annex B.1.4 of ITU-T Rec. P.800. + +![Diagram illustrating the format for single sentence speech samples. It shows a timeline starting at 'Start of File' and ending at 'End of File'. A central box labeled 'Sentence 1' is flanked by two periods of silence, each labeled '~0.5 s' with arrows indicating the duration.](db7cb51aac8519daab50e2171cecae82_img.jpg) + +The diagram shows a horizontal timeline representing a speech file. It starts at a vertical line labeled "Start of File" and ends at another vertical line labeled "End of File". In the center, there is a rectangular box labeled "Sentence 1". To the left of this box, between the "Start of File" line and the box, is a period of silence indicated by a double-headed arrow and labeled "~0.5 s". Similarly, to the right of the box, between the box and the "End of File" line, is another period of silence indicated by a double-headed arrow and labeled "~0.5 s". + +Diagram illustrating the format for single sentence speech samples. It shows a timeline starting at 'Start of File' and ending at 'End of File'. A central box labeled 'Sentence 1' is flanked by two periods of silence, each labeled '~0.5 s' with arrows indicating the duration. + +**Figure 6.3.1: Example of Speech file structure for single sentences** + +It must be noted that the trailing silence of 0,5s after the end of the sentence in the file is of extreme importance, since there are (for some conditions) a series of FIR filters with large number of coefficients. If the prescribed trailing silence is not present, there is a considerable risk that speech will be clipped at the end of the file. + +#### C.6.3.4 Format for Short Speech Samples + +Each source speech file will contain one pair of sentences and will last nominally 8 seconds, with a flexible time interval between the two sentences. All source speech files within an experiment will be exactly the same length. This enhances the ability to recognize processing problems. An approximate 0.5 seconds period of silence precedes the first sentence in the file, and a similar period of silence follows the second sentence in the file. The speech files are organized as in the example shown in Figure 6.3.2. The sentences will be simple meaningful sentences as described in Annex B.1.4 of ITU-T Rec. P.800. + +![Figure 6.3.2: Example of speech file structure for short speech samples. The diagram shows a timeline from 'Start of File' to 'End of File'. It begins with a ~0.5 s silence interval, followed by 'Sentence 1', then another ~0.5 s silence interval, followed by 'Sentence 2', and finally a ~0.5 s silence interval before the 'End of File'.](eb03559a4d92ea9ebd63ea9be663c50a_img.jpg) + +``` + + Start of File ---[~0.5 s]---> [Sentence 1] ---[~0.5 s]---> [Sentence 2] ---[~0.5 s]---> End of File + +``` + +Figure 6.3.2: Example of speech file structure for short speech samples. The diagram shows a timeline from 'Start of File' to 'End of File'. It begins with a ~0.5 s silence interval, followed by 'Sentence 1', then another ~0.5 s silence interval, followed by 'Sentence 2', and finally a ~0.5 s silence interval before the 'End of File'. + +**Figure 6.3.2: Example of speech file structure for short speech samples** + +It must be noted that the trailing silence of 0.5s after the end of the second sentence in the file is of extreme importance, since there are (for some conditions) a series of FIR filters with large number of coefficients. If the prescribed trailing silence is not present, there is a considerable risk that speech will be clipped at the end of the file. + +#### C.6.3.5 Format for Long Speech Samples + +Each sample will contain 4 different sentences and will last nominally 16 seconds, with a time interval between sentences as described in Annex B1.4 of ITU-T Rec. P.800. All source speech files within an experiment will be exactly the same length. An approximate 0.3-0.5 seconds period of silence precedes the first sentence in the file, and a similar period of silence follows the last sentence in the file. The speech files are organized as in the example shown in Figure 6.3.3. The sentences will be simple meaningful sentences as described in Annex B1.4 of ITU-T Rec. P.800. Active speech in each source speech file should be present for not less than 9 seconds and not more than 12s. *{note – this last requirement may be hard to meet for some speech data bases. The typical English Harvard Sentence is less than 2 seconds long. Four of these would be less than the required 9 seconds of active speech. Therefore a reasonable relaxation of this last requirement should be tolerated.}* + +![Figure 6.3.3: Example of speech file structure for long speech samples. The diagram shows a timeline from 'Start of File' to 'End of File'. It begins with a ~0.3 to 0.5s silence interval, followed by 'Sentence 1', then a 'pause', 'Sentence 2', another 'pause', 'Sentence 3', a third 'pause', 'Sentence 4', and finally a ~0.3 to 0.5s silence interval before the 'End of File'.](6f31cdb576d2f15c35c3f266e5f59211_img.jpg) + +``` + + Start of File ---[~0.3 to 0.5s]---> [Sentence 1] ---[pause]---> [Sentence 2] ---[pause]---> [Sentence 3] ---[pause]---> [Sentence 4] ---[~0.3 to 0.5s]---> End of File + +``` + +Figure 6.3.3: Example of speech file structure for long speech samples. The diagram shows a timeline from 'Start of File' to 'End of File'. It begins with a ~0.3 to 0.5s silence interval, followed by 'Sentence 1', then a 'pause', 'Sentence 2', another 'pause', 'Sentence 3', a third 'pause', 'Sentence 4', and finally a ~0.3 to 0.5s silence interval before the 'End of File'. + +**Figure 6.3.3: Example of speech file structure for long speech samples** + +These samples could be built by the addition of two of the 8-sec sentences described in clause 6.3.3, providing that the constraint for the active speech described above is (reasonably) fulfilled. + +#### C.6.3.6 Processing of the Speech Files + +All speech files will need to be pre-processed prior to being processed through the experimental conditions. This pre-processing ensures that the speech is at the correct level and has the correct input characteristic. Full details on the processing required are given in [2]. Speech levels will be measured with the P.56 [7] algorithm and level adjusted with the gain/loss algorithm to the level required for each test condition as defined in the test plans for the individual experiments. Where the nominal level is specified, this level should be set to 26dB ( $\pm 1$ dB) below digital overload (-26dBovl). + +Some of the experiments require that the source speech material has background noise added. Details of the process to be followed are given in [2]. Noise levels will be measured with the rms. computation algorithm and level adjusted with the gain/loss algorithm to the required level. The following procedure will be followed: + +The environmental noise will be Delta SM filtered to incorporate a near field microphone response. + +The environmental noises will be passed through the GSM send characteristic (see [2]). + +The noise levels will be adjusted using the r.m.s. measure to the mean level dictated by following test plans. For each type of noise, six segments will be taken from the noise file. The segments will be numbered from N1 to N6. + +The source speech material will be passed through the GSM send characteristic [2] and normalized (level equalized to -26dB) using the speech level meter complying with ITU-T Recommendation P.56 [7]. This is the responsibility of the Host Laboratories. + +Finally, the noise will be digitally mixed with the normalized speech material. If the resulting signal amplitude exceeds the overload point of the A/D converter, it should be limited to the peak value and the clipping effect should be controlled by expert observation. The following mixing scheme details the combining of speech and noise samples for each speaker. + +**Table 6.3.4: Speech vs. Noise samples mixing scheme** + +| | M1 | M2 | F1 | F2 | +|---------------------------------------|----|----|----|----| +| Speech sample 1 | N1 | N2 | N3 | N4 | +| Speech sample 2 | N2 | N3 | N4 | N5 | +| Speech sample 3 | N3 | N4 | N5 | N6 | +| Speech sample 4 | N4 | N5 | N6 | N1 | +| Speech sample 5 | N5 | N6 | N1 | N2 | +| Speech sample 6 | N6 | N1 | N2 | N3 | +| Speech sample 7
(practice) | N1 | N3 | N5 | N2 | + +### C.6.4 Listening Environment + +For all experiments, subjects should be seated in a quiet environment; 30dBA Hoth Spectrum (as defined by ITU-T Recommendation P.800 [8] , Annex A, clause A.1.1.2.2.1 Room Noise, with table A.1 and Figure A.1) measured at the head position of the subject. This will help ensure consistency between the different subjects in the same laboratory as well as across the different laboratories in which these experiments will be performed. + +The following points should be adhered to: + +Where the experiment design and the listening environment allows for multiple subjects in each listening session, the requirements stated above apply to each of the positions the subjects will occupy. + +Where there are multiple simultaneous subjects, they should not be able to see the responses made by other subjects. + +All test stimuli will be presented to the subjects over a telephone handset with Modified IRS receiving response (exclusive of the SRAEN filter). Any deviation shall be reported, e.g. use of one ear-piece in a headphone. + +Subjects should be told not to discuss the experiment with subjects who are yet to participate. + +Any test house performing multiple experiments must use different listening subjects for each experiment or sub-experiment. + +### C.6.5 Experimental Procedure + +Initially the experimenter should present and explain the experiment instructions to the subjects. When the subject has understood the instructions, they will first listen and give score to the preliminary conditions. After the preliminaries have been completed, there should be sufficient time allowed for answering possible questions from the subjects. Any questions about the procedure or the meaning of the instructions should be answered, but any technical questions on matters such as the experimental methodology or details of the types of distortions they are listening to must not be answered until they have completed the experiment. + +### C.6.6 Preliminary Conditions + +Preliminary conditions are included in the experiment to help acclimatize the subjects with the experimental procedure and to help reduce learning effects of the subjects, by ensuring that the subjects hear a full range of the potential qualities at the start of the experiment. No suggestions should be made to the subjects that the preliminary samples include the best or worst in the range to be covered, or exhaust the range of conditions they can expect to hear. + +### C.6.7 Reference Conditions + +Four types of reference conditions are used in these experiments: + +**AMR without NS References:** These are to be used to determine how the AMR Noise Suppressor performs in relation to these. + +**Direct unprocessed speech plus noise source material.** + +**MNRU references:** These are included as standard references of known and well understood performance and will allow the results to be expressed in terms of Equivalent Q as well as MOS for the ACR tests. MNRUs are also included in the CCR tests as references to estimate the test sensitivity and explore most of the CMOS range. For the CCR experiments, relative MNRU comparisons are used to estimate the test sensitivity. For example an MNRU of 12 may be compared to an MNRU of 16. If this difference is just above the significance level, it represents the test sensitivity. + +**Ideal noise suppression levels:** These are represented by varying the SNR level between the speech and noise. These conditions are AMR processed. This is an attempt to define equivalent noise suppression levels. + +For the Tests involving background noise conditions, the MNRU references will use noisy speech (i.e. background noise will be used with the MNRU). The exact number of each of these types of reference in each experiment can be found in the experiment plans in the clauses 7-10. + +### C.6.8 Noise Material + +Most of the Noise Suppressor Selection Test Experiments require the addition of noise to the speech material. The following types of noise are identified in this test plan: + +**Car Noise:** This represents stationary (static) background noise and will be typical of the noise experienced when inside a moving vehicle (car) at a constant speed. + +**Street Noise:** This represents non-stationary (dynamic) noise and will be typical of noise which might be experienced by someone using a mobile on a city street. + +**Babble Noise:** This represents non-stationary (dynamic) noise and will be typical of the background noise encountered in public places: restaurant, cafeteria, open offices. + +Noise files available free of charge from ARCON solely for the purposes of SMG11/S4 work shall be used. Contact ETSI (Paolo Usai) for further information (Paolo.Usai@ETSI.FR). + +## --- C.7 Experiment 1: Degradation in Clean Speech (Pair Comparison Test) + +### C.7.1 Introduction + +This PC (Paired-Comparison) experiment was prepared to test the '**No degradation in clean speech**' requirement in the Recommended Minimum Performance Requirements specification ([1], TS 3GPP TS 06.77 [4]), i.e.. This PC experiment will be run for the whole set of bit rates of the base vocoder, in single and tandem connection. + +The test methodology is direct, paired, forced choice comparison (i.e. A versus B test method with forced choice). The question that we are trying to answer with this test is not "What is the rank order of several coders?" but rather "Does the quality of coder with noise suppression (+NS) meet or exceed the quality of the coder without NS for a given condition?" The direct comparison A/B test methodology can answer this question by considering the proportion (or percent) of the measures where the candidate was preferred over the standard. Each individual judgement is a binary + +decision. A rank order approach could be taken as noted in the Handbook of Telephonometry [3] regarding Paired Comparisons but notes: "In the scaling modulus is included the common standard deviation, which is, however, unknown and so does not permit calculating confidence limits for the scale positions obtained." + +For the A/B experiment proposed here, with 24 subjects each making two independent measures (A/B and B/A) of the preference of the candidate coder over the standard coder for four talkers (two male and two female) each condition and with one repeat, the effective N is 384. In order to accommodate the repeat measure, single sentence samples will be used. This provides the additional benefit of directly adjacent A/B comparisons during presentation. The repeat measure will be made using a unique second sentence. + +### C.7.2 Test Factors and Conditions + +The PC test will be run for the following basic vocoder conditions: + +- Bit Rates of 4.75 kbit/s, 5.15 kbit/s, 5.9 kbit/s, 6.7 kbit/s, 7.4 kbit/s, 7.95 kbit/s, 10.2 kbit/s and 12.2 kbit/s. +- Single codec. + +This results in a single PC experiment with clean source speech and no channel impairments. The speech material used in these experiments are 4s samples (single sentence). + +The following table (Table 7.1) shows the testing factors to be used in this experiment. Due to the limited number of conditions tested within this experiment, it is possible to design a more balanced test structure and introduce some dummy conditions where the perceived difference in quality within the pairs of stimuli should be obvious for the subjects. A list of test conditions is given in Table 7.3. + +**Table 7.1: Factors and conditions for Experiment 1** + +| Main Codec Conditions | # | Notes | +|----------------------------|----------|------------------------------------------------------------| +| Noise Suppressor Candidate | 1 | | +| Codec | 1 | AMR | +| Codec Modes (FR/HR) | HR | All 8 AMR modes | +| | FR | | +| BERs | 0 | Clear channel, no transmission errors | +| Input level | 1 | nominal: -26dB relative to OVL | +| Acoustic Background Noise | 0 | None | +| Tandeming | 0 | No tandeming condition | +| Input Characteristic | 1 | GSM Filtered | +| Codec references | # | Notes | +| Test vocoders | 1 | AMR with NS | +| Reference vocoder | 8 | AMR at 12.2, 10.2, 7.95, 7.4, 6.7, 5.9, 5.15 & 4.75 | +| Other references | # | Notes | +| Direct | | Nominal level, GSM Filtered | +| MNRU | 2 | Q = 5 dB & 20 dB, other Q values in preliminaries | +| Ideal Noise Suppression | 0 | None | +| Common Conditions | # | Notes | +| GSM Channel | 0 | NO channel model | +| Number of talkers | 4 | 2 male + 2 female | +| Number of speech samples | 52 | 12/talker + 1 practice/talker | +| Sentences/sample | 1 | Single sentence stimuli | +| Listening Level | 1 | -15dBPa (79dB SPL) at ERP | +| Listeners | 24 | Naive Listeners | +| Randomizations | 6 | 6 groups of 4 listeners | +| Rating Scale | 1 | PC Instructions | +| Replications | 2 | Original Presentation + repeat w/ 2 nd sentence | + +### C.7.3 Preliminary Conditions + +The following 16 preliminary test conditions are recommended. + +**Table 7.2: List of preliminary conditions for Experiment 1** + +| Cond. | Presentation order | Reference Codec | Trans-codings | Processed Codec | Trans-codings | Talker and Sample Number | +|-------|--------------------|-----------------|---------------|-----------------|---------------|--------------------------| +| P1 | 5 | Direct | - | MNRU-20 | - | F1S13 | +| P2 | 1 | MNRU-18 | - | MNRU-22 | - | M1S13 | +| P3 | 3 | MNRU-19 | - | MNRU-21 | - | F2S13 | +| P4 | 7 | AMR-12.2 | 1 | AMR-12.2 | 1 | M2S13 | +| P5 | 6 | AMR-12.2 | 1 | AMR-5.9 | 1 | F1S13 | +| P6 | 2 | AMR-5.9 | 1 | AMR-5.9 | 1 | M1S13 | +| P7 | 4 | AMR-4.75 | 1 | AMR-7.95 | 1 | F2S13 | +| P8 | 8 | MNRU-5 | - | MNRU-20 | - | M2S13 | +| P9 | 14 | MNRU-20 | - | Direct | - | F1S13 | +| P10 | 10 | MNRU-22 | - | MNRU-18 | - | M1S13 | +| P11 | 12 | MNRU-21 | - | MNRU-19 | - | F2S13 | +| P12 | 16 | AMR-12.2 | 1 | AMR-12.2 | 1 | M2S13 | +| P13 | 13 | AMR-5.9 | 1 | AMR-12.2 | 1 | F1S13 | +| P14 | 9 | AMR-5.9 | 1 | AMR-5.9 | 1 | M1S13 | +| P15 | 11 | AMR-7.95 | 1 | AMR-4.75 | 1 | F2S13 | +| P16 | 15 | MNRU-20 | - | MNRU-5 | - | M2S13 | + +### C.7.4 Speech Material + +Single sentences. For the 4 talkers, 2 male and 2 female there are: + +- 13 stimuli / talker, each stimuli 4sec long w/ 1 sentence +- 12 unique sentences / talker for test plus one for practice + +To reduce the speech material effect, each talkers' samples must be unique. For this experiment, the unique samples are not balanced across all condition, candidates and subject groups. The same sample numbers for each talker are used for common conditions within a subject group and changed across subject groups. + +### C.7.5 Experimental Design + +The design is based on a restricted randomization philosophy using 6 different randomizations, each one covered by a group of 4 of the 24 subjects. This means that up to 4 subjects can perform the experiment simultaneously. + +Each subject will hear all of the conditions 16 times, four times with speech from each of the four talkers. Each of two stimuli for a talker will be presented in both the A/B and B/A order. Over the experiment as a whole, each of the conditions will be paired with twelve different samples from each of the four talkers. Each of the six groups of subjects will hear different combinations of source material and condition. + +### C.7.6 Processing + +Every condition has to be processed for each of the twelve stimuli of each of the four talkers. The actual samples used for each condition by each subject group are presented in Clause 7.12 Test Conditions. + +### C.7.7 Randomizations + +Separate randomizations for each of the six subject groups shall be provided to reduce order effects and to minimize differences between the laboratories. There shall be six randomizations for the experiment, one for each subject group. Each one will therefore be used by four of the 24 subjects. + +### C.7.8 Duration of the PC Experiment + +Each stimuli is 4 sec reference + 4 sec speech sample + 4 s voting time or 12 seconds. For this experiment there are 16 preliminary conditions x 12 seconds or 3.2 minutes for an introductory block. The presentation set for the experiment consists of 40 conditions (A/B+B/A) x 2 repeats x 4 talkers x 12 seconds or 64 minutes. The experiment is presented as the 16 preliminary conditions followed by the test itself divided in several sessions, i.e. 67,2 minutes testing time / + +subject group. The 6 groups of 4 subjects require 7 hours and 30 minutes total testing time for the experiment (6 x 1h 15 env.) + +To reduce the effects of subject fatigue, sessions should be separated by short comfort breaks. + +Note that the above calculations do not include the time needed to give the subjects their instructions, or for comfort breaks. + +### C.7.9 Votes Per Condition + +Every condition will have 24 subjects vote on four stimulus from each of four talkers, giving: + +$$(24 \text{ subjects} \times 4 \text{ talkers} \times 4 \text{ Presentations}) = 384 \text{ votes per condition}$$ + +From past experience of PC tests, this is the minimum number of votes per condition needed to give enough statistical certainty to differentiate the performance of one candidate process from another candidate process over the conditions and against the references. + +### C.7.10 Test Procedure + +Factors important for the experimental environment are specified in clauses 6.4, 6.5, and 6.6. As specified in clause 7.8, comfort breaks should be provided to reduce the effects of subject fatigue. + +### C.7.11 Opinion Scale + +The question asked of the subject is according to the Paired-Comparison binary scale. The specific wording is designed to evaluate the relative quality of the test sample in relation to the reference sample. In order to minimise presentation bias, the samples will be presented in both the A/B and B/A directions within the experiment. The subjects will listen to each pair of samples, and after presentation is completed, they will be asked to give their opinion. Annex A.1 contains an example of the instructions for the subjects in English. + +### C.7.12 Statistical Analysis + +The statistics to be reported for this pair-comparison experiment [4] are the proportion $P$ of subjects preferring the test stimulus over the reference stimulus (as defined in Table 2) for a total of $N$ votes per condition, the standard deviation $s$ : + +$$s = \sqrt{\frac{P \cdot (1-P)}{N}} \quad (\text{Eq.1})$$ + +and the upper and lower confidence limits, as calculated by: + +$$CI_{1-\alpha} = \frac{N}{N + z_{1-\alpha/2}^2} \cdot \left( P + \frac{z_{1-\alpha/2}^2}{2N} \pm z_{1-\alpha/2} \sqrt{\frac{P \cdot (1-P)}{N} + \frac{z_{1-\alpha/2}^2}{4N^2}} \right) \quad (\text{Eq.2})$$ + +where $z_{1-\alpha/2}$ is the standardized score for a normal distribution cutting off the lower $\alpha/2$ proportion of cases. + +Additionally, a hypothesis to test was whether the preference for the noise reduction-enabled AMR codec was statistically different from the ideal proportion $\pi=0.5$ , i.e. that the AMR with noise suppression is equally preferred to AMR without noise suppression (for quiet background). In other words, + +$$H_0 : \pi = 0.5$$ + +$$H_1 : \pi \neq 0.5$$ + +The null hypothesis $H_0$ is tested using a $z$ test where: + +$$z = \frac{P - \pi}{\sqrt{\frac{\pi(1-\pi)}{N}}} = \frac{P - 0.5}{0.5 / \sqrt{384}} = 2\sqrt{384} \cdot (P - 0.5) = 39.192 \cdot (P - 0.5) \quad (\text{Eq.3})$$ + +Hence, the null hypothesis is rejected if + +$$|z| \geq z_{1-\alpha/2}$$ + +Or accepted if: + +$$0.5 - \frac{z_{1-\alpha/2}}{39.19} < P < 0.5 + \frac{z_{1-\alpha/2}}{39.19} \quad (\text{Eq.4})$$ + +For a 95% confidence level, Equations 2 and 4 are reduced to ( $z_{1-\alpha/2} = 1.96$ , $N=384$ ): + +$$CI_{95\%} = \frac{N}{N+3.84} \left[ P + \frac{1.92}{N} \pm 1.96 \sqrt{\frac{P \cdot (1-P)}{N} + \frac{0.96}{N^2}} \right] \approx 0.99 \left[ P + 0.005 \pm 0.1 \sqrt{P(1-P)} \right] \quad (\text{Eq.5})$$ + +$$0.45 < P < 0.55 \quad (\text{Eq.6})$$ + +### C.7.13 Test Conditions for Experiment 1 + +**Table 7.3: Test conditions for Experiment 1** + +| Cond. | Reference Codec | Processed Codec | Trans-codings | Speech sample number (6 sequences) | +|---------|-----------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------|-----------------|---------------|------------------------------------| +| 1 | AMR@12.2 | AMR@12.2 | 1 | 2 3 4 5 6 1 | +| 2 | AMR@10.2 | AMR@10.2 | 1 | 3 4 5 6 1 2 | +| 3 | AMR@7.95 | AMR@7.95 | 1 | 1 2 3 4 5 6 | +| 4 | AMR@7.4 | AMR@7.4 | 1 | 4 5 6 1 2 3 | +| 5 | AMR@6.7 | AMR@6.7 | 1 | 5 6 1 2 3 4 | +| 6 | AMR@5.9 | AMR@5.9 | 1 | 6 1 2 3 4 5 | +| 7 | AMR@5.15 | AMR@5.15 | 1 | 2 3 4 5 6 1 | +| 8 | AMR@4.75 | AMR@4.75 | 1 | 3 4 5 6 1 2 | +| 9 | AMR@12.2 | AMR@5.9 | 1 | 1 2 3 4 5 6 | +| 10 | AMR@4.75 | AMR@7.95 | 1 | 4 5 6 1 2 3 | +| 11 | DIRECT | MNRU Q= 20 dB | 1 | 5 6 1 2 3 4 | +| 12 | MNRU Q= 5 dB | MNRU Q= 20 dB | 1 | 6 1 2 3 4 5 | +| 13 | AMR@12.2 | AMR/NS@12.2 | 1 | 2 3 4 5 6 1 | +| 14 | AMR@10.2 | AMR/NS@10.2 | 1 | 3 4 5 6 1 2 | +| 15 | AMR@7.95 | AMR/NS@7.95 | 1 | 1 2 3 4 5 6 | +| 16 | AMR@7.4 | AMR/NS@7.4 | 1 | 4 5 6 1 2 3 | +| 17 | AMR@6.7 | AMR/NS@6.7 | 1 | 5 6 1 2 3 4 | +| 18 | AMR@5.9 | AMR/NS@5.9 | 1 | 6 1 2 3 4 5 | +| 19 | AMR@5.15 | AMR/NS@5.15 | 1 | 1 2 3 4 5 6 | +| 20 | AMR@4.75 | AMR/NS@4.75 | 1 | 3 4 5 6 1 2 | +| 21 – 40 | Reversed order of the reference and processed speech samples in cond. 1-20 | | | | +| 41 – 60 | Repeat of conditions 1 – 20 with Speech Sample Number +6 | | | | +| 61 - 80 | Reversed order of the reference and processed speech samples in cond. 41 - 60 | | | | +| NOTES: | 4 talkers are used for all conditions: 2 male and 2 female
12 speech samples (4 s) are used for each talker
AMR@12.2 means AMR at 12.2 kbit/s
AMR/NS@12.2 means NS candidate x with AMR at 12.2 kbit/s | | | | + +## --- C.8 Experiments 2a, 2b & 2c: No degradation of Speech and no Undesirable Effects in Residual Noise in Conditions with Background Noise (ACR) + +### C.8.1 Introduction + +These ACR experiments are designed to test the requirement "No degradation of Speech and no Undesirable Effects in Residual Noise" in the Minimum Performance Requirements for Noise Suppressor Application to the AMR Speech Encoder, [1]. These ACR experiments will be run for three types of acoustic background noise. + +### C.8.2 Test Factors and Conditions + +The ACR test will be run for the following three types of acoustic background noise: + +- A car noise that is stationary both in level and in spectrum. +- A street noise that is non-stationary in level but fairly stationary in spectrum. +- A babble noise that is fairly stationary in level but non-stationary in spectrum. + +This results in a total of three ACR experiments with the different noise types in separate experiments. Within each experiment, a low, a medium and a high SNR level will be tested. The values for the low SNR are $SNR\_C = 6$ dB for the car noise, $SNR\_S = 9$ dB for the street noise, and $SNR\_B = 9$ dB for the babble noise. The higher SNR will be equal to $SNR + 6$ dB and $SNR + 12$ dB for all three noise types. The noise samples will have been recorded in scenarios representative of the respective low SNR value for each noise type (i.e. $SNR = 6$ or $9$ dB). + +All three experiments are run at AMR bit rate 12.2 kbit/s and 5.9 kbit/s. + +The following table shows the testing factors to be used in these experiments. A full list of test conditions is given in Clause 8.12. + +**Table 8.2.1: Factors and conditions for Experiments 2a, 2b, 2c** + +| Main Codec Conditions | # | Notes | +|------------------------------|----------|------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------| +| Noise Suppressor Algorithms | 1 | | +| Codec | 1 | AMR | +| Codec Modes | 2 | 12.2 kbps rate, 5.9 kbps rate | +| BERs | 0 | Clear channel, no transmission errors | +| Input level | 3 | nominal (high, low): -26dB (-16 dB, -36 dB) relative to OVL | +| Acoustic Background Noise | 3 | Static Car @ 6dB, 12dB, 18dB
Street @ 9dB, 15dB, 21dB
Babble @ 9dB, 15dB, 21dB | +| Input Characteristic | 1 | GSM Filtered | +| VAD/CNG/DTX | 2 | ON only at the nominal level, medium SNR values, zero value of Ideal NS
OFF for other conditions
One VAD/CNG/DTX will be used ; either VAD Option 1 or 2, depending on the implementers choice | +| Codec references | # | Notes | +| All Experiments | 1 | AMR wo/ NS | +| Other references | # | Notes | +| Direct | | Nominal level, GSM Filtered | +| MNRU, Exp 2a, 2b, 2c | 5 | Nominal level, with background noise, GSM Filtered, Q= 6, 12, 18, 24, 30dB | +| Ideal Noise Suppression | 6 | 3 levels for each SNR | +| Common Conditions | # | Notes | +| GSM Channel | 0 | NO channel model | +| Number of talkers | 4 | 2 male + 2 female | +| Number of speech samples | 28 | 6/ talker for the main test + 1/ talker for the Practice session | +| Listening Level | 1 | -15dBPa (79dB SPL) at ERP | +| Listeners | 24 | Naive Listeners | +| Randomizations | 6 | 6 groups of 4 listeners | +| Rating Scale | 1 | Modified ACR Instructions | +| Replications | 1 | Original Presentation Only | + +### C.8.3 Preliminary Conditions + +The following 16 preliminary test conditions are recommended. + +**Table 8.3.1: List of preliminary conditions** + +| Cond. | Presentation on order | SNR value | Ideal NS (dB) | Codec | Talker and Sample Number | +|--------------|------------------------------|------------------|----------------------|--------------|---------------------------------| +| P1 | 5 | SNR | - | Direct | M1S07 | +| P2 | 1 | SNR | - | MNRU-12 | M2S07 | +| P3 | 3 | SNR | - | AMR@12.2 | M1S07 | +| P4 | 7 | SNR | 7 | AMR@12.2 | M2S07 | +| P5 | 6 | SNR+6 | 7 | AMR@12.2 | F1S07 | +| P6 | 2 | SNR+12 | 7 | AMR@12.2 | F2S07 | +| P7 | 4 | SNR | - | AMR@5.9 | F1S07 | +| P8 | 8 | SNR+12 | - | AMR@5.9 | F2S07 | +| P9 | 14 | SNR | - | Direct | F1S07 | +| P10 | 10 | SNR | - | MNRU-12 | F2S07 | +| P11 | 12 | SNR | - | AMR@12.2 | F1S07 | +| P12 | 16 | SNR | 7 | AMR@12.2 | F2S07 | +| P13 | 13 | SNR+6 | 7 | AMR@12.2 | M1S07 | +| P14 | 9 | SNR+12 | 7 | AMR@12.2 | M2S07 | +| P15 | 11 | SNR | - | AMR@5.9 | M1S07 | +| P16 | 15 | SNR+12 | - | AMR@5.9 | M2S07 | + +### C.8.4 Speech Material + +The speech material should be as defined in Clause 6.4 - Long Sentence Quads, with each sample containing 4 sentences. For each test condition there are: + +- 6 samples / talker, each sample 16sec long w/ 4 sentences +- 24 unique sentences / talker + +For the practice conditions there are: + +- 1 sample / talker +- 4 unique sentences / talker + +To reduce any speech material effect, each talker sample must be unique. For these experiments, the unique samples are not balanced across all condition, candidates and subject groups. The same sample numbers for each talker are used for common conditions within a subject group and changed across subject groups. For a given language, the same speech material must be used for the three experiments 2a, 2b and 2c. + +Speech samples numbered from 01 to 06 should be used for the test conditions; speech samples numbered as 07 should be used for the Practice session. + +The noise material and its mix with the speech material should be as defined in Clause 6.10 and Clause 8.2. + +### C.8.5 Experimental Design + +The design is based on a restricted randomization philosophy using 6 different randomizations, each one covered by a group of 4 of the 24 subjects. This means that up to 4 subjects can perform the experiment simultaneously. + +Each subject will hear all of the conditions four times, once with speech from each of the four talkers. Over the experiment as a whole, each of the conditions will be paired with six different samples from each of the four talkers. Each of the six groups of subjects will hear different combinations of source material and condition. + +### C.8.6 Processing + +Every condition has to be processed for each of the six stimuli of each of the four primary talkers. The actual samples used for each condition by each subject group are presented in Clause 8.12 Test Conditions. + +### C.8.7 Randomizations + +Separate randomizations for each of the six subject groups shall be provided to reduce order effects and to minimize differences between the laboratories. There shall be six randomizations for the sub-experiments, one for each subject group. The same randomizations will be used for the three experiments (2a, 2b and 2c). Each one will therefore be used by four of the 24 subjects. Each randomization shall be balanced across 4 blocks of 36 stimuli to eliminate long sequences of similar conditions or identical talkers. The sequences shall provide for alternating male-female talkers. + +### C.8.8 Duration of the ACR Experiments 2a, 2b, and 2c + +Each stimuli is 16 s speech sample + 5 s voting time or 21 seconds. For each of the three experiments there are 16 preliminary conditions x 21 seconds or 5.6 minutes for an introductory block. The test consists of 36 conditions x 4 talkers x 21 seconds or 50.4 minutes, presented as three 16.8 minute blocks of 36 stimuli for 56 minutes testing time / subject group. The 6 groups of 4 subjects require 4 hours and 24 minutes total testing time + +To reduce the effects of subject fatigue, the three blocks should be separated by short comfort breaks. + +Note that the above calculations do not include the time needed to give the subjects their instructions, or for comfort breaks. + +### C.8.9 Votes Per Condition + +In each of the three experiments, every condition will have 24 subjects vote on one stimulus from each of four talkers, giving: + +(24 subjects x 4 talkers) = 96 votes per condition + +From past experience of ACR tests, this is the minimum number of votes per condition needed to give enough statistical certainty to differentiate the performance of one candidate process from another candidate process over the conditions and against the references. + +### C.8.10 Test Procedure + +Factors important for the experimental environment are specified in clause 6.5 and 6.6. As specified in clause 9.8, comfort breaks should be provided to reduce the effects of subject fatigue. + +### C.8.11 Opinion Scale + +The question asked of the subject is a modification of the ACR Listening Quality Scale. The specific wording is designed to evaluate both the level of distortion of the speech and the presence of artefacts in the residual background noise signal. The subjects will listen to each sample and after it has completed they will be asked to give their opinion. + +Annex A contains an example of the instructions for the subjects in English. The instructions in Annex A contain a modified version of the ACR instructions. They are aimed at focusing the subjects to rate artefacts introduced by the NS device. The test administrator should have the freedom to provide guidance to the subjects to reinforce this point, provided that such instructions are consistent across all 24 subjects. This is particularly important for tests not performed in English. Any additional instructions given to the subjects should be reported as an integral part of test reports. + +### C.8.12 Test Conditions for Experiments 2a, 2b and 2c + +| Cond. | Input level | SNR value | Ideal NS (dB) | VAD/DTX | Codec | Speech sample number (6 sequences) | +|-------|-------------|-----------|---------------|---------|-------------|------------------------------------| +| 1 | nominal | SNR | - | N/A | Direct | 4 5 6 1 2 3 | +| 2 | nominal | SNR | - | N/A | MNRU-30 | 4 5 6 1 2 3 | +| 3 | nominal | SNR | - | N/A | MNRU-24 | 4 5 6 1 2 3 | +| 4 | nominal | SNR | - | N/A | MNRU-18 | 4 5 6 1 2 3 | +| 5 | nominal | SNR | - | N/A | MNRU-12 | 4 5 6 1 2 3 | +| 6 | nominal | SNR | - | N/A | MNRU-6 | 4 5 6 1 2 3 | +| 7 | nominal | SNR | - | off | AMR@12.2 | 1 2 3 4 5 6 | +| 8 | nominal | SNR | 4 | off | AMR@12.2 | 1 2 3 4 5 6 | +| 9 | nominal | SNR | 7 | off | AMR@12.2 | 1 2 3 4 5 6 | +| 10 | nominal | SNR | - | off | AMR@5.9 | 1 2 3 4 5 6 | +| 11 | high | SNR | - | off | AMR@12.2 | 1 2 3 4 5 6 | +| 12 | high | SNR | - | off | AMR@5.9 | 1 2 3 4 5 6 | +| 13 | nominal | SNR+6 | - | off | AMR@12.2 | 2 3 4 5 6 1 | +| 14 | nominal | SNR+6 | 4 | off | AMR@12.2 | 2 3 4 5 6 1 | +| 15 | nominal | SNR+6 | 7 | off | AMR@12.2 | 2 3 4 5 6 1 | +| 16 | nominal | SNR+6 | - | off | AMR@5.9 | 2 3 4 5 6 1 | +| 17 | nominal | SNR+6 | - | on | AMR@12.2 | 2 3 4 5 6 1 | +| 18 | nominal | SNR+6 | - | on | AMR@5.9 | 2 3 4 5 6 1 | +| 19 | low | SNR+6 | - | off | AMR@12.2 | 2 3 4 5 6 1 | +| 20 | low | SNR+6 | - | off | AMR@5.9 | 2 3 4 5 6 1 | +| 21 | nominal | SNR+12 | - | off | AMR@12.2 | 3 4 5 6 1 2 | +| 22 | nominal | SNR+12 | 4 | off | AMR@12.2 | 3 4 5 6 1 2 | +| 23 | nominal | SNR+12 | 7 | off | AMR@12.2 | 3 4 5 6 1 2 | +| 24 | nominal | SNR+12 | - | off | AMR@5.9 | 3 4 5 6 1 2 | +| 25 | nominal | SNR | - | off | AMR/NS@12.2 | 1 2 3 4 5 6 | +| 26 | nominal | SNR | - | off | AMR/NS@5.9 | 1 2 3 4 5 6 | +| 27 | nominal | SNR+6 | - | off | AMR/NS@12.2 | 2 3 4 5 6 1 | +| 28 | nominal | SNR+6 | - | off | AMR/NS@5.9 | 2 3 4 5 6 1 | +| 29 | nominal | SNR+12 | - | off | AMR/NS@12.2 | 3 4 5 6 1 2 | +| 30 | nominal | SNR+12 | - | off | AMR/NS@5.9 | 3 4 5 6 1 2 | +| 31 | nominal | SNR+6 | - | on | AMR/NS@12.2 | 2 3 4 5 6 1 | +| 32 | nominal | SNR+6 | - | on | AMR/NS@5.9 | 2 3 4 5 6 1 | +| 33 | low | SNR+6 | - | off | AMR/NS@12.2 | 2 3 4 5 6 1 | +| 34 | low | SNR+6 | - | off | AMR/NS@5.9 | 2 3 4 5 6 1 | +| 35 | high | SNR | - | off | AMR/NS@12.2 | 1 2 3 4 5 6 | +| 36 | high | SNR | - | off | AMR/NS@5.9 | 1 2 3 4 5 6 | + +NOTE: Experiment 2a: Car noise with SNR = SNR\_C = 6 dB, +Experiment 2b: Street noise with SNR = SNR\_S = 9 dB +Experiment 2c: Babble noise with SNR = SNR\_B = 9 dB + +### C.8.13 Statistical Analysis + +The statistics to be reported from this ACR test are the averaged MOS ( $MOS_k$ ) scores and the standard deviations ( $S_k$ ) for all the conditions. + +Additionally, the requirement in [1, Clause 6.1.3] should be checked using a hypothesis test for the conditions 25-36 if the mean MOS score is greater or equal to the MOS score for the corresponding equivalent (all being equal except NS activated) reference condition for AMR without NS within a 95 % confidence. + +The hypothesis test should be performed using a 2-tailed T-test. The NS algorithm has failed the requirement if, for any of test condition, + +$$t < -t_{N,0.05}$$ + +where + +$$t = \frac{\text{MOS}_{\text{test}} - \text{MOS}_{\text{ref}}}{\sqrt{\frac{S_{\text{test}}^2 + S_{\text{ref}}^2}{N}}}$$ + +and the subscripts *test* and *ref* denotes the test condition and corresponding reference condition, respectively, *N* is the number of votes, and $t_{N,0.05}$ is the inverse of the Student's t-distribution with *N* degrees of freedom and probability 0.05. + +## C.9 Experiments 3a & 3b: Performances in Background Noise Conditions (Mod-CCR) + +### C.9.1 Introduction + +These experiments are designed to test Requirements in the associated Clause in the Recommended Minimum Performance Requirements Specification ([1], TS 3GPP TS 06.77 [4]). Specifically, the AMR with noise suppression should, in a certain number of conditions, be preferred to the AMR without noise suppression in a background noise environment and should provide a reasonable level of SNR improvement. Experiment 3a examines the performance of the noise suppression with the half-rate codec, while Experiment 3b examines the noise suppression with the full rate codec. Both experiments will use the Modified Comparison Category Rating (Mod-CCR, Note 1) method with a seven-point rating scale. Listeners will judge the relative quality of samples processed through the codec with noise suppression, compared to those without the noise suppression applied (example instructions for listeners are given in Annex A.3). The samples will have background noise of various types and levels mixed into the source speech before processing through the codec. + +The factors for each of the four sub-experiments are presented in Table 9.1. + +**Table 9.1: Factors for Experiments 3a and 3b** + +| Factor | Expt 3a | Expt 3b | +|-------------|-------------------------------------------------------------------|-------------------------------------------------------------------| +| codec | AMR 5.9 kb/s | AMR 12.2 kb/s | +| noise types | car (6 and 15 dB)
street (9 and 18 dB)
babble (9 and 18 dB) | car (6 and 15 dB)
street (9 and 18 dB)
babble (9 and 18 dB) | + +**NOTE:** The standard Comparison Category Rating method (CCR) which is described in Annex E of Rec. P.800 [8] is similar to the Degradation Category Rating method (DCR, Annex D). In Annex E, it is explicitly said : "Listeners are presented with a pair of speech samples on each trial. In the DCR procedure, a reference (unprocessed) sample is presented **first**, followed by the same speech sample, which has been processed by some technique. In the CCR procedure, the order of the processed and unprocessed samples is chosen at random for each trial. Listeners use the seven-point CCR scale to judge the quality of the second sample relative to that of the first. **The DCR and the CCR methods are particularly useful for assessing the performance of telecommunications systems when the input has been corrupted by background noise. However, an advantage of the CCR method over the DCR procedure is the possibility to assess speech processing that either degrades or improves the quality of the speech.** + +Here we are using a different application of the standard CCR method. The modified CCR method uses processed reference samples (but without noise suppression applied) whereas the standard CCR method uses unprocessed reference samples. + +### C.9.2 Test Factors and Conditions + +Three types of background noise will be used, at two different SNRs: + +A car noise that is stationary both in level and in spectrum. + +- A street noise that is non-stationary in level, but fairly stationary in spectrum. + +- A babble noise that is fairly stationary in level, but non-stationary in spectrum. + +The noise samples will be those utilised during the AMR Noise Suppression Selection Phase. + +The codec is held constant for each experiment, with two SNR classes ('SNR' and 'SNR+9dB') per experiment. All of the noise types are used in each experiment. The noise samples will have been recorded in scenarios representative of the respective SNR value for each noise. + +The factors and conditions to be used in Experiments 3a and 3b are presented in Table 9.2. The expanded set of test conditions is given in Clause 9.12. + +**Table 9.2: Factors and conditions for Experiments 3a and 3b** + +| Main Codec Conditions | # | Notes | +|------------------------------------|----------|-----------------------------------------------------------------------| +| Noise Suppresser Candidates | 1 | | +| Codec | 1 | AMR | +| Codec Modes (HR/FR) | HR
FR | 5.9 kbit/s rate for Experiment 3a
12.2 kbps rate for Experiment 3b | +| BERs | 0 | Clear channel, no transmission errors | +| Input level | 1 | nominal: -26dB relative to OVL | +| Acoustic Background Noise | 3 | car, street, and babble noise | +| Background noise SNRs | 2 | low, high for each (see Table 9.1) | +| Input Characteristic | 1 | GSM transmit filtered | +| Codec references | # | Notes | +| All Experiments | 1 | the same AMR rate w/o NS | +| Other references | # | Notes | +| Direct | | nominal level, GSM transmit filtered | +| MNRU, Exp 3a and 3b | | nominal level, GSM transmit filtered, Q= 12, $\Delta Q= 4$ | +| Ideal noise suppression simulation | | | +| Common Conditions | # | Notes | +| GSM Channel | 0 | NO channel model | +| Number of talkers | 4 | 2 male + 2 female primary talkers | +| Number of speech samples | 28 | 7 Sentence-pairs/primary talker (6 for Test, 1 for Practice) | +| Listening Level | 1 | -15dBPa (79dB SPL) at ERP | +| Listeners | 24 | Naive Listeners | +| Randomizations | 6 | 6 groups of 4 listeners | +| Rating Scale | 1 | CCR Instructions | +| Replications | 1 | Original Presentation Only | + +### C.9.3 Preliminary Conditions + +The following 16 preliminary test conditions are recommended, for presentation, before proceeding to the test samples. The samples shall be presented in the random order given in Table 9.3. + +**Table 9.3: List of preliminary conditions** + +| Cond. | Presentation order | Noise | SNR (dB) | Reference | Processed | | Speech Sample Number | +|-------|--------------------|--------|----------|-----------|-----------|---------|----------------------| +| | | | | | Ideal NS | Codec | | +| P1 | 9 | Car | 6 | Direct | - | Direct | M1S07 | +| P2 | 5 | Car | 15 | AMR@x | - | AMR@x | F1S07 | +| P3 | 12 | Car | 6 | MNRU-12 | - | MNRU-16 | M2S07 | +| P4 | 13 | Car | 15 | MNRU-12 | - | Direct | F2S07 | +| P5 | 2 | Street | 9 | AMR@x | - | AMR@x | M1S07 | +| P6 | 4 | Street | 18 | MNRU-12 | - | MNRU-16 | F1S07 | +| P7 | 8 | Street | 18 | MNRU-12 | - | Direct | M2S07 | +| P8 | 16 | Babble | 9 | AMR@x | - | AMR@x | F2S07 | +| P9 | 7 | Babble | 9 | MNRU-12 | - | MNRU-16 | M1S07 | +| P10 | 1 | Babble | 18 | MNRU-12 | - | Direct | F1S07 | +| P11 | 11 | Car | 6 | AMR@x | 4 | AMR@x | M2S07 | +| P12 | 3 | Car | 15 | AMR@x | 10 | AMR@x | F2S07 | +| P13 | 15 | Street | 18 | AMR@x | 4 | AMR@x | M1S07 | +| P14 | 6 | Street | 9 | AMR@x | 10 | AMR@x | F1S07 | +| P15 | 10 | Babble | 9 | AMR@x | 4 | AMR@x | M2S07 | +| P16 | 14 | Babble | 18 | AMR@x | 10 | AMR@x | F2S07 | + +NOTE: The bit rate for the AMR processing for the preliminary samples shall be the same as that used for the test samples, 5.9 kbit/s for Experiment 3a, 12.2 kbit/s for Experiment 3b. + +### C.9.4 Speech Material + +The source speech material shall be as defined in Clause 6.3 and will consist of the material used during the AMR Noise Suppression Selection phase: Each sample consists of two sentences. Only primary talkers are needed. For the four talkers, the following source material should be prepared: + +Seven samples for each talker, six for the test samples and one for the preliminaries, + +Each sample to be eight seconds long, + +Unique sentences-pairs in each sample (i.e., no repeated across the talkers). + +To reduce any speech material effect, the samples for each talker must be unique. For these experiments, these unique stimuli are not balanced across all conditions, candidates and subject groups. The same sample numbers for each talker are used for common conditions within a subject group and changed across subject groups (these sample numbers are arbitrarily assigned to samples). For a given language, the same speech material must be used for the two experiments 3a and 3b. The noise material and its mix with the speech material should be as defined in Clause 6.8 and Clause 6.3.7 respectively. + +### C.9.5 Experimental Design + +The design is based on a restricted randomization philosophy using six different randomizations, each of which is used with a group of four of the 24 listeners. This means that up to four subjects can perform the experiment simultaneously. + +Each listener will hear all of the conditions four times, once with speech from each of the four talkers. Over the experiment as a whole, each of the conditions will be paired with six different samples from each of the four talkers. Each of the six groups of subjects will hear different combinations of source material and condition. + +### C.9.6 Processing + +Every condition is processed with each of the six samples of each of the four primary talkers. The actual samples to be used for each condition, within with each subject group, are presented in Clause 9.12, *Test Conditions*. + +### C.9.7 Randomizations + +The test shall be completed using the randomizations provided by the experimenter. There shall be six randomizations for the sub-experiments, one for each subject group. The same randomizations shall be used for the two experiments (3a and 3b). Each one will therefore be used by four of the 24 subjects. Each randomization is balanced across four blocks of 48 stimuli to eliminate long sequences of similar conditions or identical talkers. The sequences shall provide for alternating male-female talkers. Use of these randomizations will allow presentation order to be used as a factor in a global analysis, should that be necessary. The randomization shall be constrained to a randomized block design, which controls practice and fatigue effects that may occur over the course of a test session. + +### C.9.8 Duration of the CCR Experiments 3a and 3b + +Each trial consists of an eight-second reference sample + an eight-second test sample + five second voting time, totalling 21 seconds. For each of the four experiments there are 16 preliminary conditions x 21 seconds or 5.6 minutes for an introductory block. Each presentation set within an experiment consists of 52 conditions (A/B+B/A) x 4 talkers x 21 seconds or 70 minutes, presented as eight 8.75 minute blocks of 25 stimuli for 75.6 minutes testing time / subject group / experiment. The total testing time for each experiment will be 7 hours and 34 minutes, if four listeners are tested at one time. + +To reduce the effects of subject fatigue, each 8.75 minute block should be separated by short comfort breaks. + +Note that the above calculations do not include the time needed to give the subjects their instructions, or time taken for comfort breaks. + +### C.9.9 Votes Per Condition + +In each of the three experiments, 24 listeners rate every condition with four talkers in each of two presentation orders (A/B and B/A), giving: + +(24 subjects x 4 talkers x 2 presentations) = 192 votes per condition + +From past experience with CCR tests, this is the minimum number of votes per condition needed to give enough statistical certainty to differentiate the performance of one candidate process from another candidate process over the conditions and against the references. + +### C.9.10 Test Procedure + +Factors important for the experimental environment are specified in Clauses 6.4, 6.5, and 6.6. As specified in Clause 9.8, comfort breaks should be provided to reduce the effects of subject fatigue. + +### C.9.11 Opinion Scale + +The question asked of the subject is a based on of the CCR Listening Quality Comparison Scale. The listening subjects will judge the quality of the second sample with regard to quality of the first sample. The subjects will listen to each pair of samples and after these have been played, they will be asked to give their comparative opinion. Annex A contains an example of the instructions for the subjects in English. Changes to the instructions may be needed to specify the method of data collection being used (button-press, paper & pencil, etc.). + +### C.9.12 Test Conditions for Experiments 3a and 3b + +| Cond. | Noise | SNR (dB) | Reference | Processed | | Speech sample number | +|--------|----------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------|----------|-----------|-----------|-----------|----------------------| +| | | | | Ideal NS | Codec | | +| 1 | Car | 6 | AMR@x | - | AMR@x | 4 5 6 1 2 3 | +| 2 | Street | 9 | AMR@x | - | AMR@x | 4 5 6 1 2 3 | +| 3 | Babble | 9 | AMR@x | - | AMR@x | 4 5 6 1 2 3 | +| 4 | Car | 6 | MNRU-16 | - | MNRU-12 | 4 - - 1 - - | +| 5 | Car | 6 | Direct | - | MNRU-12 | 4 - - 1 - - | +| 4' | Street | 9 | MNRU-16 | - | MNRU-12 | - 5 - - 2 - | +| 5' | Street | 9 | Direct | - | MNRU-12 | - 5 - - 2 - | +| 4" | Babble | 9 | MNRU-16 | - | MNRU-12 | - - 6 - - 3 | +| 5" | Babble | 9 | Direct | - | MNRU-12 | - - 6 - - 3 | +| 6 | Car | 6 | AMR@x | 3 | AMR@x | 1 2 3 4 5 6 | +| 7 | Car | 6 | AMR@x | 6 | AMR@x | 1 2 3 4 5 6 | +| 8 | Car | 6 | AMR@x | 9 | AMR@x | 1 2 3 4 5 6 | +| 9 | Street | 9 | AMR@x | 3 | AMR@x | 2 3 4 5 6 1 | +| 10 | Street | 9 | AMR@x | 6 | AMR@x | 2 3 4 5 6 1 | +| 11 | Street | 9 | AMR@x | 9 | AMR@x | 2 3 4 5 6 1 | +| 12 | Babble | 9 | AMR@x | 3 | AMR@x | 3 4 5 6 1 2 | +| 13 | Babble | 9 | AMR@x | 6 | AMR@x | 3 4 5 6 1 2 | +| 14 | Babble | 9 | AMR@x | 9 | AMR@x | 3 4 5 6 1 2 | +| 15 | Car | 6 | AMR@x | - | AMR/NS1@x | 1 2 3 4 5 6 | +| 16 | Street | 9 | AMR@x | - | AMR/NS1@x | 2 3 4 5 6 1 | +| 17 | Babble | 9 | AMR@x | - | AMR/NS1@x | 3 4 5 6 1 2 | +| 18 | Car | 15 | AMR@x | 3 | AMR@x | 1 2 3 4 5 6 | +| 19 | Car | 15 | AMR@x | 6 | AMR@x | 1 2 3 4 5 6 | +| 20 | Street | 18 | AMR@x | 3 | AMR@x | 2 3 4 5 6 1 | +| 21 | Street | 18 | AMR@x | 6 | AMR@x | 2 3 4 5 6 1 | +| 22 | Babble | 18 | AMR@x | 3 | AMR@x | 3 4 5 6 1 2 | +| 23 | Babble | 18 | AMR@x | 6 | AMR@x | 3 4 5 6 1 2 | +| 24 | Car | 15 | AMR@x | - | AMR/NS1@x | 1 2 3 4 5 6 | +| 25 | Street | 18 | AMR@x | - | AMR/NS1@x | 2 3 4 5 6 1 | +| 26 | Babble | 18 | AMR@x | - | AMR/NS1@x | 3 4 5 6 1 2 | +| 27-52 | Reversed order of the reference and processed speech samples in cond. 1-26 | | | | | | +| NOTES: | AMR@x denotes AMR at bit rate x, AMR/NS1@x denotes the NS candidate at bit rate x;
5.9 kbit/s for Experiment 3a, 12.2 kbit/s for Experiment 3b
SNR(dB) denotes SNR for noise
4 talkers are used for all conditions: 2 male and 2 female
6 speech samples (8 s) are used for each talker
- 'multiple' conditions "4s" and "5s" (e.g. 4 and 4') are only presented to a subset of listeners (e.g. to the first and the fourth groups of randomisation), | | | | | | + +### C.9.13 Statistical Analysis + +The statistics to be reported from this CCR test are the averaged CMOS ( $CMOS_k$ ) scores and the standard deviations ( $S_k$ ) for all the conditions. + +Additionally, the requirement in [1, Clause 6.1.4] should be checked using hypothesis tests for the conditions 15-17 and 24-26 if the mean CMOS score is greater than zero (the NS performance is preferred) and greater or equal to zero (the NS performance is equivalent) within a 95 % confidence. + +The hypothesis test should be performed using a 1-tailed T-test. The NS algorithm has failed the requirement at level "preferred" for any of test condition if: + +$$t < t_{N,0.05}$$ + +where + +$$t = \frac{\text{CMOS}_{test}}{S_{test} / \sqrt{N}}$$ + +and the subscripts *test* denotes the test condition, $N$ is the number of votes, and $t_{N,0.05}$ is the inverse of the Student's t-distribution with $N$ degrees of freedom and probability 0.05. + +Similarly, the NS algorithm has failed the requirement at level "equal" if + +$$t < -t_{N,0.05}$$ + +## C.10 Experiments 4: Influence of Input Level, Voice Activity Detection and Discontinuous Transmission (CCR) + +### C.10.1 Introduction + +This experiment is designed to test Requirements in the associated Clause in the Recommended Minimum Performance Requirements Specification ([1], TS 3GPP TS 06.77[4]). Specifically, the AMR with noise suppression should, in a certain number of conditions, be preferred to the AMR without noise suppression in a background noise environment and should provide a reasonable level of SNR improvement. + +### C.10.2 Test Factors and Conditions + +Three types of background noise will be used, at two different SNRs: + +A car noise that is stationary both in level and in spectrum. + +- A street noise that is non-stationary in level, but fairly stationary in spectrum. +- A babble noise that is fairly stationary in level, but non-stationary in spectrum. + +The factors and conditions to be used in Experiment 4 are presented in Table 10.2. The expanded set of test conditions is given in Clause 10.12. + +**Table 10.2: Factors and conditions for Experiment 4** + +| Main Codec Conditions | # | Notes | +|------------------------------|----------|-----------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------| +| Noise Suppressor Candidates | 1 | NS algorithm under test | +| Codec | 1 | AMR | +| Codec Modes | 1 | 12.2 kbit/s rate | +| BERs | 0 | Clear channel, no transmission errors | +| Input level | 3 | Nominal: -26dBov; High-level (-16 dBov); Low-level (-36 dBov) | +| Acoustic Background Noise | 3 | Car noise at 6 dB SNR; Street and Babble noise at 9 dB SNR | +| Input Characteristic | 1 | GSM transmit filtered | +| VAD/CNG/DTX | 2 | ON for all noise/level combinations
OFF for all noise types but only at the nominal level
One VAD/CNG/DTX will be used ; either VAD Option 1 or 2, depending on the implementers choice | +| Codec references | # | Notes | +| | 1 | AMR 12.2 kbit/s rate without NS | +| Other references | # | Notes | +| Direct | 3 | Nominal level, GSM transmit filtered | +| MNRU | 6 | Nominal level, GSM transmit filtered, Q=30, 24, 21, 18, 12, 6 dB, compared against Q=18 dB | +| Common Conditions | # | Notes | +| GSM Channel | 0 | NO channel model | +| Number of talkers | 4 | 2 male + 2 female primary talkers | +| Number of speech samples | 28 | 7 Sentence-pairs/primary talker (6 for Test, 1 for Practice) | +| Listening Level | 1 | -15dBPa (79dB SPL) at ERP | +| Listeners | 24 | Naive Listeners | +| Randomizations | 6 | 6 groups of 4 listeners | +| Rating Scale | 1 | CCR Instructions | +| Replications | 1 | | + +### C.10.3 Preliminary Conditions + +The following 16 preliminary test conditions are recommended, for presentation, before proceeding to the test samples. The samples shall be presented in the random order given in Table 10.3. + +**Table 10.3: List of preliminary conditions [TO BE REVISED]** + +| Cond. | Presentation order | Noise | Input level | SNR (dB) | VAD/DTX | Reference | Processed | Speech Sample Number | +|--------------|---------------------------|--------------|--------------------|-----------------|----------------|------------------|------------------|-----------------------------| +| P1 | 9 | Car | nominal | | | | | M1S07 | +| P2 | 5 | Car | nominal | 6 | | Direct | MNRU-12 | F1S07 | +| P3 | 12 | Car | nominal | 6 | | MNRU-16 | MNRU-12 | M2S07 | +| P4 | 13 | Car | nominal | 15 | | | | F2S07 | +| P5 | 2 | Street | nominal | 9 | | Direct | MNRU-12 | M1S07 | +| P6 | 4 | Street | nominal | 9 | | MNRU-16 | MNRU-12 | F1S07 | +| P7 | 8 | Street | nominal | | | | | M2S07 | +| P8 | 16 | Babble | nominal | 9 | | Direct | MNRU-12 | F2S07 | +| P9 | 7 | Babble | nominal | 9 | | MNRU-16 | MNRU-12 | M1S07 | +| P10 | 1 | Babble | nominal | | | | | F1S07 | +| P11 | 11 | Car | nominal | 6 | off | AMR@12.2 | AMR@12.2 | M2S07 | +| P12 | 3 | Car | nominal | 6 | on | AMR@12.2 | AMR@12.2 | F2S07 | +| P13 | 15 | Street | nominal | 9 | off | AMR@12.2 | AMR@12.2 | M1S07 | +| P14 | 6 | Street | nominal | 9 | on | AMR@12.2 | AMR@12.2 | F1S07 | +| P15 | 10 | Babble | nominal | 9 | off | AMR@12.2 | AMR@12.2 | M2S07 | +| P16 | 14 | Babble | nominal | 9 | on | AMR@12.2 | AMR@12.2 | F2S07 | + +### C.10.4 Speech Material + +The source speech material shall be as defined in Clause 6.3 and will consist of the material used during the AMR Noise Suppression Selection phase: Each sample consists of two sentences. Only primary talkers are needed. For the four talkers, the following source material should be prepared: + +- Seven samples for each talker, six for the test samples and one for the preliminaries, +- Each sample to be eight seconds long, +- Unique sentences-pairs in each sample (i.e., no repeated across the talkers) + +To reduce any speech material effect, the samples for each talker must be unique. For these experiments, these unique stimuli are balanced across all conditions, candidates and subject groups. The noise material and its mix with the speech material should be as defined in Clause 6.8 and Clause 6.3.7 respectively. + +### C.10.5 Experimental Design + +The design is based on a restricted randomization philosophy using six different randomizations, each of which is used with a group of four of the 24 listeners. This means that up to four subjects can perform the experiment simultaneously. + +Each listener will hear all of the conditions four times, once with speech from each of the four talkers. Over the experiment as a whole, each of the conditions will be paired with six different samples from each of the four talkers. Each of the six groups of subjects will hear different combinations of source material and condition. + +### C.10.6 Processing + +Every condition is processed with each of the six samples of each of the four primary talkers. Every speech file will be processed through all test conditions. + +### C.10.7 Randomizations + +The test shall be completed using the randomizations provided by the experimenter. There shall be six randomizations for the sub-experiments, one for each group of four subjects. Each randomization shall be balanced across four blocks of 30 stimuli to eliminate long sequences of similar conditions or identical talkers. The sequences shall provide for alternating male-female talkers. Use of these randomizations will allow presentation order to be used as a factor in a global analysis, should that be necessary. The randomization shall be constrained to a randomized block design, which controls practice and fatigue effects that may occur over the course of a test session. + +### C.10.8 Duration of the Experiment + +Each trial consists of an eight-second reference sample + an eight-second test sample + five second voting time, totalling 21 seconds. For each of the four experiments there are 16 preliminary conditions x 21 seconds or 5.6 minutes for an introductory block. Each presentation set within an experiment consists of 60 conditions (A/B+B/A) x 4 talkers x 21 seconds or approximately 1h30min. The total testing time for each experiment will be 9 hours and 34 minutes, if four listeners are tested at one time. + +Note that the above calculations do not include the time needed to give the subjects their instructions, or time taken for comfort breaks. + +### C.10.9 Votes Per Condition + +In each of the three experiments, 24 listeners rate every condition with four talkers in each of two presentation orders (A/B and B/A), giving: + +(24 subjects x 4 talkers x 2 presentations) = 192 votes per condition + +From past experience with CCR tests, this is the minimum number of votes per condition needed to give enough statistical certainty to differentiate the performance of one candidate process from another candidate process over the conditions and against the references. + +### C.10.10 Test Procedure + +Factors important for the experimental environment are specified in Clauses 6.4, 6.5, and 6.6. Comfort breaks should be provided to reduce the effects of subject fatigue. + +### C.10.11 Opinion Scale + +The question asked of the subject is a based on of the CCR Listening Quality Comparison Scale. The listening subjects will judge the quality of the second sample with regard to quality of the first sample. The subjects will listen to each pair of samples and after these have been played, they will be asked to give their comparative opinion. Annex A contains an example of the instructions for the subjects in English. Changes to the instructions may be needed to specify the method of data collection being used (button-press, paper & pencil, etc.). + +### C.10.12 Test Conditions for Experiment 4 + +| Cond. | Noise | Input level | SNR (dB) | VAD/DTX | Reference | Processed | Speech sample | +|-------|-----------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------|-------------|----------|---------|-----------|-------------|---------------| +| | | | | | | Codec | number | +| 1 | Car | nominal | 6 | off | AMR@12.2 | AMR@12.2 | 4 5 6 1 2 3 | +| 2 | Street | nominal | 9 | off | AMR@12.2 | AMR@12.2 | 4 5 6 1 2 3 | +| 3 | Babble | nominal | 9 | off | AMR@12.2 | AMR@12.2 | 4 5 6 1 2 3 | +| 4 | Car | nominal | 6 | on | AMR@12.2 | AMR@12.2 | 4 - - 1 - - | +| 5 | Car | nominal | 6 | N/A | Direct | MNRU-12 | 4 - - 1 - - | +| 6 | Car | nominal | 6 | N/A | MNRU-16 | MNRU-12 | 4 - - 1 - - | +| 4' | Street | nominal | 9 | on | AMR@12.2 | AMR@12.2 | - 5 - - 2 - | +| 5' | Street | nominal | 9 | N/A | Direct | MNRU-12 | - 5 - - 2 - | +| 6' | Street | nominal | 9 | N/A | MNRU-16 | MNRU-12 | - 5 - - 2 - | +| 4" | Babble | nominal | 9 | on | AMR@12.2 | AMR@12.2 | - - 6 - - 3 | +| 5" | Babble | nominal | 9 | N/A | Direct | MNRU-12 | - - 6 - - 3 | +| 6" | Babble | nominal | 9 | N/A | MNRU-16 | MNRU-12 | - - 6 - - 3 | +| 7 | Car | nominal | 6 | on | AMR@12.2 | AMR/NS@12.2 | 1 2 3 4 5 6 | +| 8 | Street | nominal | 9 | on | AMR@12.2 | AMR/NS@12.2 | 2 3 4 5 6 1 | +| 9 | Babble | nominal | 9 | on | AMR@12.2 | AMR/NS@12.2 | 3 4 5 6 1 2 | +| 10 | Car | low | 6 | off | AMR@12.2 | AMR/NS@12.2 | 5 6 1 2 3 4 | +| 11 | Street | low | 9 | off | AMR@12.2 | AMR/NS@12.2 | 6 1 2 3 4 5 | +| 12 | Babble | low | 9 | off | AMR@12.2 | AMR/NS@12.2 | 1 2 3 4 5 6 | +| 13 | Car | high | 6 | off | AMR@12.2 | AMR/NS@12.2 | 2 3 4 5 6 1 | +| 14 | Street | high | 9 | off | AMR@12.2 | AMR/NS@12.2 | 3 4 5 6 1 2 | +| 15 | Babble | high | 9 | off | AMR@12.2 | AMR/NS@12.2 | 5 6 1 2 3 4 | +| 16 | Car | nominal | 15 | on | AMR@12.2 | AMR/NS@12.2 | 6 1 2 3 4 5 | +| 17 | Street | nominal | 18 | on | AMR@12.2 | AMR/NS@12.2 | 1 2 3 4 5 6 | +| 18 | Babble | nominal | 18 | on | AMR@12.2 | AMR/NS@12.2 | 2 3 4 5 6 1 | +| 19 | Car | low | 15 | off | AMR@12.2 | AMR/NS@12.2 | 3 4 5 6 1 2 | +| 20 | Street | low | 18 | off | AMR@12.2 | AMR/NS@12.2 | 5 6 1 2 3 4 | +| 21 | Babble | low | 18 | off | AMR@12.2 | AMR/NS@12.2 | 6 1 2 3 4 5 | +| 22 | Car | high | 15 | off | AMR@12.2 | AMR/NS@12.2 | 1 2 3 4 5 6 | +| 23 | Street | high | 18 | off | AMR@12.2 | AMR/NS@12.2 | 2 3 4 5 6 1 | +| 24 | Babble | high | 18 | off | AMR@12.2 | AMR/NS@12.2 | 3 4 5 6 1 2 | +| 25-48 | Reversed order of the reference and processed speech samples in cond. 1-24 | | | | | | | +| NOTE | 4 talkers are used for all conditions: 2 male and 2 female
6 speech samples (8 s) are used for each talker
- 'multiple' conditions "4s", "5s" and "6s" (e.g. 4, 4' and 4") are only presented to a subset of listeners (e.g. to the first and the fourth groups of randomisation) | | | | | | | + +### C.10.13 Statistical Analysis + +The statistics to be reported from this CCR test are the averaged CMOS ( $CMOS_k$ ) scores and the standard deviations ( $S_k$ ) for all the conditions. + +Additionally, the requirement in [1, Clause 6.1.4] should be checked using hypothesis tests for the conditions 7-24 if the mean CMOS score is greater than zero (the NS performance is preferred) and greater or equal to zero (the NS performance is equivalent) within a 95 % confidence. + +The hypothesis test should be performed using a 1-tailed T-test. The NS algorithm has failed the requirement at level "preferred" for any of test condition if + +$$t < t_{N,0.05}$$ + +where + +$$t = \frac{\text{CMOS}_{test}}{S_{test} / \sqrt{N}}$$ + +and the subscripts *test* denotes the test condition, *N* is the number of votes, and *tN,0.05* is the inverse of the Student's t-distribution with *N* degrees of freedom and probability 0.05. + +Similarly, the NS algorithm has failed the requirement at level "equal" if + +$$t < -t_{N,0.05}$$ + +## C.11 Instructions to subjects and data collection + +The instructions given to the subjects will to some extent depend on the method used to collect opinion data. In this clause, example instructions are given for Pair Comparison, Modified ACR and CCR experiments. To ensure consistency, the actual instructions given to the subjects should be as close as possible to these examples, adapted for the number of speech files and length of the actual experiment, data collection method, and translated into the native language. + +The instructions must be given prior to the commencement of the experiment, and the experimenter should ensure that the subject has understood them before starting the experiment. Questions asked by the subjects on procedural aspects of the experiment can, and should, be answered. However questions about the experiment design or what the experiment is investigating should not be answered until the subjects have completed the experiment. Subjects must be told not to give such information to subjects who are yet to participate in the experiment. + +Subjects' responses may be collected by any convenient method: e.g. pencil and paper, press buttons controlling lamps recorded by the operator, or automatic data-logging equipment. Whichever method is used, care must be taken that subjects should not be able to observe other subjects' responses, nor should they be able to see the record of their own responses made in a previous session. Apart from the inevitable memory effects, each response should be independent of every other. + +### C.11.1 Example Instructions for Experiment 1 + +In this test we are evaluating systems that might be used for a type of communications between separate places under a variety of conditions. You are going to hear a number of samples of speech reproduced in the earpieces of the handset. Each sample will consist of a sentence that was produced with two different communication systems. The first is identified as A and the second is identified as B. + +Please listen to both A and B and then decide which of the two you prefer. Preference is strictly your decision and the decision should be based on your opinion of the quality of the speech samples. Some of the A/B pairs will seem clearly different and your decisions will be effortless. Others may be more difficult. **ALWAYS MAKE A DECISION BETWEEN THE TWO. "I DON'T KNOW" and "I don't like either one" ARE NOT OPTIONS.** Make your decisions independently. You should always compare the two current sentences, and not use any other presentation. Nor, should you be rating whether you like one talker better than another. This is not a test of you in any way; it is an evaluation of the systems. There is no right or wrong. Do not discuss how you are making your ratings during breaks or stretching periods. + +For indicating your opinion you are requested to use the button box at your test station. . After listening to the two sentences, all lights on the box will flash. At that time, please press the appropriate single button that represents your opinion of the communication quality of the sample just + +heard. Use the leftmost button if you preferred the first sample (or A). Use the button on the far right if you preferred the second speech sample (or B). The corresponding light will be activated when a choice has been indicated. Once the button has been pushed, you will not be allowed to change your mind, so please respond carefully. + +After you have given your opinion there will be a short pause before the next sample begins. + +For practice, you will hear a series and provide an opinion on each; then there will be a break to make sure that everything is clear. An administrator will be in the room to answer questions. From then on you will have a break after each test block (approximately xx minutes). After the test block there will be a three -minute break during which you may leave the room. This series will continue for the duration of the test. + +It is imperative that you do a good job with each rating by giving a true opinion of the communication system samples. The ratings you make later in the day are as important as those made earlier in the day. Please stay alert and do your best each time you make a decision. + +Thank you for participating in this research. Feel free to ask any procedural questions at this time. + +### C.11.2 Example Modified ACR Instructions for Experiment 2 + +#### Instructions to the listeners + +In this experiment we evaluate systems that might be used for telecommunication service between separate places. + +You will hear speech samples reproduced in a telephone handset. Every sample consists of four short unconnected sentences in an environment with varying amounts of background noise. Your task is to indicate your opinion of the overall sound quality with respect to any unnatural sounds leading to unpleasant effects in the sample. Please make your judgement of the sample considering unnatural sound during the complete sample. + +Use the following 5-point scale: + +| | | +|-------------------|-----------------------------| +| Excellent: | no unpleasant effects | +| Good: | slightly unpleasant effects | +| Fair: | somewhat unpleasant effects | +| Poor: | very unpleasant effects | +| Bad: | severely unpleasant effects | + +After each stimuli there will be a short pause for you to give your opinion. As a practice, you will first hear several samples and give an opinion on each. Then we will check that everything is clear before we start the test. Don't hesitate to ask questions if you have any. The experiment is divided in four parts with breaks in between. The parts last approximately 15 minutes each. Please do not discuss your opinions with the other participants in the experiment. + +Thank you for your participation. + +### C.11.3 Example Instructions for Experiment 3 and 4 + +# INSTRUCTIONS TO SUBJECTS + +In this experiment we are evaluating systems that might be used for telecommunication services. + +You are going to hear through the handset pairs of speech samples, most of which have been recorded in different noisy environments (for example inside a car, in an office, or on the street). The first sample you will hear will be the reference sample. You will then hear the same sample again, but this time it will have passed through a telecommunications system. These samples will each consist of two short unconnected sentences. + +You should listen carefully to each pair of samples. When they have finished, please record your opinion of the second sample with regard to the first one using the following scale: + +Much better + +Better + +Slightly better + +About the same + +Slightly worse + +Worse + +Much worse + +For practice, you will first hear [n] sample pairs and give an opinion on each. There will then be a short break to make sure that everything is clear. + +From then on you will have a break approximately every [p] minutes. The test will last a total of approximately [q] minutes. + +Please do not discuss your opinions with other listeners participating in the experiment. + +## --- C.12 Processing Tables + +To be provided by the listening laboratory This shall be reported along with the results of the experiments. + +## --- C.13 Presentation Orders + +To be provided by the listening laboratory This shall be reported along with the results of the experiments. + +# Annex D (informative): Change history + +| Change history | | | | | | | | +|----------------|-------|-----------|------|-----|---------------------------------------------------------------------------|--------|--------| +| Date | TSG # | TSG Doc. | CR | Rev | Subject/Comment | Old | New | +| 03-2001 | 11 | SP-010102 | A001 | 4 | Addition of test plan and tidying | 8.0.0 | 8.1.0 | +| 03-2001 | 11 | SP-010102 | A002 | 1 | Update of C code for objective measures for NS algorithm characterization | 8.0.0 | 8.1.0 | +| 03-2001 | 11 | SP-010102 | A003 | 1 | Correction of Annex A | 8.0.0 | 8.1.0 | +| 03-2001 | 11 | | | | Release 4 version | | 4.0.0 | +| 06-2002 | 16 | | | | Release 5 version | 4.0.0 | 5.0.0 | +| 06-2003 | | | | | Correct spelling mistake in title | 5.0.0 | 5.0.1 | +| 12-2004 | 26 | | | | Release 6 version | 5.0.1 | 6.0.0 | +| 06-2007 | 36 | | | | Release 7 version | 6.0.0 | 7.0.0 | +| 12-2008 | 42 | | | | Release 8 version | 7.0.0 | 8.0.0 | +| 12-2009 | 46 | | | | Release 9 version | 8.0.0 | 9.0.0 | +| 03-2011 | 51 | | | | Release 10 version | 9.0.0 | 10.0.0 | +| 09-2012 | 57 | | | | Release 11 version | 10.0.0 | 11.0.0 | \ No newline at end of file diff --git a/marked/Rel-11/26_series/26090/187d05bf7ead21e1394b61320d8b3632_img.jpg b/marked/Rel-11/26_series/26090/187d05bf7ead21e1394b61320d8b3632_img.jpg new file mode 100644 index 0000000000000000000000000000000000000000..8d0561a284e2540baa60101f59dca2d44ed5035d --- /dev/null +++ b/marked/Rel-11/26_series/26090/187d05bf7ead21e1394b61320d8b3632_img.jpg @@ -0,0 +1,3 @@ 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+|-----------------------------------------------------------------------------|----| +| Foreword ..... | 5 | +| 1 Scope..... | 6 | +| 2 References..... | 6 | +| 3 Definitions, symbols and abbreviations..... | 6 | +| 3.1 Definitions..... | 6 | +| 3.2 Symbols..... | 8 | +| 3.3 Abbreviations ..... | 11 | +| 4 Outline description..... | 12 | +| 4.1 Functional description of audio parts..... | 12 | +| 4.2 Preparation of speech samples ..... | 13 | +| 4.2.1 PCM format conversion ..... | 13 | +| 4.3 Principles of the adaptive multi-rate speech encoder ..... | 13 | +| 4.4 Principles of the adaptive multi-rate speech decoder ..... | 15 | +| 4.5 Sequence and subjective importance of encoded parameters ..... | 16 | +| 5 Functional description of the encoder..... | 16 | +| 5.1 Pre-processing (all modes)..... | 16 | +| 5.2 Linear prediction analysis and quantization ..... | 16 | +| 5.2.1 Windowing and auto-correlation computation..... | 17 | +| 5.2.2 Levinson-Durbin algorithm (all modes)..... | 18 | +| 5.2.3 LP to LSP conversion (all modes)..... | 18 | +| 5.2.4 LSP to LP conversion (all modes)..... | 20 | +| 5.2.5 Quantization of the LSP coefficients ..... | 20 | +| 5.2.6 Interpolation of the LSPs..... | 22 | +| 5.2.7 Monitoring resonance in the LPC spectrum (all modes)..... | 23 | +| 5.3 Open-loop pitch analysis..... | 23 | +| 5.4 Impulse response computation (all modes)..... | 27 | +| 5.5 Target signal computation (all modes) ..... | 27 | +| 5.6 Adaptive codebook ..... | 27 | +| 5.6.1 Adaptive codebook search ..... | 27 | +| 5.6.2 Adaptive codebook gain control (all modes) ..... | 31 | +| 5.7 Algebraic codebook ..... | 31 | +| 5.7.1 Algebraic codebook structure..... | 31 | +| 5.7.2 Algebraic codebook search ..... | 34 | +| 5.8 Quantization of the adaptive and fixed codebook gains ..... | 37 | +| 5.8.1 Adaptive codebook gain limitation in quantization ..... | 37 | +| 5.8.2 Quantization of codebook gains..... | 37 | +| 5.8.3 Update past quantized adaptive codebook gain buffer (all modes) ..... | 40 | +| 5.9 Memory update (all modes)..... | 40 | +| 6 Functional description of the decoder..... | 40 | +| 6.1 Decoding and speech synthesis..... | 40 | +| 6.2 Post-processing ..... | 43 | +| 6.2.1 Adaptive post-filtering (all modes) ..... | 43 | +| 6.2.2 High-pass filtering and up-scaling (all modes) ..... | 44 | +| 7 Detailed bit allocation of the adaptive multi-rate codec ..... | 45 | +| 8 Homing sequences ..... | 49 | +| 8.1 Functional description..... | 49 | +| 8.2 Definitions..... | 50 | +| 8.3 Encoder homing ..... | 50 | +| 8.4 Decoder homing..... | 50 | +| 9 Bibliography ..... | 54 | +| Annex A (informative): Change history ..... | 55 | + +# --- Foreword + +This Technical Specification has been produced by the 3rd Generation Partnership Project (3GPP). + +The contents of the present document are subject to continuing work within the TSG and may change following formal TSG approval. Should the TSG modify the contents of the present document, it will be re-released by the TSG with an identifying change of release date and an increase in version number as follows: + +Version x.y.z + +where: + +- x the first digit: + - 1 presented to TSG for information; + - 2 presented to TSG for approval; + - 3 or greater indicates TSG approved document under change control. +- y the second digit is incremented for all changes of substance, i.e. technical enhancements, corrections, updates, etc. +- z the third digit is incremented when editorial only changes have been incorporated in the document. + +# --- 1 Scope + +The present document describes the detailed mapping from input blocks of 160 speech samples in 13-bit uniform PCM format to encoded blocks of 95, 103, 118, 134, 148, 159, 204, and 244 bits and from encoded blocks of 95, 103, 118, 134, 148, 159, 204, and 244 bits to output blocks of 160 reconstructed speech samples. The sampling rate is 8 000 samples/s leading to a bit rate for the encoded bit stream of 4.75, 5.15, 5.90, 6.70, 7.40, 7.95, 10.2 or 12.2 kbit/s. The coding scheme for the multi-rate coding modes is the so-called Algebraic Code Excited Linear Prediction Coder, hereafter referred to as ACELP. The multi-rate ACELP coder is referred to as MR-ACELP. + +In the case of discrepancy between the requirements described in the present document and the fixed point computational description (ANSI-C code) of these requirements contained in [4], the description in [4] will prevail. The ANSI-C code is not described in the present document, see [4] for a description of the ANSI-C code. + +The transcoding procedure specified in the present document is mandatory for systems using the AMR speech codec. + +# --- 2 References + +The following documents contain provisions which, through reference in this text, constitute provisions of the present document. + +- References are either specific (identified by date of publication, edition number, version number, etc.) or non-specific. + - For a specific reference, subsequent revisions do not apply. + - For a non-specific reference, the latest version applies. In the case of a reference to a 3GPP document (including a GSM document), a non-specific reference implicitly refers to the latest version of that document *in the same Release as the present document*. +- [1] GSM 03.50: " Digital cellular telecommunications system (Phase 2+); Transmission planning aspects of the speech service in the GSM Public Land Mobile Network (PLMN) system". +- [2] 3GPP TS 26.101 : "Frame Structure". +- [3] 3GPP TS 26.094: "AMR Speech Codec; Voice Activity Detector". +- [4] 3GPP TS 26.073: "Adaptive Multi-Rate (AMR); ANSI C source code". +- [5] 3GPP TS 26.074: "Adaptive Multi-Rate (AMR); Test sequences". +- [6] ITU-T Recommendation G.711 (1988): "Pulse code modulation (PCM) of voice frequencies". +- [7] ITU-T Recommendation G.726: "40, 32, 24, 16 kbit/s Adaptive Differential Pulse Code Modulation (ADPCM)". +- [8] ITU-T Recommendation G.712 + +# --- 3 Definitions, symbols and abbreviations + +## 3.1 Definitions + +For the purposes of the present document, the following terms and definitions apply: + +**adaptive codebook:** contains excitation vectors that are adapted for every subframe. The adaptive codebook is derived from the long-term filter state. The lag value can be viewed as an index into the adaptive codebook + +**adaptive postfilter:** this filter is applied to the output of the short-term synthesis filter to enhance the perceptual quality of the reconstructed speech. In the adaptive multi-rate codec, the adaptive postfilter is a cascade of two filters: a formant postfilter and a tilt compensation filter + +**algebraic codebook:** fixed codebook where algebraic code is used to populate the excitation vectors (innovation vectors). The excitation contains a small number of nonzero pulses with predefined interlaced sets of positions + +**anti-sparseness processing:** adaptive post-processing procedure applied to the fixed codebook vector in order to reduce perceptual artefacts from a sparse fixed codebook vector + +**closed-loop pitch analysis:** adaptive codebook search, i.e., a process of estimating the pitch (lag) value from the weighted input speech and the long term filter state. In the closed-loop search, the lag is searched using error minimization loop (analysis-by-synthesis). In the adaptive multi-rate codec, closed-loop pitch search is performed for every subframe + +**direct form coefficients:** One of the formats for storing the short term filter parameters. In the adaptive multi-rate codec, all filters which are used to modify speech samples use direct form coefficients. + +**fixed codebook:** The fixed codebook contains excitation vectors for speech synthesis filters. The contents of the codebook are non-adaptive (i.e., fixed). In the adaptive multi-rate codec, the fixed codebook is implemented using an algebraic codebook. + +**fractional lags:** A set of lag values having sub-sample resolution. In the adaptive multi-rate codec a sub-sample resolution of $1/6^{\text{th}}$ or $1/3^{\text{rd}}$ of a sample is used. + +**frame:** time interval equal to 20 ms (160 samples at an 8 kHz sampling rate) + +**integer lags:** set of lag values having whole sample resolution + +**interpolating filter:** FIR filter used to produce an estimate of subsample resolution samples, given an input sampled with integer sample resolution + +**inverse filter:** this filter removes the short term correlation from the speech signal. The filter models an inverse frequency response of the vocal tract + +**lag:** long term filter delay. This is typically the true pitch period, or its multiple or sub-multiple + +**Line Spectral Frequencies:** (see Line Spectral Pair) + +**Line Spectral Pair:** transformation of LPC parameters. Line Spectral Pairs are obtained by decomposing the inverse filter transfer function $A(z)$ to a set of two transfer functions, one having even symmetry and the other having odd symmetry. The Line Spectral Pairs (also called as Line Spectral Frequencies) are the roots of these polynomials on the z-unit circle + +**LP analysis window:** for each frame, the short term filter coefficients are computed using the high pass filtered speech samples within the analysis window. In the adaptive multi-rate codec, the length of the analysis window is always 240 samples. For each frame, two asymmetric windows are used to generate two sets of LP coefficient in the 12.2 kbit/s mode. For the other modes, only a single asymmetric window is used to generate a single set of LP coefficients. In the 12.2 kbit/s mode, no samples of the future frames are used (no lookahead). The other modes use a 5 ms lookahead + +**LP coefficients:** linear Prediction (LP) coefficients (also referred as Linear Predictive Coding (LPC) coefficients) is a generic descriptive term for the short term filter coefficients + +**mode:** when used alone, refers to the source codec mode, i.e., to one of the source codecs employed in the AMR codec + +**open-loop pitch search:** process of estimating the near optimal lag directly from the weighted speech input. This is done to simplify the pitch analysis and confine the closed-loop pitch search to a small number of lags around the open-loop estimated lags. In the adaptive multi-rate codec, an open-loop pitch search is performed in every other subframe + +**residual:** the output signal resulting from an inverse filtering operation + +**short term synthesis filter:** this filter introduces, into the excitation signal, short term correlation which models the impulse response of the vocal tract + +**perceptual weighting filter:** this filter is employed in the analysis-by-synthesis search of the codebooks. The filter exploits the noise masking properties of the formants (vocal tract resonances) by weighting the error less in regions near the formant frequencies and more in regions away from them + +**subframe:** time interval equal to 5 ms (40 samples at 8 kHz sampling rate) + +**vector quantization:** method of grouping several parameters into a vector and quantizing them simultaneously + +**zero input response:** output of a filter due to past inputs, i.e. due to the present state of the filter, given that an input of zeros is applied + +**zero state response:** output of a filter due to the present input, given that no past inputs have been applied, i.e., given that the state information in the filter is all zeroes + +## 3.2 Symbols + +For the purposes of the present document, the following symbols apply: + +| | | +|----------------------------------------------------------------|---------------------------------------------------------------------------------------| +| $A(z)$ | The inverse filter with unquantized coefficients | +| $\hat{A}(z)$ | The inverse filter with quantized coefficients | +| $H(z) = \frac{1}{\hat{A}(z)}$ | The speech synthesis filter with quantized coefficients | +| $a_i$ | The unquantized linear prediction parameters (direct form coefficients) | +| $\hat{a}_i$ | The quantified linear prediction parameters | +| $m$ | The order of the LP model | +| $\frac{1}{B(z)}$ | The long-term synthesis filter | +| $W(z)$ | The perceptual weighting filter (unquantized coefficients) | +| $\gamma_1, \gamma_2$ | The perceptual weighting factors | +| $F_E(z)$ | Adaptive pre-filter | +| $T$ | The integer pitch lag nearest to the closed-loop fractional pitch lag of the subframe | +| $\beta$ | The adaptive pre-filter coefficient (the quantified pitch gain) | +| $H_f(z) = \frac{\hat{A}(z / \gamma_n)}{\hat{A}(z / \gamma_d)}$ | The formant postfilter | +| $\gamma_n$ | Control coefficient for the amount of the formant post-filtering | +| $\gamma_d$ | Control coefficient for the amount of the formant post-filtering | +| $H_t(z)$ | Tilt compensation filter | +| $\gamma_t$ | Control coefficient for the amount of the tilt compensation filtering | +| $\mu = \gamma_t k_1'$ | A tilt factor, with $k_1'$ being the first reflection coefficient | +| $h_f(n)$ | The truncated impulse response of the formant postfilter | +| $L_h$ | The length of $h_f(n)$ | +| $r_h(i)$ | The auto-correlations of $h_f(n)$ | +| $\hat{A}(z/\gamma_n)$ | The inverse filter (numerator) part of the formant postfilter | +| $1/\hat{A}(z/\gamma_d)$ | The synthesis filter (denominator) part of the formant postfilter | +| $\hat{r}(n)$ | The residual signal of the inverse filter $\hat{A}(z/\gamma_n)$ | +| $h_t(n)$ | Impulse response of the tilt compensation filter | +| $\beta_{sc}(n)$ | The AGC-controlled gain scaling factor of the adaptive postfilter | + +| | | +|--------------------------------------------|--------------------------------------------------------------------------------------| +| $\alpha$ | The AGC factor of the adaptive postfilter | +| $H_{h1}(z)$ | Pre-processing high-pass filter | +| $w_I(n), w_{II}(n)$ | LP analysis windows | +| $L_1^{(I)}$ | Length of the first part of the LP analysis window $w_I(n)$ | +| $L_2^{(I)}$ | Length of the second part of the LP analysis window $w_I(n)$ | +| $L_1^{(II)}$ | Length of the first part of the LP analysis window $w_{II}(n)$ | +| $L_2^{(II)}$ | Length of the second part of the LP analysis window $w_{II}(n)$ | +| $r_{ac}(k)$ | The auto-correlations of the windowed speech $s'(n)$ | +| $w_{lag}(i)$ | Lag window for the auto-correlations (60 Hz bandwidth expansion) | +| $f_0$ | The bandwidth expansion in Hz | +| $f_s$ | The sampling frequency in Hz | +| $r'_{ac}(k)$ | The modified (bandwidth expanded) auto-correlations | +| $E_{LD}(i)$ | The prediction error in the $i$ th iteration of the Levinson algorithm | +| $k_i$ | The $i$ th reflection coefficient | +| $a_j^{(i)}$ | The $j$ th direct form coefficient in the $i$ th iteration of the Levinson algorithm | +| $F'_1(z)$ | Symmetric LSF polynomial | +| $F'_2(z)$ | Antisymmetric LSF polynomial | +| $F_1(z)$ | Polynomial $F'_1(z)$ with root $z = -1$ eliminated | +| $F_2(z)$ | Polynomial $F'_2(z)$ with root $z = 1$ eliminated | +| $q_i$ | The line spectral pairs (LSPs) in the cosine domain | +| $\mathbf{q}$ | An LSP vector in the cosine domain | +| $\hat{\mathbf{q}}_i^{(n)}$ | The quantified LSP vector at the $i$ th subframe of the frame $n$ | +| $\omega_i$ | The line spectral frequencies (LSFs) | +| $T_m(x)$ | A $m$ th order Chebyshev polynomial | +| $f_1(i), f_2(i)$ | The coefficients of the polynomials $F_1(z)$ and $F_2(z)$ | +| $f'_1(i), f'_2(i)$ | The coefficients of the polynomials $F'_1(z)$ and $F'_2(z)$ | +| $f(i)$ | The coefficients of either $F_1(z)$ or $F_2(z)$ | +| $C(x)$ | Sum polynomial of the Chebyshev polynomials | +| $x$ | Cosine of angular frequency $\omega$ | +| $\lambda_k$ | Recursion coefficients for the Chebyshev polynomial evaluation | +| $f_i$ | The line spectral frequencies (LSFs) in Hz | +| $\mathbf{f}^t = [f_1 f_2 \dots f_{10}]$ | The vector representation of the LSFs in Hz | +| $\mathbf{z}^{(1)}(n), \mathbf{z}^{(2)}(n)$ | The mean-removed LSF vectors at frame $n$ | +| $\mathbf{r}^{(1)}(n), \mathbf{r}^{(2)}(n)$ | The LSF prediction residual vectors at frame $n$ | +| $\mathbf{p}(n)$ | The predicted LSF vector at frame $n$ | + +| | | +|------------------------------------------------------------------|----------------------------------------------------------------------------------------------------------------| +| $\hat{\mathbf{r}}^{(2)}(n-1)$ | The quantified second residual vector at the past frame | +| $\hat{\mathbf{f}}^k$ | The quantified LSF vector at quantization index $k$ | +| $E_{LSP}$ | The LSP quantization error | +| $w_i, i=1, \dots, 10$ | LSP-quantization weighting factors | +| $d_i$ | The distance between the line spectral frequencies $f_{i+1}$ and $f_{i-1}$ | +| $h(n)$ | The impulse response of the weighted synthesis filter | +| $O_k$ | The correlation maximum of open-loop pitch analysis at delay $k$ | +| $O_{t_i}, i=1, \dots, 3$ | The correlation maxima at delays $t_i, i=1, \dots, 3$ | +| $(M_i, t_i), i=1, \dots, 3$ | The normalized correlation maxima $M_i$ and the corresponding delays $t_i, i=1, \dots, 3$ | +| $H(z)W(z) = \frac{A(z/\gamma_1)}{\hat{A}(z)\hat{A}(z/\gamma_2)}$ | The weighted synthesis filter | +| $A(z/\gamma_1)$ | The numerator of the perceptual weighting filter | +| $1/A(z/\gamma_2)$ | The denominator of the perceptual weighting filter | +| $T_1$ | The integer nearest to the fractional pitch lag of the previous (1 st or 3 rd ) subframe | +| $s'(n)$ | The windowed speech signal | +| $s_w(n)$ | The weighted speech signal | +| $\hat{s}(n)$ | Reconstructed speech signal | +| $\hat{s}'(n)$ | The gain-scaled post-filtered signal | +| $\hat{s}_f(n)$ | Post-filtered speech signal (before scaling) | +| $x(n)$ | The target signal for adaptive codebook search | +| $x_2(n) \quad \mathbf{x}_2^t$ | The target signal for algebraic codebook search | +| $res_{LP}(n)$ | The LP residual signal | +| $c(n)$ | The fixed codebook vector | +| $v(n)$ | The adaptive codebook vector | +| $y(n) = v(n) * h(n)$ | The filtered adaptive codebook vector | +| $y_k(n)$ | The past filtered excitation | +| $u(n)$ | The excitation signal | +| $\hat{u}(n)$ | The emphasized adaptive codebook vector | +| $\hat{u}'(n)$ | The gain-scaled emphasized excitation signal | +| $T_{op}$ | The best open-loop lag | +| $t_{min}$ | Minimum lag search value | +| $t_{max}$ | Maximum lag search value | +| $R(k)$ | Correlation term to be maximized in the adaptive codebook search | +| $b_{24}$ | The FIR filter for interpolating the normalized correlation term $R(k)$ | +| $R(k)_t$ | The interpolated value of $R(k)$ for the integer delay $k$ and fraction $t$ | + +| | | +|------------------------------------------|---------------------------------------------------------------------------------------------------------------------| +| $b_{60}$ | The FIR filter for interpolating the past excitation signal $u(n)$ to yield the adaptive codebook vector $v(n)$ | +| $A_k$ | Correlation term to be maximized in the algebraic codebook search at index $k$ | +| $C_k$ | The correlation in the numerator of $A_k$ at index $k$ | +| $E_{Dk}$ | The energy in the denominator of $A_k$ at index $k$ | +| $\mathbf{d} = \mathbf{H}^t \mathbf{x}_2$ | The correlation between the target signal $x_2(n)$ and the impulse response $h(n)$ , i.e., backward filtered target | +| $\mathbf{H}$ | The lower triangular Toeplitz convolution matrix with diagonal $h(0)$ and lower diagonals $h(1), \dots, h(39)$ | +| $\Phi = \mathbf{H}^t \mathbf{H}$ | The matrix of correlations of $h(n)$ | +| $d(n)$ | The elements of the vector $\mathbf{d}$ | +| $\phi(i, j)$ | The elements of the symmetric matrix $\Phi$ | +| $\mathbf{c}_k$ | The innovation vector | +| $C$ | The correlation in the numerator of $A_k$ | +| $m_i$ | The position of the $i$ th pulse | +| $g_i$ | The amplitude of the $i$ th pulse | +| $N_p$ | The number of pulses in the fixed codebook excitation | +| $E_D$ | The energy in the denominator of $A_k$ | +| $res_{LTP}(n)$ | The normalized long-term prediction residual | +| $b(n)$ | The signal used for presetting the signs in algebraic codebook search | +| $s_b(n)$ | The sign signal for the algebraic codebook search | +| $d'(n)$ | Sign extended backward filtered target | +| $\phi'(i, j)$ | The modified elements of the matrix $\Phi$ , including sign information | +| $\mathbf{z}^t, z(n)$ | The fixed codebook vector convolved with $h(n)$ | +| $E(n)$ | The mean-removed innovation energy (in dB) | +| $\bar{E}$ | The mean of the innovation energy | +| $\tilde{E}(n)$ | The predicted energy | +| $[b_1 \ b_2 \ b_3 \ b_4]$ | The MA prediction coefficients | +| $\hat{R}(k)$ | The quantified prediction error at subframe $k$ | +| $E_I$ | The mean innovation energy | +| $R(n)$ | The prediction error of the fixed-codebook gain quantization | +| $E_Q$ | The quantization error of the fixed-codebook gain quantization | +| $e(n)$ | The states of the synthesis filter $1/\hat{A}(z)$ | +| $e_w(n)$ | The perceptually weighted error of the analysis-by-synthesis search | +| $\eta$ | The gain scaling factor for the emphasized excitation | +| $g_c$ | The fixed-codebook gain | +| $g'_c$ | The predicted fixed-codebook gain | + +| | | +|----------------------------|-------------------------------------------------------------------------| +| $\hat{g}_c$ | The quantified fixed codebook gain | +| $g_p$ | The adaptive codebook gain | +| $\hat{g}_p$ | The quantified adaptive codebook gain | +| $\gamma_{gc} = g_c / g'_c$ | A correction factor between the gain $g_c$ and the estimated one $g'_c$ | +| $\hat{\gamma}_{gc}$ | The optimum value for $\gamma_{gc}$ | +| $\gamma_{sc}$ | Gain scaling factor | + +## 3.3 Abbreviations + +For the purposes of the present document, the following abbreviations apply. + +| | | +|-------|-----------------------------------------------| +| ACELP | Algebraic Code Excited Linear Prediction | +| AGC | Adaptive Gain Control | +| AMR | Adaptive Multi-Rate | +| CELP | Code Excited Linear Prediction | +| EFR | Enhanced Full Rate | +| FIR | Finite Impulse Response | +| ISPP | Interleaved Single-Pulse Permutation | +| LP | Linear Prediction | +| LPC | Linear Predictive Coding | +| LSF | Line Spectral Frequency | +| LSP | Line Spectral Pair | +| LTP | Long Term Predictor (or Long Term Prediction) | +| MA | Moving Average | + +# --- 4 Outline description + +The present document is structured as follows: + +Clause 4.1 contains a functional description of the audio parts including the A/D and D/A functions. Clause 4.2 describes the conversion between 13-bit uniform and 8-bit A-law or $\mu$ -law samples. Clauses 4.3 and 4.4 present a simplified description of the principles of the AMR codec encoding and decoding process respectively. In clause 4.5, the sequence and subjective importance of encoded parameters are given. + +Clause 5 presents the functional description of the AMR codec encoding, whereas clause 6 describes the decoding procedures. In clause 7, the detailed bit allocation of the AMR codec is tabulated. + +## 4.1 Functional description of audio parts + +The analogue-to-digital and digital-to-analogue conversion will in principle comprise the following elements: + +- 1) Analogue to uniform digital PCM + - microphone; + - input level adjustment device; + - input anti-aliasing filter; + - sample-hold device sampling at 8 kHz; + - analogue-to-uniform digital conversion to 13-bit representation. + +The uniform format shall be represented in two's complement. + +- 2) Uniform digital PCM to analogue + - conversion from 13-bit/8 kHz uniform PCM to analogue; + +- a hold device; +- reconstruction filter including $x/\sin(x)$ correction; +- output level adjustment device; +- earphone or loudspeaker. + +In the terminal equipment, the A/D function may be achieved either: + +- by direct conversion to 13-bit uniform PCM format; +- or by conversion to 8-bit A-law or $\mu$ -law compounded format, based on a standard A-law or $\mu$ -law codec/filter according to ITU-T Recommendations G.711 [6] and G.714, followed by the 8-bit to 13-bit conversion as specified in clause 4.2.1. + +For the D/A operation, the inverse operations take place. + +In the latter case it should be noted that the specifications in ITU-T G.714 (superseded by G.712) are concerned with PCM equipment located in the central parts of the network. When used in the terminal equipment, the present document does not on its own ensure sufficient out-of-band attenuation. The specification of out-of-band signals is defined in [1] in clause 2. + +## 4.2 Preparation of speech samples + +The encoder is fed with data comprising of samples with a resolution of 13 bits left justified in a 16-bit word. The three least significant bits are set to '0'. The decoder outputs data in the same format. Outside the speech codec further processing must be applied if the traffic data occurs in a different representation. + +### 4.2.1 PCM format conversion + +The conversion between 8-bit A-Law or $\mu$ -law compressed data and linear data with 13-bit resolution at the speech encoder input shall be as defined in ITU-T Rec. G.711 [6]. + +ITU-T Rec. G.711 [6] specifies the A-Law or $\mu$ -law to linear conversion and vice versa by providing table entries. Examples on how to perform the conversion by fixed-point arithmetic can be found in ITU-T Rec. G.726 [7]. Clause 4.2.1 of G.726 [7] describes A-Law or $\mu$ -law to linear expansion and clause 4.2.8 of G.726 [7] provides a solution for linear to A-Law or $\mu$ -law compression. + +## 4.3 Principles of the adaptive multi-rate speech encoder + +The AMR codec consists of eight source codecs with bit-rates of 12.2, 10.2, 7.95, 7.40, 6.70, 5.90, 5.15 and 4.75 kbit/s. + +The codec is based on the code-excited linear predictive (CELP) coding model. A 10th order linear prediction (LP), or short-term, synthesis filter is used which is given by: + +$$H(z) = \frac{1}{\hat{A}(z)} = \frac{1}{1 + \sum_{i=1}^m \hat{a}_i z^{-i}}, \quad (1)$$ + +where $\hat{a}_i, i = 1, \dots, m$ , are the (quantified) linear prediction (LP) parameters, and $m = 10$ is the predictor order. The long-term, or pitch, synthesis filter is given by: + +$$\frac{1}{B(z)} = \frac{1}{1 - g_p z^{-T}}, \quad (2)$$ + +where $T$ is the pitch delay and $g_p$ is the pitch gain. The pitch synthesis filter is implemented using the so-called adaptive codebook approach. + +The CELP speech synthesis model is shown in figure 2. In this model, the excitation signal at the input of the short-term LP synthesis filter is constructed by adding two excitation vectors from adaptive and fixed (innovative) codebooks. The speech is synthesized by feeding the two properly chosen vectors from these codebooks through the short-term synthesis filter. The optimum excitation sequence in a codebook is chosen using an analysis-by-synthesis search procedure in which the error between the original and synthesized speech is minimized according to a perceptually weighted distortion measure. + +The perceptual weighting filter used in the analysis-by-synthesis search technique is given by: + +$$W(z) = \frac{A(z/\gamma_1)}{A(z/\gamma_2)}, \quad (3)$$ + +where $A(z)$ is the unquantized LP filter and $0 < \gamma_2 < \gamma_1 \leq 1$ are the perceptual weighting factors. The values $\gamma_1 = 0.9$ (for the 12.2 and 10.2 kbit/s mode) or $\gamma_1 = 0.94$ (for all other modes) and $\gamma_2 = 0.6$ are used. The weighting filter uses the unquantized LP parameters. + +The coder operates on speech frames of 20 ms corresponding to 160 samples at the sampling frequency of 8 000 sample/s. At each 160 speech samples, the speech signal is analysed to extract the parameters of the CELP model (LP filter coefficients, adaptive and fixed codebooks' indices and gains). These parameters are encoded and transmitted. At the decoder, these parameters are decoded and speech is synthesized by filtering the reconstructed excitation signal through the LP synthesis filter. + +The signal flow at the encoder is shown in figure 3. LP analysis is performed twice per frame for the 12.2 kbit/s mode and once for the other modes. For the 12.2 kbit/s mode, the two sets of LP parameters are converted to line spectrum pairs (LSP) and jointly quantized using split matrix quantization (SMQ) with 38 bits. For the other modes, the single set of LP parameters is converted to line spectrum pairs (LSP) and vector quantized using split vector quantization (SVQ). The speech frame is divided into 4 subframes of 5 ms each (40 samples). The adaptive and fixed codebook parameters are transmitted every subframe. The quantized and unquantized LP parameters or their interpolated versions are used depending on the subframe. An open-loop pitch lag is estimated in every other subframe (except for the 5.15 and 4.75 kbit/s modes for which it is done once per frame) based on the perceptually weighted speech signal. + +Then the following operations are repeated for each subframe: + +The target signal $x(n)$ is computed by filtering the LP residual through the weighted synthesis filter $W(z)H(z)$ with the initial states of the filters having been updated by filtering the error between LP residual and excitation (this is equivalent to the common approach of subtracting the zero input response of the weighted synthesis filter from the weighted speech signal). + +The impulse response, $h(n)$ of the weighted synthesis filter is computed. + +Closed-loop pitch analysis is then performed (to find the pitch lag and gain), using the target $x(n)$ and impulse response $h(n)$ , by searching around the open-loop pitch lag. Fractional pitch with 1/6th or 1/3rd of a sample resolution (depending on the mode) is used. + +The target signal $x(n)$ is updated by removing the adaptive codebook contribution (filtered adaptive codevector), and this new target, $x_2(n)$ , is used in the fixed algebraic codebook search (to find the optimum innovation). + +The gains of the adaptive and fixed codebook are scalar quantified with 4 and 5 bits respectively or vector quantified with 6-7 bits (with moving average (MA) prediction applied to the fixed codebook gain). + +Finally, the filter memories are updated (using the determined excitation signal) for finding the target signal in the next subframe. + +The bit allocation of the AMR codec modes is shown in table 1. In each 20 ms speech frame, 95, 103, 118, 134, 148, 159, 204 or 244 bits are produced, corresponding to a bit-rate of 4.75, 5.15, 5.90, 6.70, 7.40, 7.95, 10.2 or + +12.2 kbit/s. More detailed bit allocation among the codec parameters is given in tables 9a-9h. Note that the most significant bits (MSB) are always sent first. + +**Table 1: Bit allocation of the AMR coding algorithm for 20 ms frame** + +| Mode | Parameter | 1 st subframe | 2 nd subframe | 3 rd subframe | 4 th subframe | total per frame | +|-----------------------------------|----------------|--------------------------|--------------------------|--------------------------|--------------------------|-----------------| +| 12.2 kbit/s
(GSM EFR) | 2 LSP sets | | | | | 38 | +| | Pitch delay | 9 | 6 | 9 | 6 | 30 | +| | Pitch gain | 4 | 4 | 4 | 4 | 16 | +| | Algebraic code | 35 | 35 | 35 | 35 | 140 | +| | Codebook gain | 5 | 5 | 5 | 5 | 20 | +| | Total | | | | | 244 | +| 10.2 kbit/s | LSP set | | | | | 26 | +| | Pitch delay | 8 | 5 | 8 | 5 | 26 | +| | Algebraic code | 31 | 31 | 31 | 31 | 124 | +| | Gains | 7 | 7 | 7 | 7 | 28 | +| | Total | | | | | 204 | +| 7.95 kbit/s | LSP sets | | | | | 27 | +| | Pitch delay | 8 | 6 | 8 | 6 | 28 | +| | Pitch gain | 4 | 4 | 4 | 4 | 16 | +| | Algebraic code | 17 | 17 | 17 | 17 | 68 | +| | Codebook gain | 5 | 5 | 5 | 5 | 20 | +| | Total | | | | | 159 | +| 7.40 kbit/s
(TDMA EFR) | LSP set | | | | | 26 | +| | Pitch delay | 8 | 5 | 8 | 5 | 26 | +| | Algebraic code | 17 | 17 | 17 | 17 | 68 | +| | Gains | 7 | 7 | 7 | 7 | 28 | +| | Total | | | | | 148 | +| 6.70 kbit/s
(PDC EFR) | LSP set | | | | | 26 | +| | Pitch delay | 8 | 4 | 8 | 4 | 24 | +| | Algebraic code | 14 | 14 | 14 | 14 | 56 | +| | Gains | 7 | 7 | 7 | 7 | 28 | +| | Total | | | | | 134 | +| 5.90 kbit/s | LSP set | | | | | 26 | +| | Pitch delay | 8 | 4 | 8 | 4 | 24 | +| | Algebraic code | 11 | 11 | 11 | 11 | 44 | +| | Gains | 6 | 6 | 6 | 6 | 24 | +| | Total | | | | | 118 | +| 5.15 kbit/s | LSP set | | | | | 23 | +| | Pitch delay | 8 | 4 | 4 | 4 | 20 | +| | Algebraic code | 9 | 9 | 9 | 9 | 36 | +| | Gains | 6 | 6 | 6 | 6 | 24 | +| | Total | | | | | 103 | +| 4.75 kbit/s | LSP set | | | | | 23 | +| | Pitch delay | 8 | 4 | 4 | 4 | 20 | +| | Algebraic code | 9 | 9 | 9 | 9 | 36 | +| | Gains | 8 | | 8 | | 16 | +| | Total | | | | | 95 | + +## 4.4 Principles of the adaptive multi-rate speech decoder + +The signal flow at the decoder is shown in figure 4. At the decoder, based on the chosen mode, the transmitted indices are extracted from the received bitstream. The indices are decoded to obtain the coder parameters at each transmission frame. These parameters are the LSP vectors, the fractional pitch lags, the innovative codevectors, and the pitch and innovative gains. The LSP vectors are converted to the LP filter coefficients and interpolated to obtain LP filters at each subframe. Then, at each 40-sample subframe: + +- the excitation is constructed by adding the adaptive and innovative codevectors scaled by their respective gains; +- the speech is reconstructed by filtering the excitation through the LP synthesis filter. + +Finally, the reconstructed speech signal is passed through an adaptive postfilter. + +## 4.5 Sequence and subjective importance of encoded parameters + +The encoder will produce the output information in a unique sequence and format, and the decoder must receive the same information in the same way. In table 9a-9h, the sequence of output bits and the bit allocation for each parameter is shown. + +The different parameters of the encoded speech and their individual bits have unequal importance with respect to subjective quality. The output and input frame formats for the AMR speech codec are given in [2], where a reordering of bits take place. + +# 5 Functional description of the encoder + +In this clause, the different functions of the encoder represented in figure 3 are described. + +## 5.1 Pre-processing (all modes) + +Two pre-processing functions are applied prior to the encoding process: high-pass filtering and signal down-scaling. + +Down-scaling consists of dividing the input by a factor of 2 to reduce the possibility of overflows in the fixed-point implementation. + +The high-pass filter serves as a precaution against undesired low frequency components. A filter with a cut off frequency of 80 Hz is used, and it is given by: + +$$H_{h1}(z) = \frac{0.927246093 - 1.8544941z^{-1} + 0.927246903z^{-2}}{1 - 1.906005859z^{-1} + 0.911376953z^{-2}} \quad (4)$$ + +Down-scaling and high-pass filtering are combined by dividing the coefficients at the numerator of $H_{h1}(z)$ by 2. + +## 5.2 Linear prediction analysis and quantization + +### 12.2 kbit/s mode + +Short-term prediction, or linear prediction (LP), analysis is performed twice per speech frame using the auto-correlation approach with 30 ms asymmetric windows. No lookahead is used in the auto-correlation computation. + +The auto-correlations of windowed speech are converted to the LP coefficients using the Levinson-Durbin algorithm. Then the LP coefficients are transformed to the Line Spectral Pair (LSP) domain for quantization and interpolation purposes. The interpolated quantified and unquantized filter coefficients are converted back to the LP filter coefficients (to construct the synthesis and weighting filters at each subframe). + +#### 10.2, 7.95, 7.40, 6.70, 5.90, 5.15, 4.75 kbit/s modes + +Short-term prediction, or linear prediction (LP), analysis is performed once per speech frame using the auto-correlation approach with 30 ms asymmetric windows. A lookahead of 40 samples (5 ms) is used in the auto-correlation computation. + +The auto-correlations of windowed speech are converted to the LP coefficients using the Levinson-Durbin algorithm. Then the LP coefficients are transformed to the Line Spectral Pair (LSP) domain for quantization and interpolation purposes. The interpolated quantified and unquantized filter coefficients are converted back to the LP filter coefficients (to construct the synthesis and weighting filters at each subframe). + +### 5.2.1 Windowing and auto-correlation computation + +#### 12.2 kbit/s mode + +LP analysis is performed twice per frame using two different asymmetric windows. The first window has its weight concentrated at the second subframe and it consists of two halves of Hamming windows with different sizes. The window is given by: + +$$w_I(n) = \begin{cases} 0.54 - 0.46 \cos\left(\frac{\pi n}{L_1^{(I)} - 1}\right), & n = 0, \dots, L_1^{(I)} - 1, \\ 0.54 + 0.46 \cos\left(\frac{\pi(n - L_1^{(I)})}{L_2^{(I)} - 1}\right), & n = L_1^{(I)}, \dots, L_1^{(I)} + L_2^{(I)} - 1. \end{cases} \quad (5)$$ + +The values $L_1^{(I)} = 160$ and $L_2^{(I)} = 80$ are used. The second window has its weight concentrated at the fourth subframe and it consists of two parts: the first part is half a Hamming window and the second part is a quarter of a cosine function cycle. The window is given by: + +$$w_{II}(n) = \begin{cases} 0.54 - 0.46 \cos\left(\frac{2\pi n}{2L_1^{(II)} - 1}\right), & n = 0, \dots, L_1^{(II)} - 1, \\ \cos\left(\frac{2\pi(n - L_1^{(II)})}{4L_2^{(II)} - 1}\right), & n = L_1^{(II)}, \dots, L_1^{(II)} + L_2^{(II)} - 1 \end{cases} \quad (6)$$ + +where the values $L_1^{(II)} = 232$ and $L_2^{(II)} = 8$ are used. + +Note that both LP analyses are performed on the same set of speech samples. The windows are applied to 80 samples from past speech frame in addition to the 160 samples of the present speech frame. No samples from future frames are used (no lookahead). A diagram of the two LP analysis windows is depicted below. + +![Figure 1: LP analysis windows. A graph showing two window functions, w_I(n) and w_II(n), plotted against time t. The horizontal axis is divided into 'frame n-1' (20 ms, 160 samples) and 'frame n' (40 samples, 5 ms sub frame). w_I(n) is a curve that starts at the beginning of frame n-1, rises to a peak in frame n, and then drops to zero at the end of frame n. w_II(n) is a curve that starts at the beginning of frame n-1, rises to a peak in frame n, and then drops to zero at the end of frame n. The curves overlap significantly in frame n.](aaba634667a4ad2369fe5478c594ff58_img.jpg) + +Figure 1: LP analysis windows. A graph showing two window functions, w\_I(n) and w\_II(n), plotted against time t. The horizontal axis is divided into 'frame n-1' (20 ms, 160 samples) and 'frame n' (40 samples, 5 ms sub frame). w\_I(n) is a curve that starts at the beginning of frame n-1, rises to a peak in frame n, and then drops to zero at the end of frame n. w\_II(n) is a curve that starts at the beginning of frame n-1, rises to a peak in frame n, and then drops to zero at the end of frame n. The curves overlap significantly in frame n. + +Figure 1: LP analysis windows + +The auto-correlations of the windowed speech $s'(n)$ , $n = 0, \dots, 239$ , are computed by: + +$$r_{ac}(k) = \sum_{n=k}^{239} s'(n)s'(n-k), \quad k = 0, \dots, 10, \quad (7)$$ + +and a 60 Hz bandwidth expansion is used by lag windowing the auto-correlations using the window: + +$$w_{lag}(i) = \exp \left[ -\frac{1}{2} \left( \frac{2\pi f_0 i}{f_s} \right)^2 \right], \quad i = 1, \dots, 10 \quad (8)$$ + +where $f_0 = 60$ Hz is the bandwidth expansion and $f_s = 8000$ Hz is the sampling frequency. Further, $r_{ac}(0)$ is multiplied by the white noise correction factor 1.0001 which is equivalent to adding a noise floor at -40 dB. + +#### 10.2, 7.95, 7.40, 6.70, 5.90, 5.15, 4.75 kbit/s modes + +LP analysis is performed once per frame using an asymmetric window. The window has its weight concentrated at the fourth subframe and it consists of two parts: the first part is half a Hamming window and the second part is a quarter of a cosine function cycle. The window is given by equation (6) where the values $L_1 = 200$ and $L_2 = 40$ are used. + +The auto-correlations of the windowed speech $s'(n)$ , $n = 0, \dots, 239$ , are computed by equation (7) and a 60 Hz bandwidth expansion is used by lag windowing the auto-correlations using the window of equation (8). Further, $r_{ac}(0)$ is multiplied by the white noise correction factor 1.0001 which is equivalent to adding a noise floor at -40 dB. + +### 5.2.2 Levinson-Durbin algorithm (all modes) + +The modified auto-correlations $r'_{ac}(0) = 1.0001 r_{ac}(0)$ and $r'_{ac}(k) = r_{ac}(k)w_{lag}(k)$ , $k = 1, \dots, 10$ , are used to obtain the direct form LP filter coefficients $a_k$ , $k = 1, \dots, 10$ , by solving the set of equations. + +$$\sum_{k=1}^{10} a_k r'_{ac}(|i-k|) = -r'_{ac}(i), \quad i = 1, \dots, 10. \quad (9)$$ + +The set of equations in (9) is solved using the Levinson-Durbin algorithm. This algorithm uses the following recursion: + +$$\begin{aligned} E_{LD}(0) &= r'_{ac}(0) \\ \text{for } i &= 1 \text{ to } 10 \text{ do} \\ & \quad a_0^{(i-1)} = 1 \\ & \quad k_i = -\left[ \sum_{j=0}^{i-1} a_j^{(i-1)} r'_{ac}(i-j) \right] / E_{LD}(i-1) \\ & \quad a_i^{(i)} = k_i \\ & \quad \text{for } j = 1 \text{ to } i-1 \text{ do} \\ & \quad \quad a_j^{(i)} = a_j^{(i-1)} + k_i a_{i-j}^{(i-1)} \\ & \quad \text{end} \\ & \quad E_{LD}(i) = (1 - k_i^2) E_{LD}(i-1) \\ \text{end} \end{aligned}$$ + +The final solution is given as $a_j = a_j^{(10)}$ , $j = 1, \dots, 10$ . + +The LP filter coefficients are converted to the line spectral pair (LSP) representation for quantization and interpolation purposes. The conversions to the LSP domain and back to the LP filter coefficient domain are described in the next clause. + +### 5.2.3 LP to LSP conversion (all modes) + +The LP filter coefficients $a_k, k = 1, \dots, 10$ , are converted to the line spectral pair (LSP) representation for quantization and interpolation purposes. For a 10th order LP filter, the LSPs are defined as the roots of the sum and difference polynomials: + +$$F_1'(z) = A(z) + z^{-11} A(z^{-1}) \quad (10)$$ + +and + +$$F_2'(z) = A(z) - z^{-11} A(z^{-1}), \quad (11)$$ + +respectively. The polynomial $F_1'(z)$ and $F_2'(z)$ are symmetric and anti-symmetric, respectively. It can be proven that all roots of these polynomials are on the unit circle and they alternate each other. $F_1'(z)$ has a root $z = -1$ ( $\omega = \pi$ ) and $F_2'(z)$ has a root $z = 1$ ( $\omega = 0$ ). To eliminate these two roots, we define the new polynomials: + +$$F_1(z) = F_1'(z) / (1 + z^{-1}) \quad (12)$$ + +and + +$$F_2(z) = F_2'(z) / (1 - z^{-1}) \quad (13)$$ + +Each polynomial has 5 conjugate roots on the unit circle $(e^{\pm j\omega_i})$ , therefore, the polynomials can be written as + +$$F_1(z) = \prod_{i=1,3,\dots,9} (1 - 2q_i z^{-1} + z^{-2}) \quad (14)$$ + +and + +$$F_2(z) = \prod_{i=2,4,\dots,10} (1 - 2q_i z^{-1} + z^{-2}), \quad (15)$$ + +where $q_i = \cos(\omega_i)$ with $\omega_i$ being the line spectral frequencies (LSF) and they satisfy the ordering property $0 < \omega_1 < \omega_2 < \dots < \omega_{10} < \pi$ . We refer to $q_i$ as the LSPs in the cosine domain. + +Since both polynomials $F_1(z)$ and $F_2(z)$ are symmetric only the first 5 coefficients of each polynomial need to be computed. The coefficients of these polynomials are found by the recursive relations (for $i = 0$ to 4): + +$$\begin{aligned} f_1(i+1) &= a_{i+1} + a_{m-i} - f_1(i) \\ f_2(i+1) &= a_{i+1} - a_{m-i} + f_2(i) \end{aligned} \quad (16)$$ + +where $m = 10$ is the predictor order. + +The LSPs are found by evaluating the polynomials $F_1(z)$ and $F_2(z)$ at 60 points equally spaced between 0 and $\pi$ and checking for sign changes. A sign change signifies the existence of a root and the sign change interval is then divided 4 times to better track the root. The Chebyshev polynomials are used to evaluate $F_1(z)$ and $F_2(z)$ . In this + +method the roots are found directly in the cosine domain $\{q_i\}$ . The polynomials $F_1(z)$ or $F_2(z)$ evaluated at $z = e^{j\omega}$ can be written as: + +$$F(\omega) = 2e^{-j5\omega} C(x),$$ + +with: + +$$C(x) = T_5(x) + f(1)T_4(x) + f(2)T_3(x) + f(3)T_2(x) + f(4)T_1(x) + f(5)/2, \quad (17)$$ + +where $T_m(x) = \cos(m\omega)$ is the $m$ th order Chebyshev polynomial, and $f(i), i = 1, \dots, 5$ are the coefficients of either $F_1(z)$ or $F_2(z)$ , computed using the equations in (16). The polynomial $C(x)$ is evaluated at a certain value of $x = \cos(\omega)$ using the recursive relation: + +for $k = 4$ down to 1 + +$$\lambda_k = 2x\lambda_{k+1} - \lambda_{k+2} + f(5 - k)$$ + +end + +$$C(x) = x\lambda_1 - \lambda_2 + f(5) / 2,$$ + +with initial values $\lambda_5 = 1$ and $\lambda_6 = 0$ . The details of the Chebyshev polynomial evaluation method are found in P. Kabal and R.P. Ramachandran [4]. + +### 5.2.4 LSP to LP conversion (all modes) + +Once the LSPs are quantified and interpolated, they are converted back to the LP coefficient domain $\{a_k\}$ . The conversion to the LP domain is done as follows. The coefficients of $F_1(z)$ or $F_2(z)$ are found by expanding equations (14) and (15) knowing the quantified and interpolated LSPs $q_i, i = 1, \dots, 10$ . The following recursive relation is used to compute $f_1(i)$ : + +for $i = 1$ to 5 + +$$f_1(i) = -2q_{2i-1}f_1(i-1) + 2f_1(i-2)$$ + +for $j = i - 1$ down to 1 + +$$f_1(j) = f_1(j) - 2q_{2i-1}f_1(j-1) + f_1(j-2)$$ + +end + +end + +with initial values $f_1(0)=1$ and $f_1(-1)=0$ . The coefficients $f_2(i)$ are computed similarly by replacing $q_{2i-1}$ by $q_{2i}$ . + +Once the coefficients $f_1(i)$ and $f_2(i)$ are found, $F_1(z)$ and $F_2(z)$ are multiplied by $1 + z^{-1}$ and $1 - z^{-1}$ , respectively, to obtain $F_1'(z)$ and $F_2'(z)$ ; that is: + +$$\begin{aligned} f_1'(i) &= f_1(i) + f_1(i-1), & i &= 1, \dots, 5 \\ f_2'(i) &= f_2(i) - f_2(i-1), & i &= 1, \dots, 5 \end{aligned} \quad (18)$$ + +Finally the LP coefficients are found by: + +$$a_i = \begin{cases} 0.5f_1'(i) + 0.5f_2'(i), & i = 1, \dots, 5 \\ 0.5f_1'(11-i) - 0.5f_2'(11-i), & i = 6, \dots, 10 \end{cases} \quad (19)$$ + +This is directly derived from the relation $A(z) = (F_1'(z) + F_2'(z))/2$ , and considering the fact that $F_1'(z)$ and $F_2'(z)$ are symmetric and anti-symmetric polynomials, respectively. + +### 5.2.5 Quantization of the LSP coefficients + +#### 12.2 kbit/s mode + +The two sets of LP filter coefficients per frame are quantified using the LSP representation in the frequency domain; that is: + +$$f_i = \frac{f_s}{2\pi} \arccos(q_i), \quad i=1, \dots, 10, \quad (20)$$ + +where $f_i$ are the line spectral frequencies (LSF) in Hz [0,4 000] and $f_s=8000$ is the sampling frequency. The LSF vector is given by $\mathbf{f}^t = [f_1 f_2 \dots f_{10}]$ , with $t$ denoting transpose. + +A 1st order MA prediction is applied, and the two residual LSF vectors are jointly quantified using split matrix quantization (SMQ). The prediction and quantization are performed as follows. Let $\mathbf{z}^{(1)}(n)$ and $\mathbf{z}^{(2)}(n)$ denote the mean-removed LSF vectors at frame $n$ . The prediction residual vectors $\mathbf{r}^{(1)}(n)$ and $\mathbf{r}^{(2)}(n)$ are given by: + +$$\begin{aligned} \mathbf{r}^{(1)}(n) &= \mathbf{z}^{(1)}(n) - \mathbf{p}(n), \quad \text{and} \\ \mathbf{r}^{(2)}(n) &= \mathbf{z}^{(2)}(n) - \mathbf{p}(n), \end{aligned} \quad (21)$$ + +where $\mathbf{p}(n)$ is the predicted LSF vector at frame $n$ . First order moving-average (MA) prediction is used where: + +$$\mathbf{p}(n) = 0.65\hat{\mathbf{r}}^{(2)}(n-1), \quad (22)$$ + +where $\hat{\mathbf{r}}^{(2)}(n-1)$ is the quantified second residual vector at the past frame. + +The two LSF residual vectors $\mathbf{r}^{(1)}$ and $\mathbf{r}^{(2)}$ are jointly quantified using split matrix quantization (SMQ). The matrix $\begin{pmatrix} \mathbf{r}^{(1)} & \mathbf{r}^{(2)} \end{pmatrix}$ is split into 5 submatrices of dimension 2 x 2 (two elements from each vector). For example, the first submatrix consists of the elements $r_1^{(1)}$ , $r_2^{(1)}$ , $r_1^{(2)}$ , and $r_2^{(2)}$ . The 5 submatrices are quantified with 7, 8, 8+1, 8, and 6 bits, respectively. The third submatrix uses a 256-entry signed codebook (8-bit index plus 1-bit sign). + +A weighted LSP distortion measure is used in the quantization process. In general, for an input LSP vector $\mathbf{f}$ and a quantified vector at index $k$ , $\hat{\mathbf{f}}^k$ , the quantization is performed by finding the index $k$ which minimizes: + +$$E_{LSP} = \sum_{i=1}^{10} \left[ f_i w_i - \hat{f}_i^k w_i \right]^2. \quad (23)$$ + +The weighting factors $w_i, i=1, \dots, 10$ , are given by + +$$\begin{aligned} +w_i &= 3.347 - \frac{1.547}{450} d_i \quad \text{for } d_i < 450, \\ +&= 1.8 - \frac{0.8}{1050} (d_i - 450) \quad \text{otherwise,} +\end{aligned} \tag{24}$$ + +where $d_i = f_{i+1} - f_{i-1}$ with $f_0 = 0$ and $f_{11} = 4000$ . Here, two sets of weighting coefficients are computed for the two LSF vectors. In the quantization of each submatrix, two weighting coefficients from each set are used with their corresponding LSFs. + +#### 10.2, 7.95, 7.40, 6.70, 5.90, 5.15, 4.75 kbit/s modes + +The set of LP filter coefficients per frame is quantified using the LSP representation in the frequency domain using equation (20). + +A 1st order MA prediction is applied, and the residual LSF vector is quantified using split vector quantization. The prediction and quantization are performed as follows. Let $\mathbf{z}(n)$ denote the mean-removed LSF vectors at frame $n$ . The prediction residual vectors $\mathbf{r}(n)$ is given by: + +$$\mathbf{r}(n) = \mathbf{z}(n) - \mathbf{p}(n) \tag{25}$$ + +where $\mathbf{p}(n)$ is the predicted LSF vector at frame $n$ . First order moving-average (MA) prediction is used where: + +$$p_j(n) = \alpha_j \hat{r}_j(n-1) \quad j = 1, \dots, 10, \tag{26}$$ + +where $\hat{\mathbf{r}}(n-1)$ is the quantified residual vector at the past frame and $\alpha_j$ is the prediction factor for the $j$ th LSF. + +The LSF residual vectors $\mathbf{r}$ is quantified using split vector quantization. The vector $\mathbf{r}$ is split into 3 subvectors of dimension 3, 3, and 4. The 3 subvectors are quantified with 7-9 bits according to table 2. + +**Table 2. Bit allocation split vector quantization of LSF residual vector.** + +| Mode | Subvector 1 | Subvector 2 | Subvector 3 | +|-------------|-------------|-------------|-------------| +| 10.2 kbit/s | 8 | 9 | 9 | +| 7.95 kbit/s | 9 | 9 | 9 | +| 7.40 kbit/s | 8 | 9 | 9 | +| 6.70 kbit/s | 8 | 9 | 9 | +| 5.90 kbit/s | 8 | 9 | 9 | +| 5.15 kbit/s | 8 | 8 | 7 | +| 4.75 kbit/s | 8 | 8 | 7 | + +The weighted LSP distortion measure of equation (23) with the weighting of equation (24) is used in the quantization process. + +### 5.2.6 Interpolation of the LSPs + +#### 12.2 kbit/s mode + +The two sets of quantified (and unquantized) LP parameters are used for the second and fourth subframes whereas the first and third subframes use a linear interpolation of the parameters in the adjacent subframes. The interpolation + +is performed on the LSPs in the $\mathbf{q}$ domain. Let $\hat{\mathbf{q}}_4^{(n)}$ be the LSP vector at the 4th subframe of the present frame $n$ , $\hat{\mathbf{q}}_2^{(n)}$ be the LSP vector at the 2nd subframe of the present frame $n$ , and $\hat{\mathbf{q}}_4^{(n-1)}$ the LSP vector at the 4th subframe of the past frame $n-1$ . The interpolated LSP vectors at the 1st and 3rd subframes are given by: + +$$\begin{aligned}\hat{q}_1^{(n)} &= 0.5\hat{q}_4^{(n-1)} + 0.5\hat{q}_2^{(n)}, \\ \hat{q}_3^{(n)} &= 0.5\hat{q}_2^{(n)} + 0.5\hat{q}_4^{(n)}.\end{aligned}\quad (27)$$ + +The interpolated LSP vectors are used to compute a different LP filter at each subframe (both quantified and unquantized coefficients) using the LSP to LP conversion method described in clause 5.2.4. + +#### 10.2, 7.95, 7.40, 6.70, 5.90, 5.15, 4.75 kbit/s modes + +The set of quantified (and unquantized) LP parameters is used for the fourth subframe whereas the first, second, and third subframes use a linear interpolation of the parameters in the adjacent subframes. The interpolation is performed on the LSPs in the **q** domain. The interpolated LSP vectors at the 1st, 2nd, and 3rd subframes are given by: + +$$\begin{aligned}\hat{q}_1^{(n)} &= 0.75\hat{q}_4^{(n-1)} + 0.25\hat{q}_4^{(n)}, \\ \hat{q}_2^{(n)} &= 0.5\hat{q}_4^{(n-1)} + 0.5\hat{q}_4^{(n)}, \\ \hat{q}_3^{(n)} &= 0.25\hat{q}_4^{(n-1)} + 0.75\hat{q}_4^{(n)}.\end{aligned}\quad (28)$$ + +The interpolated LSP vectors are used to compute a different LP filter at each subframe (both quantified and unquantized coefficients) using the LSP to LP conversion method described in clause 5.2.4. + +### 5.2.7 Monitoring resonance in the LPC spectrum (all modes) + +Resonances in the LPC filter are monitored to detect possible problem areas where divergence between the adaptive codebook memories in the encoder and the decoder could cause unstable filters in areas with highly correlated continuous signals. Typically, this divergence is due to channel errors. + +The monitoring of resonance signals is performed using unquantized LSPs $q_i, i = 1, \dots, 10$ . The LSPs are available after the LP to LSP conversion in clause 5.2.3. The algorithm utilises the fact that LSPs are closely located at a peak in the spectrum. First, two distances $dist_1$ and $dist_2$ are calculated in two different regions, defined as $dist_1 = \min(q_i - q_{i+1}), i = 4, \dots, 8$ and $dist_2 = \min(q_i - q_{i+1}), i = 2, 3$ . + +Either of these two minimum distance conditions must be fulfilled to classify the frame as a resonance frame and increase the resonance counter. + +``` +if ( $dist_1 < TH_1$ ) OR if ( $dist_2 < TH_2$ ) + counter = counter + 1 +else + counter = 0 +``` + +$TH_1 = 0.046$ is a fixed threshold while the second one is depending on $q_2$ according to: + +$$TH_2 = \begin{cases} 0.018, & q_2 > 0.98 \\ 0.024, & 0.93 < q_2 \leq 0.98 \\ 0.034, & otherwise \end{cases}$$ + +12 consecutive resonance frames are needed to indicate possible problem conditions, otherwise the LSP\_flag is cleared. + +``` + +if (counter $\geq 12$ ) + counter = 12 + LSP_flag = 1 +else + LSP_flag = 0 + +``` + +## 5.3 Open-loop pitch analysis + +Open-loop pitch analysis is performed in order to simplify the pitch analysis and confine the closed-loop pitch search to a small number of lags around the open-loop estimated lags. + +Open-loop pitch estimation is based on the weighted speech signal $s_w(n)$ which is obtained by filtering the input speech signal through the weighting filter $W(z) = A(z/\gamma_1)/A(z/\gamma_2)$ . That is, in a subframe of size $L$ , the weighted speech is given by: + +$$s_w(n) = s(n) + \sum_{i=1}^{10} a_i \gamma_1^i s(n-i) - \sum_{i=1}^{10} a_i \gamma_2^i s_w(n-i), \quad n = 0, \dots, L-1 \quad (29)$$ + +#### 12.2 kbit/s mode + +Open-loop pitch analysis is performed twice per frame (each 10 ms) to find two estimates of the pitch lag in each frame. + +Open-loop pitch analysis is performed as follows. In the first step, 3 maxima of the correlation: + +$$O_k = \sum_{n=0}^{79} s_w(n) s_w(n-k) \quad (30)$$ + +are found in the three ranges: + +$$i=3: \quad 18, \dots, 35,$$ + +$$i=2: \quad 36, \dots, 71,$$ + +$$i=1: \quad 72, \dots, 143.$$ + +The retained maxima $O_{t_i}, i=1, \dots, 3$ , are normalized by dividing by $\sqrt{\sum_n s_w^2(n-t_i)}, i=1, \dots, 3$ , respectively. The normalized maxima and corresponding delays are denoted by $(M_i, t_i), i=1, \dots, 3$ . The winner, $T_{op}$ , among the three normalized correlations is selected by favouring the delays with the values in the lower range. This is performed by weighting the normalized correlations corresponding to the longer delays. The best open-loop delay $T_{op}$ is determined as follows: + +``` + + $T_{op} = t_1$ + $M(T_{op}) = M_1$ +if $M_2 > 0.85M(T_{op})$ + $M(T_{op}) = M_2$ + $T_{op} = t_2$ +end +if $M_3 > 0.85M(T_{op})$ + $M(T_{op}) = M_3$ + $T_{op} = t_3$ +end + +``` + +This procedure of dividing the delay range into 3 clauses and favouring the lower clauses is used to avoid choosing pitch multiples. + +#### 10.2 kbit/s mode + +Open-loop pitch analysis is performed twice per frame (every 10 ms) to find two estimates of the pitch lag in each frame. + +The open-loop pitch analysis is performed as follows. First, the correlation of weighted speech is determined for each pitch lag value $d$ by: + +$$C(d) = \sum_{n=0}^{79} s_w(n)s_w(n-d)w(d), \quad d = 20, \dots, 143 \quad (31)$$ + +where $w(d)$ is a weighting function. The estimated pitch-lag is the delay that maximises the weighted correlation function $C(d)$ . The weighting emphasises lower pitch lag values reducing the likelihood of selecting a multiple of the correct delay. The weighting function consists of two parts: a low pitch lag emphasis function, $w_l(d)$ , and a previous frame lag neighbouring emphasis function, $w_n(d)$ : + +$$w(d) = w_l(d)w_n(d) \quad (32)$$ + +The low pitch lag emphasis function is a given by: + +$$w_l(d) = cw(d) \quad (33)$$ + +where $cw(d)$ is defined by a table in the fixed point computational description (ANSI-C code) in [4]. The previous frame lag neighbouring emphasis function depends on the pitch lag of previous speech frames: + +$$w_n(d) = \begin{cases} cw(|T_{old} - d| + d_L), & v > 0.3, \\ 1.0, & \text{otherwise,} \end{cases} \quad (34)$$ + +where $d_L = 20$ , $T_{old}$ is the median filtered pitch lag of 5 previous voiced speech half-frames, and $v$ is an adaptive parameter. If the frame is classified as voiced by having the open-loop gain $g > 0.4$ , the $v$ -value is set to 1.0 for the next frame. Otherwise, the $v$ -value is updated by $v = 0.9v$ . The open loop gain is given by: + +$$g = \frac{\sum_{n=0}^{79} s_w(n) s_w(n - d_{\max})}{\sum_{n=0}^{79} s_w^2(n)} \quad (35)$$ + +where $d_{\max}$ is the pitch delay that maximizes $C(d)$ . The median filter is updated only during voiced speech frames. The weighting depends on the reliability of the old pitch lags. If previous frames have contained unvoiced speech or silence, the weighting is attenuated through the parameter $v$ . + +#### 7.95, 7.40, 6.70, 5.90 kbit/s modes + +Open-loop pitch analysis is performed twice per frame (each 10 ms) to find two estimates of the pitch lag in each frame. + +Open-loop pitch analysis is performed as follows. In the first step, 3 maxima of the correlation in equation (30) are found in the three ranges: + +$$i=3: \quad 20, \dots, 39,$$ + +$$i=2: \quad 40, \dots, 79,$$ + +$$i=1: \quad 80, \dots, 143.$$ + +The retained maxima $O_{t_i}, i=1, \dots, 3$ , are normalized by dividing by $\sqrt{\sum_n s_w^2(n - t_i)}, i=1, \dots, 3$ , respectively. + +The normalized maxima and corresponding delays are denoted by $(M_i, t_i), i=1, \dots, 3$ . The winner, $T_{op}$ , among the three normalized correlations is selected by favouring the delays with the values in the lower range. This is performed by weighting the normalized correlations corresponding to the longer delays. The best open-loop delay $T_{op}$ is determined as follows: + +$$\begin{aligned} T_{op} &= t_1 \\ M(T_{op}) &= M_1 \\ \text{if } M_2 &> 0.85M(T_{op}) \\ M(T_{op}) &= M_2 \\ T_{op} &= t_2 \\ \text{end} \\ \text{if } M_3 &> 0.85M(T_{op}) \\ M(T_{op}) &= M_3 \\ T_{op} &= t_3 \\ \text{end} \end{aligned}$$ + +This procedure of dividing the delay range into 3 clauses and favouring the lower clauses is used to avoid choosing pitch multiples. + +#### 5.15, 4.75 kbit/s modes + +Open-loop pitch analysis is performed once per frame (each 20 ms) to find an estimate of the pitch lag in each frame. + +Open-loop pitch analysis is performed as follows. In the first step, 3 maxima of the correlation in equation (30) are found in the three ranges: + +$$i=3: 20, \dots, 39,$$ + +$$i=2: 40, \dots, 79,$$ + +$$i=1: 79, \dots, 143.$$ + +The retained maxima $O_{t_i}, i=1, \dots, 3$ , are normalized by dividing by $\sqrt{\sum_n s_w^2(n-t_i)}, i=1, \dots, 3$ , respectively. + +The normalized maxima and corresponding delays are denoted by $(M_i, t_i), i=1, \dots, 3$ . The winner, $T_{op}$ , among the three normalized correlations is selected by favouring the delays with the values in the lower range. This is performed by weighting the normalized correlations corresponding to the longer delays. The best open-loop delay $T_{op}$ is determined as follows: + +$$\begin{aligned} T_{op} &= t_1 \\ M(T_{op}) &= M_1 \\ \text{if } M_2 &> 0.85M(T_{op}) \\ M(T_{op}) &= M_2 \\ T_{op} &= t_2 \\ \text{end} \\ \text{if } M_3 &> 0.85M(T_{op}) \\ M(T_{op}) &= M_3 \\ T_{op} &= t_3 \\ \text{end} \end{aligned}$$ + +This procedure of dividing the delay range into 3 clauses and favouring the lower clauses is used to avoid choosing pitch multiples. + +## 5.4 Impulse response computation (all modes) + +The impulse response, $h(n)$ , of the weighted synthesis filter $H(z)W(z) = A(z/\gamma_1)/[\hat{A}(z)A(z/\gamma_2)]$ is computed each subframe. This impulse response is needed for the search of adaptive and fixed codebooks. The impulse response $h(n)$ is computed by filtering the vector of coefficients of the filter $A(z/\gamma_1)$ extended by zeros through the two filters $1/\hat{A}(z)$ and $1/A(z/\gamma_2)$ . + +## 5.5 Target signal computation (all modes) + +The target signal for adaptive codebook search is usually computed by subtracting the zero input response of the weighted synthesis filter $H(z)W(z) = A(z/\gamma_1)/[\hat{A}(z)A(z/\gamma_2)]$ from the weighted speech signal $s_w(n)$ . This is performed on a subframe basis. + +An equivalent procedure for computing the target signal, which is used in the present document, is the filtering of the LP residual signal $res_{LP}(n)$ through the combination of synthesis filter $1/\hat{A}(z)$ and the weighting filter $A(z/\gamma_1)/A(z/\gamma_2)$ . After determining the excitation for the subframe, the initial states of these filters are updated by filtering the difference between the LP residual and excitation. The memory update of these filters is explained in clause 5.9. + +The residual signal $res_{LP}(n)$ which is needed for finding the target vector is also used in the adaptive codebook search to extend the past excitation buffer. This simplifies the adaptive codebook search procedure for delays less than the subframe size of 40 as will be explained in the next clause. The LP residual is given by: + +$$res_{LP}(n) = s(n) + \sum_{i=1}^{10} \hat{a}_i s(n-i). \quad (36)$$ + +## 5.6 Adaptive codebook + +### 5.6.1 Adaptive codebook search + +Adaptive codebook search is performed on a subframe basis. It consists of performing closed-loop pitch search, and then computing the adaptive codevector by interpolating the past excitation at the selected fractional pitch lag. + +The adaptive codebook parameters (or pitch parameters) are the delay and gain of the pitch filter. In the adaptive codebook approach for implementing the pitch filter, the excitation is repeated for delays less than the subframe length. In the search stage, the excitation is extended by the LP residual to simplify the closed-loop search. + +#### 12.2 kbit/s mode + +In the first and third subframes, a fractional pitch delay is used with resolutions: 1/6 in the range $[17 \ 3/6, 94 \ 3/6]$ and integers only in the range [95, 143]. For the second and fourth subframes, a pitch resolution of 1/6 is always used in the range $[T_1 - 5 \ 3/6, T_1 + 4 \ 3/6]$ , where $T_1$ is nearest integer to the fractional pitch lag of the previous (1st or 3rd) subframe, bounded by 18...143. + +Closed-loop pitch analysis is performed around the open-loop pitch estimates on a subframe basis. In the first (and third) subframe the range $T_{op} \pm 3$ , bounded by 18...143, is searched. For the other subframes, closed-loop pitch analysis is performed around the integer pitch selected in the previous subframe, as described above. The pitch delay is encoded with 9 bits in the first and third subframes and the relative delay of the other subframes is encoded with 6 bits. + +The closed-loop pitch search is performed by minimizing the mean-square weighted error between the original and synthesized speech. This is achieved by maximizing the term: + +$$R(k) = \frac{\sum_{n=0}^{39} x(n)y_k(n)}{\sqrt{\sum_{n=0}^{39} y_k(n)y_k(n)}}, \quad (37)$$ + +where $x(n)$ is the target signal and $y_k(n)$ is the past filtered excitation at delay $k$ (past excitation convolved with $h(n)$ ). Note that the search range is limited around the open-loop pitch as explained earlier. + +The convolution $y_k(n)$ is computed for the first delay $t_{min}$ in the searched range, and for the other delays in the search range $k = t_{min} + 1, \dots, t_{max}$ , it is updated using the recursive relation: + +$$y_k(n) = y_{k-1}(n-1) + u(-k)h(n), \quad 1 \leq n \leq 39 \quad (38)$$ + +and $y_k(0) = u(-k)h(0)$ , where $u(n), n = -(143 + 11), \dots, 39$ , is the excitation buffer. Note that in search stage, the samples $u(n), n = 0, \dots, 39$ , are not known, and they are needed for pitch delays less than 40. To simplify the search, the LP residual is copied to $u(n)$ in order to make the relation in equation (38) valid for all delays. + +Once the optimum integer pitch delay is determined, the fractions from $-3/6$ to $3/6$ with a step of $1/6$ around that integer are tested. The fractional pitch search is performed by interpolating the normalized correlation in equation (37) and searching for its maximum. The interpolation is performed using an FIR filter $b_{24}$ based on a Hamming + +windowed $\sin(x)/x$ function truncated at $\pm 23$ and padded with zeros at $\pm 24$ ( $b_{24}(24)=0$ ). The filter has its cut-off frequency (-3 dB) at 3 600 Hz in the over-sampled domain. The interpolated values of $R(k)$ for the fractions $-3/6$ to $3/6$ are obtained using the interpolation formula: + +$$R(k)_t = \sum_{i=0}^3 R(k-i) b_{24}(t+i \cdot 6) + \sum_{i=0}^3 R(k+1+i) b_{24}(6-t+i \cdot 6), \quad t=0, \dots, 5, \quad (39)$$ + +where $t=0, \dots, 5$ corresponds to the fractions $0, 1/6, 2/6, 3/6, -2/6$ , and $-1/6$ , respectively. Note that it is necessary to compute the correlation terms in equation (37) using a range $t_{\min} - 4, t_{\max} + 4$ , to allow for the proper interpolation. + +Once the fractional pitch lag is determined, the adaptive codebook vector $v(n)$ is computed by interpolating the past excitation signal $u(n)$ at the given integer delay $k$ and phase (fraction) $t$ : + +$$v(n) = \sum_{i=0}^9 u(n-k-i) b_{60}(t+i \cdot 6) + \sum_{i=0}^9 u(n-k+1+i) b_{60}(6-t+i \cdot 6), \quad n=0, \dots, 39, \quad t=0, \dots, 5. \quad (40)$$ + +The interpolation filter $b_{60}$ is based on a Hamming windowed $\sin(x)/x$ function truncated at $\pm 59$ and padded with zeros at $\pm 60$ ( $b_{60}(60)=0$ ). The filter has a cut-off frequency (-3 dB) at 3 600 Hz in the over-sampled domain. + +The adaptive codebook gain is then found by: + +$$g_p = \frac{\sum_{n=0}^{39} x(n)y(n)}{\sum_{n=0}^{39} y(n)y(n)}, \quad \text{bounded by } 0 \leq g_p \leq 1.2 \quad (41)$$ + +where $y(n) = v(n)*h(n)$ is the filtered adaptive codebook vector (zero state response of $H(z)W(z)$ to $v(n)$ ). + +The computed adaptive codebook gain is quantified using 4-bit non-uniform scalar quantization in the range $[0.0, 1.2]$ . + +#### 7.95 kbit/s mode + +In the first and third subframes, a fractional pitch delay is used with resolutions: $1/3$ in the range $[191/3, 842/3]$ and integers only in the range $[85, 143]$ . For the second and fourth subframes, a pitch resolution of $1/3$ is always used in the range $[T_1 - 102/3, T_1 + 92/3]$ , where $T_1$ is nearest integer to the fractional pitch lag of the previous (1st or 3rd) subframe, bounded by $20 \dots 143$ . + +Closed-loop pitch analysis is performed around the open-loop pitch estimates on a subframe basis. In the first (and third) subframe the range $T_{op} \pm 3$ , bounded by $20 \dots 143$ , is searched. For the other subframes, closed-loop pitch analysis is performed around the integer pitch selected in the previous subframe, as described above. The pitch delay is encoded with 8 bits in the first and third subframes and the relative delay of the other subframes is encoded with 6 bits. + +The closed-loop pitch search is performed by minimizing the mean-square weighted error between the original and synthesized speech. This is achieved by maximizing the term of equation (37). Note that the search range is limited around the open-loop pitch as explained earlier. + +The convolution $y_k(n)$ is computed for the first delay $t_{\min}$ in the searched range, and for the other delays in the search range $k = t_{\min} + 1, \dots, t_{\max}$ , it is updated using the recursive relation of equation (38). + +Once the optimum integer pitch delay is determined, the fractions from $-2/3$ to $2/3$ with a step of $1/3$ around that integer are tested. The fractional pitch search is performed by interpolating the normalized correlation in equation (37) and searching for its maximum. Once the fractional pitch lag is determined, the adaptive codebook vector $v(n)$ is computed by interpolating the past excitation signal $u(n)$ at the given integer delay and phase (fraction). The interpolation is performed using two FIR filters (Hamming windowed sinc functions); one for interpolating the term in equation (37) with the sinc truncated at $\pm 11$ and the other for interpolating the past excitation with the sinc truncated at $\pm 29$ . The filters have their cut-off frequency ( $-3$ dB) at $3\,600$ Hz in the over-sampled domain. + +The adaptive codebook gain is then found as in equation (41). + +The computed adaptive codebook gain is quantified using 4-bit non-uniform scalar quantization as described in clause 5.8. + +#### 10.2, 7.40 kbit/s mode + +In the first and third subframes, a fractional pitch delay is used with resolutions: $1/3$ in the range $[19\ 1/3, 84\ 2/3]$ and integers only in the range $[85, 143]$ . For the second and fourth subframes, a pitch resolution of $1/3$ is always used in the range $[T_1 - 5\ 2/3, T_1 + 4\ 2/3]$ , where $T_1$ is nearest integer to the fractional pitch lag of the previous (1st or 3rd) subframe, bounded by $20 \dots 143$ . + +Closed-loop pitch analysis is performed around the open-loop pitch estimates on a subframe basis. In the first (and third) subframe the range $T_{op} \pm 3$ , bounded by $20 \dots 143$ , is searched. For the other subframes, closed-loop pitch analysis is performed around the integer pitch selected in the previous subframe, as described above. The pitch delay is encoded with 8 bits in the first and third subframes and the relative delay of the other subframes is encoded with 5 bits. + +The closed-loop pitch search is performed by minimizing the mean-square weighted error between the original and synthesized speech. This is achieved by maximizing the term of equation (37). Note that the search range is limited around the open-loop pitch as explained earlier. + +The convolution $y_k(n)$ is computed for the first delay $t_{\min}$ in the searched range, and for the other delays in the search range $k = t_{\min} + 1, \dots, t_{\max}$ , it is updated using the recursive relation of equation (38). + +Once the optimum integer pitch delay is determined, the fractions from $-2/3$ to $2/3$ with a step of $1/3$ around that integer are tested. The fractional pitch search is performed by interpolating the normalized correlation in equation (37) and searching for its maximum. Once the fractional pitch lag is determined, the adaptive codebook vector $v(n)$ is computed by interpolating the past excitation signal $u(n)$ at the given integer delay and phase (fraction). The interpolation is performed using two FIR filters (Hamming windowed sinc functions); one for interpolating the term in equation (37) with the sinc truncated at $\pm 11$ and the other for interpolating the past excitation with the sinc truncated at $\pm 29$ . The filters have their cut-off frequency ( $-3$ dB) at $3\,600$ Hz in the over-sampled domain. + +The adaptive codebook gain is then found as in equation (41). + +The computed adaptive codebook gain (and the fixed codebook gain) is quantified using 7-bit non-uniform vector quantization as described in clause 5.8. + +#### 6.70, 5.90 kbit/s modes + +In the first and third subframes, a fractional pitch delay is used with resolutions: $1/3$ in the range $[19\ 1/3, 84\ 2/3]$ and integers only in the range $[85, 143]$ . For the second and fourth subframes, integer pitch resolution is used in the range $[T_1 - 5, T_1 + 4]$ , where $T_1$ is nearest integer to the fractional pitch lag of the previous (1st or 3rd) subframe, bounded by $20 \dots 143$ . Additionally, a fractional resolution of $1/3$ is used in the range $[T_1 - 12/3, T_1 + 2/3]$ . + +Closed-loop pitch analysis is performed around the open-loop pitch estimates on a subframe basis. In the first (and third) subframe the range $T_{op} \pm 3$ , bounded by 20...143, is searched. For the other subframes, closed-loop pitch analysis is performed around the integer pitch selected in the previous subframe, as described above. The pitch delay is encoded with 8 bits in the first and third subframes and the relative delay of the other subframes is encoded with 4 bits. + +The closed-loop pitch search is performed by minimizing the mean-square weighted error between the original and synthesized speech. This is achieved by maximizing the term of equation (37). Note that the search range is limited around the open-loop pitch as explained earlier. + +The convolution $y_k(n)$ is computed for the first delay $t_{min}$ in the searched range, and for the other delays in the search range $k = t_{min} + 1, \dots, t_{max}$ , it is updated using the recursive relation of equation (38). + +Once the optimum integer pitch delay is determined, the fractions from $-2/3$ to $2/3$ with a step of $1/3$ around that integer are tested. The fractional pitch search is performed by interpolating the normalized correlation in equation (37) and searching for its maximum. Once the fractional pitch lag is determined, the adaptive codebook vector $v(n)$ is computed by interpolating the past excitation signal $u(n)$ at the given integer delay and phase (fraction). The interpolation is performed using two FIR filters (Hamming windowed sinc functions); one for interpolating the term in equation (37) with the sinc truncated at $\pm 11$ and the other for interpolating the past excitation with the sinc truncated at $\pm 29$ . The filters have their cut-off frequency ( $-3$ dB) at 3 600 Hz in the over-sampled domain. + +The adaptive codebook gain is then found as in equation (41). + +The computed adaptive codebook gain (and the fixed codebook gain) is quantified using vector quantization as described in clause 5.8. + +#### 5.15, 4.75 kbit/s modes + +In the first subframe, a fractional pitch delay is used with resolutions: $1/3$ in the range $[19\ 1/3, 84\ 2/3]$ and integers only in the range $[85, 143]$ . For the second, third, and fourth subframes, integer pitch resolution is used in the range $[T_1 - 5, T_1 + 4]$ , where $T_1$ is nearest integer to the fractional pitch lag of the previous subframe, bounded by 20...143. Additionally, a fractional resolution of $1/3$ is used in the range $[T_1 - 1\ 2/3, T_1 + 2/3]$ . + +Closed-loop pitch analysis is performed around the open-loop pitch estimates on a subframe basis. In the first subframe the range $T_{op} \pm 5$ , bounded by 20...143, is searched. For the other subframes, closed-loop pitch analysis is performed around the integer pitch selected in the previous subframe, as described above. The pitch delay is encoded with 8 bits in the first subframe and the relative delay of the other subframes is encoded with 4 bits. + +The closed-loop pitch search is performed by minimizing the mean-square weighted error between the original and synthesized speech. This is achieved by maximizing the term of equation (37). Note that the search range is limited around the open-loop pitch as explained earlier. + +The convolution $y_k(n)$ is computed for the first delay $t_{min}$ in the searched range, and for the other delays in the search range $k = t_{min} + 1, \dots, t_{max}$ , it is updated using the recursive relation of equation (38). + +Once the optimum integer pitch delay is determined, the fractions from $-2/3$ to $2/3$ with a step of $1/3$ around that integer are tested. The fractional pitch search is performed by interpolating the normalized correlation in equation (37) and searching for its maximum. Once the fractional pitch lag is determined, the adaptive codebook vector $v(n)$ is computed by interpolating the past excitation signal $u(n)$ at the given integer delay and phase (fraction). The interpolation is performed using two FIR filters (Hamming windowed sinc functions); one for interpolating the term in equation (37) with the sinc truncated at $\pm 11$ and the other for interpolating the past excitation with the sinc truncated at $\pm 29$ . The filters have their cut-off frequency ( $-3$ dB) at 3 600 Hz in the over-sampled domain. + +The adaptive codebook gain is then found as in equation (41). + +The computed adaptive codebook gain (and the fixed codebook gain) is quantified using vector quantization as described in clause 5.8. + +### 5.6.2 Adaptive codebook gain control (all modes) + +The average adaptive codebook gain is calculated if the *LSP\_flag* is set and the unquantized adaptive codebook gain exceeds the gain threshold $GP_{th} = 0.95$ + +The average gain is calculated from the present unquantized gain and the quantized gains of the seven previous subframes. That is $GP_{ave} = \text{mean}\{g_p(n), \hat{g}_p(n-1), \hat{g}_p(n-2), \dots, \hat{g}_p(n-7)\}$ where $n$ is the current + +subframe. If the average adaptive codebook gain exceeds the $GP_{th}$ the unquantized gain is limited to the threshold value and the *GpC\_flag* is set to indicate the limitation. + +``` +if (GP_ave > GP_th) + g_p = GP_th + GpC_flag = 1 +else + GpC_flag = 0 +``` + +The *GpC\_flag* is used in the gain quantization in clause 5.8. + +## 5.7 Algebraic codebook + +### 5.7.1 Algebraic codebook structure + +The algebraic codebook structure is based on interleaved single-pulse permutation (ISPP) design. + +#### 12.2 kbit/s mode + +In this codebook, the innovation vector contains 10 non-zero pulses. All pulses can have the amplitudes +1 or -1. The 40 positions in a subframe are divided into 5 tracks, where each track contains two pulses, as shown in table 3. + +**Table 3: Potential positions of individual pulses in the algebraic codebook, 12.2 kbit/s.** + +| Track | Pulse | Positions | +|-------|--------|------------------------------| +| 1 | i0, i5 | 0, 5, 10, 15, 20, 25, 30, 35 | +| 2 | i1, i6 | 1, 6, 11, 16, 21, 26, 31, 36 | +| 3 | i2, i7 | 2, 7, 12, 17, 22, 27, 32, 37 | +| 4 | i3, i8 | 3, 8, 13, 18, 23, 28, 33, 38 | +| 5 | i4, i9 | 4, 9, 14, 19, 24, 29, 34, 39 | + +Each two pulse positions in one track are encoded with 6 bits (total of 30 bits, 3 bits for the position of every pulse), and the sign of the first pulse in the track is encoded with 1 bit (total of 5 bits). + +For two pulses located in the same track, only one sign bit is needed. This sign bit indicates the sign of the first pulse. The sign of the second pulse depends on its position relative to the first pulse. If the position of the second pulse is smaller, then it has opposite sign, otherwise it has the same sign than in the first pulse. + +All the 3-bit pulse positions are Gray coded in order to improve robustness against channel errors. This gives a total of 35 bits for the algebraic code. + +#### 10.2 kbit/s mode + +In this codebook, the innovation vector contains 8 non-zero pulses. All pulses can have the amplitudes +1 or -1. The 40 positions in a subframe are divided into 4 tracks, where each track contains two pulses, as shown in table 4. + +**Table 4: Potential positions of individual pulses in the algebraic codebook, 10.2 kbit/s.** + +| Track | Pulse | Positions | +|-------|------------|--------------------------------------| +| 1 | $i_0, i_4$ | 0, 4, 8, 12, 16, 20, 24, 28, 32, 36 | +| 2 | $i_1, i_5$ | 1, 5, 9, 13, 17, 21, 25, 29, 33, 37 | +| 3 | $i_2, i_6$ | 2, 6, 10, 14, 18, 22, 26, 30, 34, 38 | +| 4 | $i_3, i_7$ | 3, 7, 11, 15, 19, 23, 27, 31, 35, 39 | + +The pulses are grouped into 3, 3, and 2 pulses and their positions are encoded with 10, 10, and 7 bits, respectively (total of 27 bits). The sign of the first pulse in each track is encoded with 1 bit (total of 4 bits). + +For two pulses located in the same track, only one sign bit is needed. This sign bit indicates the sign of the first pulse. The sign of the second pulse depends on its position relative to the first pulse. If the position of the second pulse is smaller, then it has opposite sign, otherwise it has the same sign than in the first pulse. + +This gives a total of 31 bits for the algebraic code. + +#### 7.95, 7.40 kbit/s modes + +In this codebook, the innovation vector contains 4 non-zero pulses. All pulses can have the amplitudes +1 or -1. The 40 positions in a subframe are divided into 4 tracks, where each track contains one pulse, as shown in table 5. + +**Table 5: Potential positions of individual pulses in the algebraic codebook, 7.95, 7.40 kbit/s.** + +| Track | Pulse | Positions | +|-------|-------|---------------------------------------------------------------| +| 1 | $i_0$ | 0, 5, 10, 15, 20, 25, 30, 35 | +| 2 | $i_1$ | 1, 6, 11, 16, 21, 26, 31, 36 | +| 3 | $i_2$ | 2, 7, 12, 17, 22, 27, 32, 37 | +| 4 | $i_3$ | 3, 8, 13, 18, 23, 28, 33, 38,
4, 9, 14, 19, 24, 29, 34, 39 | + +The pulse positions are encoded with 3, 3, 3, and 4 bits (total of 13 bits), and the sign of the each pulse is encoded with 1 bit (total of 4 bits). This gives a total of 17 bits for the algebraic code. + +#### 6.70 kbit/s mode + +In this codebook, the innovation vector contains 3 non-zero pulses. All pulses can have the amplitudes +1 or -1. The 40 positions in a subframe are divided into 3 tracks, where each track contains one pulse, as shown in table 6. + +**Table 6: Potential positions of individual pulses in the algebraic codebook, 6.70 kbit/s.** + +| Track | Pulse | Positions | +|-------|-------|---------------------------------------------------------------| +| 1 | $i_0$ | 0, 5, 10, 15, 20, 25, 30, 35 | +| 2 | $i_1$ | 1, 6, 11, 16, 21, 26, 31, 36,
3, 8, 13, 18, 23, 28, 33, 38 | +| 3 | $i_2$ | 2, 7, 12, 17, 22, 27, 32, 37,
4, 9, 14, 19, 24, 29, 34, 39 | + +The pulse positions are encoded with 3, 4, and 4 bits (total of 11 bits), and the sign of the each pulse is encoded with 1 bit (total of 3 bits). This gives a total of 14 bits for the algebraic code. + +#### 5.90 kbit/s mode + +In this codebook, the innovation vector contains 2 non-zero pulses. All pulses can have the amplitudes +1 or -1. The 40 positions in a subframe are divided into 2 tracks, where each track contains one pulse, as shown in table 7. + +**Table 7: Potential positions of individual pulses in the algebraic codebook, 5.90 kbit/s.** + +| Track | Pulse | Positions | +|-------|-------|---------------------------------------------------------------------------------------------------------------------------------| +| 1 | $i_0$ | 1, 6, 11, 16, 21, 26, 31, 36,
3, 8, 13, 18, 23, 28, 33, 38 | +| 2 | $i_1$ | 0, 5, 10, 15, 20, 25, 30, 35,
1, 6, 11, 16, 21, 26, 31, 36,
2, 7, 12, 17, 22, 27, 32, 37,
4, 9, 14, 19, 24, 29, 34, 39 | + +The pulse positions are encoded with 4 and 5 bits (total of 9 bits), and the sign of the each pulse is encoded with 1 bit (total of 2 bits). This gives a total of 11 bits for the algebraic code. + +#### 5.15, 4.75 kbit/s modes + +In this codebook, the innovation vector contains 2 non-zero pulses. All pulses can have the amplitudes +1 or -1. The 40 positions in a subframe are divided into 5 tracks. Two subsets of 2 tracks each are used for each subframe with one pulse in each track. Different subsets of tracks are used for each subframe. The pulse positions used in each subframe are shown in table 8. + +**Table 8: Potential positions of individual pulses in the algebraic codebook, 5.15, 4.75 kbit/s.** + +| Subframe | Subset | Pulse | Positions | +|----------|--------|-------|------------------------------| +| 1 | 1 | $i_0$ | 0, 5, 10, 15, 20, 25, 30, 35 | +| | | $i_1$ | 2, 7, 12, 17, 22, 27, 32, 37 | +| | 2 | $i_0$ | 1, 6, 11, 16, 21, 26, 31, 36 | +| | | $i_1$ | 3, 8, 13, 18, 23, 28, 33, 38 | +| 2 | 1 | $i_0$ | 0, 5, 10, 15, 20, 25, 30, 35 | +| | | $i_1$ | 3, 8, 13, 18, 23, 28, 33, 38 | +| | 2 | $i_0$ | 2, 7, 12, 17, 22, 27, 32, 37 | +| | | $i_1$ | 4, 9, 14, 19, 24, 29, 34, 39 | +| 3 | 1 | $i_0$ | 0, 5, 10, 15, 20, 25, 30, 35 | +| | | $i_1$ | 2, 7, 12, 17, 22, 27, 32, 37 | +| | 2 | $i_0$ | 1, 6, 11, 16, 21, 26, 31, 36 | +| | | $i_1$ | 4, 9, 14, 19, 24, 29, 34, 39 | +| 4 | 1 | $i_0$ | 0, 5, 10, 15, 20, 25, 30, 35 | +| | | $i_1$ | 3, 8, 13, 18, 23, 28, 33, 38 | +| | 2 | $i_0$ | 1, 6, 11, 16, 21, 26, 31, 36 | +| | | $i_1$ | 4, 9, 14, 19, 24, 29, 34, 39 | + +One bit is needed to encoded the subset used. The two pulse positions are encoded with 3 bits each (total of 6 bits), and the sign of the each pulse is encoded with 1 bit (total of 2 bits). This gives a total of 9 bits for the algebraic code. + +### 5.7.2 Algebraic codebook search + +The algebraic codebook is searched by minimizing the mean square error between the weighted input speech and the weighted synthesized speech. The target signal used in the closed-loop pitch search is updated by subtracting the adaptive codebook contribution. That is: + +$$x_2(n) = x(n) - \hat{g}_p y(n), \quad n = 0, \dots, 39 \quad (42)$$ + +where $y(n) = v(n) * h(n)$ is the filtered adaptive codebook vector and $\hat{g}_p$ is the quantified adaptive codebook gain. If $\mathbf{c}^k$ is the algebraic codevector at index $k$ , then the algebraic codebook is searched by maximizing the term: + +$$A_k = \frac{(C_k)^2}{E_{Dk}} = \frac{(\mathbf{d}^t \mathbf{c}_k)^2}{\mathbf{c}_k^t \Phi \mathbf{c}_k}, \quad (43)$$ + +where $\mathbf{d} = \mathbf{H}^t \mathbf{x}_2$ is the correlation between the target signal $x_2(n)$ and the impulse response $h(n)$ , $\mathbf{H}$ is a the lower triangular Toeplitz convolution matrix with diagonal $h(0)$ and lower diagonals $h(1), \dots, h(39)$ , and $\Phi = \mathbf{H}^t \mathbf{H}$ is the matrix of correlations of $h(n)$ . The vector $\mathbf{d}$ (backward filtered target) and the matrix $\Phi$ are computed prior to the codebook search. The elements of the vector $\mathbf{d}$ are computed by + +$$d(n) = \sum_{i=n}^{39} x_2(i)h(i-n), \quad n = 0, \dots, 39, \quad (44)$$ + +and the elements of the symmetric matrix $\Phi$ are computed by: + +$$\phi(i, j) = \sum_{n=j}^{39} h(n-i)h(n-j), \quad (j \geq i) \quad (45)$$ + +The algebraic structure of the codebooks allows for very fast search procedures since the innovation vector $\mathbf{c}_k$ contains only a few nonzero pulses. The correlation in the numerator of Equation (43) is given by: + +$$C = \sum_{i=0}^{N_p-1} \vartheta_i d(m_i) \quad (46)$$ + +where $m_i$ is the position of the $i$ th pulse, $\vartheta_i$ is its amplitude, and $N_p$ is the number of pulses ( $N_p = 10$ ). The energy in the denominator of equation (43) is given by: + +$$E_D = \sum_{i=0}^{N_p-1} \phi(m_i, m_i) + 2 \sum_{i=0}^{N_p-2} \sum_{j=i+1}^{N_p-1} \vartheta_i \vartheta_j \phi(m_i, m_j). \quad (47)$$ + +To simplify the search procedure, the pulse amplitudes are preset by the mere quantization of an appropriate signal $b(n)$ . This is simply done by setting the amplitude of a pulse at a certain position equal to the sign of $b(n)$ at that position. The simplification proceeds as follows (prior to the codebook search). First, the sign signal + +$s_b(n) = \text{sign}[b(n)]$ and the signal $d'(n) = d(n)s_b(n)$ are computed. Second, the matrix $\Phi$ is modified by including the sign information; that is, $\phi'(i, j) = s_b(i)s_b(j)\phi(i, j)$ . The correlation in equation (46) is now given by: + +$$C = \sum_{i=0}^{N_p-1} d'(m_i) \quad (48)$$ + +and the energy in equation (47) is given by: + +$$E_D = \sum_{i=0}^{N_p-1} \phi'(m_i, m_i) + 2 \sum_{i=0}^{N_p-2} \sum_{j=i+1}^{N_p-1} \phi'(m_i, m_j). \quad (49)$$ + +#### 12.2 kbit/s mode + +In this case the signal $b(n)$ , used for presetting the amplitudes, is a sum of the normalized $d(n)$ vector and normalized long-term prediction residual $res_{LTP}(n)$ : + +$$b(n) = \frac{res_{LTP}(n)}{\sqrt{\sum_{i=0}^{39} res_{LTP}(i) res_{LTP}(i)}} + \frac{d(n)}{\sqrt{\sum_{i=0}^{39} d(i) d(i)}}, \quad n = 0, \dots, 39, \quad (50)$$ + +is used. Having preset the pulse amplitudes, as explained above, the optimal pulse positions are determined using an efficient non-exhaustive analysis-by-synthesis search technique. In this technique, the term in equation (43) is tested for a small percentage of position combinations. + +First, for each of the five tracks the pulse positions with maximum absolute values of $b(n)$ are searched. From these the global maximum value for all the pulse positions is selected. The first pulse $i_0$ is always set into the position corresponding to the global maximum value. + +Next, four iterations are carried out. During each iteration the position of pulse $i_1$ is set to the local maximum of one track. The rest of the pulses are searched in pairs by sequentially searching each of the pulse pairs $\{i_2, i_3\}$ , $\{i_4, i_5\}$ , $\{i_6, i_7\}$ and $\{i_8, i_9\}$ in nested loops. Every pulse has 8 possible positions, i.e., there are four 8x8-loops, resulting in 256 different combinations of pulse positions for each iteration. + +In each iteration all the 9 pulse starting positions are cyclically shifted, so that the pulse pairs are changed and the pulse $i_1$ is placed in a local maximum of a different track. The rest of the pulses are searched also for the other positions in the tracks. At least one pulse is located in a position corresponding to the global maximum and one pulse is located in a position corresponding to one of the 4 local maxima. + +A special feature incorporated in the codebook is that the selected codevector is filtered through an adaptive pre-filter $F_E(z)$ which enhances special spectral components in order to improve the synthesized speech quality. Here the filter $F_E(z) = 1/(1 - \beta z^{-T})$ is used, where $T$ is the nearest integer pitch lag to the closed-loop fractional pitch lag of the subframe, and $\beta$ is the quantized pitch gain of the current subframe bounded by $[0.0, 1.0]$ . Note that prior to the codebook search, the impulse response $h(n)$ must include the pre-filter $F_E(z)$ . That is, for values of $T$ less than 40, the impulse $h(n)$ is modified according to + +$$h(n) = \begin{cases} h(n) & n = 0, \dots, T-1 \\ 0 & n = T, \dots, 39 \end{cases} \quad (50a)$$ + +The fixed codebook gain is then found by: + +$$g_c = \frac{\mathbf{x}_2^t \mathbf{z}}{\mathbf{z}^t \mathbf{z}} \quad (51)$$ + +where $\mathbf{x}_2$ is the target vector for fixed codebook search and $\mathbf{z}$ is the fixed codebook vector convolved with $h(n)$ , + +$$z(n) = \sum_{i=0}^n c(i) h(n-i), \quad n = 0, \dots, 39. \quad (52)$$ + +#### 10.2 kbit/s mode + +In this case the signal $b(n)$ , used for presetting the amplitudes, is given by eq. (50). Having preset the pulse amplitudes, as explained above, the optimal pulse positions are determined using an efficient non-exhaustive analysis-by-synthesis search technique. In this technique, the term in equation (43) is tested for a small percentage of position combinations. + +A special feature incorporated in the codebook is that the selected codevector is filtered through an adaptive pre-filter $F_E(z)$ which enhances special spectral components in order to improve the synthesized speech quality. + +Here the filter $F_E(z) = 1/(1 - \beta z^{-T})$ is used, where $T$ is the nearest integer pitch lag to the closed-loop fractional pitch lag of the subframe, and $\beta$ is the quantized pitch gain of the previous subframe bounded by $[0.0, 0.8]$ . Note that prior to the codebook search, the impulse response $h(n)$ must include the pre-filter $F_E(z)$ . That is, for values of $T$ less than 40, the impulse $h(n)$ is modified according to equation (50a). + +The fixed codebook gain is then found by equation (51). + +#### 7.95, 7.40 kbit/s modes + +In this case the signal $b(n)$ , used for presetting the amplitudes, is equal to the signal $d(n)$ . Having preset the pulse amplitudes, as explained above, the optimal pulse positions are determined using an efficient non-exhaustive analysis-by-synthesis search technique. In this technique, the term in equation (43) is tested for a small percentage of position combinations. + +A special feature incorporated in the codebook is that the selected codevector is filtered through an adaptive pre-filter $F_E(z)$ which enhances special spectral components in order to improve the synthesized speech quality. Here the filter $F_E(z) = 1/(1 - \beta z^{-T})$ is used, where $T$ is the nearest integer pitch lag to the closed-loop fractional pitch lag of the subframe, and $\beta$ is the quantized pitch gain of the previous subframe bounded by $[0.0, 0.8]$ . Note that prior to the codebook search, the impulse response $h(n)$ must include the pre-filter $F_E(z)$ . That is, for values of $T$ less than 40, the impulse $h(n)$ is modified according to equation (50a). + +The fixed codebook gain is then found by equation (51). + +#### 6.70 kbit/s mode + +In this case the signal $b(n)$ , used for presetting the amplitudes, is equal to the signal $d(n)$ . Having preset the pulse amplitudes, as explained above, the optimal pulse positions are determined using an efficient non-exhaustive analysis-by-synthesis search technique. In this technique, the term in equation (43) is tested for a small percentage of position combinations. + +A special feature incorporated in the codebook is that the selected codevector is filtered through an adaptive pre-filter $F_E(z)$ which enhances special spectral components in order to improve the synthesized speech quality. Here the filter $F_E(z) = 1/(1 - \beta z^{-T})$ is used, where $T$ is the nearest integer pitch lag to the closed-loop fractional pitch lag of the subframe, and $\beta$ is the quantized pitch gain of the previous subframe bounded by $[0.0, 0.8]$ . Note that prior to the codebook search, the impulse response $h(n)$ must include the pre-filter $F_E(z)$ . That is, for values of $T$ less than 40, the impulse $h(n)$ is modified according to equation (50a). + +The fixed codebook gain is then found by equation (51). + +#### 5.90 kbit/s mode + +In this case the signal $b(n)$ , used for presetting the amplitudes, is equal to the signal $d(n)$ . Having preset the pulse amplitudes, as explained above, the optimal pulse positions are determined using an exhaustive analysis-by-synthesis search technique. + +A special feature incorporated in the codebook is that the selected codevector is filtered through an adaptive pre-filter $F_E(z)$ which enhances special spectral components in order to improve the synthesized speech quality. Here the filter $F_E(z) = 1/(1 - \beta z^{-T})$ is used, where $T$ is the nearest integer pitch lag to the closed-loop fractional pitch lag of the subframe, and $\beta$ is the quantized pitch gain of the previous subframe bounded by $[0.0, 0.8]$ . Note that prior to the codebook search, the impulse response $h(n)$ must include the pre-filter $F_E(z)$ . That is, for values of $T$ less than 40, the impulse $h(n)$ is modified according to equation (50a). + +The fixed codebook gain is then found by equation (51). + +#### 5.15, 4.75 kbit/s modes + +In this case the signal $b(n)$ , used for presetting the amplitudes, is equal to the signal $d(n)$ . Having preset the pulse amplitudes, as explained above, the optimal pulse positions are determined using an exhaustive analysis-by-synthesis search technique. Note that both subsets are searched. + +A special feature incorporated in the codebook is that the selected codevector is filtered through an adaptive pre-filter $F_E(z)$ which enhances special spectral components in order to improve the synthesized speech quality. Here the filter $F_E(z) = 1/(1 - \beta z^{-T})$ is used, where $T$ is the nearest integer pitch lag to the closed-loop fractional pitch lag of the subframe, and $\beta$ is the quantized pitch gain of the previous subframe for the 5.15 kbit/s mode and the previous odd subframe for the 4.75 kbit/s mode bounded by [0.0, 0.8]. Note that prior to the codebook search, the impulse response $h(n)$ must include the pre-filter $F_E(z)$ . That is, for values of $T$ less than 40, the impulse $h(n)$ is modified according to equation (50a). + +The fixed codebook gain is then found by equation (51). + +## 5.8 Quantization of the adaptive and fixed codebook gains + +### 5.8.1 Adaptive codebook gain limitation in quantization + +If the *GpC\_flag* is set, the limited adaptive codebook gain is used in the gain quantization in clause 5.8.2. The quantization codebook search range is limited to only include adaptive codebook gain values less than $GP_{th}$ . This is performed in the quantization search for all modes. + +### 5.8.2 Quantization of codebook gains + +#### Prediction of the fixed codebook gain (all modes) + +The fixed codebook gain quantization is performed using MA prediction with fixed coefficients. The 4th order MA prediction is performed on the innovation energy as follows. Let $E(n)$ be the mean-removed innovation energy (in dB) at subframe $n$ , and given by: + +$$E(n) = 10 \log \left( \frac{1}{N} \sum_{i=0}^{N-1} c^2(i) \right) - \bar{E}, \quad (53)$$ + +where $N=40$ is the subframe size, $c(i)$ is the fixed codebook excitation, and $\bar{E}$ (in dB) is the mean of the innovation energy. The predicted energy is given by: + +$$\tilde{E}(n) = \sum_{i=1}^4 b_i \hat{R}(n-i), \quad (54)$$ + +where $[b_1 \ b_2 \ b_3 \ b_4] = [0.68 \ 0.58 \ 0.34 \ 0.19]$ are the MA prediction coefficients, and $\hat{R}(k)$ is the quantified prediction error at subframe $k$ . The predicted energy is used to compute a predicted fixed-codebook gain $g'_c$ as in equation (53) (by substituting $E(n)$ by $\tilde{E}(n)$ and $g_c$ by $g'_c$ ). This is done as follows. First, the mean innovation energy is found by: + +$$E_I = 10 \log \left( \frac{1}{N} \sum_{j=0}^{N-1} c^2(j) \right) \quad (55)$$ + +and then the predicted gain $g'_c$ is found by: + +$$g'_c = 10^{0.05(\tilde{E}(n) + \bar{E} - E_I)} \quad (56)$$ + +A correction factor between the gain $g_c$ and the estimated one $g'_c$ is given by: + +$$\gamma_{gc} = g_c / g'_c \quad (57)$$ + +Note that the prediction error is given by: + +$$R(n) = E(n) - \tilde{E}(n) = 20 \log(\gamma_{gc}) \quad (58)$$ + +#### 12.2 kbit/s mode + +The correction factor $\gamma_{gc}$ is computed using a mean energy value, $\bar{E} = 36$ dB. The correction factor $\gamma_{gc}$ is quantified using a 5-bit codebook. The quantization table search is performed by minimizing the error: + +$$E_Q = \left( g_c - \hat{\gamma}_{gc} g'_c \right)^2 \quad (59)$$ + +Once the optimum value $\hat{\gamma}_{gc}$ is chosen, the quantified fixed codebook gain is given by $\hat{g}_c = \hat{\gamma}_{gc} g'_c$ . + +#### 10.2 kbit/s mode + +The correction factor $\gamma_{gc}$ is computed using a mean energy value, $\bar{E} = 33$ dB. The adaptive codebook gain $g_p$ and the correction factor $\gamma_{gc}$ are jointly vector quantized using a 7-bit codebook. The gain codebook search is performed by minimizing equation (63). + +#### 7.95 kbit/s mode + +The correction factor $\gamma_{gc}$ is computed using a mean energy value, $\bar{E} = 36$ dB. The same scalar codebooks as for the 12.2 kbit/s mode is used for quantization of the adaptive codebook gain $g_p$ and the correction factor $\gamma_{gc}$ . The search of the codebooks starts with finding 3 candidates for the adaptive codebook gain. These candidates are the best codebook value in scalar quantization and the two adjacent codebook values. These 3 candidates are searched together with the correction factor codebook minimizing the term of equation (63). + +An adaptor based on the coding gain in the adaptive codebook decides if the coding gain is low. If this is the case, the correction factor codebook is searched once more minimizing a modified criterion in order to find a new quantized fixed codebook gain. The modified criterion is given by: + +$$E_{\text{mod}} = (1 - \alpha) \cdot \|\mathbf{c}\|^2 \cdot \left( g_c - \hat{\gamma}_{gc} \cdot g'_c \right)^2 + \alpha \cdot \left( \sqrt{E_{\text{res}}} - \sqrt{E_{\text{exc}}} \right)^2 \quad (60)$$ + +where $E_{\text{res}}$ and $E_{\text{exc}}$ are the energy (the squared norm) of the LP residual and the total excitation, respectively. + +The criterion is searched with the already quantized adaptive codebook gain and the correction factor $\hat{\gamma}_{gc}$ that minimizes (60) is selected. The balance factor $\alpha$ decides the amount of energy matching in the modified criterion. This factor is adaptively decided based on the coding gain in the adaptive codebook as computed by: + +$$ag = 10 \cdot \log_{10} \frac{\|\mathbf{res}_{LP}\|^2}{\|\mathbf{res}_{LP} - \mathbf{v}\|^2} \quad (61)$$ + +If the coding gain $ag$ is less than 1 dB, the modified criterion is employed, except when an onset is detected. An onset is said to be detected if the fixed codebook gain in the current subframe is more than twice the value of the + +fixed codebook gain in the previous subframe. A hangover of 8 subframes is used in the onset detection so that the modified criterion is not used for the next 7 subframes either if an onset is detected. The balance factor $\alpha$ is computed from the median filtered adaptive coding gain. The current and the $ag$ -values for the previous 4 subframes are median filtered to get $ag_m$ . The $\alpha$ -factor is computed by: + +$$\alpha = \begin{cases} 0 & ag_m > 2 \\ 0.5 \cdot (1 - 0.5 \cdot ag_m) & 0 < ag_m < 2 \\ 0.5 & ag_m < 0 \end{cases} \quad (62)$$ + +#### 7.40 kbit/s mode + +The correction factor $\gamma^{gc}$ is computed using a mean energy value, $\bar{E} = 30$ dB. The adaptive codebook gain $g_p$ and the correction factor $\gamma^{gc}$ are jointly vector quantized using a 7-bit codebook. The gain codebook search is performed by minimizing the square of the weighted error between original and reconstructed speech which is given by + +$$E = \|\mathbf{x} - g_p \mathbf{y} - g_c \mathbf{z}\|^2 = \mathbf{x}^t \mathbf{x} + g_p^2 \mathbf{y}^t \mathbf{y} + g_c^2 \mathbf{z}^t \mathbf{z} - 2g_p \mathbf{x}^t \mathbf{y} - 2g_c \mathbf{x}^t \mathbf{z} + 2g_p g_c \mathbf{y}^t \mathbf{z} \quad (63)$$ + +where $\mathbf{x}$ is the target vector, $\mathbf{y}$ is the filtered adaptive codebook vector, and $\mathbf{z}$ is the filtered fixed codebook vector. + +#### 6.70 kbit/s mode + +The correction factor $\gamma^{gc}$ is computed using a mean energy value, $\bar{E} = 28.75$ dB. The adaptive codebook gain $g_p$ and the correction factor $\gamma^{gc}$ are jointly vector quantized using a 7-bit codebook. The gain codebook search is performed by minimizing equation (63). + +#### 5.90, 5.15 kbit/s modes + +The correction factor $\gamma^{gc}$ is computed using a mean energy value, $\bar{E} = 33$ dB. The adaptive codebook gain $g_p$ and the correction factor $\gamma^{gc}$ are jointly vector quantized using a 6-bit codebook. The gain codebook search is performed by minimizing equation (63). + +#### 4.75 kbit/s mode + +The correction factors are computed using a mean energy value, $\bar{E} = 33$ dB. The adaptive codebook gains and the correction factors are jointly vector quantized every 10 ms. This is done by minimizing a weighted sum of the error criterion (63) for each of the two subframes. The default values on the weighing factors are 1. If the energy of the second subframe is more than two times the energy of the first subframe, the weight of the first subframe is set to 2. If the energy of the first subframe is more than four times the energy of the second subframe, the weight of the second subframe is set to 2. + +### 5.8.3 Update past quantized adaptive codebook gain buffer (all modes) + +After the gain quantization, the buffer with past adaptive codebook gains is updated, regardless of the value of the $GpC\_flag$ . That is: + +$$\hat{g}_p(n-i) = \hat{g}_p(n-i+1), \quad i = 7, \dots, 1$$ + +## 5.9 Memory update (all modes) + +An update of the states of the synthesis and weighting filters is needed in order to compute the target signal in the next subframe. + +After the two gains are quantified, the excitation signal, $u(n)$ , in the present subframe is found by: + +$$u(n) = \hat{g}_p v(n) + \hat{g}_c c(n), \quad n = 0, \dots, 39, \quad (64)$$ + +where $\hat{g}_p$ and $\hat{g}_c$ are the quantified adaptive and fixed codebook gains, respectively, $v(n)$ the adaptive codebook vector (interpolated past excitation), and $c(n)$ is the fixed codebook vector (algebraic code including pitch sharpening). The states of the filters can be updated by filtering the signal $res_{LP}(n) - u(n)$ (difference between residual and excitation) through the filters $1/\hat{A}(z)$ and $A(z/\gamma_1)/A(z/\gamma_2)$ for the 40-sample subframe and saving the states of the filters. This would require 3 filterings. A simpler approach which requires only one filtering is as follows. The local synthesized speech, $\hat{s}(n)$ , is computed by filtering the excitation signal through $1/\hat{A}(z)$ . The output of the filter due to the input $res_{LP}(n) - u(n)$ is equivalent to $e(n) = s(n) - \hat{s}(n)$ . So the states of the synthesis filter $1/\hat{A}(z)$ are given by $e(n)$ , $n = 30, \dots, 39$ . Updating the states of the filter $e(n) = s(n) - \hat{s}(n)$ can be done by filtering the error signal $e(n)$ through this filter to find the perceptually weighted error $e_w(n)$ . However, the signal $e_w(n)$ can be equivalently found by: + +$$e_w(n) = x(n) - \hat{g}_p y(n) - \hat{g}_c z(n), \quad (65)$$ + +Since the signals $x(n)$ , $y(n)$ , and $z(n)$ are available, the states of the weighting filter are updated by computing $e_w(n)$ as in equation (65) for $n = 30, \dots, 39$ . This saves two filterings. + +#### 4.75 kbit/s mode + +The memory update in the first and third subframes use the unquantized gains in equation (64). After the second and fourth subframes respectively, when the gains are quantized, the state is recalculated using the quantized gains. + +# 6 Functional description of the decoder + +The function of the decoder consists of decoding the transmitted parameters (LP parameters, adaptive codebook vector, adaptive codebook gain, fixed codebook vector, fixed codebook gain) and performing synthesis to obtain the reconstructed speech. The reconstructed speech is then post-filtered and upscaled. The signal flow at the decoder is shown in figure 4. + +## 6.1 Decoding and speech synthesis + +The decoding process is performed in the following order: + +**Decoding of LP filter parameters:** The received indices of LSP quantization are used to reconstruct the quantified LSP vectors. The interpolation described in clause 5.2.6 is performed to obtain 4 interpolated LSP vectors (corresponding to 4 subframes). For each subframe, the interpolated LSP vector is converted to LP filter coefficient domain $a_k$ , which is used for synthesizing the reconstructed speech in the subframe. + +The following steps are repeated for each subframe: + +- 1) **Decoding of the adaptive codebook vector:** The received pitch index (adaptive codebook index) is used to find the integer and fractional parts of the pitch lag. The adaptive codebook vector $v(n)$ is found by interpolating the past excitation $u(n)$ (at the pitch delay) using the FIR filter described in clause 5.6. +- 2) **Decoding of the innovative codebook vector:** The received algebraic codebook index is used to extract the positions and amplitudes (signs) of the excitation pulses and to find the algebraic codevector $c(n)$ . If the integer part of the pitch lag, $T$ , is less than the subframe size 40, the pitch sharpening procedure is applied + +which translates into modifying $c(n)$ by $c(n) = c(n) + \beta c(n - T)$ , where $\beta$ is the decoded pitch gain, $\hat{g}_p$ , bounded by $[0.0, 1.0]$ or $[0.0, 0.8]$ , depending on mode. + +- 3) **Decoding of the adaptive and fixed codebook gains:** In case of scalar quantization of the gains (12.2 kbit/s and 7.95 kbit/s modes) the received indices are used to readily find the quantified adaptive codebook gain, $\hat{g}_p$ , and the quantified fixed codebook gain correction factor, $\hat{\gamma}_{gc}$ , from the corresponding quantization tables. In case of vector quantization of the gains (all other modes), the received index gives both the quantified adaptive codebook gain, $\hat{g}_p$ , and the quantified fixed codebook gain correction factor, $\hat{g}_{gc}$ . The estimated fixed codebook gain $g'_c$ is found as described in clause 5.7. First, the predicted energy is found by: + +$$\tilde{E}(n) = \sum_{i=1}^4 b_i \hat{R}(n - i) \quad (66)$$ + +and then the mean innovation energy is found by: + +$$E_I = 10 \log \left( \frac{1}{N} \sum_{j=0}^{N-1} c^2(j) \right) \quad (67)$$ + +The predicted gain $g'_c$ is found by: + +$$g'_c = 10^{0.05(\tilde{E}(n) + \bar{E} - E_I)} \quad (68)$$ + +The quantified fixed codebook gain is given by: + +$$\hat{g}_c = \gamma_{gc} g'_c \quad (69)$$ + +- 4) **Smoothing of the fixed codebook gain (10.2, 6.70, 5.90, 5.15, 4.75 kbit/s modes):** An adaptive smoothing of the fixed codebook gain is performed to avoid unnatural fluctuations in the energy contour. The smoothing is based on a measure of the stationarity of the short-term spectrum in the **q** domain. The smoothing strength is computed from this measure. An averaged **q**-value is computed for each frame $n$ by: + +$$\bar{q}(n) = 0.84 \cdot \bar{q}(n - 1) + 0.16 \cdot \hat{q}_4(n) \quad (70)$$ + +For each subframe $m$ , a difference measure between the averaged vector and the quantized and interpolated vector is computed by: + +$$diff_m = \sum_j \sum_m \frac{|\bar{q}^{(j)}(n) - \hat{q}_m^{(j)}(n)|}{\bar{q}^{(j)}(n)} \quad (71)$$ + +where $j$ runs over the 10 LSPs. Furthermore, a smoothing factor, $k_m$ , is computed by: + +$$k_m = \min(K_2, \max(0, diff_m - K_1)) / K_2 \quad (72)$$ + +where the constants are set to $K_1 = 0.4$ and $K_2 = 0.25$ . A hangover period of 40 subframes is used where the $k_m$ -value is set 1.0 if the $diff_m$ has been above 0.65 for 10 consecutive frames. A value of 1.0 corresponds to no smoothing. An averaged fixed codebook gain value is computed for each subframe by: + +$$\bar{g}(m) = \frac{1}{5} \sum_{i=0}^4 \hat{g}_c(m-i) \quad (73)$$ + +The fixed codebook gain used for synthesis is now replaced by a smoothed value given by: + +$$\hat{g}_c = \hat{g}_c \cdot k_m + \bar{g}_c \cdot (1 - k_m) \quad (74)$$ + +- 5) **Anti-sparseness processing (7.95, 6.70, 5.90, 5.15, 4.75 kbit/s modes):** An adaptive anti-sparseness post-processing procedure is applied to the fixed codebook vector $c(n)$ in order to reduce perceptual artefacts arising from the sparseness of the algebraic fixed codebook vectors with only a few non-zero samples per subframe. The anti-sparseness processing consists of circular convolution of the fixed codebook vector with an impulse response. Three pre-stored impulse responses are used and a number $impNr = 0, 1, 2$ is set to select one of them. A value of 2 corresponds to no modification, a value of 1 corresponds to medium modification, while a value of 0 corresponds to strong modification. The selection of the impulse response is performed adaptively from the adaptive and fixed codebook gains. The following procedure is employed: + +if $\hat{g}_p < 0.6$ then + $impNr = 0$ ; +else if $\hat{g}_p < 0.9$ then + $impNr = 1$ ; +else + $impNr = 2$ ; + +Detect onset by comparing the fixed codebook gain to the previous fixed codebook gain. If the current value is more than twice the previous value an onset is detected. + +If not onset and $impNr = 0$ , the median filtered value of the current and the previous 4 adaptive codebook gains are computed. If this value is less than 0.6, $impNr = 0$ . + +If not onset, the $impNr$ -value is restricted to increase by one step from the previous subframe. + +If an onset is declared, the $impNr$ -value is increased by one if it is less than 2. + +- 6) **Computing the reconstructed speech:** The excitation at the input of the synthesis filter is given by: + +$$u(n) = \hat{g}_p v(n) + \hat{g}_c c(n) \quad (75)$$ + +Before the speech synthesis, a post-processing of excitation elements is performed. This means that the total excitation is modified by emphasizing the contribution of the adaptive codebook vector: + +$$\hat{u}(n) = \begin{cases} u(n) + 0.25 \beta \hat{g}_p v(n), & \hat{g}_p > 0.5, 12.2 \text{ kbit/s mode} \\ u(n) + 0.5 \beta \hat{g}_p v(n), & \hat{g}_p > 0.5, \text{ all other modes} \\ u(n) & \hat{g}_p \leq 0.5 \end{cases} \quad (76)$$ + +Adaptive gain control (AGC) is used to compensate for the gain difference between the non-emphasized excitation $u(n)$ and emphasized excitation $\hat{u}(n)$ . The gain scaling factor $\eta$ for the emphasized excitation is computed by: + +$$\eta = \begin{cases} \sqrt{\frac{\sum_{n=0}^{39} u^2(n)}{\sum_{n=0}^{39} \hat{u}^2(n)}}, & \hat{g}_p > 0.5, \\ 1.0, & \hat{g}_p \leq 0.5. \end{cases} \quad (77)$$ + +The gain-scaled emphasized excitation signal $\hat{u}'(n)$ is given by: + +$$\hat{u}'(n) = \hat{u}(n)\eta. \quad (78)$$ + +The reconstructed speech for the subframe of size 40 is given by: + +$$\hat{s}(n) = \hat{u}'(n) - \sum_{i=1}^{10} \hat{a}_i \hat{s}(n-i), \quad n = 0, \dots, 39 \quad (79)$$ + +where $\hat{a}_i$ are the interpolated LP filter coefficients. + +- 7) **Additional instability protection:** An additional instability protection is implemented in the speech decoder which is monitoring overflows in the synthesis filter. If an overflow has occurred in the synthesis part, the whole adaptive codebook memory, $v(n), n=-(143+11), \dots, 39$ is scaled down by a factor of 4, and the synthesis filtering is repeated using this down-scaled memory. I.e. in this case step 6) is repeated, except that the post-processing in (76) - (78) of the excitation signal is by-passed. + +The synthesized speech $\hat{s}(n)$ is then passed through an adaptive postfilter which is described in the following clause. + +## 6.2 Post-processing + +### 6.2.1 Adaptive post-filtering (all modes) + +The adaptive postfilter is the cascade of two filters: a formant postfilter, and a tilt compensation filter. The postfilter is updated every subframe of 5 ms. + +The formant postfilter is given by: + +$$H_f(z) = \frac{\hat{A}(z/\gamma_n)}{\hat{A}(z/\gamma_d)} \quad (80)$$ + +where $\hat{A}(z)$ is the received quantified (and interpolated) LP inverse filter (LP analysis is not performed at the decoder), and the factors $\gamma_n$ and $\gamma_d$ control the amount of the formant post-filtering. + +Finally, the filter $H_t(z)$ compensates for the tilt in the formant postfilter $H_f(z)$ and is given by: + +$$H_t(z) = 1 - \mu z^{-1} \quad (81)$$ + +where $\mu = \gamma_t k_1'$ is a tilt factor, with $k_1'$ being the first reflection coefficient calculated on the truncated ( $L_h = 22$ ) impulse response, $h_f(n)$ , of the filter $\hat{A}(z/\gamma_n)/\hat{A}(z/\gamma_d)$ . $k_1'$ is given by: + +$$k'_1 = \frac{r_h(1)}{r_h(0)}; \quad r_h(i) = \sum_{j=0}^{L_h-i-1} h_f(j)h_f(j+i) \quad (82)$$ + +The post-filtering process is performed as follows. First, the synthesized speech $\hat{s}(n)$ is inverse filtered through $\hat{A}(z/\gamma_n)$ to produce the residual signal $\hat{r}(n)$ . The signal $\hat{r}(n)$ is filtered by the synthesis filter $1/\hat{A}(z/\gamma_d)$ . Finally, the signal at the output of the synthesis filter $1/\hat{A}(z/\gamma_d)$ is passed to the tilt compensation filter $H_t(z)$ resulting in the post-filtered speech signal $\hat{s}_f(n)$ . + +Adaptive gain control (AGC) is used to compensate for the gain difference between the synthesized speech signal $\hat{s}(n)$ and the post-filtered signal $\hat{s}_f(n)$ . The gain scaling factor $\gamma_{sc}$ for the present subframe is computed by: + +$$\gamma_{sc} = \sqrt{\frac{\sum_{n=0}^{39} \hat{s}^2(n)}{\sum_{n=0}^{39} \hat{s}_f^2(n)}} \quad (83)$$ + +The gain-scaled post-filtered signal $\hat{s}'(n)$ is given by: + +$$\hat{s}'(n) = \beta_{sc}(n)\hat{s}_f(n) \quad (84)$$ + +where $\beta_{sc}(n)$ is updated in sample-by-sample basis and given by: + +$$\beta_{sc}(n) = \alpha\beta_{sc}(n-1) + (1-\alpha)\gamma_{sc} \quad (85)$$ + +where $\alpha$ is a AGC factor with value of 0.9. + +#### 12.2, 10.2 kbit/s modes + +The adaptive post-filtering factors are given by: $\gamma_n = 0.7$ , $\gamma_d = 0.75$ and + +$$\gamma_t = \begin{cases} 0.8, & k'_1 > 0, \\ 0, & \text{otherwise.} \end{cases} \quad (86)$$ + +#### 7.95, 7.40, 6.70, 5.90, 5.15, 4.75 kbit/s modes + +The adaptive post-filtering factors are given by: $\gamma_n = 0.55$ , $\gamma_d = 0.7$ and $\gamma_t = 0.8$ . + +### 6.2.2 High-pass filtering and up-scaling (all modes) + +The high-pass filter serves as a precaution against undesired low frequency components. A filter cut-off frequency of 60 Hz is used, and the filter is given by + +$$H_{h2}(z) = \frac{0.939819335 - 1.879638672z^{-1} + 0.939819335z^{-2}}{1 - 1.933105469z^{-1} + 0.935913085z^{-2}} \quad (87)$$ + +Up-scaling consists of multiplying the post-filtered speech by a factor of 2 to compensate for the down-scaling by 2 which is applied to the input signal. + +# 7 Detailed bit allocation of the adaptive multi-rate codec + +The detailed allocation of the bits in the adaptive multi-rate speech encoder is shown for each mode in table 9a-9h. These tables show the order of the bits produced by the speech encoder. Note that the most significant bit (MSB) of each codec parameter is always sent first. + +**Table 9a: Source encoder output parameters in order of occurrence and bit allocation within the speech frame of 244 bits/20 ms, 12.2 kbit/s mode.** + +| Bits (MSB-LSB) | Description | +|----------------|------------------------------------------------------------------| +| s1 - s7 | index of 1 st LSF submatrix | +| s8 - s15 | index of 2 nd LSF submatrix | +| s16 - s23 | index of 3 rd LSF submatrix | +| s24 | sign of 3 rd LSF submatrix | +| s25 - s32 | index of 4 th LSF submatrix | +| s33 - s38 | index of 5 th LSF submatrix | +| subframe 1 | | +| s39 - s47 | adaptive codebook index | +| s48 - s51 | adaptive codebook gain | +| s52 | sign information for 1 st and 6 th pulses | +| s53 - s55 | position of 1 st pulse | +| s56 | sign information for 2 nd and 7 th pulses | +| s57 - s59 | position of 2 nd pulse | +| s60 | sign information for 3 rd and 8 th pulses | +| s61 - s63 | position of 3 rd pulse | +| s64 | sign information for 4 th and 9 th pulses | +| s65 - s67 | position of 4 th pulse | +| s68 | sign information for 5 th and 10 th pulses | +| s69 - s71 | position of 5 th pulse | +| s72 - s74 | position of 6 th pulse | +| s75 - s77 | position of 7 th pulse | +| s78 - s80 | position of 8 th pulse | +| s81 - s83 | position of 9 th pulse | +| s84 - s86 | position of 10 th pulse | +| s87 - s91 | fixed codebook gain | +| subframe 2 | | +| s92 - s97 | adaptive codebook index (relative) | +| s98 - s141 | same description as s48 - s91 | +| subframe 3 | | +| s142 - s194 | same description as s39 - s91 | +| subframe 4 | | +| s195 - s244 | same description as s92 - s141 | + +**Table 9b: Source encoder output parameters in order of occurrence and bit allocation within the speech frame of 204 bits/20 ms, 10.2 kbit/s mode.** + +| Bits (MSB-LSB) | Description | +|----------------|-----------------------------------------------------------------------------| +| s1 – s8 | index of 1 st LSF subvector | +| s9 – s17 | index of 2 nd LSF subvector | +| s18 – s26 | index of 3 rd LSF subvector | +| subframe 1 | | +| s27 – s34 | adaptive codebook index | +| s35 | sign information for 1 st and 5 th pulses | +| s36 | sign information for 2 nd and 6 th pulses | +| s37 | sign information for 3 rd and 7 th pulses | +| s38 | sign information for 4 th and 8 th pulses | +| s39-s48 | position for 1 st , 2 nd , and 5 th pulses | +| s49-s58 | position for 3 rd , 6 th , and 7 th pulses | +| s59-s65 | position for 4 th and 8 th pulses | +| s66 – s72 | codebook gains | +| subframe 2 | | +| s73 – s77 | adaptive codebook index (relative) | +| s78 – s115 | same description as s35 – s72 | +| subframe 3 | | +| s116 – s161 | same description as s27 – s72 | +| subframe 4 | | +| s162 – s204 | same description as s73 – s115 | + +**Table 9c: Source encoder output parameters in order of occurrence and bit allocation within the speech frame of 159 bits/20 ms, 7.95 kbit/s mode.** + +| Bits (MSB-LSB) | Description | +|----------------|--------------------------------------------| +| s1 – s9 | index of 1 st LSF subvector | +| s10 – s18 | index of 2 nd LSF subvector | +| s19 – s27 | index of 3 rd LSF subvector | +| subframe 1 | | +| s28 – s35 | adaptive codebook index | +| s36 – s39 | position of 4 th pulse | +| s40 – s42 | position of 3 rd pulse | +| s43 – s45 | position of 2 nd pulse | +| s46 – s48 | position of 1 st pulse | +| s49 | sign information for 4 th pulse | +| s50 | sign information for 3 rd pulse | +| s51 | sign information for 2 nd pulse | +| s52 | sign information for 1 st pulse | +| s53 – s56 | adaptive codebook gain | +| s57 – s61 | fixed codebook gain | +| subframe 2 | | +| s62 – s67 | adaptive codebook index (relative) | +| s68 – s93 | same description as s36 – s61 | +| subframe 3 | | +| s94 – s127 | same description as s28 – s61 | +| subframe 4 | | +| s128 – s159 | same description as s62 – s93 | + +**Table 9d: Source encoder output parameters in order of occurrence and bit allocation within the speech frame of 148 bits/20 ms, 7.40 kbit/s mode.** + +| Bits (MSB-LSB) | Description | +|----------------|--------------------------------------------| +| s1 – s8 | index of 1 st LSF subvector | +| s9 - s17 | index of 2 nd LSF subvector | +| s18 – s26 | index of 3 rd LSF subvector | +| subframe 1 | | +| s27 – s34 | adaptive codebook index | +| s35 – s38 | position of 4 th pulse | +| s39 – s41 | position of 3 rd pulse | +| s42 - s44 | position of 2 nd pulse | +| s45 – s47 | position of 1 st pulse | +| s48 | sign information for 4 th pulse | +| s49 | sign information for 3 rd pulse | +| s50 | sign information for 2 nd pulse | +| s51 | sign information for 1 st pulse | +| s52 – s58 | codebook gains | +| subframe 2 | | +| s59 – s63 | adaptive codebook index (relative) | +| s64 – s87 | same description as s35 – s58 | +| subframe 3 | | +| s88 – s119 | same description as s27 – s58 | +| subframe 4 | | +| s120 – s148 | same description as s59 – s87 | + +**Table 9e: Source encoder output parameters in order of occurrence and bit allocation within the speech frame of 134 bits/20 ms, 6.70 kbit/s mode.** + +| Bits (MSB-LSB) | Description | +|----------------|--------------------------------------------| +| s1 – s8 | index of 1 st LSF subvector | +| s9 - s17 | index of 2 nd LSF subvector | +| s18 – s26 | index of 3 rd LSF subvector | +| subframe 1 | | +| s27 – s34 | adaptive codebook index | +| s35 – s38 | position of 3 rd pulse | +| s39 – s42 | position of 2 nd pulse | +| s43 – s45 | position of 1 st pulse | +| s46 | sign information for 3 rd pulse | +| s47 | sign information for 2 nd pulse | +| s48 | sign information for 1 st pulse | +| s49 – s55 | codebook gains | +| subframe 2 | | +| s56 – s59 | adaptive codebook index (relative) | +| s60 – s80 | same description as s35 – s55 | +| subframe 3 | | +| s81 – s109 | same description as s27 – s55 | +| subframe 4 | | +| s110 – s134 | same description as s56 – s80 | + +**Table 9f: Source encoder output parameters in order of occurrence and bit allocation within the speech frame of 118 bits/20 ms, 5.90 kbit/s mode.** + +| Bits (MSB-LSB) | Description | +|----------------|--------------------------------------------| +| s1 – s8 | index of 1 st LSF subvector | +| s9 - s17 | index of 2 nd LSF subvector | +| s18 – s26 | index of 3 rd LSF subvector | +| subframe 1 | | +| s27 – s34 | adaptive codebook index | +| s35 – s39 | position of 2 nd pulse | +| s40 – s43 | position of 1 st pulse | +| s44 | sign information for 2 nd pulse | +| s45 | sign information for 1 st pulse | +| s46 – s51 | codebook gains | +| subframe 2 | | +| s52 – s55 | adaptive codebook index (relative) | +| s56 – s72 | same description as s35 – s51 | +| subframe 3 | | +| s73 – s97 | same description as s27 – s51 | +| subframe 4 | | +| s98 – s118 | same description as s52 – s72 | + +**Table 9g: Source encoder output parameters in order of occurrence and bit allocation within the speech frame of 103 bits/20 ms, 5.15 kbit/s mode.** + +| Bits (MSB-LSB) | Description | +|----------------|--------------------------------------------| +| s1 – s8 | index of 1 st LSF subvector | +| s9 - s16 | index of 2 nd LSF subvector | +| s17 – s23 | index of 3 rd LSF subvector | +| subframe 1 | | +| s24 – s31 | adaptive codebook index | +| s32 | position subset | +| s33 – s35 | position of 2 nd pulse | +| s36 – s38 | position of 1 st pulse | +| s39 | sign information for 2 nd pulse | +| s40 | sign information for 1 st pulse | +| s41 – s46 | codebook gains | +| subframe 2 | | +| s47 – s50 | adaptive codebook index (relative) | +| s51 – s65 | same description as s32 – s46 | +| subframe 3 | | +| s66 – s84 | same description as s47 – s65 | +| subframe 4 | | +| s85 – s103 | same description as s47 – s65 | + +**Table 9h: Source encoder output parameters in order of occurrence and bit allocation within the speech frame of 95 bits/20 ms, 4.75 kbit/s mode.** + +| Bits (MSB-LSB) | Description | +|----------------|--------------------------------------------| +| s1 – s8 | index of 1 st LSF subvector | +| s9 - s16 | index of 2 nd LSF subvector | +| s17 – s23 | index of 3 rd LSF subvector | +| subframe 1 | | +| s24 – s31 | adaptive codebook index | +| s32 | position subset | +| s33 – s35 | position of 2 nd pulse | +| s36 – s38 | position of 1 st pulse | +| s39 | sign information for 2 nd pulse | +| s40 | sign information for 1 st pulse | +| s41 – s48 | codebook gains | +| subframe 2 | | +| s49 – s52 | adaptive codebook index (relative) | +| s53 – s61 | same description as s32 – s40 | +| subframe 3 | | +| s62 - s65 | same description as s49 – s52 | +| s66 – s82 | same description as s32– s48 | +| subframe 4 | | +| s83 – s95 | same description as s49 – s61 | + +# 8 Homing sequences + +## 8.1 Functional description + +The adaptive multi-rate speech codec is described in a bit-exact arithmetic to allow for easy type approval as well as general testing purposes of the adaptive multi-rate speech codec. + +The response of the codec to a predefined input sequence can only be foreseen if the internal state variables of the codec are in a predefined state at the beginning of the experiment. Therefore, the codec has to be put in a so called home state before a bit-exact test can be performed. This is usually done by a reset (a procedure in which the internal state variables of the codec are set to their defined initial values). The codec mode of the speech encoder and speech decoder shall be set to the tested codec mode by external means at reset. + +To allow a reset of the codec in remote locations, special homing frames have been defined for the encoder and the decoder, thus enabling a codec homing by inband signalling. + +The codec homing procedure is defined in such a way, that in either direction (encoder or decoder) the homing functions are called after processing the homing frame that is input. The output corresponding to the first homing frame is therefore dependent on the used codec mode and the codec state when receiving that frame and hence usually not known. The response of the encoder to any further homing frame is by definition the corresponding decoder homing frame for the used codec mode. The response of the decoder to any further homing frame is by definition the encoder homing frame. This procedure allows homing of both, the encoder and decoder from either side, if a loop back configuration is implemented, taking proper framing into account. + +## 8.2 Definitions + +**Encoder homing frame:** The encoder homing frame consists of 160 identical samples, each 13 bits long, with the least significant bit set to "one" and all other bits set to "zero". When written to 16-bit words with left justification, the samples have a value of 0008 hex. The speech decoder has to produce this frame as a response to the second and any further decoder homing frame if at least two decoder homing frames were input to the decoder consecutively. The encoder homing frame is identical for all codec modes. + +**Decoder homing frame:** There exist eight different decoder homing frames, which correspond to the eight AMR codec modes. Using one of these codec modes, the corresponding decoder homing frame is the natural response of the speech encoder to the second and any further encoder homing frame if at least two encoder homing frames were input to the encoder consecutively. In [4], for each decoder homing frame the parameter values are given. + +## 8.3 Encoder homing + +Whenever the adaptive multi-rate speech encoder receives at its input an encoder homing frame exactly aligned with its internal speech frame segmentation, the following events take place: + +- Step 1: The speech encoder performs its normal operation including VAD and SCR and produces in accordance with the used codec mode a speech parameter frame at its output which is in general unknown. But if the speech encoder was in its home state at the beginning of that frame, then the resulting speech parameter frame is identical to that decoder homing frame, which corresponds to the used codec mode (this is the way how the decoder homing frames were constructed). +- Step 2: After successful termination of that operation the speech encoder provokes the homing functions for all sub-modules including VAD and SCR and sets all state variables into their home state. On the reception of the next input frame, the speech encoder will start from its home state. + +NOTE: Applying a sequence of N encoder homing frames will cause at least N-1 decoder homing frames at the output of the speech encoder. + +## 8.4 Decoder homing + +Whenever the speech decoder receives at its input a decoder homing frame, which corresponds to the used codec mode, then the following events take place: + +- Step 1: The speech decoder performs its normal operation and produces a speech frame at its output which is in general unknown. But if the speech decoder was in its home state at the beginning of that frame, then the resulting speech frame is replaced by the encoder homing frame. This would not naturally be the case but is forced by this definition here. +- Step 2: After successful termination of that operation the speech decoder provokes the homing functions for all sub-modules including the comfort noise generator and sets all state variables into their home state. On the reception of the next input frame, the speech decoder will start from its home state. + +NOTE 1: Applying a sequence of N decoder homing frames will cause at least N-1 encoder homing frames at the output of the speech decoder. + +NOTE 2: By definition (!) the first frame of each decoder test sequence must differ from the decoder homing frame at least in one bit position within the parameters for LPC and first subframe. Therefore, if the decoder is in its home state, it is sufficient to check only these parameters to detect a subsequent decoder homing frame. This definition is made to support a delay-optimized implementation in the TRAUP uplink direction. + +![Simplified block diagram of the CELP synthesis model showing adaptive and fixed codebooks, gain factors gp and gc, a summation node, LP synthesis filter 1/A(z), and post-filtering.](187d05bf7ead21e1394b61320d8b3632_img.jpg) + +The diagram illustrates the CELP synthesis model. It features two input codebooks: an *adaptive codebook* (represented by a waveform) and a *fixed codebook* (represented by discrete impulses). The adaptive codebook output is scaled by a gain factor $g_p$ to produce $v(n)$ . The fixed codebook output is scaled by a gain factor $g_c$ to produce $c(n)$ . These two signals are summed at a node labeled with a '+' to form the excitation signal $u(n)$ . This signal is then processed by an *LP synthesis* filter, represented by the transfer function $\frac{1}{A(z)}$ , resulting in $\hat{s}(n)$ . Finally, $\hat{s}(n)$ is passed through a *post-filtering* block to produce the final output $\hat{s}'(n)$ . A feedback loop is shown from the output $\hat{s}'(n)$ back to the input of the adaptive codebook. + +Simplified block diagram of the CELP synthesis model showing adaptive and fixed codebooks, gain factors gp and gc, a summation node, LP synthesis filter 1/A(z), and post-filtering. + +Figure 2: Simplified block diagram of the CELP synthesis model + +![Simplified block diagram of the adaptive multi-rate encoder showing frame and subframe processing stages.](40a8c30f7ea5ecea4912e040c97c5b9c_img.jpg) + +The diagram illustrates the adaptive multi-rate encoder architecture, divided into **frame** and **subframe** processing stages. + +**frame** processing includes: + +- Pre-processing**: Input $s(n)$ is processed here. +- LPC analysis (twice per frame)**: + - $s(n)$ is processed through **windowing and autocorrelation $R[\cdot]$** , **Levinson-Durbin $R[\cdot] \rightarrow A(z)$** , and **$A(z) \rightarrow LSP$** blocks. + - LSP quantization** produces **LSP indices**. + - interpolation for the 4 subframes $LSP \rightarrow \hat{A}(z)$** generates the interpolated LPC filter coefficients. +- Open-loop pitch search (twice per frame)**: + - $s(n)$ is processed through **interpolation for the 4 subframes $LSP \rightarrow A(z)$** to get $A(z)$ . + - $A(z)$ is used to **compute weighted speech (4 subframes)** and **find open-loop pitch**. + +**subframe** processing (repeated for each subframe) includes: + +- Adaptive codebook search**: + - Inputs: $A(z)$ , $\hat{A}(z)$ , $T_o$ , $h(n)$ , and $x(n)$ . + - Blocks: **compute target for adaptive codebook**, **find best delay and gain**, **quantize LTP-gain**, and **compute adaptive codebook contribution**. + - Outputs: **pitch index**, **LTP gain index**, and $x(n)$ . +- Innovative codebook search**: + - Inputs: $x(n)$ , $x_2(n)$ , and $h(n)$ . + - Blocks: **compute target for innovation** and **find best innovation**. + - Outputs: **code index** and $x_2(n)$ . +- Filter memory update**: + - Inputs: **compute excitation** and **fixed codebook gain quantization**. + - Block: **update filter memories for next subframe**. + +Additional blocks and signals: + +- compute impulse response**: Takes $A(z)$ and $\hat{A}(z)$ as input to produce $h(n)$ . +- fixed codebook gain quantization**: Produces the **fixed codebook gain index**. + +Simplified block diagram of the adaptive multi-rate encoder showing frame and subframe processing stages. + +Figure 3: Simplified block diagram of the adaptive multi-rate encoder + +![Simplified block diagram of the adaptive multi-rate decoder. The diagram is divided into three sections: frame, subframe, and post-processing. The 'frame' section takes 'LSP indices' as input to a 'decode LSP' block. This block outputs to an 'interpolation of LSP for the 4 subframes' block, which in turn outputs to an 'LSP -> A_hat(z)' block. The 'subframe' section takes 'pitch index' to a 'decode adaptive codebook' block, 'gains indices' to a 'decode gains' block, and 'code index' to a 'decode innovative codebook' block. The outputs of these three blocks are combined in a 'construct excitation' block. The 'LSP -> A_hat(z)' block from the 'frame' section and the 'construct excitation' block from the 'subframe' section both feed into a 'synthesis filter' block. The output of the 'synthesis filter' is labeled s_hat(n). This signal then enters the 'post-processing' section, which contains a 'post filter' block that outputs the final signal s'_hat(n).](a0e8fe7862a6d7341faf5dac275277cc_img.jpg) + +Simplified block diagram of the adaptive multi-rate decoder. The diagram is divided into three sections: frame, subframe, and post-processing. The 'frame' section takes 'LSP indices' as input to a 'decode LSP' block. This block outputs to an 'interpolation of LSP for the 4 subframes' block, which in turn outputs to an 'LSP -> A\_hat(z)' block. The 'subframe' section takes 'pitch index' to a 'decode adaptive codebook' block, 'gains indices' to a 'decode gains' block, and 'code index' to a 'decode innovative codebook' block. The outputs of these three blocks are combined in a 'construct excitation' block. The 'LSP -> A\_hat(z)' block from the 'frame' section and the 'construct excitation' block from the 'subframe' section both feed into a 'synthesis filter' block. The output of the 'synthesis filter' is labeled s\_hat(n). This signal then enters the 'post-processing' section, which contains a 'post filter' block that outputs the final signal s'\_hat(n). + +Figure 4: Simplified block diagram of the adaptive multi-rate decoder + +# --- 9 Bibliography + +- 1) M.R. Schroeder and B.S. Atal, "Code-Excited Linear Prediction (CELP): High quality speech at very low bit rates," in *Proc. ICASSP'85*, pp. 937-940, 1985. +- 2) L.R. Rabiner and R.W. Schaefer. *Digital processing of speech signals*. Prentice-Hall Int., 1978. +- 3) F. Itakura, "Line spectral representation of linear predictive coefficients of speech signals," *J. Acoust. Soc. Amer.*, vol. 57, Supplement no. 1, S35, 1975. +- 4) F.K. Soong and B.H. Juang, "Line spectrum pair (LSP) and speech data compression", in *Proc. ICASSP'84*, pp. 1.10.1-1.10.4. +- 5) K.K Paliwal and B.S. Atal, "Efficient vector quantization of LPC parameters at 24 bits/frame", *IEEE Trans. Speech and Audio Processing*, vol. 1, no 1, pp. 3-14, 1993. +- 6) P. Kabal and R.P. Ramachandran, "The computation of line spectral frequencies using Chebyshev polynomials", *IEEE Trans. on ASSP*, vol. 34, no. 6, pp. 1419-1426, Dec. 1986. +- 7) K. Järvinen, J. Vainio, P. Kapanen, T. Honkanen, P. Haavisto, R. Salami, C. Laflamme, and J.-P. Adoul, "GSM enhanced full rate speech codec", in *Proc. ICASSP'97*, pp. 771-774. +- 8) T. Honkanen, J. Vainio, K. Järvinen, P. Haavisto, R. Salami, C. Laflamme, and J.-P. Adoul, "Enhanced full rate speech codec for IS-136 digital cellular system", in *Proc. ICASSP'97*, pp. 731-734. +- 9) R. Hagen, E. Ekudden, B. Johansson, and W.B. Kleijn, "Removal of sparse-excitation artefacts in CELP", in *Proc. ICASSP'98*, pp. I-145-I-148. + +# Annex A (informative): Change history + +| Tdoc | SPEC | CR | RE | VER | SUBJECT | CAT | NEW | +|-----------------------|--------------|-----------------|-----------|------------|---------------------------------------------------------------------------------------|------------|------------| +| SP-99570 | 26.090 | A001 | | 3.0.1 | Bit allocation of the adaptive multi-rate codec | F | 3.1.0 | +| Change history | | | | | | | | +| Date | TSG # | TSG Doc. | CR | Rev | Subject/Comment | Old | New | +| 03-2001 | 11 | | | | Version for Release 4 | | 4.0.0 | +| 06-2002 | 16 | | | | Version for Release 5 | 4.0.0 | 5.0.0 | +| 12-2004 | 26 | | | | Version for Release 6 | 5.0.0 | 6.0.0 | +| 06-2007 | 36 | | | | Version for Release 7 | 6.0.0 | 7.0.0 | +| 12-2008 | 42 | | | | Version for Release 8 | 7.0.0 | 8.0.0 | +| 06-2009 | 44 | SP-090249 | 0002 | 1 | Corrections to Quantization of codebook gains in sub-clause 5.8.2 | 8.0.0 | 8.1.0 | +| 06-2009 | 44 | SP-090249 | 0003 | 1 | Correction to recursive equation for the past filtered excitation in sub-clause 5.6.1 | 8.0.0 | 8.1.0 | +| 12-2009 | 46 | | | | Version for Release 9 | 8.1.0 | 9.0.0 | +| 03-2011 | 51 | | | | Version for Release 10 | 9.0.0 | 10.0.0 | +| 09-2011 | 53 | SP-110548 | 0007 | 1 | Correction of equation for pitch sharpening | 10.0.0 | 10.1.0 | +| 09-2012 | 57 | | | | Version for Release 11 | 10.1.0 | 11.0.0 | \ No newline at end of file diff --git a/marked/Rel-11/26_series/26091/35a7554182eb055209552843f341a1ae_img.jpg b/marked/Rel-11/26_series/26091/35a7554182eb055209552843f341a1ae_img.jpg new file mode 100644 index 0000000000000000000000000000000000000000..199b0fccba86861d4fcec8d224dd9692dd932107 --- /dev/null +++ b/marked/Rel-11/26_series/26091/35a7554182eb055209552843f341a1ae_img.jpg @@ -0,0 +1,3 @@ +version https://git-lfs.github.com/spec/v1 +oid sha256:824d3160c50e2da13d65971a393e5fa75c7e829687963796b00ea91cef231510 +size 55574 diff --git a/marked/Rel-11/26_series/26091/raw.md b/marked/Rel-11/26_series/26091/raw.md new file mode 100644 index 0000000000000000000000000000000000000000..b3bbda82d48ebf824b7db33c73c192f5092131d7 --- /dev/null +++ b/marked/Rel-11/26_series/26091/raw.md @@ -0,0 +1,360 @@ + + + + + + +# Contents + +| | | +|---------------------------------------------------------------------------|----| +| Foreword ..... | 4 | +| 1 Scope..... | 5 | +| 2 References..... | 5 | +| 3 Definitions and abbreviations ..... | 5 | +| 3.1 Definitions..... | 5 | +| 3.2 Abbreviations ..... | 5 | +| 4 General..... | 6 | +| 5 Requirements ..... | 6 | +| 5.1 Error detection..... | 6 | +| 5.2 Lost speech frames..... | 6 | +| 5.3 First lost SID frame ..... | 6 | +| 5.4 Subsequent lost SID frames ..... | 6 | +| 6 Example ECU/BFH Solution 1 ..... | 6 | +| 6.1 State Machine..... | 7 | +| 6.2 Assumed Active Speech Frame Error Concealment Unit Actions ..... | 8 | +| 6.2.1 BFI = 0, prevBFI = 0, State = 0..... | 8 | +| 6.2.2 BFI = 0, prevBFI = 1, State = 0 or 5 ..... | 8 | +| 6.2.3 BFI = 1, prevBFI = 0 or 1, State = 1...6 ..... | 9 | +| 6.2.3.1 LTP-lag update ..... | 9 | +| 6.2.3.2 Innovation sequence ..... | 9 | +| 6.3 Assumed Non-Active Speech Signal Error Concealment Unit Actions ..... | 10 | +| 6.3.1 General ..... | 10 | +| 6.3.2 Detectors..... | 10 | +| 6.3.2.1 Background detector..... | 10 | +| 6.3.2.2 Voicing detector..... | 10 | +| 6.3.3 Background ECU Actions ..... | 10 | +| 6.4 Substitution and muting of lost SID frames ..... | 10 | +| 7 Example ECU/BFH Solution 2 ..... | 11 | +| 7.1 State Machine..... | 11 | +| 7.2 Substitution and muting of lost speech frames..... | 11 | +| 7.2.1 BFI = 0, prevBFI = 0, State = 0..... | 11 | +| 7.2.2 BFI = 0, prevBFI = 1, State = 0 or 5 ..... | 11 | +| 7.2.3 BFI = 1, prevBFI = 0 or 1, State = 1...6 ..... | 12 | +| 7.2.3.1 LTP-lag update ..... | 12 | +| 7.2.4 Innovation sequence ..... | 12 | +| 7.3 Substitution and muting of lost SID frames ..... | 12 | +| Annex A (informative): Change history..... | 13 | + +# --- Foreword + +This Technical Specification has been produced by the 3rd Generation Partnership Project (3GPP). + +The contents of the present document are subject to continuing work within the TSG and may change following formal TSG approval. Should the TSG modify the contents of the present document, it will be re-released by the TSG with an identifying change of release date and an increase in version number as follows: + +Version x.y.z + +where: + +- x the first digit: + - 1 presented to TSG for information; + - 2 presented to TSG for approval; + - 3 or greater indicates TSG approved document under change control. +- y the second digit is incremented for all changes of substance, i.e. technical enhancements, corrections, updates, etc. +- z the third digit is incremented when editorial only changes have been incorporated in the document. + +# --- 1 Scope + +The present document defines an error concealment procedure, also termed frame substitution and muting procedure, which shall be used by the AMR speech codec receiving end when one or more lost speech or lost Silence Descriptor (SID) frames are received. + +The requirements of the present document are mandatory for implementation in all networks and User Equipment (UE)s capable of supporting the AMR speech codec. It is not mandatory to follow the bit exact implementation outlined in the present document and the corresponding C source code. + +# --- 2 References + +The following documents contain provisions which, through reference in this text, constitute provisions of the present document. + +- References are either specific (identified by date of publication, edition number, version number, etc.) or non-specific. +- For a specific reference, subsequent revisions do not apply. +- For a non-specific reference, the latest version applies. In the case of a reference to a 3GPP document (including a GSM document), a non-specific reference implicitly refers to the latest version of that document *in the same Release as the present document*. + +- [1] 3GPP TS 26.102: "AMR Speech Codec; Interface to Iu and Uu". +- [2] 3GPP TS 26.090: "Transcoding functions". +- [3] 3GPP TS 26.093: "Source Controlled Rate operation". +- [4] 3GPP TS 26.101: "Frame Structure". + +# --- 3 Definitions and abbreviations + +## 3.1 Definitions + +For the purposes of the present document, the following terms and definitions apply: + +**N-point median operation:** consists of sorting the N elements belonging to the set for which the median operation is to be performed in an ascending order according to their values, and selecting the $(\text{int}(N/2) + 1)$ -th largest value of the sorted set as the median value + +Further definitions of terms used in the present document can be found in the references. + +## 3.2 Abbreviations + +For the purposes of the present document, the following abbreviations apply: + +| | | +|----------|----------------------------------------------------------------| +| AN | Access Network | +| BFH | Bad Frame Handling | +| BFI | Bad Frame Indication from AN | +| BSI_netw | Bad Sub-block Indication obtained from AN interface CRC checks | +| CRC | Cyclic Redundancy Check | +| ECU | Error Concealment Unit | +| medianN | N-point median operation | +| PDFI | Potentially Degraded Frame Indication | +| prevBFI | Bad Frame Indication of previous frame | +| RX | Receive | +| SCR | Source Controlled Rate (operation) | +| SID | Silence Descriptor frame (Background descriptor) | + +# --- 4 General + +The purpose of the error concealment procedure is to conceal the effect of lost AMR speech frames. The purpose of muting the output in the case of several lost frames is to indicate the breakdown of the channel to the user and to avoid generating possible annoying sounds as a result from the error concealment procedure. + +The network shall indicate lost speech or lost SID frames by setting the RX\_TYPE values [3] to SPEECH\_BAD or SID\_BAD. If these flags are set, the speech decoder shall perform parameter substitution to conceal errors. + +The network should also indicate potentially degraded frames using the flag RX\_TYPE value SPEECH\_PROBABLY\_DEGRADED. This flag may be derived from channel quality indicators. It may be used by the speech decoder selectively depending on the estimated signal type. + +The example solutions provided in paragraphs 6 and 7 apply only to bad frame handling on a complete speech frame basis. Sub-frame based error concealment may be derived using similar methods. + +# --- 5 Requirements + +## 5.1 Error detection + +If the most sensitive bits of the AMR speech data (class A in [4]) are received in error, the network shall indicate RX\_TYPE = SPEECH\_BAD in which case the BFI flag is set. If a SID frame is received in error, the network shall indicate RX\_TYPE = SID\_BAD in which case the BFI flag is also set. The RX\_TYPE = SPEECH\_PROBABLY\_DEGRADED flag should be set appropriately using quality information from the channel decoder, in which case the PDFI flag is set. + +## 5.2 Lost speech frames + +Normal decoding of lost speech frames would result in very unpleasant noise effects. In order to improve the subjective quality, lost speech frames shall be substituted with either a repetition or an extrapolation of the previous good speech frame(s). This substitution is done so that it gradually will decrease the output level, resulting in silence at the output. Clauses 6, and 7 provide example solutions. + +## 5.3 First lost SID frame + +A lost SID frame shall be substituted by using the SID information from earlier received valid SID frames and the procedure for valid SID frames be applied as described in [3]. + +## 5.4 Subsequent lost SID frames + +For many subsequent lost SID frames, a muting technique shall be applied to the comfort noise that will gradually decrease the output level. For subsequent lost SID frames, the muting of the output shall be maintained. Clauses 6 and 7 provide example solutions. + +# --- 6 Example ECU/BFH Solution 1 + +The C code of the following example is embedded in the bit exact software of the codec. In the code the ECU is designed to allow subframe-by-subframe synthesis, thereby reducing the speech synthesis delay to a minimum. + +## 6.1 State Machine + +This example solution for substitution and muting is based on a state machine with seven states (Figure 1). + +The system starts in state 0. Each time a bad frame is detected, the state counter is incremented by one and is saturated when it reaches 6. Each time a good speech frame is detected, the state counter is reset to zero, except when we are in state 6, where we set the state counter to 5. The state indicates the quality of the channel: the larger the value of the state counter, the worse the channel quality is. The control flow of the state machine can be described by the following C code (**BFI** = bad frame indicator, **State** = state variable): + +``` +if(BFI != 0 ) + State = State + 1; +else if(State == 6) + State = 5; +else + State = 0; +if(State > 6 ) + State = 6; +``` + +In addition to this state machine, the **Bad Frame Flag** from the previous frame is checked (**prevBFI**). The processing depends on the value of the **State**-variable. In states 0 and 5, the processing depends also on the two flags **BFI** and **prevBFI**. + +The procedure can be described as follows: + +![Figure 1: State machine for controlling the bad frame substitution. The diagram shows a vertical sequence of seven rectangular boxes representing states. Each box contains three lines of text: 'S' followed by a number (0-6), 'BFI', and 'KBF1'. Arrows indicate transitions between states: a downward arrow from each state to the next, and a self-loop arrow at the top of the first state. Horizontal arrows point from the right side of each state to the right side of the state above it. From the bottom of the sixth state, two horizontal arrows point to two separate horizontal ovals containing 'K' and 'BFI'.](35a7554182eb055209552843f341a1ae_img.jpg) + +Figure 1: State machine for controlling the bad frame substitution. The diagram shows a vertical sequence of seven rectangular boxes representing states. Each box contains three lines of text: 'S' followed by a number (0-6), 'BFI', and 'KBF1'. Arrows indicate transitions between states: a downward arrow from each state to the next, and a self-loop arrow at the top of the first state. Horizontal arrows point from the right side of each state to the right side of the state above it. From the bottom of the sixth state, two horizontal arrows point to two separate horizontal ovals containing 'K' and 'BFI'. + +Figure 1: State machine for controlling the bad frame substitution + +## 6.2 Assumed Active Speech Frame Error Concealment Unit Actions + +### 6.2.1 BFI = 0, prevBFI = 0, State = 0 + +No error is detected in the received or in the previous received speech frame. The received speech parameters are used in the normal way in the speech synthesis. The current frame of speech parameters is saved. + +### 6.2.2 BFI = 0, prevBFI = 1, State = 0 or 5 + +No error is detected in the received speech frame, but the previous received speech frame was bad. The LTP gain and fixed codebook gain are limited below the values used for the last received good + +$$\text{subframe: } g^p = \begin{cases} g^p, & g^p \leq g^p(-1) \\ g^p(-1), & g^p > g^p(-1) \end{cases} \quad (1)$$ + +where $g^p$ = current decoded LTP gain, $g^p(-1)$ = LTP gain used for the last good subframe (BFI = 0), and + +$$g^c = \begin{cases} g^c, & g^c \leq g^c(-1) \\ g^c(-1), & g^c > g^c(-1) \end{cases} \quad (2)$$ + +where $g^c$ = current decoded fixed codebook gain and $g^c(-1)$ = fixed codebook gain used for the last good subframe (BFI = 0). + +The rest of the received speech parameters are used normally in the speech synthesis. The current frame of speech parameters is saved. + +### 6.2.3 + +#### BFI = 1, prevBFI = 0 or 1, State = 1...6 + +An error is detected in the received speech frame and the substitution and muting procedure is started. The LTP gain and fixed codebook gain are replaced by attenuated values from the previous subframes: + +$$g^p = \begin{cases} P(state) g^p(-1), & g^p(-1) \leq median5(g^p(-1), \dots, g^p(-5)) \\ P(state) median5(g^p(-1), \dots, g^p(-5)), & g^p(-1) > median5(g^p(-1), \dots, g^p(-5)) \end{cases} \quad (3)$$ + +where $g^p$ = current decoded LTP gain, $g^p(-1), \dots, g^p(-n)$ = LTP gains used for the last n subframes, $median5()$ = 5-point median operation, $P(state)$ = attenuation factor ( $P(1) = 0.98, P(2) = 0.98, P(3) = 0.8, P(4) = 0.3, P(5) = 0.2, P(6) = 0.2$ ), $state$ = state number, and + +$$g^c = \begin{cases} C(state) g^c(-1), & g^c(-1) \leq median5(g^c(-1), \dots, g^c(-5)) \\ C(state) median5(g^c(-1), \dots, g^c(-5)), & g^c(-1) > median5(g^c(-1), \dots, g^c(-5)) \end{cases} \quad (4)$$ + +where $g^c$ = current decoded fixed codebook gain, $g^c(-1), \dots, g^c(-n)$ = fixed codebook gains used for the last n subframes, $median5()$ = 5-point median operation, $C(state)$ = attenuation factor ( $C(1) = 0.98, C(2) = 0.98, C(3) = 0.98, C(4) = 0.98, C(5) = 0.98, C(6) = 0.7$ ), and $state$ = state number. + +The higher the state value is, the more the gains are attenuated. Also the memory of the predictive fixed codebook gain is updated by using the average value of the past four values in the memory: + +$$ener(0) = \frac{1}{4} \sum_{i=1}^4 ener(-i) \quad (5)$$ + +The past LSFs are shifted towards their mean: + +$$lsf\_q1(i) = lsf\_q2(i) = \alpha past\_lsf\_q(i) + (1 - \alpha)mean\_lsf(i), \quad i = 0 \dots 9 \quad (6)$$ + +where $\alpha = 0.95$ , $lsf\_q1$ and $lsf\_q2$ are two sets of LSF-vectors for current frame, $past\_lsf\_q$ is $lsf\_q2$ from the previous frame, and $mean\_lsf$ is the average LSF-vector. Note that two sets of LSFs are available only in the 12.2 mode. + +#### 6.2.3.1 LTP-lag update + +The LTP-lag values are replaced by the past value from the 4th subframe of the previous frame (12.2 mode) or slightly modified values based on the last correctly received value (all other modes). + +#### 6.2.3.2 Innovation sequence + +The received fixed codebook innovation pulses from the erroneous frame are used in the state in which they were received when corrupted data are received. In the case when no data were received random fixed codebook indices should be employed. + +## 6.3 Assumed Non-Active Speech Signal Error Concealment Unit Actions + +### 6.3.1 + +### General + +The Non-Active Speech ECU is used to reduce the negative impact of amplitude variations and tonal artefacts when using the conventional Active Speech ECU in non-voiced signals such as background noise and unvoiced speech. The background ECU actions are only used for the lower rate Speech Coding modes. + +The Non-Active Speech ECU actions are done as postprocessing actions of the Active Speech ECU, actions thus ensuring that the Active Speech ECU states are continuously updated. This will guarantee instant and seamless switching to the Active Speech ECU. The detectors and state updates have to be running continuously for all speech coding modes to avoid switching problems. + +Only the differences to the Active Speech ECU are stated below. + +### 6.3.2 + +### Detectors + +#### 6.3.2.1 Background detector + +An energy level and energy change detector is used to monitor the signal. If the signal is considered to contain background noise and only shows minor energy level changes, a flag is set. The resulting indicator is the **inBackgroundNoise** flag which indicates the signal state of the previous frame. + +#### 6.3.2.2 Voicing detector + +The received LTP gain is monitored and used to prevent the use of the background ECU actions in possibly voiced segments. A median filtered LTP gain value with a varying filter memory length is thresholded to provide the correct voicing decision. Additionally, a counter **voicedHangover** is used to monitor the time since a frame was presumably voiced. + +### 6.3.3 + +#### Background ECU Actions + +The BFI, and DFI indications are used together with the flag **inBackgroundNoise** and the counter **voicedHangover** to adjust the LTP part and the innovation part of the excitation. The actions are only taken if the previous frame has been classified as background noise and sufficient time has passed since the last voiced frame was detected. + +The background ECU actions are: energy control of the excitation signal, relaxed LTP lag control, stronger limitation of the LTP gain, adjusted adaptation of the Gain-Contour-Smoothing algorithm and modified adaptation of the Anti-Sparseness Procedure. + +## 6.4 + +### Substitution and muting of lost SID frames + +In the speech decoder a single frame classified as SID\_BAD shall be substituted by the last valid SID frame information and the procedure for valid SID frames be applied. If the time between SID information updates (updates are specified by SID\_UPDATE arrivals and occasionally by SID\_FIRST arrivals see 06.92) is greater than one second this shall lead to attenuation. + +# 7 Example ECU/BFH Solution 2 + +This is an alternative example solution which is a simplified version of Example ECU/BFH Solution 1. + +## 7.1 + +### State Machine + +This example solution for substitution and muting is based on a state machine with seven states (Figure 1, same state machine as in Example 1). + +The system starts in state 0. Each time a bad frame is detected, the state counter is incremented by one and is saturated when it reaches 6. Each time a good speech frame is detected, the state counter is reset to zero, except when we are in state 6, where we set the state counter to 5. The state indicates the quality of the channel: the larger the state counter, the + +worse the channel quality is. The control flow of the state machine can be described by the following C code (**BFI** = bad frame indicator, **State** = state variable): + +``` + +if(BFI != 0 ) + State = State + 1; +else if(State == 6) + State = 5; +else + State = 0; +if(State > 6 ) + State = 6; + +``` + +In addition to this state machine, the **Bad Frame Flag** from the previous frame is checked (**prevBFI**). The processing depends on the value of the **State**-variable. In states 0 and 5, the processing depends also on the two flags **BFI** and **prevBFI**. + +## 7.2 Substitution and muting of lost speech frames + +### 7.2.1 BFI = 0, prevBFI = 0, State = 0 + +No error is detected in the received or in the previous received speech frame. The received speech parameters are used normally in the speech synthesis. The current frame of speech parameters is saved. + +### 7.2.2 BFI = 0, prevBFI = 1, State = 0 or 5 + +No error is detected in the received speech frame but the previous received speech frame was bad. The LTP gain and fixed codebook gain are limited below the values used for the last received good subframe: + +$$g^p = \begin{cases} g^p, & g^p \leq g^p(-1) \\ g^p(-1), & g^p > g^p(-1) \end{cases} \quad (7)$$ + +where $g^p$ = current decoded LTP gain, $g^p(-1)$ = LTP gain used for the last good subframe (BFI = 0), and + +$$g^c = \begin{cases} g^c, & g^c \leq g^c(-1) \\ g^c(-1), & g^c > g^c(-1) \end{cases} \quad (8)$$ + +where $g^c$ = current decoded fixed codebook-gain and $g^c(-1)$ = fixed codebook gain used for the last good subframe (BFI = 0). + +The rest of the received speech parameters are used normally in the speech synthesis. The current frame of speech parameters is saved. + +### 7.2.3 BFI = 1, prevBFI = 0 or 1, State = 1...6 + +An error is detected in the received speech frame and the substitution and muting procedure is started. The LTP gain and fixed codebook gain are replaced by attenuated values from the previous subframes: + +$$g^p = \begin{cases} P(state) g^p(-1), & g^p(-1) \leq median5(g^p(-1), \dots, g^p(-5)) \\ P(state) median5(g^p(-1), \dots, g^p(-5)), & g^p(-1) > median5(g^p(-1), \dots, g^p(-5)) \end{cases} \quad (9)$$ + +where $g^p$ = current decoded LTP gain, $g^p(-1), \dots, g^p(-n)$ = LTP gains used for the last n subframes, $median5()$ = 5-point median operation, $P(state)$ = attenuation factor ( $P(1) = 0.98, P(2) = 0.98, P(3) = 0.8, P(4) = 0.3, P(5) = 0.2, P(6) = 0.2$ ), $state$ = state number, and + +$$g^c = \begin{cases} C(state) g^c(-1), & g^c(-1) \leq median5(g^c(-1), \dots, g^c(-5)) \\ C(state) median5(g^c(-1), \dots, g^c(-5)), & g^c(-1) > median5(g^c(-1), \dots, g^c(-5)) \end{cases} \quad (10)$$ + +where $g^c$ = current decoded fixed codebook gain, $g^c(-1), \dots, g^c(-n)$ = fixed codebook gains used for the last $n$ subframes, $median5()$ = 5-point median operation, $C(state)$ = attenuation factor ( $C(1) = 0.98$ , $C(2) = 0.98$ , $C(3) = 0.98$ , $C(4) = 0.98$ , $C(5) = 0.98$ , $C(6) = 0.7$ ), and $state$ = state number. + +The higher the state value is, the more the gains are attenuated. Also the memory of the predictive fixed codebook gain is updated by using the average value of the past four values in the memory: + +$$ener(0) = \frac{1}{4} \sum_{i=1}^4 ener(-i) \quad (11)$$ + +The past LSFs are used by shifting their values towards their mean: + +$$lsf\_q1(i) = lsf\_q2(i) = \alpha past\_lsf\_q(i) + (1 - \alpha)mean\_lsf(i), \quad i = 0 \dots 9 \quad (12)$$ + +where $\alpha = 0.95$ , $lsf\_q1$ and $lsf\_q2$ are two sets of LSF-vectors for current frame, $past\_lsf\_q$ is $lsf\_q2$ from the previous frame, and $mean\_lsf$ is the average LSF-vector. Note that two sets of LSFs are available only in the 12.2 mode. + +#### 7.2.3.1 LTP-lag update + +The LTP-lag values are replaced by the past value from the 4th subframe of the previous frame (12.2 mode) or slightly modified values based on the last correctly received value (all other modes). + +### 7.2.4 Innovation sequence + +The received fixed codebook innovation pulses from the erroneous frame are used in the state in which they were received when corrupted data are received. In the case when no data were received random fixed codebook indices should be employed. + +## 7.3 Substitution and muting of lost SID frames + +In the speech decoder a single frame classified as SID\_BAD shall be substituted by the last valid SID frame information and the procedure for valid SID frames be applied. If the time between SID information updates (updates are specified by SID\_UPDATE arrivals and occasionally by SID\_FIRST arrivals) is greater than one second this shall lead to attenuation. + +# Annex A (informative): Change history + +| Tdoc | SPEC | CR | RE | VER | SUBJECT | CAT | NEW | +|-----------------------|--------|----------|----|-------|--------------------------------------------------------|--------|--------| +| SP-99570 | 26.091 | A001 | | 3.0.1 | Use of random excitation when RX_NODATA and not in DTX | F | 3.1.0 | +| Change history | | | | | | | | +| Date | TSG # | TSG Doc. | CR | Rev | Subject/Comment | Old | New | +| 03-2001 | 11 | | | | Version for Release 4 | | 4.0.0 | +| 06-2002 | 16 | | | | Version for Release 5 | 4.0.0 | 5.0.0 | +| 12-2004 | 26 | | | | Version for Release 6 | 5.0.0 | 6.0.0 | +| 06-2007 | 36 | | | | Version for Release 7 | 6.0.0 | 7.0.0 | +| 12-2008 | 42 | | | | Version for Release 8 | 7.0.0 | 8.0.0 | +| 12-2009 | 46 | | | | Version for Release 9 | 8.0.0 | 9.0.0 | +| 03-2011 | 51 | | | | Version for Release 10 | 9.0.0 | 10.0.0 | +| 09-2012 | 57 | | | | Version for Release 11 | 10.0.0 | 11.0.0 | \ No newline at end of file diff --git a/marked/Rel-11/26_series/26102/raw.md b/marked/Rel-11/26_series/26102/raw.md new file mode 100644 index 0000000000000000000000000000000000000000..e4e07942b1b1b4684eae96cf9e95a2768f9ec8bd --- /dev/null +++ b/marked/Rel-11/26_series/26102/raw.md @@ -0,0 +1,1021 @@ + + + + + + +# Contents + +| | | +|--------------------------------------------------------------------------------------|----| +| Foreword ..... | 5 | +| 1 Scope..... | 6 | +| 2 References..... | 6 | +| 3 Definitions and abbreviations ..... | 7 | +| 3.1 Definitions..... | 7 | +| 3.2 Abbreviations ..... | 8 | +| 4 General..... | 8 | +| 5 RAB aspects..... | 10 | +| 6 Iu Interface User Plane (RAN) ..... | 11 | +| 6.1 Frame structure on the Iu UP transport protocol..... | 11 | +| 6.1.1 Initialisation..... | 11 | +| 6.1.2 Time Alignment Procedure ..... | 11 | +| 6.2 Mapping of the bits ..... | 11 | +| 6.3 Frame handlers ..... | 13 | +| 6.3.1 Handling of frames from TC to Iu interface (downlink)..... | 13 | +| 6.3.1.1 Frame Quality Indicator..... | 13 | +| 6.3.1.2 Frame Type..... | 13 | +| 6.3.1.3 Codec Mode Indication..... | 13 | +| 6.3.1.4 Codec Mode Request..... | 13 | +| 6.3.1.5 Optional internal 8 bits CRC ..... | 14 | +| 6.3.1.6 Mapping of Speech or Comfort Noise parameter bits ..... | 14 | +| 6.3.2 Handling of frames from Iu interface to TC (uplink)..... | 14 | +| 6.3.2.1 Frame Quality Indicator..... | 14 | +| 6.3.2.2 Frame Type..... | 14 | +| 6.3.2.3 Codec Mode Indication..... | 14 | +| 6.3.2.4 Codec Mode Request..... | 15 | +| 6.3.2.5 Optional internal 8 bits CRC ..... | 15 | +| 6.3.2.6 Speech and Comfort noise parameter bits ..... | 15 | +| 7 Uu Interface User Plane (UE)..... | 15 | +| 8 Nb Interface User Plane (CN) of a BICC-based Circuit Switched Core Network ..... | 15 | +| 8.1 Frame structure on the Nb UP transport protocol ..... | 15 | +| 8.1.1 Initialisation..... | 16 | +| 8.1.2 Time Alignment Procedure ..... | 16 | +| 8.1.3 SID Frame Generation..... | 16 | +| 8.2 Mapping of the bits ..... | 16 | +| 8.2.1 Mapping for AMR frames..... | 16 | +| 8.2.2 Mapping for PCM Coded Speech ..... | 16 | +| 8.2.3 Mapping for GSM_EFR frames ..... | 17 | +| 8.2.4 Mapping for GSM_FR frames ..... | 18 | +| 8.2.5 Mapping for GSM_HR frames..... | 18 | +| 8.3 Frame handlers ..... | 20 | +| 9 Nb Interface User Plane (CN) of a SIP-I -based Circuit Switched Core Network ..... | 20 | +| 9.1 Overview ..... | 20 | +| 9.1.1 Time Alignment Procedure ..... | 21 | +| 9.1.2 SID Frame Generation..... | 21 | +| 9.1.3 Initial Codec Mode..... | 22 | +| 9.2 AMR..... | 22 | +| 9.3 AMR-WB..... | 23 | +| 9.4 GSM_EFR..... | 24 | +| 9.5 GSM_FR ..... | 24 | +| 9.6 GSM_HR..... | 24 | +| 9.7 PCM ..... | 25 | +| 9.8 Telephone-Event ..... | 25 | + +10 A-Interface User Plane over IP .....25 +10.1 Overview ..... 25 +10.1.1 Time Alignment Procedure ..... 26 +10.1.2 SID Frame Generation..... 26 +10.1.3 Initial Codec Mode ..... 26 +10.2 AMR..... 27 +10.3 AMR-WB..... 27 +10.4 GSM\_EFR..... 28 +10.5 GSM\_FR ..... 28 +10.6 GSM\_HR..... 28 +10.7 PCM ..... 28 +Annex A (informative): Change history .....29 + +# --- Foreword + +This Technical Specification (TS) has been produced by the 3rd Generation Partnership Project (3GPP). + +The contents of the present document are subject to continuing work within the TSG and may change following formal TSG approval. Should the TSG modify the contents of the present document, it will be re-released by the TSG with an identifying change of release date and an increase in version number as follows: + +Version x.y.z + +where: + +- x the first digit: + - 1 presented to TSG for information; + - 2 presented to TSG for approval; + - 3 or greater indicates TSG approved document under change control. +- y the second digit is incremented for all changes of substance, i.e. technical enhancements, corrections, updates, etc. +- z the third digit is incremented when editorial only changes have been incorporated in the document. + +# --- 1 Scope + +The present document specifies the mapping of the AMR generic frame format (3GPP TS 26.101) to the Iu Interface (3GPP TS 25.415 [7]), the Uu Interface and the Nb Interface (3GPP TS 29.415). It further specifies the mapping of Enhanced Full Rate (GSM\_EFR) coded speech and of PCM 64 kBit/s (ITU-T G.711 [9]) coded speech to the Nb Interface in a BICC-based circuit switched core network. + +The present document also specifies the mapping of Full Rate (GSM\_FR) coded speech and of Half Rate (GSM\_HR) coded speech to the Nb Interface in a BICC-based circuit switched core network. + +The present document also specifies the transport of the AMR Codec Types, the AMR-WB Codec Types, the GSM\_EFR Codec, the GSM\_FR Codec, the GSM\_HR Codec and the ITU-T G.711 Codec over the A-Interface over IP (3GPP TS 48.002 [11]) and the Nb-Interface in a SIP-I-based circuit switched core network (3GPP TS 23.231 [12]). + +# --- 2 References + +The following documents contain provisions which, through reference in this text, constitute provisions of the present document. + +- References are either specific (identified by date of publication, edition number, version number, etc.) or non-specific. +- For a specific reference, subsequent revisions do not apply. +- For a non-specific reference, the latest version applies. In the case of a reference to a 3GPP document (including a GSM document), a non-specific reference implicitly refers to the latest version of that document *in the same Release as the present document*. + +- [1] 3GPP TS 25.415: "Iu Interface CN-UTRAN User plane Protocols". +- [2] 3GPP TS 26.101: "AMR Speech Codec, Frame structure". +- [3] 3GPP TS 23.107: "QoS Concept and Architecture". +- [4] 3GPP TS 46.051: "Enhanced Full Rate (EFR) speech processing functions; General Description" +- [5] 3GPP TS 28.062: "Inband Tandem Free Operation (TFO) of speech codecs; Service description; Stage 3". +- [6] 3GPP TS 23.153: "Out of band transcoder control, Stage 2". +- [7] 3GPP TS 29.415: "Core Network Nb Interface User Plane Protocols". +- [8] ITU-T I.366.2: "AAL type 2 service specific convergence sublayer for trunking". +- [9] ITU-T Recommendation G.711: "Pulse code modulation (PCM) of voice frequencies". +- [10] 3GPP TS 29.414: "Core Network Nb data transport and transport signalling". +- [11] 3GPP TS 48.002: "Base Station System - Mobile-services Switching Centre (BSS - MSC) interface; Interface principles". +- [12] 3GPP TS 23.231: "SIP-I based circuit-switched core network; Stage 2". +- [13] 3GPP TS 29.007: "General requirements on interworking between the Public Land Mobile Network (PLMN) and the Integrated Services Digital Network (ISDN) or Public Switched Telephone Network (PSTN)". +- [14] 3GPP TS 26.103: "Speech codec list for GSM and UMTS ". +- [15] IETF RFC 3264 (2002): "An Offer/Answer Model with the Session Description Protocol (SDP)", J. Rosenberg and H. Schulzrinne. + +- [16] IETF RFC 3550 (2003): "RTP: A Transport Protocol for Real-Time Applications", H. Schulzrinne, S. Casner, R. Frederick and V. Jacobson. +- [17] IETF RFC 3551 (2003): "RTP Profile for Audio and Video Conferences with Minimal Control", H. Schulzrinne and S. Casner. +- [18] void +- [19] IETF RFC 4566 (2006): "SDP: Session Description Protocol", M. Handley, V. Jacobson and C. Perkins. +- [20] IETF RFC 4733 (2006): "RTP Payload for DTMF Digits, Telephony Tones, and Telephony Signals", H. Schulzrinne and T. Taylor. +- [21] IETF RFC 4867 (2007): "RTP Payload Format and File Storage Format for the Adaptive Multi-Rate (AMR) and Adaptive Multi-Rate Wideband (AMR-WB) Audio Codecs", J. Sjoberg, M. Westerlund, A. Lakanieniemi and Q. Xie. +- [22] IETF RFC 5993 (2010) "RTP Payload Format for Global System for Mobile Communications Half Rate (GSM-HR)". +- [23] 3GPP TS 46.010: "Full rate speech; Transcoding". +- [24] 3GPP TS 46.020: "Half rate speech; Half rate speech transcoding". +- [25] 3GPP TS 46.041: "Half rate speech; Discontinuous Transmission (DTX) for half rate speech traffic channels". +- [26] 3GPP TS 48.060: "In-band control of remote transcoders and rate adaptors for full rate traffic channels". +- [27] 3GPP TS 48.061: "In band control of remote transcoders and rate adaptors for half rate traffic channels". +- [28] 3GPP TS 46.012: "Full rate speech; Comfort noise aspect for full rate speech traffic channels". +- [29] 3GPP TS 46.022: "Half rate speech; Comfort noise aspects for half rate speech traffic channels". +- [30] 3GPP TS 46.062: "Comfort noise aspects for Enhanced Full Rate (EFR) speech traffic channels". +- [31] 3GPP TS 26.093: "Adaptive Multi-Rate (AMR) speech codec; Source controlled rate operation". +- [32] 3GPP TS 26.193: "Adaptive Multi-Rate - Wideband (AMR-WB) speech codec; Source controlled rate operation". +- [33] 3GPP TS 48.008: "Mobile Switching Centre - Base Station System (MSC-BSS) interface". +- [34] 3GPP TS 48.103: "Base Station System – Media GateWay (BSS-MGW) interface; User Plane transport mechanism". +- [35] 3GPP TS 45.009: "Radio Access Network; Link adaptation" +- [36] 3GPP TS 46.060: "EFR Speech Codec; Speech Transcoding Functions" + +# --- 3 Definitions and abbreviations + +## 3.1 Definitions + +For the purposes of the present document the following terms and definitions apply: + +**AMR Generic Frame Interface:** this interface transports the AMR IF1 generic frame as defined in 3GPP TS 26.101 + +## 3.2 Abbreviations + +For the purposes of the present document, the following abbreviations apply: + +| | | +|---------|----------------------------------------------------------------------------------| +| AAL2 | ATM Adaptation Layer 2 | +| ACS | Active Codec Set | +| AMR | Adaptive Multi-Rate | +| AoIP | A-Interface user plane transport over RTP/UDP/IP | +| AS | Access Stratum | +| ATM | Asynchronous Transfer Mode | +| BFH | Bad Frame Handling | +| CDMA | Code Division Multiple Access | +| CMI | Codec Mode Indication | +| CMR/CMC | Codec Mode Request or Codec Mode Command | +| CN | Core Network | +| DRC | Downlink Rate Command | +| FDD | Frequency Duplex Division | +| FQC | Frame Quality Classification (Iu Interface) | +| FQI | Frame Quality Indication (AMR IF1) | +| GSM | Global System for Mobile communications | +| ITU-T | International Telecommunication Union – Telecommunication standardisation sector | +| MGW | Media GateWay | +| NboIP | Nb-Interface user plane transport over RTP/UDP/IP when SIP-I is used on Nc | +| PCM | Pulse Code Modulation, synonym for 64 kBit/s coded speech (see ITU-T G.711 [9]) | +| PDC | Personal Digital Communication | +| PLMN | Public Land Mobile Network | +| QoS | Quality of Service | +| RAB | Radio Access Bearer | +| RAN | Radio Access Network | +| RF | Radio Frequency | +| RFC | RAB sub-flow Combination | +| RFCI | RFC Indicator | +| RFCS | RFC Set | +| RX | Receive | +| SCR | Source Controlled Rate | +| SDU | Source Data Unit | +| SID | Silence Insertion Descriptor | +| SMpSDU | Support Mode for Predefined SDU sizes | +| SPD | SPeech Decoder | +| SPE | SPeech Encoder | +| TC | Transcoder | +| TDD | Time Duplex Division | +| TDMA | Time Division Multiple Access | +| TFO | Tandem Free Operation | +| TrFO | Transcoder Free Operation | +| TX | Transmit | +| UE | User Equipment (terminal) | +| URC | Uplink Rate Command | + +# --- 4 General + +The Iu-Interface is defined in two different variants for speech telephony, + +- for the ATM bearer with Iu-framing and +- for the IP bearer with Iu-framing. + +The Nb-Interface is defined in three different variants for speech telephony, + +- for the ATM bearer with Nb-framing in a BICC-based Core Network, +- for the IP bearer with Nb-framing in a BICC-based Core Network and +- for the IP bearer with RTP packetization in a SIP-I -based Core Network, also called NboIP. + +The mapping of the AMR Speech Codec parameters to the Iu interface specifies the frame structure of the speech data exchanged between the RNC and the TC inside the MGW in case of normal operation. This mapping is independent from the radio interface in the sense that it has the same structure for both FDD and TDD modes of the UTRAN. + +The mapping between the Speech Codec and the Radio Access Network within the UE is not an open interface and need not to be detailed. + +The mapping on the Nb Interface in a BICC based Core Network is identical to the one on the Iu Interface in case of Transcoder Free Operation, with the MGW relaying the SDUs unaltered between Iu and Nb Interfaces. + +PCM coded speech is mapped onto the Nb-Interface in packets of 40 octets (5ms packetization time) or 160 octets (20ms packetization time). With Nb-framing (i.e. in a BICC-based Circuit Switched Core Network, IP or ATM) the default packetization time for PCM-coded speech is 5ms; 20ms is an additional option. For NbIP (i.e. RTP packetization in a SIP-I -based Circuit Switched Core Network) the default packetization time for PCM-coded speech is 20ms; 5ms is an additional option. + +The packetization time of PCM-coded speech for AoIP is 20ms without any other option. + +For the 3GPP Codec Types (GSM\_FR, GSM\_HR, GSM\_EFR, AMR and AMR-WB) the framing is always 20ms and also the packetization time is 20ms in all versions of the Nb-Interface and the A-Interface over IP. The mapping of GSM\_FR, GSM\_HR and GSM\_EFR Speech Codec parameters is defined on the A Interface over IP and all versions of the Nb-Interface, but not on the Iu Interface. + +# --- 5 RAB aspects + +During the RAB Assignment procedure initiated by the CN to establish the RAB for AMR, the RAB parameters are defined. The AMR RAB is established with one or more RAB co-ordinated sub-flows with predefined sizes and QoS parameters. In this way, each RAB sub-flow Combination corresponds to one AMR frame type. For AMR, subject to operator tuning, the first RAB sub-flow (sub-flow 1) corresponds with the Class A bits. In case there are three RAB + +sub-flows, subject to operator tuning, the third RAB sub-flow (sub-flow 3) corresponds with the Class C bits. On the Iu interface, these RAB parameters define the corresponding parameters regarding the transport of AMR frames. + +Some of the QoS parameters in the RAB assignment procedure are determined from the Bearer Capability Information Element used at call set up. These QoS parameters as defined in [3], can be set as follows: + +Table 5-1: Example of mapping of BC IE into QoS parameters for UMTS AMR + +| RAB service attribute | RAB service attribute value | | | Comments | +|------------------------------|-----------------------------------------------------------|----------------------------------|----------------------------------|-------------------------------------------------------------------------------------------------------------------------| +| Traffic Class | Conversational | | | | +| RAB Asymmetry Indicator | Symmetric, bidirectional | | | Symmetric RABs are used for uplink and downlink | +| Maximum bit rate | 12.2 / 10.2 / 7.95 / 7.4 / 6.7 / 5.9 / 5.15 / 4.75 kbit/s | | | This value depends on the highest mode rate in the RFCS | +| Guaranteed bit rate | 12.2 / 10.2 / 7.95 / 7.4 / 6.7 / 5.9 / 5.15 / 4.75 kbit/s | | | One of the values is chosen, depending on the lowest rate controllable SDU format (note 2) | +| Delivery Order | Yes | | | (note 1) | +| Maximum SDU size | 244 / 204 / 159 / 148 / 134 / 118 / 103 / 95 bits | | | Maximum size of payload field in Iu UP, according to the highest mode rate in the RFCS | +| Traffic Handling Priority | Not applicable | | | Parameter not applicable for the conversational traffic class. (note 1) | +| Source statistics descriptor | Speech | | | (note 1) | +| SDU Parameters | RAB sub-flow 1
(Class A bits) | RAB sub-flow 2
(Class B bits) | RAB sub-flow 3
(Class C bits) | The number of SDU, their number of RAB sub-flow and their relative sub-flow size is subject to operator tuning (note 3) | +| SDU error ratio | $7 \cdot 10^{-3}$ | - | - | (note 3) | +| Residual bit error ratio | $10^{-6}$ | $10^{-3}$ | $5 \cdot 10^{-3}$ | (note 3 – applicable for every sub-flow) | +| Delivery of erroneous SDUs | yes | - | - | Class A bits are delivered with error indication;
Class B and C bits are delivered without any error indication. | +| SDU format information 1-9 | | | | (note 4) | +| Sub-flow SDU size 1-9 | (note 5) | (note 5) | (note 5) | | + +NOTE 1: These parameters apply to all UMTS speech codec types. +NOTE 2: The guaranteed bit rate depends on the periodicity and the lowest rate controllable SDU size. +NOTE 3: These parameters are subject to operator tuning. +NOTE 4: SDU format information has to be specified for each AMR core frame type (i.e. with speech bits and comfort noise bits) included in the RFCS as defined in [2]. +NOTE 5: The sub-flow SDU size corresponding to an AMR core frame type indicates the number of bits in the class A, class B and class C fields. The assigned SDU sizes shall be set so that the SCR operation is always possible. + +The RAB parameters shall be set so that the SCR operation is always possible. + +The conversational traffic class shall be used for the speech service, which is identified by the ITC parameter of the bearer capability information element in the SETUP message. This shall apply for all UMTS speech codec types. + +The parameters traffic class, transfer delay, traffic handling priority and source statistics descriptor shall be the same for all speech codec types applicable for UMTS. + +# 6 Iu Interface User Plane (RAN) + +The data structure exchanged on the Iu interface are symmetrical, i.e. the structure of the uplink data frames is identical to that of the downlink data frames. + +## 6.1 Frame structure on the Iu UP transport protocol + +### 6.1.1 Initialisation + +At the initialisation of the SMpSDU mode of operation, several parameters are set by the CN. The initialisation procedure is described in [1]. + +- RFCS: + +In the case of AMR, the RFCS corresponds to the Active Codec Set (ACS) plus potentially SCR authorised in the communication. Annex A of [1] gives an illustration of the usage of RFCI for AMR speech RAB. RFCS used in downlink may differ from that in uplink. + +- Delivery of erroneous SDUs: + +This parameter shall be set to YES. Erroneous speech frames may be used to assist the error concealment procedures. Therefore, according to [1], PDU type 0 (containing a payload CRC) shall be used for transport of AMR data. + +### 6.1.2 Time Alignment Procedure + +The TC should adjust the timing of the speech data transmission in downlink direction according to the time alignment frames sent by the RNC. + +Time alignment procedure shall be dismissed in case of TFO and TrFO. + +## 6.2 Mapping of the bits + +The mapping of the bits between the generic AMR frames and the PDU is the same for both uplink and downlink frames. + +The following table gives the correspondence of the bit fields between the generic AMR frames at the TC interface and the PDU exchanged with the Iu transport layer. + +**Table 6-1: Mapping of generic AMR frames onto Iu PDUs** + +| PDU field | Corresponding field within the generic AMR frame | Comment | +|------------------------------|--------------------------------------------------|-----------| +| PDU Type | N/A | Type 0 | +| Frame Number | N/A | | +| FQC | Frame Quality Indicator | | +| RFCI | Frame Type | | +| Payload CRC | N/A | | +| Header CRC | N/A | | +| | | | +| Payload Fields (N Sub-flows) | Class A or SID payload
Class B
Class C | | +| SDU #1 | Most important speech bits come first | Mandatory | +| SDU #2 | Next bits follow | Optional | +| ... | ... | Optional | +| SDU #N | Least important speech bits | Optional | + +The number of RAB sub-flows, their corresponding sizes, and their attributes such as "Delivery of erroneous SDUs" shall be defined at the RAB establishment and signalled in the RANAP RAB establishment request, as proposed in clause 5. The number of RAB sub-flows is corresponding to the desired bit protection classes. The total number of bits in all sub-flows for one RFC shall correspond to the total number given in 3GPP TS 26.101, generic AMR frame, format IF1, for the corresponding Codec Mode, respectively Frame Type. + +The class division and relative subjective importance of the encoded bits is given in 3GPP TS 26.101 and provides guidance for setting the number of bits in each RAB sub-flow. + +The following two tables are examples of mapping of RAB sub-flows. + +Table 6-2 gives three examples of sub-flow mapping. + +The RFCI definition is given in order of increasing SDU sizes. + +- Example 1 describes Codec Type UMTS\_AMR, with all eight codec modes foreseen in the Active Codec Set (ACS) and provision for Source Controlled Rate operation (SCR). In this example, Blind Transport Format Detection is supported and the sub-flow mapping follows the 3GPP TS 26.101 class division. +- Example 2 describes Codec Type GSM\_EFR, with one codec mode, including SCR. + +- Example 3 describes Codec Type FR\_AMR, including AMR SCR + +**Table 6-2: Example for AMR with SCR and three sub-flows, according to subjective class division indication of 3GPP TS 26.101** + +| UMTS_AMR
RFCI
Example 1 | GSM_EFR
RFCI
Example 2 | FR_AMR
RFCI
Example 3 | RAB sub-flows | | | Total size of
bits/RAB sub-
flows
combination
(Mandatory) | Source rate | +|-------------------------------|------------------------------|-----------------------------|----------------------------------|----------------------------------|----------------------------------|-----------------------------------------------------------------------|---------------| +| | | | RAB sub-
flow 1
(Optional) | RAB sub-
flow 2
(Optional) | RAB sub-
flow 3
(Optional) | | | +| 2 | | 2 | 42 | 53 | 0 | 95 | AMR 4,75 kbps | +| 3 | | | 49 | 54 | 0 | 103 | AMR 5,15 kbps | +| 4 | | 3 | 55 | 63 | 0 | 118 | AMR 5,9 kbps | +| 5 | | 4 | 58 | 76 | 0 | 134 | AMR 6,7 kbps | +| 6 | | | 61 | 87 | 0 | 148 | AMR 7,4 kbps | +| 7 | | | 75 | 84 | 0 | 159 | AMR 7,95 kbps | +| 8 | | 5 | 65 | 99 | 40 | 204 | AMR 10,2 kbps | +| 9 | 2 | | 81 | 103 | 60 | 244 | AMR 12,2 kbps | +| 1 | | 1 | 39 | 0 | 0 | 39 | AMR SID | +| | 1 | | 43 | 0 | 0 | 43 | GSM-EFR SID | + +Table 6-3 gives one example of sub-flow mapping that supports Equal Error Protection. The RFCI definition is given in order of increasing SDU sizes. + +- Example 4 describes Codec Type PDC\_EFR and the corresponding Source Controlled Rate operation (SCR). + +**Table 6-3: Example of SDU sizes for PDC\_EFR with SCR and Equal Error Protection** + +| PDC_EFR
RFCI
Example 4 | RAB sub-flow
RAB sub-
Flow 1
(Mandatory) | Total size of
bits/RAB sub-flows
combination
(Mandatory) | Source rate | +|------------------------------|---------------------------------------------------|-------------------------------------------------------------------|--------------| +| | | | | +| | 95 | 95 | AMR 4,75kbps | +| | 103 | 103 | AMR 5,15kbps | +| | 118 | 118 | AMR 5,9kbps | +| 2 | 134 | 134 | AMR 6,7kbps | +| | 148 | 148 | AMR 7,4kbps | +| | 159 | 159 | AMR 7,95kbps | +| | 204 | 204 | AMR 10,2kbps | +| | 244 | 244 | AMR 12,2kbps | +| | 39 | 39 | AMR SID | +| | 43 | 43 | GSM-EFR SID | +| | 38 | 38 | TDMA-EFR SID | +| 1 | 37 | 37 | PDC-EFR SID | + +## 6.3 Frame handlers + +In PDU Frame handling functions are described in 3GPP TS 25.415 [1]. This section describes the mandatory frame handling functions at the AMR Generic frame interface. + +### 6.3.1 Handling of frames from TC to Iu interface (downlink) + +The frames from the TC in generic AMR frame format IF1 are mapped onto the Iu PDU as follows. + +#### 6.3.1.1 Frame Quality Indicator + +The Frame Quality Indicator (FQI) from the TC is directly mapped to the Frame Quality Classification (FQC) of the Iu frame according to Table 6-4. + +**Table 6-4: FQI AMR to FQC Iu PDU mapping** + +| FQI AMR | FQI value
(1 bit) | FQC PDU | FQC value
(2 bit) | +|---------|----------------------|---------|----------------------| +| GOOD | 1 | GOOD | 00 | +| BAD | 0 | BAD | 01 | + +#### 6.3.1.2 Frame Type + +The received Frame Type Index $l$ is mapped onto the RFCI $j$ thanks to the assigned RFCS table: the correspondence between Codec Mode, Frame Type Index $l$ and RFCI $j$ is defined at RAB assignment. + +#### 6.3.1.3 Codec Mode Indication + +The Codec Mode Indication is not used. + +#### 6.3.1.4 Codec Mode Request + +Codec Mode Request (CMR) in downlink direction is forwarded to the rate control procedure when it changes, or when it is commanded so by the TC in case of TFO, see 3G TS 28.062. + +#### 6.3.1.5 Optional internal 8 bits CRC + +The internal AMR Codec CRC is not used on the Iu interface. + +#### 6.3.1.6 Mapping of Speech or Comfort Noise parameter bits + +Let us define the $N$ payload fields of the $N$ sub-flows for RFCI $j$ as follows: + +$U_i(k)$ shall be the bits in sub-flow $i$ , for $k = 1$ to $M_i$ + +$M_i$ shall be the size of sub-flow $i$ , for $i = 1$ to $N$ + +$d(k)$ shall be the bits of the speech or comfort noise parameters of the corresponding Frame Type $l$ in decreasing subjective importance, as defined in the generic AMR frame format IF1, see TS 26.101 [2]. + +Then the following mapping in pseudo code applies: + +$U_1(k) = d(k-1)$ with $k = 1, \dots, M_1$ + +$U_2(k) = d(k-1+M_1)$ with $k = 1, \dots, M_2$ + +$U_3(k) = d(k-1+M_2)$ with $k = 1, \dots, M_3$ + +... + +$U_N(k) = d(k-1+M_{N-1})$ with $k = 1, \dots, M_N$ + +### 6.3.2 Handling of frames from Iu interface to TC (uplink) + +The uplink Iu frames are mapped onto generic AMR frames, format IF1, as follows. + +#### 6.3.2.1 Frame Quality Indicator + +At reception of Iu PDU the Iu frame handler function set the Frame Quality Classification according to the received FQC, Header-CRC check, and Payload-CRC check (see 25.415). AMR Frame Type and Frame Quality Indicator are determined according to the following table: + +**Table 6-5: FQC Iu PDU type 0 to AMR FQI and AMR Frame Type mapping** + +| FQC | FQC value (2 bits) | Resulting FQI | FQI value (1 bit) | resulting Frame Type | +|-----------|--------------------|---------------|-------------------|----------------------| +| GOOD | 00 | GOOD | 1 | from RFCI | +| BAD | 01 | BAD | 0 | NO_DATA | +| BAD Radio | 10 | BAD | 0 | from RFCI | +| Reserved | 11 | BAD | 0 | Reserved | + +#### 6.3.2.2 Frame Type + +The received RFCI j is mapped onto the Frame Type Index I thanks to the RFCS table. + +#### 6.3.2.3 Codec Mode Indication + +The Codec Mode Indication is not used. + +#### 6.3.2.4 Codec Mode Request + +The received Downlink Rate Control command (DRC) is mapped onto the Codec Mode Request (CMR) towards the AMR Codec. In case a new DRC is received it is mapped into the corresponding CMR of the generic AMR frame format. It is remembered by the TC until the next DRC is received. In each new frame that is sent to the AMR Codec, the stored CMR is resent, in order to control the Codec Mode for the downlink direction. + +#### 6.3.2.5 Optional internal 8 bits CRC + +The internal AMR Codec CRC is not used on the Iu interface. + +#### 6.3.2.6 Speech and Comfort noise parameter bits + +The speech and Comfort noise parameter bits are mapped from the sub-flows to the payload of the generic AMR frames with the reverse function of clause 6.3.1.6. + +# 7 Uu Interface User Plane (UE) + +The interface between the UE AMR speech codec (see 3GPP TS 26.101) and the Radio Access Network is an internal UE interface and is not detailed. The mapping is corresponding to the mapping described in clause 6 for the Iu interface. + +Even though the details of Uu interface are not detailed, there are some functional requirements for the UE that need to be considered, when an AMR codec type (i.e. UMTS AMR2) is being used in a conversational speech call. These requirements are related to the mapping of AMR Generic frame format handling functions. The requirements are + +1. The set of available codec modes (bitrates) that the UE may use are configured by UTRAN. The UE shall select, from the configured set of codec modes, a mode that is supported by the current TX power conditions as defined in 3GPP TS25.133. The highest available mode should be used for best speech quality. +2. The lowest configured codec mode is always to be considered supported. + +3. When the codec mode is being adapted during a call, the used mode should be changed in a step-by-step fashion within the configured set of codec modes, i.e. by stepping one mode up or down within the configured set. This avoids disruptions on AMR decoding in GSM side, if TFO or TrFO operation is ongoing. + +# --- 8 Nb Interface User Plane (CN) of a BICC-based Circuit Switched Core Network + +The data structures exchanged on the Nb interface are symmetrical, i.e. the structures of the sent and received data frames are identical. + +The Nb-Interface is defined in a BICC-based Core Network in two different variants, a) for the ATM bearer with Nb-framing, b) for the IP bearer with Nb-framing. The Nb-framing and the use of PDU Type 0 for the speech payload and PDU Type 14 [7] for AMR Rate Control is common for both versions of the Nb-Interface here. These two versions also share the principle of "Nb\_Init", where the Nb-Interface is initialized on User Plane level and where the Initial Codec Mode for AMR and/or AMR-WB is signalled. + +## 8.1 Frame structure on the Nb UP transport protocol + +Delivery of erroneous SDUs for AMR- and AMR-WB-coded speech, as well as for GSM\_FR-, GSM\_HR-, GSM\_EFR-coded speech and for PCM-coded speech on the Nb-Interface shall be set to: "YES" in a BICC-based Circuit Switched Core Network. Erroneous speech frames may be used to assist the error concealment procedures. Therefore, according to [1] and [7], PDU Type 0 (with payload CRC) shall be used for the transport of AMR, AMR-WB, GSM\_FR, GSM\_HR and GSM\_EFR coded speech on the Nb interface. PDU Type 0 (with payload CRC) shall be used for the transport of PCM coded speech on the Nb interface, too. + +### 8.1.1 Initialisation + +The initialisation procedure is used for support mode. At the initialisation several parameters are set by the CN. The initialisation procedure for the Nb Interface is described in [7]. + +### 8.1.2 Time Alignment Procedure + +The handling of Time Alignment on the Nb Interface is described in [7]. + +The Time alignment procedure shall be dismissed in case of TFO and TrFO. + +### 8.1.3 SID Frame Generation + +All 3GPP Codec Types include a standardized Discontinuous Transmission (DTX) with Voice Activity Detection, Silence Description (by SID frames) and Comfort Noise Generation to fill the speech pauses. If speech inactivity is detected by the Encoder, then (speech) frames are not transmitted, but the transmission is suspended in order to save battery life time in the mobile station, reduce interference on the radio interface and reduce load on all links. The receiving Decoder fills these transmission pauses with Comfort Noise to minimize the contrast between pauses and active speech. Silence Descriptor (SID) frames need to be sent during speech inactivity to keep the Comfort Noise decently well aligned with the background noise at sender side. This is especially important at the onset of the next talkspurt and therefore SID frames should not be too old, when speech starts again. + +The generation of SID frames for the AMR and AMR-WB families of Codecs is determined by the Speech Encoder as specified in TS 26.093 [31], respectively TS 26.193 [32]. The radio subsystem does not influence this timing! SID frames come during speech pauses in uplink and downlink about every 160ms. Also an AMR Encoder in the Media Gateway generates and sends SID frames about every 160ms. + +The generation of SID frames for GSM\_FR, GSM\_HR and GSM\_EFR in the GSM radio network is determined by the GSM mobile station and the GSM radio subsystem, not primarily by the Speech Encoder! SID frames come during speech pauses in uplink from the mobile station about every 480ms. In downlink to the mobile station, when they are generated by the Speech Encoder of the GSM radio subsystem, SID frames are sent every 20ms to the GSM base station, which then picks only one every 480ms for downlink radio transmission. For other applications, like transport over Nb, it is more appropriate to send the SID frames less often than every 20ms, but 480ms may be too sparse. As a compromise it is recommended that an Encoder in the Media Gateway should generate and send SID frames every 160ms. + +## 8.2 Mapping of the bits + +### 8.2.1 Mapping for AMR frames + +The mapping of the bits between the generic AMR frames and the PDU for the Nb Interface is identical to the mapping on the Iu Interface. In case of TrFO the MGW relays the AMR frames from the Iu Interface unaltered to the Nb Interface and vice versa, as described in [7]. + +### 8.2.2 Mapping for PCM Coded Speech + +The mapping for the PCM coded speech in 5ms frames on the Nb Interface shall be as defined in Table 8-1. + +**Table 8-1: Mapping of PCM Coded Speech in 5 ms frames onto Nb PDU, Type 0** + +| PDU field | Comment | +|---------------|----------------------------------------------------------| +| PDU Type | Type 0 (with Payload CRC) | +| Frame Number | as defined in [7] | +| FQC | set to "good", i.e. value 0 | +| RFCI | initialise by MGW, see [7],
one value required | +| Header CRC | as defined in [7] | +| Payload CRC | as defined in [7] | +| | | +| Payload Field | 320 bits of PCM coded speech,
in accordance with [8]. | + +The mapping for the PCM coded speech in 20ms frames on the Nb Interface shall be as defined in Table 8-2. + +**Table 8-2: Mapping of PCM Coded Speech in 20ms frames onto Nb PDU, Type 0** + +| PDU field | Comment | +|---------------|------------------------------------------------------------| +| PDU Type | Type 0 (with Payload CRC) | +| Frame Number | as defined in [7] | +| FQC | set to "good", i.e. value 0 | +| RFCI | initialised by MGW, see [7],
one value required | +| Header CRC | as defined in [7] | +| Payload CRC | as defined in [7] | +| | | +| Payload Field | 4x320 bits of PCM coded speech,
in accordance with [8]. | + +5ms is the default packetization time to be supported for PCM encoded speech over Nb in a BICC based Core Network. 20ms is an additional optional packetization time for PCM encoded speech in a BICC based Core Network over IP Nb bearer that may be negotiated during bearer establishment as specified in [10]. + +NOTE: the use of 20ms packetization time will result in some call scenarios in higher delays over the speech path compared to the 5ms packetization time. This potentially higher delay should be taken into account in the overall end to end (ear to mouth) delay budget. + +### 8.2.3 Mapping for GSM\_EFR frames + +The mapping of the bits between the generic GSM\_EFR frames and the PDUs for the Nb Interface follows the same principles as the mapping of AMR frames. The PDU for the GSM\_EFR speech frame is identical to the PDU for AMR Mode 12.2 kbps. + +The PDU for the GSM\_EFR SID frame is similar to the PDU for AMR SID, with 43 instead of 39 bits in the payload field. The contents of GSM\_EFR SID is the Comfort Noise Parameter set ( $s(i)$ ) as defined in [36]. The Comfort noise parameters are computed as described in [30] by the GSM\_EFR speech encoder and are denoted as $s(i) = \{s(1), s(2), \dots, s(38), s(87), s(88), \dots, s(91)\}$ . The notation $s(i)$ follows that of [36] (Table 6). The notation $d(j) = \{d(1) \dots d(43)\}$ of the SID frame is local to the present document and is formed as defined by the pseudo code below. + +``` + +for j = 1 to 38 + d(j) := s(j); /* LSP parameters in s(1) to s(38) */; + +for j = 39 to 43 + d(j) := s(j+48); /* fixed codebook gain parameter in s(87)-s(91) */ + +``` + +The payload within the PDU shall be the vector $d(j)$ constructed above. The first bit in the vector $d(j)$ shall be supplied first in the payload within the PDU. + +NOTE: The Payload field for Nb frames for GSM\_EFR in a BICC-based Circuit Switched Core Network is filled differently to the RTP payload according to RFC3551 [17], used in AoIP and NboIP. + +### 8.2.4 Mapping for GSM\_FR frames + +The mapping of GSM\_FR-coded speech in 20ms frames on the Nb Interface shall be as defined in Table 8.2.4.1. + +**Table 8.2.4.1: Mapping of GSM\_FR-coded speech in 20ms frames onto Nb PDU, Type 0** + +| PDU field | Comment | +|---------------|--------------------------------------------------------------------------------------------------------------| +| PDU Type | Type 0 (with Payload CRC) | +| Frame Number | as defined in [7] | +| FQC | see below | +| RFCI | initialise by MGW, see [7], two values required (Speech and SID) | +| Header CRC | as defined in [7] | +| Payload CRC | as defined in [7] | +| | | +| Payload Field | 264 bits if Nb PDU contains a speech frame, see below.
42 bits if Nb PDU contains a SID frame, see below. | + +#### Payload field with speech frame: + +The 260 bits of GSM\_FR-coded speech (b1...b260) are defined in TS 46.010, chapter 1.7. They and a "signature" are copied into the 33 octets of the Payload field as follows. The four most significant bits (bit 8...5) of the first octet (octet 1) of the Nb Payload field are set to a "signature" of 0b1101 = 0xD. Then the four most significant bits (b6...b3) of the first GSM\_FR parameter (LAR 1) are copied into the next bits (bit 4...1) of the first octet. The two least significant bits of the first GSM\_FR parameter (LAR 1) are copied into the next octet (octet 2) into the 2 MSBs (bit 8...7), and so on. Each GSM\_FR parameter is copied bit by bit with its most significant bit first. The least significant bit of the last GSM\_FR parameter (b258 of RPE-pulse no.13) is placed in the LSB (bit 1) of octet 33. + +#### Payload field with SID frame: + +The GSM\_FR SID frames are defined in chapter 5.2 of [28] and in chapter 1.7 of [23] and are denoted as $b(i) = \{b(1), b(2), \dots, b(36), b(48), b(49), \dots, b(53)\}$ . Each GSM\_FR SID parameter is copied bit by bit with its most significant bit first. The notation $d(j) = \{d(1) \dots d(42)\}$ of the SID frame is local to the present document and is formed by the pseudo code below. + +``` + +for j = 1 to 36 + d(j) := b(j); /* averaged log area coefficients in b(1) to b(36) */; + +for j = 37 to 42 + d(j) := b(j+11); /* averaged block amplitude values in b(48)-b(53) */ + +``` + +The payload within the PDU shall be the vector $d(j)$ constructed above. The first bit in the vector $d(j)$ shall be supplied first in the payload within the PDU. + +NOTE: The Payload field for Nb frames for GSM\_FR containing SID frame in a BICC-based Circuit Switched Core Network is filled differently to the Payload field in RTP Packets according to RFC3551 [17], used in AoIP and NboIP. + +#### FQC: + +The FQC (see Nb UP [7]) is set by the MGW depending on the call case: + +1. FQC is set to "good", i.e. value 0, if the GSM\_FR-compression and coding is performed within the MGW +2. FQC is set to "good", i.e. value 0, if GSM\_FR-coded speech is received without frame quality indication + +3. FQC is derived from the input frame, if FQC or a similar frame quality indication is specified there. In case of GSM\_FR-coded speech received via TFO frames (see TS 28.062 [5]) the FQC is derived from the "Bad Frame Indication" (BFI) of these TFO frames. Speech frames and SID frames marked with BFI set to "good" shall be sent with FQC set to "good", i.e. value 0. Speech frames and SID frames marked with BFI set to "bad" shall not be sent in order to save bandwidth on Nb. + +### 8.2.5 Mapping for GSM\_HR frames + +The mapping of GSM\_HR-coded speech in 20ms frames on the Nb Interface shall be as defined in Table 8.2.5.1. + +**Table 8.2.5.1: Mapping of GSM\_HR-coded speech in 20ms frames onto Nb PDU, Type 0** + +| PDU field | Comment | +|---------------|--------------------------------------------------------------------------------------------------------------| +| PDU Type | Type 0 (with Payload CRC) | +| Frame Number | as defined in [7] | +| FQC | see below | +| RFCI | initialise by MGW, see [7], two values required (Speech and SID) | +| Header CRC | as defined in [7] | +| Payload CRC | as defined in [7] | +| Payload Field | 112 bits if Nb PDU contains a Speech frame, see below.
33 bits if Nb PDU contains a SID frame, see below. | + +#### Payload field with speech frame: + +The 112 bits of GSM\_HR-coded speech ( $b(1)\dots b(112)$ ) are defined in TS 46.020, Annex B, in the order of occurrence. The first bit ( $b(1)$ ) of the first parameter is placed in bit 8 (the MSB) of the first octet (octet 1) of the Nb Payload field; the second bit is placed in bit 7 of the first octet and so on. The last bit ( $b(112)$ ) is placed in the LSB (bit 1) of octet 14. + +NOTE 1: The Payload field for Nb frames for GSM\_HR with speech frame in a BICC-based Circuit Switched Core Network is identical to the "speech data section" of the RTP payload. It is done according to [22], used in AoIP and NboIP. + +#### Payload field with SID frame: + +The GSM\_HR SID frames are defined in [24] and [29] and are denoted as $b(i) = \{b(1), b(2), \dots, b(33)\}$ . The notation $d(j) = \{d(1) \dots d(33)\}$ of the SID frame is local to the present document and is formed by the pseudo code as follows. + +``` + +for $j = 1$ to 5 + $d(j) := b(j)$ ; /* R0 parameter in $b(1)$ to $b(5)$ */ + +for $j = 6$ to 16 + $d(j) := b(j)$ ; /* LPC1 parameter in $b(6)$ - $b(16)$ */ + +for $j = 17$ to 25 + $d(j) := b(j)$ ; /* LPC2 parameter in $b(17)$ - $b(25)$ */ + +for $j = 26$ to 33 + $d(j) := b(j)$ ; /* LPC3 parameter in $b(26)$ - $b(33)$ */ + +``` + +The payload within the PDU shall be the vector $d(j)$ constructed above. The first bit in the vector $d(j)$ shall be supplied first in the payload within the PDU. + +NOTE 2: The Payload field for Nb frames for GSM\_HR with SID frame in a BICC-based Circuit Switched Core Network is not filled in the same way as the "speech data section" of the RTP payload used in AoIP and NboIP. For both AoIP and NboIP packing is according to [22]. + +#### FQC: + +The FQC (see Nb UP [7]) is set by the MGW depending on the call case: + +1. FQC is set to "good", i.e. value 0, if the GSM\_HR-compression and coding is performed within the MGW. +2. FQC is set to "good", i.e. value 0, if GSM\_HR-coded speech is received without frame quality indication. +3. FQC is derived from the input frame, if FQC or a similar frame quality indication is specified there. In case of GSM\_HR-coded speech received via TFO frames (see TS 28.062 [5]) the FQC is derived from + +the "Extended control bits" (XC1 to XC5) for 8kbps submultiplexing (specified in TS 48.061, chapter 5.2.4.1.1 and partly reprinted here for ease of reading) as defined in table 8.2.5.2. + +**Table 8.2.5.2: The FQC for GSM\_HR-coded Nb frames derived from TFO frames** + +| FQC | XC1 | XC2 | XC3 | XC4 | XC5 | Meaning (in Abis frames with 8kbps submultiplexing) | +|------|-----|-----|-----|-----|-----|----------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------| +| good | 0 | 0 | 0 | 0 | 0 | Good speech frame with UFI = 0
(BFI=0, SID=0, TAF=1)
(BFI=0, SID=0, TAF=0) | +| bad* | 0 | 0 | 0 | 0 | 1 | Unreliable speech frame (if speech decoder is in speech decoding mode) or unusable frame (if speech decoder is in comfort noise insertion mode) with UFI = 1
(BFI=0, SID=0, TAF=1)
(BFI=0, SID=0, TAF=0) | +| good | 0 | 0 | 0 | 1 | 0 | Valid SID frame with UFI = 0
(BFI=0, SID=2, TAF=1)
(BFI=0, SID=2, TAF=0) | +| bad | 0 | 0 | 0 | 1 | 1 | Invalid SID frame with UFI = 1
(BFI=0, SID=2, TAF=1)
(BFI=0, SID=2, TAF=0) | +| bad | 0 | 1 | 0 | 0 | 0 | Invalid SID frame at TAF=0 with UFI = 0
(BFI=0, SID=1, TAF=0)
(BFI=1, SID=1, TAF=0)
(BFI=1, SID=2, TAF=0) | +| bad | 0 | 1 | 0 | 0 | 1 | Invalid SID frame at TAF=0 with UFI = 1
(BFI=0, SID=1, TAF=0)
(BFI=1, SID=1, TAF=0)
(BFI=1, SID=2, TAF=0) | +| bad | 0 | 1 | 0 | 1 | 0 | Invalid SID frame at TAF=1 with UFI = 0
(BFI=0, SID=1, TAF=1)
(BFI=1, SID=1, TAF=1)
(BFI=1, SID=2, TAF=1) | +| bad | 0 | 1 | 0 | 1 | 1 | Invalid SID frame at TAF=1 with UFI = 1
(BFI=0, SID=1, TAF=1)
(BFI=1, SID=1, TAF=1)
(BFI=1, SID=2, TAF=1) | +| bad* | 0 | 1 | 1 | 0 | 0 | Bad speech frame or unusable frame at TAF = 0 with UFI = 0
(BFI=1, SID=0, TAF=0) | +| bad* | 0 | 1 | 1 | 0 | 1 | Bad speech frame or unusable frame at TAF = 0 with UFI = 1
(BFI=1, SID=0, TAF=0) | +| bad* | 0 | 1 | 1 | 1 | 0 | Bad speech frame or unusable frame at TAF = 1 with UFI = 0
(BFI=1, SID=0, TAF=1) | +| bad* | 0 | 1 | 1 | 1 | 1 | Bad speech frame or unusable frame at TAF = 1 with UFI = 1
(BFI=1, SID=0, TAF=1) | + +Speech frames and SID frames marked in Table 8.2.5.2 with FQC set to "good", i.e. value 0, shall be sent. + +Frames marked in Table 8.2.5.2 with FQC set to "bad\*" or "bad" shall not be sent in order to save bandwidth on Nb. + +NOTE 3: the abbreviations "UFI" (unreliable frame indication), "BFI" (bad frame indication), "SID" (Silence Descriptor) and "TAF" (Time Alignment Flag) are defined in 3GPP TS 46.041 [25]. + +## 8.3 Frame handlers + +Nb PDU Frame handling functions are described in [7]. + +# 9 Nb Interface User Plane (CN) of a SIP-I -based Circuit Switched Core Network + +## 9.1 Overview + +The SIP-I -based Circuit Switched Core Network is specified in 3GPP TS 23.231 [12]. The User Plane in this Core Network is further specified in 3GPP TS 29.414 [10]. RTP is specified in IETF RFC 3550 [16]. + +RTP is used in a SIP-I -based Circuit Switched Core Network as framing protocol at the Nb-Interface (without Nb-framing protocol). The rules for the usage of RTP and RTCP in 3GPP TS 29.414 [10] are applicable in combination with further Codec-specific rules provided in the present specification. + +Table 9.1.1 lists the applicable 3GPP Codec Types for a SIP-I -based Circuit Switched Core Network. Codecs for data transport are described in 3GPP TS 29.007 [13]. + +**Table 9.1.1 Supported Codec Types in a SIP-I -based Circuit Switched Core Network** + +| Payload Type Name | References | Remarks | Support | +|-----------------------|--------------------|----------------------------------------------------------------|-----------------------------------------------------------------------------------------------------------------| +| audio/AMR | IETF RFC 4867 [21] | Applicable for FR_AMR, HR_AMR, OHR_AMR, UMTS_AMR and UMTS_AMR2 | Mandatory.
Not all AMR Configurations are mandatory. Some Configurations are preferred, see below. | +| audio/AMR-WB | IETF RFC 4867 [21] | Applicable for FR_AMR-WB, OHR_AMR-WB, OFR_AMR-WB, UMTS_AMR-WB | Optional.
AMR-WB is mandatory, if WB speech is supported. Not all WB Configurations are mandatory, see below | +| audio/GSM_EFR | IETF RFC 3551 [17] | Useful if an A-interface over IP is attached or TFO is used. | Optional | +| audio/GSM_FR | IETF RFC 3551 [17] | Useful if an A-interface over IP is attached or TFO is used. | Optional | +| audio/GSM_HR | IETF RFC 5993 [22] | Useful if an A-interface over IP is attached or TFO is used. | Optional | +| audio/PCMA | IETF RFC 3551 [17] | ITU-T G.711 Alaw | Mandatory | +| audio/PCMU | IETF RFC 3551 [17] | ITU-T G.711 ulaw | Mandatory | +| audio/telephone-event | IETF RFC 4733 [20] | Used to transport DTMF | Mandatory | + +The RTP "Payload Type" number for the Nb-Interface is either static (for PCMA, PCMU and GSM\_FR) or determined by the MSC-S (dynamic Payload Type). + +### 9.1.1 Time Alignment Procedure + +Time Alignment (and AMR Phase Alignment) is not specified in a SIP-I -based Circuit Switched Core Network. + +### 9.1.2 SID Frame Generation + +All 3GPP Codec Types include standardized Discontinuous Transmission (DTX) with Voice Activity Detection, Silence Description (by SID frames) and Comfort Noise Generation to fill the speech pauses. If speech inactivity is detected by the Encoder, then (speech) frames are not transmitted, but the transmission is suspended in order to save battery life time in the mobile station, reduce interference on the radio interface and reduce load on all links. The receiving Decoder fills these transmission pauses with Comfort Noise to minimize the contrast between pauses and active speech. Silence Descriptor frames need to be sent during speech inactivity to keep the Comfort Noise decently well aligned with the background noise at sender side. This is especially important at the onset of the next talkspurt and therefore SID frames should not be too old, when speech starts again. + +The generation of SID frames for the AMR and AMR-WB families of Codecs is determined by the Speech Encoder as specified in TS 26.093 [31], respectively TS 26.193 [32]. The radio subsystem does not influence this timing! SID frames come during speech pauses in uplink and downlink about every 160ms. Also an AMR Encoder in the Media Gateway generates and sends SID frames about every 160ms. + +The generation of SID frames for GSM\_FR, GSM\_HR and GSM\_EFR in the GSM radio network is determined by the GSM mobile station and the GSM radio subsystem, not primarily by the Speech Encoder! SID frames come during speech pauses in uplink from the mobile station about every 480ms. In downlink to the mobile station, when they are generated by the Speech Encoder of the GSM radio subsystem, SID frames are sent every 20ms to the GSM base station, which then picks only one every 480ms for downlink radio transmission. For other applications, like transport over Nb, it is more appropriate to send the SID frames less often than every 20ms, but 480ms may be too sparse. As a compromise it is recommended that an Encoder in the Media Gateway should generate and send SID frames every 160ms. + +### 9.1.3 Initial Codec Mode + +NOTE: At the Nb-Interface in a SIP-I -based Core Network, direct RTP packetization without Nb-framing is applied. Therefore the use of PDU Type 0 for the speech payload and PDU Type 14 [7] for AMR Rate Control is here not applicable. Also the principle of "Nb\_Init" is not applicable for a SIP-I -based Core Network. + +The Initial Codec Mode for AMR and AMR-WB shall be derived by pre-defined rules from the AMR Configuration (Active Codec Set), see TS 45.009 [35], chapter 3.4.3 "Initial Codec Mode Selection at Call Setup and Handover". + +Start of extract from TS 45.009 [35] for information and ease of reading: + +"If the Initial Codec Mode is not signalled, then the default Initial Codec Mode is given by the following implicit rule. If the Active Codec Set contains: + +- 1 mode, then this shall be the Initial Codec Mode; +- 2 or 3 modes, then the Initial Codec Mode shall be the most robust mode of the set (with lowest bit rate); +- 4 modes, then the Initial Codec Mode shall be the second most robust mode of the set (with second lowest bit rate." + +End of extract from TS 45.009 [35]. + +In case of FR\_AMR (Set 1), i.e. Config-NB-Code 1, see below, this is the AMR Mode with 5.90kbps. + +## 9.2 AMR + +AMR (FR\_AMR, HR\_AMR, OHR\_AMR, UMTS\_AMR and UMTS\_AMR2) shall be packed according to IETF RFC 4867 [21]. + +The AMR Codec Types can be used in conversational speech telephony services in a number of different Codec Configurations. The set of preferred AMR Codec Configurations is defined in TS 28.062 [5], Table 7.11.3.1.3-2. One of these preferred Configurations, **Config-NB-Code 1**, is recommended for TFO-TrFO harmonisation between GSM and UMTS networks. This Configuration shall be supported in a SIP-I based circuit switched core network to ensure interoperability with an AoIP-based BSS. However, it is recommended that nodes in the core network support all AMR modes for maximum interoperability. + +The bandwidth efficient mode of RFC 4867 shall be used. CRC and robust sorting shall not be applied. + +To avoid delay, a single frame (Speech or SID\_FIRST or SID\_UPDATE or ONSET) shall be included in one RTP packet, Interleaving (redundancy) shall not be used, and a packetization time of 20ms shall be applied. No\_Data frames should not be sent, except when needed for urgent Rate Control. + +Nodes in the core network (e.g. MGWs) transcoding between AMR and some other Codec shall observe the following rules: + +- An AMR Encoder (sender) in the core network shall obey an AMR Codec Mode change period of 40ms, i.e. Codec Mode changes by the AMR Encoder (sender) in this core network node are only permissible at every second frame. This ensures maximum interoperability with any AMR receiver. +- An AMR Decoder (receiver) shall, however, be able to accept Codec Mode changes at any time. Variations of the Codec Mode period in receive direction may happen due to handover or other events during a conversation. The UMTS\_AMR Codec Type (only allowed in R99 UTRAN-only terminals) may change its Codec Mode any time. Other application of the AMR Codec Types (e.g. MTSI) may perform Codec Mode changes any time. This ensures maximum interoperability with any AMR sender. +- An AMR Encoder shall only change the Codec Mode to a neighbouring mode of the defined AMR Configuration (one step up or one step down), regardless which Rate Control command it receives. If necessary the AMR Encoder shall apply several Codec Mode changes in a row, if the received Rate Control command requests a change of more than one step. This ensures maximum interoperability with any AMR receiver, especially within GSM terminals. +- An AMR Decoder (receiver) shall, however, be able to accept Codec Mode changes in any step size. Variations of the Codec Mode in receive direction may happen due to handover or other events during a + +conversation. Other application of the AMR Codec Types (e.g. MTSI) may perform any Codec Mode changes. This ensures maximum interoperability with any AMR sender. + +- DTX (SCR) shall be supported in send and receive direction. + +AMR Rate Control shall use the CMR bits inside the RTP payload, both, in send and receive direction. RTCP shall not be used for AMR Rate Control in a CS core network. + +Rate Control Commands coming from an Nb-Interface of a BICC-based Core Network, or an Iu-Interface, or an IMS-Interface, or an general VoIP-Interface, shall be converted to CMR and shall be send continuously inside RTP packets together with the next Speech or SID\_FIRST or SID\_UPDATE or ONSET frame. + +NOTE: In a SIP-I -based Circuit Switched Core Network no Nb-framing is applied and so also no "PDU Type 14" [7] exists for Rate Control Commands. + +It is allowed to send an artificially inserted No\_Data frame to transport an urgent CMR in RTP. Please note that a GSM radio subsystem connected via AoIP can not send No\_Data frames across the radio interface and will typically ignore such No\_Data frames. The use of No\_Data frames for CMR is especially helpful inside the Core Network at call setup to control the downlink mode for the Encoder inside the terminating MGW for the compression of the ringback tone or an announcement, when the originating MGW still blocks the speech path in forward direction to prevent fraud. + +## 9.3 AMR-WB + +AMR-WB (FR\_AMR-WB, OHR\_AMR-WB, OFR\_AMR-WB, UMTS\_AMR-WB) shall be packed according to IETF RFC 4867 [21]. + +The AMR-WB Codec Types can be used in conversational speech telephony services in a number of different Codec Configurations. The set of AMR-WB Codec Configurations is defined in TS 26.103 [14], Table 5.7-1. One of these Configurations, **Config-WB-Code 0**, shall be supported by all nodes supporting the AMR-WB codec in a SIP-I based circuit switched core network to ensure interoperability. However, it is recommended that nodes in the core network support all AMR-WB modes for maximum interoperability. + +The bandwidth efficient mode of RFC 4867 [21] shall be used. CRC and robust sorting shall not be applied. + +To avoid delay, a single frame (Speech or SID\_FIRST or SID\_UPDATE or ONSET) shall be included in one RTP packet, Interleaving (redundancy) shall not be used, and a packetization time of 20ms shall be applied. No\_Data frames should not be sent, except when needed for urgent Rate Control. + +Nodes in the core network (e.g. MGWs) transcoding between AMR-WB and some other Codec shall observe the following rules: + +- An AMR-WB Encoder (sender) in the core network shall obey an AMR-WB Codec Mode change period of 40ms, i.e. Codec Mode changes by the AMR-WB Encoder (sender) in this core network node are only permissible at every second frame. This ensures maximum interoperability with any AMR-WB receiver. +- An AMR-WB Decoder (receiver) shall, however, be able to accept Codec Mode changes at any time. Variations of the Codec Mode period in receive direction may happen due to handover or other events during a conversation. Other application of the AMR-WB Codec Types (e.g. MTSI) may perform Codec Mode changes any time. This ensures maximum interoperability with any AMR-WB sender. +- An AMR-WB Encoder shall only change the Codec Mode to a neighbouring mode of the defined AMR-WB Configuration (one step up or one step down), regardless which Rate Control command it receives. If necessary the AMR-WB Encoder shall apply several Codec Mode changes in a row, if the received Rate Control command requests a change of more than one step. This ensures maximum interoperability with any AMR-WB receiver, especially within GSM terminals. +- An AMR-WB Decoder (receiver) shall, however, be able to accept Codec Mode changes in any step size. Variations of the Codec Mode in receive direction may happen due to handover or other events during a conversation. Other application of the AMR-WB Codec Types (e.g. MTSI) may perform any Codec Mode changes. This ensures maximum interoperability with any AMR-WB sender. +- DTX (SCR) shall be supported in send and receive direction. + +AMR-WB Rate Control shall use the CMR bits inside the RTP payload, both, in send and receive direction. RTCP shall not be used for AMR-WB Rate Control in a CS core network. + +Rate Control Commands coming from an Nb-Interface of a BICC-based Core Network, or an Iu-Interface, or an IMS-Interface, or an general VoIP-Interface, shall be converted to CMR and shall be sent continuously inside RTP packets together with the next Speech or SID\_FIRST or SID\_UPDATE or ONSET frame. + +NOTE: In a SIP-I -based Circuit Switched Core Network no Nb-framing is applied and so also no "PDU Type 14" [7] exists for Rate Control Commands. + +It is allowed to send an artificially inserted No\_Data frame to transport an urgent CMR in RTP. Please note that a GSM radio subsystem connected via AoIP can not send No\_Data frames across the radio interface and will typically ignore such No\_Data frames. The use of No\_Data frames for CMR is especially helpful on the AoIP-Interface in uplink and inside the Core Network at call setup to control the downlink mode for the Encoder inside the terminating MGW for the compression of the ringback tone or an announcement, when the originating MGW still blocks the speech path in forward direction to prevent fraud. + +## 9.4 GSM\_EFR + +GSM\_EFR shall be packed according to IETF RFC 3551 [17]. + +The coding of SID frames is based on the coding of Speech frames by setting the 95 bits of the so called "SID-Codeword" all to "1", see TS 46.062 [30]. + +To avoid delay, a single frame (Speech or SID) shall be included in one RTP packet, Interleaving (redundancy) shall not be used, and a packetization time of 20 ms shall be applied. No\_Data frames shall not be sent. + +DTX shall be supported in send and receive direction. + +GSM\_EFR frames received from some interface (e.g. a GSM radio interface via TFO) with a bad frame indication set to "bad" shall not be forwarded on the Nb-Interface in a SIP-I -based Circuit Switched Core Network, but silently discarded. + +NOTE: RFC 3551 [17] does not support the concept of Bad Frame Indication. + +## 9.5 GSM\_FR + +GSM\_FR shall be packed according to IETF RFC 3551 [17]. + +The coding of SID frames is based on the coding of Speech frames by setting the 95 bits of the so called "SID-Codeword" all to "0", see TS 46.012 [28]. + +To avoid delay, a single frame (Speech or SID) shall be included in one RTP packet, Interleaving (redundancy) shall not be used, and a packetization time of 20ms shall be applied. No\_Data frames shall not be sent. + +DTX shall be supported in send and receive direction. + +GSM\_FR frames received from some interface (e.g. a GSM radio interface via TFO) with a bad frame indication set to "bad" shall not be forwarded on the Nb-Interface in a SIP-I based Circuit Switched Core Network, but silently discarded. + +NOTE: RFC 3551 [17] does not support the concept of Bad Frame Indication. + +## 9.6 GSM\_HR + +GSM\_HR shall be packed according to [22]. + +The options specified in [22] are not applied inside the Circuit Switched Core Network, but set to pre-defined values as follows: + +A single frame (Speech or SID) shall be included in one RTP packet, FEC and Interleaving (redundancy) shall not be used, Encryption shall not be used, a packetization time of 20ms shall be applied. No\_Data frames shall not be sent. + +DTX shall be supported in send and receive direction. + +GSM\_HR frames received from some interface (e.g. a GSM radio interface via TFO) with a bad frame indication set to "bad" shall not be forwarded on the Nb-Interface in a SIP-I -based Circuit Switched Core Network, but silently discarded. + +NOTE: [22] does not support the concept of Bad Frame Indication. + +## 9.7 PCM + +PCMU and PCMA shall be packed according to IETF RFC 3551 [17]. + +The PCM packetization time for a SIP-I -based Core Network is negotiated via SDP. The mandatory, default value is 20ms; 5ms is one other, optional value; no other packetization time shall be used. To avoid delay, a single frame of length equal to the packetization time shall be included in one RTP packet, Interleaving (redundancy) shall not be used. + +The usage of DTX for PCM-coded speech is not recommended for NboIP. + +## 9.8 Telephone-Event + +Telephony-Event (DTMF) shall be encoded according to IETF RFC 4733 [20]. + +The audio/telephone-event payload type in IETF RFC 4733 [20] with default events and default rate shall be used to encode DTMF, if compressed speech is used in a SIP-I -based Core Network. Only in case of PCM-coded speech on NboIP the Telephone-Event is optional, i.e. also inband DTMF tones may be used (see TS 23.231 [12]). + +# 10 A-Interface User Plane over IP + +## 10.1 Overview + +The A interface User Plane over IP (AoIP) is standardised in the 3GPP TS 48-series (mainly in TS 48.008) [33] for the Control Plane and in TS 48.103 [34] for the User Plane). + +For AoIP the same Codecs as described in Clause 9 are applicable, except telephone-event, see table 10.1.1. Those Codecs shall also be applied in the same manner as described in Clause 9, unless otherwise specified in the present Clause 10. + +**Table 10.1.1 Supported Codec Types for the A interface User Plane over IP** + +| Payload Type Name | References | Remarks | Support | +|-------------------|--------------------|--------------------------------------------------|----------------------------------------------------------------------------------------------------------------------------| +| audio/AMR | IETF RFC 4867 [21] | Applicable for FR_AMR, HR_AMR, OHR_AMR | Optional.
Not all AMR Configurations are mandatory. Some Configurations are preferred, see chapter 9. | +| audio/AMR-WB | IETF RFC 4867 [21] | Applicable for FR_AMR-WB, OHR_AMR-WB, OFR_AMR-WB | Optional.
AMR-WB is mandatory, if WB speech is supported.
Not all AMR-WB Configurations are mandatory, see chapter 9 | +| audio/GSM_EFR | IETF RFC 3551 [17] | | Optional | +| audio/GSM_FR | IETF RFC 3551 [17] | | Mandatory | +| audio/GSM_HR | IETF RFC 5993 [22] | | Optional | +| audio/PCMA | IETF RFC 3551 [17] | ITU-T G.711 Alaw | Optional | +| audio/PCMU | IETF RFC 3551 [17] | ITU-T G.711 ulaw | Optional | + +The RTP "Payload Type" for AoIP is pre-determined by 3GPP TS 48.103 [34] for all Codec Types (static payload type). + +### 10.1.1 Time Alignment Procedure + +Time Alignment (and AMR Phase Alignment) is not specified for AoIP. + +### 10.1.2 SID Frame Generation + +All 3GPP Codec Types include standardized Discontinuous Transmission (DTX) with Voice Activity Detection, Silence Description (by SID frames) and Comfort Noise Generation to fill the speech pauses. If speech inactivity is detected by the Encoder, then (speech) frames are not transmitted, but the transmission is suspended in order to save battery life time in the mobile station, reduce interference on the radio interface and reduce load on all links. The receiving Decoder fills these transmission pauses with Comfort Noise to minimize the contrast between pauses and active speech. Silence Descriptor frames need to be sent during speech inactivity to keep the Comfort Noise decently well aligned with the background noise at sender side. This is especially important at the onset of the next talk spurt and therefore SID frames should not be too old, when speech starts again. + +The generation of SID frames for the AMR and AMR-WB families of Codecs is determined by the Speech Encoder as specified in TS 26.093 [31], respectively TS 26.193 [32]. The radio subsystem does not influence this timing! SID frames come during speech pauses in uplink and downlink about every 160ms. Also an AMR Encoder in the Media Gateway generates and sends SID frames about every 160ms. + +The generation of SID frames for GSM\_FR, GSM\_HR and GSM\_EFR in the GSM radio network is determined by the GSM mobile station and the GSM radio subsystem, not primarily by the Speech Encoder! SID frames come during speech pauses in uplink from the mobile station about every 480ms. In downlink to the mobile station, when they are generated by the Speech Encoder of the GSM radio subsystem, SID frames are sent every 20ms to the GSM base station, which then picks only one every 480ms for downlink radio transmission. For other applications, like transport over the A-Interface, it is more appropriate to send the SID frames less often than every 20ms, but 480ms may be too sparse. As a compromise it is recommended that an Encoder in the Media Gateway should generate and send SID frames every 160ms. + +### 10.1.3 Initial Codec Mode + +The Initial Codec Mode for AMR and AMR-WB shall be derived by pre-defined rules from the AMR Configuration (Active Codec Set), see TS 45.009 [35], chapter 3.4.3 "Initial Codec Mode Selection at Call Setup and Handover". + +Start of extract from TS 45.009 [35] for information and ease of reading: + +"If the Initial Codec Mode is not signalled, then the default Initial Codec Mode is given by the following implicit rule. If the Active Codec Set contains: + +- 1 mode, then this shall be the Initial Codec Mode; +- 2 or 3 modes, then the Initial Codec Mode shall be the most robust mode of the set (with lowest bit rate); +- 4 modes, then the Initial Codec Mode shall be the second most robust mode of the set (with second lowest bit rate." + +End of extract from TS 45.009 [35]. + +In case of FR\_AMR (Set 1), i.e. Config-NB-Code 1, see below, this is the AMR Mode with 5.90 kbps. + +## 10.2 AMR + +AMR (FR\_AMR, HR\_AMR, OHR\_AMR) shall be packed according to IETF RFC 4867 [21]. + +The AMR Codec Types can be used in conversational speech telephony services in a number of different Codec Configurations. The set of preferred AMR Codec Configurations is defined in TS 28.062 [5], Table 7.11.3.1.3-2. One of these preferred Configurations, **Config-NB-Code 1**, is recommended for TFO-TrFO harmonisation between GSM and UMTS networks. This Configuration shall be supported in an AoIP supporting BSS and an AoIP supporting Circuit Switched Core Network to ensure interoperability. However, it is recommended that a BSS and Circuit Switched Core Network supports all AMR modes for maximum interoperability. + +The bandwidth efficient mode of RFC 4867 [21] shall be used. CRC and robust sorting shall not be applied. + +To avoid delay, a single frame (Speech or SID\_FIRST or SID\_UPDATE or ONSET) shall be included in one RTP packet, Interleaving (redundancy) shall not be used, and a packetization time of 20ms shall be applied. No\_Data frames should not be sent downlink across AoIP, except to transport an urgent CMR in RTP. The use of No\_Data frames for CMR is especially helpful on the AoIP-Interface in uplink and inside the Core Network at call setup to control the + +downlink mode for the Encoder inside the terminating MGW for the compression of the ringback tone or an announcement, when the originating MGW still blocks the speech path in forward direction to prevent fraud. + +Please note that a GSM radio subsystem can not send No\_Data frames across the radio interface and will typically ignore such No\_Data frames in downlink direction. + +DTX (SCR) shall be supported in send and receive direction. + +AMR Rate Control shall use the CMR bits inside the RTP payload, both, in send and receive direction. RTCP shall not be used for AMR Rate Control in a CS network. + +## 10.3 AMR-WB + +AMR-WB (FR\_AMR-WB, OHR\_AMR-WB, OFR\_AMR-WB) shall be packed according to IETF RFC 4867 [21]. + +The AMR-WB Codec Types can be used in conversational speech telephony services in a number of different Codec Configurations. The set of AMR-WB Codec Configurations is defined in TS 26.103 [14], Table 5.7-1. One of these Configurations, **Config-WB-Code 0**, shall be supported by all nodes supporting the AMR-WB codec in an AoIP-supporting BSS and an AoIP-supporting Circuit Switched Core Network to ensure interoperability. However, it is recommended that nodes in the Core Network support all AMR-WB modes for maximum interoperability. + +The bandwidth efficient mode of RFC 4867 [21] shall be used. CRC and robust sorting shall not be applied. + +To avoid delay, a single frame (Speech or SID\_FIRST or SID\_UPDATE or ONSET) shall be included in one RTP packet, Interleaving (redundancy) shall not be used, and a packetization time of 20ms shall be applied. No\_Data frames should not be sent downlink across AoIP, except to transport an urgent CMR in RTP. The use of No\_Data frames for CMR is especially helpful on the AoIP-Interface in uplink and inside the Core Network at call setup to control the downlink mode for the Encoder inside the terminating MGW for the compression of the ringback tone or an announcement, when the originating MGW still blocks the speech path in forward direction to prevent fraud. + +Please note that a GSM radio subsystem can not send No\_Data frames across the radio interface and will typically ignore such No\_Data frames in downlink direction. + +DTX (SCR) shall be supported in send and receive direction. + +AMR-WB Rate Control shall use the CMR bits inside the RTP payload, both, in send and receive direction. RTCP shall not be used for AMR-WB Rate Control in a Circuit Switched Core Network. + +## 10.4 GSM\_EFR + +GSM\_EFR shall be packed according to IETF RFC 3551 [17]. + +The coding of SID frames is based on the coding of Speech frames by setting the 95 bits of the so called "SID-Codeword" all to "1", see TS 46.062 [30]. + +To avoid delay, a single frame (Speech or SID) shall be included in one RTP packet, Interleaving (redundancy) shall not be used, and a packetization time of 20 ms shall be applied. No\_Data frames shall not be sent. + +DTX shall be supported in send and receive direction. + +NOTE: RFC 3551 [17] does not support the concept of Bad Frame Indication. Therefore missing GSM\_EFR frames in the AoIP downlink direction (e.g. discarded by a network node due to the missing bad frame indication) need to be properly treated within the BSS before sending downlink on the radio interface. Details are not specified. + +## 10.5 GSM\_FR + +GSM\_FR shall be packed according to IETF RFC 3551 [17]. + +The coding of SID frames is based on the coding of Speech frames by setting the 95 bits of the so called "SID-Codeword" all to "0", see TS 46.012 [28]. + +To avoid delay, a single frame (Speech or SID) shall be included in one RTP packet, Interleaving (redundancy) shall not be used, and a packetization time of 20 ms shall be applied. No\_Data frames shall not be sent. + +DTX shall be supported in send and receive direction. + +NOTE: RFC 3551 [17] does not support the concept of Bad Frame Indication. Therefore missing GSM\_EFR frames in the AoIP downlink direction (e.g. discarded by a network node due to the missing bad frame indication) need to be properly treated within the BSS before sending downlink on the radio interface. Details are not specified. + +## 10.6 GSM\_HR + +GSM\_HR shall be packed according to [22]. + +The options specified in [22] are not applied on AoIP, but set to pre-defined values as follows: + +A single frame (Speech or SID) shall be included in one RTP packet, FEC and Interleaving (redundancy) shall not be used, Encryption shall not be used, a packetization time of 20ms shall be applied. No\_Data frames shall not be sent. + +DTX shall be supported in send and receive direction. + +NOTE: [22] does not support the concept of Bad Frame Indication. Therefore missing GSM\_HR frames in the AoIP downlink direction (e.g. discarded by a network node due to the missing bad frame indication) need to be properly treated within the BSS before sending downlink on the radio interface. Details are not specified. + +## 10.7 PCM + +PCMU and PCMA shall be packed according to IETF RFC 3551 [17]. + +A packetization time of 20ms shall be applied for PCM on AoIP. The packetization time is not negotiated for AoIP. To avoid delay, a single frame of 20ms shall be included in one RTP packet, Interleaving (redundancy) shall not be used. + +The usage of DTX for PCM-coded speech is not allowed on AoIP. + +# Annex A (informative): Change history + +| Change history | | | | | | | | +|----------------|-------|-----------|------|-----|--------------------------------------------------------------------------------|--------|--------| +| Date | TSG # | TSG Doc. | CR | Rev | Subject/Comment | Old | New | +| 1999-12 | 6 | SP-99563 | | | Approved at TSG-SA#6 Plenary | | 3.0.0 | +| 2000-03 | 7 | SP-000025 | 001 | 3 | Introduction of QoS parameters used at RAB assignment | 3.0.0 | 3.1.0 | +| 2000-03 | 7 | SP-000025 | 002 | | Introduction of different RFCS set on Iu User Plane | 3.0.0 | 3.1.0 | +| 2000-03 | 7 | SP-000025 | 003 | 2 | Introduction of Time Alignment | 3.0.0 | 3.1.0 | +| 2000-12 | 10 | SP-000575 | 005 | 1 | AMR interface to Iu | 3.1.0 | 3.2.0 | +| 2001-03 | 11 | SP-010103 | 006 | 2 | Removal of TFO and TrFO from Release 99, and removal of Initial Time Alignment | 3.2.0 | 3.3.0 | +| 2001-03 | 11 | SP-010103 | 008 | 1 | Introduction of TFO and TrFO | 3.3.0 | 4.0.0 | +| 2002-06 | 16 | | | | Version for Release 5 | 4.0.0 | 5.0.0 | +| 2002-12 | 18 | SP-020689 | 012 | 2 | Correction of RAB parameter assignment for AMR | 5.0.0 | 5.1.0 | +| 2003-03 | 19 | SP-030087 | 015 | 2 | AMR Rate Adaptation of Rel-5 | 5.1.0 | 5.2.0 | +| | | | | | | | | +| 2004-04 | 25 | SP-040645 | 016 | 1 | Mapping of GSM_EFR SID on Nb Interface | 5.2.0 | 6.0.0 | +| | | | | | | | | +| 2005-12 | 30 | SP-050791 | 0017 | | 20 ms packetisation time for PCM coded speech over IP Nb | 6.0.0 | 7.0.0 | +| 2006-06 | 32 | SP-060358 | 0018 | 1 | Supplement of 20 ms packetisation time for PCM coded speech over IP Nb | 7.0.0 | 7.1.0 | +| 2008-06 | 41 | SP-080475 | 0019 | 2 | Addition of CS over IP User Plane | 7.1.0 | 8.0.0 | +| 2008-06 | 41 | SP-080475 | 0020 | 1 | Nb-framing for GSM_FR and GSM_HR | 7.1.0 | 8.0.0 | +| 2008-12 | 42 | SP-080678 | 0021 | 3 | Corrections to CS over IP User Plane | 8.0.0 | 8.1.0 | +| 2009-09 | 45 | SP-090568 | 0023 | 1 | Clarification of RAB sub-flow numbering for AMR | 8.1.0 | 8.2.0 | +| 2009-12 | 46 | SP-090703 | 0024 | 1 | Correction of payload field size for mapping of GSM FR on Nb interface | 8.2.0 | 8.3.0 | +| 2009-12 | 46 | | | | Version for Release 9 | 8.3.0 | 9.0.0 | +| 2010-03 | 47 | SP-100018 | 0026 | | Corrections and clarifications of Nb frames | 9.0.0 | 9.1.0 | +| 2011-03 | 51 | SP-110034 | 0028 | | Correction of reference for GSM-HR payload format | 9.1.0 | 9.2.0 | +| 2011-03 | 51 | | | | Version for Release 10 | 9.2.0 | 10.0.0 | +| 2012-09 | 57 | SP-120508 | 0029 | 2 | Clarification of Text Regarding Bit Protection Classes and RAB sub-flows | 10.0.0 | 11.0.0 | \ No newline at end of file diff --git a/marked/Rel-11/26_series/26103/raw.md b/marked/Rel-11/26_series/26103/raw.md new file mode 100644 index 0000000000000000000000000000000000000000..113bd2b6de2d47de55d550a2ac3a3ff537262c46 --- /dev/null +++ b/marked/Rel-11/26_series/26103/raw.md @@ -0,0 +1,822 @@ + + + + + + +# Contents + +| | | +|----------------------------------------------------------------------------------------------------------|-----------| +| Foreword ..... | 4 | +| 1 Scope..... | 5 | +| 2 Normative references ..... | 5 | +| 3 Definitions and Abbreviations ..... | 6 | +| 3.1 Definitions..... | 6 | +| 3.2 Abbreviations ..... | 6 | +| 4 General ..... | 7 | +| 5 3GPP Codec List for OoBTC in a BICC-based Circuit Switched Core Network and for AoIP ..... | 9 | +| 5.1 GSM Full Rate Codec Type (GSM FR)..... | 9 | +| 5.2 GSM Half Rate Codec Type (GSM HR) ..... | 9 | +| 5.3 GSM Enhanced Full Rate Codec Type (GSM EFR)..... | 9 | +| 5.4 Five Adaptive Multi-Rate Codec Types (FR AMR, HR AMR, UMTS AMR, UMTS AMR2, OHR AMR) ..... | 10 | +| 5.5 TDMA Enhanced Full Rate Codec Type (TDMA EFR) ..... | 13 | +| 5.6 PDC Enhanced Full Rate Codec Type (PDC_EFR) ..... | 13 | +| 5.7 Four Adaptive Multi-Rate Wideband Codec Types (FR AMR-WB, UMTS AMR-WB, OFR AMR-WB, OHR AMR-WB) ..... | 13 | +| 5.8 MuMe Dummy Codec (3G.324M)..... | 16 | +| 5.9 MuMe2 Dummy Codec (3G.324M2)..... | 17 | +| 5.10 Codec Extension..... | 17 | +| 5.11 CSData Dummy Codec (AoIP)..... | 17 | +| 6 Codec List for the Call Control Protocol ..... | 17 | +| 6.1 System Identifiers for GSM and UMTS ..... | 17 | +| 6.2 Codec Bitmap..... | 18 | +| 6.3 Selected Codec Type..... | 19 | +| 7 3GPP Codecs for OoBTC in a SIP-I -based Circuit Switched Core Network..... | 20 | +| 7.1 Overview ..... | 20 | +| 7.2 AMR..... | 20 | +| 7.3 AMR-WB..... | 21 | +| 7.4 GSM_EFR ..... | 21 | +| 7.5 GSM_FR ..... | 21 | +| 7.6 GSM_HR..... | 21 | +| 7.7 PCM ..... | 22 | +| 7.8 Telephone-Event ..... | 22 | +| Annex A (informative): Example Supported Codec List for UMTS..... | 23 | +| Annex B (informative): Change history..... | 25 | + +# --- Foreword + +This Technical Specification has been produced by the 3rd Generation Partnership Project (3GPP). + +The contents of the present document are subject to continuing work within the TSG and may change following formal TSG approval. Should the TSG modify the contents of the present document, it will be re-released by the TSG with an identifying change of release date and an increase in version number as follows: + +Version x.y.z + +where: + +- x the first digit: + - 1 presented to TSG for information; + - 2 presented to TSG for approval; + - 3 or greater indicates TSG approved document under change control. +- y the second digit is incremented for all changes of substance, i.e. technical enhancements, corrections, updates, etc. +- z the third digit is incremented when editorial only changes have been incorporated in the document. + +# --- 1 Scope + +The present Technical Specification outlines the Codec Lists in 3GPP including both systems, GSM and UMTS, to be used by the Out of Band Transcoder Control (OoBTC) protocol to set up a call or modify a call in **Transcoder Free Operation** (TrFO) and in "transcoder at the edge" scenarios. + +The TS also specifies the SDP description of 3GPP Codecs to be used within a SIP-I -based circuit switched core network as specifies in 3GPP TS 23.231 [14]. + +The TS further specifies the coding of the Supported Codec List Information Elements for the UMTS radio access technology. + +The TS further reserves the Code Point for the CSData (dummy) Codec Type for the negotiation of A-Interface Type and the RTP redundancy for CS Data and Fax services, see 3GPP TS 48.008 [23]. + +The Supported Codec List IE includes Codec\_Types from the TDMA and PDC systems, to support TFO or TrFO between UMTS and TDMA, or UMTS and PDC. + +# --- 2 Normative references + +The following documents contain provisions which, through reference in this text, constitute provisions of the present document. + +- References are either specific (identified by date of publication, edition number, version number, etc.) or non-specific. +- For a specific reference, subsequent revisions do not apply. +- For a non-specific reference, the latest version applies. In the case of a reference to a 3GPP document (including a GSM document), a non-specific reference implicitly refers to the latest version of that document *in the same Release as the present document*. + +- [1] 3GPP TS 26.090: "AMR Speech Codec; Speech Transcoding Functions". +- [2] 3GPP TS 26.093: "AMR Speech Codec; Source Controlled Rate Operation". +- [3] 3GPP TS 26.101: "Mandatory Speech Codec Speech Processing Functions; AMR Speech Codec Frame Structure". +- [4] 3GPP 46.0xx: "Enhanced Full Rate Codec Recommendations". +- [5] 3GPP 26.0xx: "Adaptive Multi-Rate Codec Recommendations". +- [6] "ITU Q.765.5: "Use of Application Transport Mechanism for Bearer Independent Call Control" +- [7] 3GPP TS 28.062: "In-band Tandem Free Operation (TFO) of Speech Codecs, Stage 3 - Service Description". +- [8] 3GPP TS 23.153: "Out of Band Transcoder Control - Stage 2". +- [9] 3GPP TS 24.008: "Mobile radio interface layer 3 specifications, Core Network Protocols" +- [10] 3GPP TS 26.190: "AMR Wideband Speech Codec; Speech Transcoding Functions". +- [11] 3GPP TS 26.193: "AMR Wideband Speech Codec; Source Controlled Rate Operation". +- [12] 3GPP TS 26.201: "Mandatory Speech Codec Speech Processing Functions; AMR Wideband Speech Codec Frame Structure". +- [13] 3GPP TS 23.172: "CS multimedia service UDI/RDI fallback and service modification; Stage 2". +- [14] 3GPP TS 23.231: "SIP-I based circuit-switched core network; Stage 2". + +- [15] 3GPP TS 29.007: "General requirements on interworking between the Public Land Mobile Network (PLMN) and the Integrated Services Digital Network (ISDN) or Public Switched Telephone Network (PSTN)". +- [16] IETF RFC 3264 (2002): "An Offer/Answer Model with the Session Description Protocol (SDP)", J. Rosenberg and H. Schulzrinne. +- [17] IETF RFC 3551 (2003): "RTP Profile for Audio and Video Conferences with Minimal Control", H. Schulzrinne and S. Casner. +- [18] void +- [19] IETF RFC 4566 (2006): "SDP: Session Description Protocol", M. Handley, V. Jacobson and C. Perkins. +- [20] IETF RFC 4733 (2006): "RTP Payload for DTMF Digits, Telephony Tones, and Telephony Signals", H. Schulzrinne and T. Taylor. +- [21] IETF RFC 4867 (2007): "RTP Payload Format and File Storage Format for the Adaptive Multi-Rate (AMR) and Adaptive Multi-Rate Wideband (AMR-WB) Audio Codecs", J. Sjoberg, M. Westerlund, A. Lakaniemi and Q. Xie. +- [22] IETF RFC 5993 (2010) "RTP Payload Format for Global System for Mobile Communications Half Rate (GSM-HR)". +- [23] 3GPP TS 48.008: "Mobile Switching Centre - Base Station System (MSC-BSS) interface". +- [24] 3GPP TS 26.102: "Adaptive Multi-Rate (AMR) speech codec; Interface to Iu, Uu and Nb".". + +# --- 3 Definitions and Abbreviations + +## 3.1 Definitions + +**Codec Type:** defines a specific type of a speech Coding algorithm, applied on a specific radio access technology (e.g. GSM FR, (GSM) FR AMR). + +**Codec Mode:** defines a specific mode of a Codec Type (e.g. 12,2 kBit/s Mode of the (GSM) FR AMR). + +**Codec Configuration:** defines a specific set of attributes to a certain Codec Type (e.g. the combination of ACS and DTX="on" for (GSM) FR AMR). + +**Organisation Identifier (OID):** Identifies the standard organisation (e.g. 3GPP) producing a specification for a Codec List. ITU-T is responsible for maintaining the list of Organisation Identifiers. + +**System Identifier (SysID):** Identifies the radio access technology (e.g. GSM or UMTS) for which the supported Codec List is defined. + +Other definitions are given in TS 23.153 [8]. + +## 3.2 Abbreviations + +For the purposes of the present document, the following abbreviations apply: + +| | | +|--------|-------------------------------------------| +| ACS | Active Codec (mode) Set | +| AoIP | A-Interface User Plane over IP | +| BWM | BandWidth Multiplier | +| CoID | Codec Identifier | +| CSData | Circuit Switched Data and Fax dummy Codec | +| DTX | Discontinuous Transmission | + +| | | +|-------|-----------------------------------------------------------------------------------------------------------------| +| GSM | Global System for Mobile communication | +| MuMe | Multi-Media | +| NbIoP | Nb-Interface User Plane transport over IP in a SIP-I -based network | +| OID | Organisation IDeNtifier (e.g. ITU-T, 3GPP) | +| OoBTC | Out of Band Transcoder Control | +| PDC | Personal Digital Communication (synonym for ...) | +| RX | Receive | +| SCR | Source Controlled Rate operation (synonym to DTX ) | +| SID | Silence Descriptor | +| SysID | System Identifier | +| TDMA | Time Division Multiple Access (synonym for ...) | +| TFO | T andem F ree O peration
(also sometimes called "Transcoder-Through" or "Codec-Bypass") | +| TrFO | T ranscoder F ree O peration | +| TX | Transmit | +| UMTS | Universal Mobile Telecommunications System | + +# --- 4 General + +The present Technical Specification (TS) outlines the 3GPP internal Codec Lists for both, GSM and UMTS, to be used by the Out of Band Transcoder Control (OoBTC) protocol a BICC-based Circuit Switched Core Network to set up a call or modify a call in Transcoder Free Operation (TrFO). The Codec List is also used in the Codec Negotiation for the A-Interface User Plane over IP (AoIP), see 3GPP TS 48.008 [23]. + +The TS specifies the SDP parameters for the 3GPP Codecs for OoBTC in a SIP-I -based Circuit Switched Core Network, see 3GPP TS 23.231 [14]. + +The TS further specifies the coding of the Supported Codec List Information Elements as defined in 3GPP TS 24.008 for the UMTS radio access technology. + +Transcoder Free Operation allows the transport of speech signals in the coded domain from one user equipment (UE) to the other user equipment through the radio access network (RAN) and core network (CN), possibly through a transit network (TN). This enables high speech quality, low transmission costs and high flexibility. + +The necessary Codec Type selection and resource allocation are negotiated out of band before and after call setup. Possible Codec (re-)configuration, Rate Control and DTX signalling may be performed after call setup by additional inband signalling or a combination of inband and out-of-band signalling. + +Up to release '99 GSM does not support Transcoder Free Operation, but specifies the Tandem Free Operation (TFO). Tandem Free Operation enables similar advantages, but is based on pure inband signalling after call setup. The parameters defined in this Technical Specification allow interaction between TrFO and TFO. They further provide an evolutionary path for GSM towards Transcoder Free Operation. + +The GERAN and UTRAN standards define fourteen different Speech Codec Types, see table 4.1. + +In addition to these Speech Codec Types some "dummy" Codec Types are defined to support the negotiation for data, fax and multimedia applications. + +**Table 4.1: Support of Codec Types in Radio Access Technologies** + +| | TDMA
EFR | UMTS
AMR 2 | UMTS
AMR | (GSM)
HR AMR | (GSM)
FR AMR | GSM
EFR | GSM
HR | GSM
FR | +|-----------------------|---------------------|----------------------------------------|---------------------|-------------------------|-------------------------|--------------------|-------------------|-------------------| +| CoID | 0x07 | 0x06 | 0x05 | 0x04 | 0x03 | 0x02 | 0x01 | 0x00 | +| GERAN
GMSK | not defined | not possible | not possible | yes, 1..4 modi | yes, 1..4 modi | yes | yes | yes | +| GERAN
8PSK | not defined | not possible | not possible | not defined | not defined | not defined | not defined | not defined | +| UTRAN | not defined | yes, 1..8 modi
1..4 modi
recomm. | R99, UTRAN-only UEs | not defined | not defined | not defined | not defined | not defined | + +| | Codec
Extension | | OHR
AMR-WB | OFR
AMR-WB | OHR
AMR | UMTS
AMR-WB | FR
AMR-WB | PDC
EFR | +|-----------------------|----------------------------|-------------|-----------------------|-----------------------|--------------------|------------------------|----------------------|--------------------| +| CoID | 0x0F | 0x0E | 0x0D | 0x0C | 0x0B | 0x0A | 0x09 | 0x08 | +| GERAN
GMSK | reserved | spare | not defined | not defined | not defined | not possible | yes3 modi | not defined | +| GERAN
8PSK | reserved | spare | yes, 3 modi | yes, 3 modi | yes, 1..4 modi | not possible | not defined | not defined | +| UTRAN | reserved | spare | not defined | not defined | not defined | yes 3..4 modi | not defined | not defined | + +CoID is reprinted here in hexadecimal notation. It is defined in section 5. + +Up to date the following Code Points are defined: + +**Table 4.2. Defined Code Points** + +| Hexadecimal
Notation | Binary
Notation | Codec Name | Remark | +|---------------------------------|-----------------------------------|-----------------------|------------------| +| 0x00h | 0x0000.0000 | GSM_FR | | +| 0x01h | 0x0000.0001 | GSM_HR | | +| 0x02h | 0x0000.0010 | GSM_EFR | | +| 0x03h | 0x0000.0011 | (GSM) FR_AMR | | +| 0x04h | 0x0000.0100 | (GSM) HR_AMR | | +| 0x05h | 0x0000.0101 | UMTS_AMR | | +| 0x06h | 0x0000.0110 | UMTS_AMR2 | | +| 0x07h | 0x0000.0111 | TDMA_EFR | | +| 0x08h | 0x0000.1000 | PDC_EFR | | +| 0x09h | 0x0000.1001 | (GSM) FR_AMR-WB | | +| 0x0Ah | 0x0000.1010 | UMTS_AMR-WB | | +| 0x0Bh | 0x0000.1011 | OHR_AMR | | +| 0x0Ch | 0x0000.1100 | OFR_AMR-WB | | +| 0x0Dh | 0x0000.1101 | OHR_AMR-WB | | +| 0x0Eh | 0x0000.1110 | Spare, for future use | | +| 0x0Fh | 0x0000.1111 | Codec Extension | For AoIP and TFO | +| 0x10h ...
0xFCh | 0x0001.0000
...
0x1111.1100 | Spare, for future use | | +| 0xFDh | 0x1111.1101 | CSData | For AoIP only | +| 0xFEh | 0x1111.1110 | MuMe2 | For OoBTC only | +| 0xFFh | 0x1111.1111 | MuMe | For OoBTC only | + +# --- 5 3GPP Codec List for OoBTC in a BICC-based Circuit Switched Core Network and for AoIP + +The definition of the common Codec List for Out of Band Transcoder Control (3GPP TS 23.153, [8]) in 3GPP for GSM and UMTS follows the specifications given in ITU Q.765.5: The most preferred Codec Type is listed first, followed by the second preferred one, and so on. An informative example for a codec list for UMTS can be found in Annex A. + +The Codec Identification codes (CoIDs) are specified in two versions: the long form (8 bits) for the use in OoBTC and the short form (the 4 LSBs of the long form) for the use in TFO and AoIP. + +## 5.1 GSM Full Rate Codec Type (GSM FR) + +The Codec Identification (CoID) code is defined to be: FR\_CoID := 0x0000.0000. + +The GSM Full Rate Codec Type has no additional parameters. + +For information (for exact details see GSM Recommendations): + +The GSM Full Rate Codec Type supports one fixed Codec Mode with 13.0 kBit/s. + +DTX may be enabled in uplink and in downlink independently of each other. DTX on or off is defined by the network on a cell basis and can not be negotiated at call setup or during the call. The DTX scheme uses one SID frame to mark the end of a speech burst and to start Comfort Noise Generation. Identical SID frames for comfort noise updates are sent in speech pauses about every 480 ms, aligned with the cell's TDMA frame structure. The defined Tandem Free Operation allows the reception of GSM FR DTX information for the downlink direction in all cases. The TFO respectively TrFO partner is prepared to receive DTX information as well. + +## 5.2 GSM Half Rate Codec Type (GSM HR) + +The Codec Identification (CoID) code is defined to be: HR\_CoID := 0x0000.0001. + +The GSM Half Rate Codec Type has no additional parameters. + +For information (for exact details see GSM Recommendations): + +The GSM Half Rate Codec Type supports one fixed Codec Mode with 5.60 kBit/s. + +DTX may be enabled in uplink and in downlink independently of each other. DTX on or off is defined by the network on a cell basis and can not be negotiated at call setup or during the call. The DTX scheme uses one SID frame to mark the end of a speech burst and to start Comfort Noise Generation. Identical SID frames for comfort noise updates are sent in speech pauses about every 480 ms, aligned with the cell's TDMA frame structure. The defined Tandem Free Operation allows the reception of GSM HR DTX information for the downlink direction in all cases. The TFO respectively TrFO partner shall be prepared to receive DTX information as well. + +## 5.3 GSM Enhanced Full Rate Codec Type (GSM EFR) + +The Codec Identification (CoID) code is defined to be: EFR\_CoID := 0x0000.0010. + +The GSM Enhanced Full Rate Codec Type has no additional parameters. + +For information (for exact details see GSM Recommendations): + +The GSM Enhanced Full Rate Codec Type supports one fixed Codec Mode with 12.2 kBit/s. + +DTX may be enabled in uplink and in downlink independently of each other. DTX on or off is defined by the network on a cell basis and can not be negotiated at call setup or during the call. The DTX scheme uses one SID frame to mark the end of a speech burst and to start Comfort Noise Generation. It is important to note that the Comfort Noise parameters for this start of the comfort noise generation are calculated at transmitter side from the previous eight speech frames. A DTX hangover period needs to be applied therefore at transmitter side before sending the first SID frame. SID frames with incremental information for comfort noise updates are sent in speech pauses about every 480 ms, aligned with the cell's TDMA frame structure. The defined Tandem Free Operation allows the reception of GSM EFR + +DTX information for the downlink direction in all cases. The TFO respectively TrFO partner shall be prepared to receive DTX information as well. + +## 5.4 Five Adaptive Multi-Rate Codec Types (FR AMR, HR AMR, UMTS AMR, UMTS AMR2, OHR AMR) + +The Adaptive Multi-Rate Codec algorithm is applied in GERAN-GMSK, GERAN-8PSK and UTRAN in five different Codec Types. + +The Codec Identification (CoID) codes are defined to be: + +FR\_AMR\_CoID := 0x0000.0011. + HR\_AMR\_CoID := 0x0000.0100. + UMTS\_AMR\_CoID := 0x0000.0101. + UMTS\_AMR\_2\_CoID := 0x0000.0110. + OHR\_AMR\_CoID := 0x0000.1011. + +The AMR Codec Types can be used in conversational speech telephony services in a number of different configurations. The set of preferred configurations is defined in TS 28.062, Table 7.11.3.1.3-2. One of these preferred configurations, Config-NB-Code 1, is recommended for TFO-TrFO harmonisation between GSM and UMTS networks, it is mandatory for an AoIP-supporting BSS, see 3GPP TS 48.008 [23], an AoIP-supporting BICC-based Circuit Switched Core Network and for any SIP-I -based Circuit Switched Core Network. + +The Single Codec Information Element for AMR Codec Types may have several additional parameters. These parameters are optional in the Supported Codec List (BICC) and in the Available Codec List (BICC), but these parameters shall specify exactly one AMR Configuration for the Selected Codec (BICC), see [8]. + +**Active Codec Set, ACS:** eight bits. + +Each bit corresponds to one AMR Mode. Setting the bit to "1" means the mode is included, setting the bit to "0" means the mode is not included in the ACS. + +Note: Except for HR\_AMR all eight AMR modes may be selected, for the HR\_AMR only the six lower modes. + +**Supported Codec Set, SCS:** eight bits. + +Each bit corresponds to one AMR Mode, as in the ACS. Setting the bit to "1" means the mode is supported, setting the bit to "0" means the mode is not supported. The SCS shall at least contain all modes of the ACS. + +**Maximal number of codec modes in the ACS, MACS:** three bits. + +MACS shall be used in the Supported Codec List (BICC) and the Available Codec List (BICC), when it is necessary to restrict the maximum number of modes for the (future) Selected Codec (BICC). + +For FR AMR, HR AMR and OHR AMR one up to four, for the UMTS AMR and UMTS AMR2 one up to eight Codec Modes are allowed. + +Coding: "001": one, "010": two, ... "111": seven, "000": eight Codec Modes allowed. + +**Optimisation Mode for ACS, OM:** one bit. + +OM indicates, whether the sending side supports the modification (optimisation) of its offered ACS for the needs of the distant side. + +Coding: "0": Optimisation of the ACS not supported, "1": Optimisation of the ACS supported. + +If OM is specified as "Optimisation of the ACS not supported", then SCS and MACS have no meaning for this Single Codec Information Element; then the SCS shall at least contain all modes of the offered ACS; MACS shall be equal to or larger than the number of modes in the offered ACS. + +**Usage of this Single Codec Information Element in OoBTC.** + +In the Single Codec Information Element for the Selected Codec (BICC) the ACS shall be specified exactly. + +For FR AMR, HR\_AMR and OHR AMR at least one, but not more than four modes shall be included. + +For UMTS AMR and UMTS AMR2 at least one, but not more than four modes should be included. + +OM shall be set to "Optimisation of the ACS not supported". + +In the Single Codec Information Element for the Supported Codec List (BICC) and the Available Codec List (BICC) one of the following codings shall be used + +- either all parameters (ACS, SCS, MACS and OM) are omitted. + +Then per default all possible AMR modes shall be treated as included in ACS and SCS, MACS shall be treated as set to its allowed maximum and OM shall be treated as set to "Optimisation of the ACS + +supported". + +- or only the ACS is specified: +Then per default all possible AMR modes shall be treated as included in the SCS, MACS shall be treated as set to its allowed maximum and OM shall be treated as set to "Optimisation of the ACS supported". +- or ACS and SCS are specified. +Then per default MACS shall be treated as set to its allowed maximum and OM shall be treated as set to "Optimisation of the ACS supported". +- or all parameters (ACS, SCS, MACS and OM) are specified. + +### Procedures in OoBTC + +The procedures for handling of these Single Codec Information Element in the originating, intermediate and terminating nodes are specified in TS 23.153 [8]. + +The "Single Codec" information element consists of 5 to 8 octets in case of the AMR Codec Types (table 5.4): + +**Table 5.4: Coding of "Single Codec" for the Adaptive Multi-Rate Codec Types** + +| Octet | Parameter | MSB 8 | 7 | 6 | 5 | 4 | 3 | 2 | 1 LSB | +|-------|-------------------|------------------------------------------------------------------------|---------|---------|---------|------|------|------|-------| +| 1 m | Single Codec | Single Codec (see ITU-T Q.765.5) | | | | | | | | +| 2 m | Length Indication | 3, 4, 5, 6 | | | | | | | | +| 3 m | Compat. Info | Compatibility Information | | | | | | | | +| 4 m | OID | ETSI OID (See ITU-T Q.765.5 [6]) | | | | | | | | +| 5 m | CoID | FR_AMR_CoID, HR_AMR_CoID, UMTS_AMR_CoID, UMTS_AMR_2_CoID, OHR_AMR_CoID | | | | | | | | +| 6 o | ACS | 12.2 | 10.2 | 7.95 | 7.40 | 6.70 | 5.90 | 5.15 | 4.75 | +| 7 o | SCS | 12.2 | 10.2 | 7.95 | 7.40 | 6.70 | 5.90 | 5.15 | 4.75 | +| 8 o | OM, MACS | (spare) | (spare) | (spare) | (spare) | OM | MACS | | | + +with "m" = mandatory and "o" = optional + +For information on GSM procedures (for exact details see GSM Recommendations): + +The GSM AMR Codec Types comprise eight (Full Rate), respectively six (Half Rate) different Codec Modes: 12,2 ... 4,75 kBit/s. + +The active Codec Mode is selected from the Active Codec Set (ACS) by the network (Codec Mode Command) with assistance by the mobile station (Codec Mode Request). This Codec Mode Adaptation, also termed Rate Control, can be performed every 40 ms by going one Codec Mode up or down within the ACS. The Codec Modes in uplink and downlink at one radio leg may be different. In Tandem Free Operation both radio legs (A and B) are considered for the optimal selection of the active Codec Mode in each direction (uplink A and then downlink B, respectively vice versa) by the "Distributed Rate Decision" algorithm. The worst of both radio legs determines the highest allowed Codec Mode, respectively the maximally allowed rate ("Maximum Rate Control"). All rate control commands are transmitted inband: on the radio interface, the BTS-TRAU interface and the TRAU-TRAU interface. + +The Active Codec Set is configured at call setup or reconfigured during the call. It consists of one up to maximally four Codec Modes (MACS) at a given time, selected from the Supported Codec Set. The maximal number of Codec Modes and the Supported Codec Set may be constrained by the network to consider resources and radio conditions. The Active Codec Sets in uplink and downlink are identical. + +First, at start up of Tandem Free Operation, Active Codec Sets, the Supported Codec Sets, the MACSs and the OMs are taken into account to determine the optimal common Active Codec Set. In a later phase the Codec Lists of both radio legs may be taken into account to find the optimum configuration. For exact details see 3GPP TS 28.062. All configuration data and update protocols are transmitted inband. + +The DTX scheme of the Adaptive Multi-Rate Codec Type marks with a specific SID\_FIRST frame the end of a speech burst. SID\_FIRST does not contain Comfort Noise parameters. This SID\_FIRST starts the comfort noise generation + +with parameters that are calculated at receiver side (!) from the latest received seven speech frames. A DTX hangover period needs to be applied therefore at transmitter side before sending of this SID\_FIRST. + +Absolutely coded SID\_UPDATE frames follow about every eighth frame (160 ms) in speech pauses. SID\_UPDATE frames are sent independently of the cell's TDMA frame structure and are related only to the source signal. + +An ONSET frame (typically) precedes in uplink direction the beginning of a new speech burst. DTX on or off is defined by the network on a cell basis. The defined Tandem Free Operation allows the reception of GSM-AMR DTX information for the downlink direction in all cases. + +Note: The DTX scheme of the Enhanced Full Rate Codec Type is not compatible with the DTX scheme of the Adaptive Multi-Rate Codec Type in Codec Mode 12.2 kBit/s, although the speech modes of these two Codec Types are bit exact identical. + +#### **Informative for terminals of R99 that support only UTRAN access ("UTRAN-only" terminals):** + +UTRAN-only terminals of R99 may either use UMTS AMR or UMTS AMR2 as default speech version in UTRAN access. + +#### **Normative for terminals that support GSM and UTRAN radio access ("dual-mode" terminals):** + +Dual-mode terminals of R99 and onwards shall use the UMTS AMR2 as the default speech version in UTRAN access. They need not to support the UMTS AMR, because the UMTS AMR2 in terminals is a fully compatible replacement. + +**Normative for all UMTS terminals of REL-4 and onwards:** The UMTS AMR2 shall be the default speech version in UTRAN access in all terminals, UTRAN-only and dual-mode (GSM and UTRAN) of REL-4 and onwards. + +For information on UMTS procedures (for exact details see 3GPP TS 28.062 (TFO) and 3GPP TS 23.153 (TrFO)): + +The active Codec Mode is selected from the Active Codec Set (ACS) by the network. This Codec Mode Adaptation, also termed Rate Control, can be performed for the UMTS AMR every 20 ms by going to another Codec Mode within the ACS. For the UMTS AMR 2 this Codec Mode Adaptation can be performed every 20ms for the downlink traffic channel, but only every 40ms for the uplink radio channel. The UE selects at call setup one of the two possible phases for Codec Mode Adaptation (odd or even frames). During the call changes of the Codec Mode in uplink direction are only allowed in this selected phase. Rate Control commands received in downlink direction are considered at the next possible phase. + +By this definition the UMTS AMR 2 Codec Type is TFO and TrFO compatible to the FR AMR, HR AMR, OHR AMR and UMTS AMR 2 Codec Types. In any multi-mode configuration the UMTS\_AMR shall be regarded as only compatible to itself, not to any other AMR codec Type, to avoid incompatibilities in TFO-TrFO-TFO interworking scenarios. In single mode configuration, UMTS AMR and UMTS AMR 2 are compatible, when both codec types use the same single rate ACS. + +The Codec Modes in uplink and downlink at one radio leg may be different. In Tandem Free Operation or Transcoder Free Operation both radio legs (A and B) are considered for the optimal selection of the active Codec Mode in each direction (uplink A and then downlink B, respectively vice versa) by a "Distributed Rate Decision" algorithm. The worst of both radio legs determine the highest allowed Codec Mode, respectively the maximally allowed rate. All rate control commands are transmitted inband on the Iu and Nb interfaces and out of band on the radio interface. + +The Active Codec Set is configured at call setup or reconfigured during the call. It consists of one up to maximally eight Codec Modes (MACS) at a given time, selected from the Supported Codec Set. The maximal number of Codec Modes and the Supported Codec Set may be constrained by the network to consider resources and radio conditions. + +The Active Codec Sets in uplink and downlink are typically identical. + +At call setup the Originating Side sends the AMR parameter set (included in the Codec List). The Terminating side then selects a suitable ACS from the given information and sends it back. In case the terminating side does not support TrFO a transcoder is allocated in the path at a suitable position, preferably as close as possible to the terminating side. This transcoder may by inband signalling install a Tandem Free Operation after call setup. Then, at start up of Tandem Free Operation, both Active Codec Sets, the Supported Codec Sets, the MACSs and the OMs are taken into account to determine the optimal common Active Codec Set. In a later phase the Codec Lists of both radio legs may be taken into account to find the optimum configuration. All configuration data and update protocols are transmitted inband on the TFO interface, but out of band within the UMTS network. For information on Tandem Free Operation see 3GPP TS 28.062 and on Transcoder Free Operation see 3GPP TS 23.153. + +The SCR scheme of the Adaptive Multi-Rate Codec Types mark with a specific SID\_FIRST frame the end of a speech burst. SID\_FIRST does not contain Comfort Noise parameters. This SID\_FIRST starts the comfort noise generation with parameters that are calculated at receiver side (!) from the latest received seven speech frames. A DTX hangover period needs to be applied therefore at transmitter side before sending of this SID\_FIRST. + +Absolutely coded SID\_UPDATE frames follow about every eighth frame (160 ms) in speech pauses. SID\_UPDATE + +frames are sent independently of the cell's timing structure and are related only to the source signal. + +An ONSET frame does (typically) not exist in UMTS networks, but may be received in TFO from the distant partner. It marks the beginning of a speech burst. The uplink SCR operation is always activated for UMTS AMR and UMTS AMR2 codec types. The defined Tandem Free Operation and Transcoder Free Operation allows the reception of AMR SCR information for the downlink direction in all cases. + +The SCR scheme of the UMTS AMR2 Codec Type is fully compatible to the SCR scheme of the UMTS AMR in UMTS and the DTX schemes of the FR AMR, HR AMR and OHR AMR Codec Types. + +## 5.5 TDMA Enhanced Full Rate Codec Type (TDMA EFR) + +The Codec IDentification (CoID) code is defined to be: TDMA\_EFR\_CoID := 0x0000.0111. + +The TDMA Enhanced Full Rate Codec Type has no additional parameters. + +For information (for exact details see TDMA Recommendations): + +The TDMA Enhanced Full Rate Codec Type supports one fixed Codec Mode with 7.4 kBit/s. This codec mode is bit exact identical with AMR codec mode at 7.4 kBit/s. + +In a TDMA system DTX may be enabled in uplink, but not in downlink. The DTX scheme uses one SID frame to mark the end of a speech burst and to start or continue Comfort Noise Generation. + +The defined Tandem Free Operation allows the reception of TDMA EFR DTX information for the downlink direction in all cases. In TDMA systems the transcoder has to generate comfort noise in speech like frames to be sent downlink. In UMTS the downlink DTX shall always be supported and the transcoder can therefore stay transparently in TFO. + +## 5.6 PDC Enhanced Full Rate Codec Type (PDC\_EFR) + +The Codec IDentification (CoID) code is defined to be: TDMA\_EFR\_CoID := 0x0000.1000. + +The PDC Enhanced Full Rate Codec Type has no additional parameters. + +For information (for exact details see PDC Recommendations): + +The PDC Enhanced Full Rate Codec Type supports one fixed Codec Mode with 6.7 kBit/s. This codec mode is bit exact identical with AMR codec mode at 6.7 kBit/s. + +In a PDC system DTX may be enabled in uplink, but not in downlink. The DTX scheme uses one SID frame to mark the end of a speech burst and to start or continue Comfort Noise Generation. + +The Tandem Free Operation allows the reception of PDC EFR DTX information for the downlink direction in all cases. In PDC systems the transcoder has to generate comfort noise in speech like frames to be sent downlink. In UMTS the downlink DTX shall always be supported and the transcoder can therefore stay transparently in TFO. + +## 5.7 Four Adaptive Multi-Rate Wideband Codec Types (FR AMR-WB, UMTS AMR-WB, OFR AMR-WB, OHR AMR-WB) + +The Adaptive Multi-Rate - WideBand Codec algorithm is applied in GERAN-GMSK, GERAN-8PSK and UTRAN in four different Codec Types. + +The Codec IDentification (CoID) codes are defined to be: + +FR\_AMR-WB\_CoID := 0x0000.1001. + +UMTS\_AMR-WB\_CoID := 0x0000.1010. + +OFR\_AMR-WB\_CoID := 0x0000.1100. + +OHR\_AMR-WB\_CoID := 0x0000.1101. + +The AMR-WB Codec Types can be used in conversational speech telephony services in a number of different configurations. The set of allowed configurations is defined in Table 5.7-1. + +**Table 5.7-1: Allowed Configurations for the Adaptive Multi-Rate – Wideband Codec Types** + +| Configuration →
(Config-WB-Code) | 0 | 1 | 2 | 3 | 4 | 5 | 6 | 7 | 8 | 9 | 10 | 11 | 12 | 13 | 14 | 15 | +|-------------------------------------|---|---|---|---|---|---|---|---|---|---|----|----|----|----|----|----| +| ↓ Codec Mode | | | | | | | | | | | | | | | | | +| 23,85 | | | | | 1 | 1 | | | | | | | | | | | +| 15,85 | | | 1 | 1 | | | | | | | | | | | | | +| 12,65 | 1 | 1 | 1 | 1 | 1 | 1 | | | | | | | | | | | +| 8,85 | 1 | 1 | 1 | 1 | 1 | 1 | | | | | | | | | | | +| 6,60 | 1 | 1 | 1 | 1 | 1 | 1 | | | | | | | | | | | +| OM | F | A | F | A | F | A | | | | | | | | | | | +| FR_AMR-WB,
OHR_AMR-WB | Y | | | | | | | | | | | | | | | | +| OFR_AMR-WB,
UMTS_AMR-WB | Y | Y | Y | Y | Y | Y | | | | | | | | | | | + +The "1" in the table indicates that the Codec Mode is included in the Active Codec Set of the Configuration. + +The parameters "OM" (Optimisation Mode) define whether the indicated Configuration can be changed to any of the other Allowed ones (OM == A) or if the change is Forbidden (OM == F). + +The "Y" in the table indicates, which Configuration is defined for which Codec Type. + +Please note that Configurations 0 to 5 are immediately fully compatible with respect to TFO/TrFO due to the specification of Maximum Rate Control. + +Table 5.7-2 defines the Coding of the "Single Codec" information element for the AMR-WB Codec Types. + +**Table 5.7-2: Coding of "Single Codec" for the Adaptive Multi-Rate - WideBand Codec Types** + +| Octet | Parameter | MSB 8 | 7 | 6 | 5 | 4 | 3 | 2 | 1 LSB | +|-------|-------------------|-----------------------------------------------------------------------------|---------|---------|---------|----------------|---|---|-------| +| 1 m | Single Codec | Single Codec (see ITU-T Q.765.5) | | | | | | | | +| 2 m | Length Indication | 4 | | | | | | | | +| 3 m | Compat. Info | Compatibility Information | | | | | | | | +| 4 m | OID | ETSI OID (See ITU-T Q.765.5 [6]) | | | | | | | | +| 5 m | CoID | FR_AMR-WB_CoID or UMTS_AMR-WB_CoID or
OHR_AMR-WB_CoID or OFR_AMR-WB_CoID | | | | | | | | +| 6 m | Config-WB | (spare) | (spare) | (spare) | (spare) | Config-WB-Code | | | | + +with "m" = mandatory + +An AMR-WB speech telephony service is only possible when the whole path allows a digitally transparent transport of the AMR-WB speech parameters end to end. + +Normative for GERAN terminals for FR\_AMR-WB, OHR\_AMR-WB and OFR\_AMR-WB. + +If a GERAN terminal offers one of these Codec Types in the capability list, then all AMR-WB Configurations that are defined for the offered Codec Type shall be supported by this terminal. + +Normative for GERAN infrastructure for FR\_AMR-WB, OHR\_AMR-WB and OFR\_AMR-WB. + +If a GERAN infrastructure supports one of these Codec Types, then at least AMR-WB Configuration 0 shall be supported. The other AMR-WB Configurations are not normative, but optional for OFR\_AMR-WB. + +For information on GERAN A/Gb mode procedures for FR\_AMR-WB, OHR\_AMR-WB and OFR\_AMR-WB (for exact details see GSM Recommendations): + +The active Codec Mode is selected from the Active Codec Set (ACS) by the network (Codec Mode Command) with assistance by the mobile station (Codec Mode Request). This Codec Mode Adaptation, also termed Rate Control, can be performed every 40 ms by going one Codec Mode up or down within the ACS. The Codec Modes in uplink and downlink at one radio leg may be different. In Tandem Free Operation both radio legs (A and B) are considered for the optimal selection of the active Codec Mode in each direction (uplink A and then downlink B, respectively vice versa) by the "Distributed Rate Decision" algorithm. The worst of both radio legs determines the highest allowed Codec Mode, respectively the maximally allowed rate ("Maximum Rate Control"). All rate control commands are transmitted inband: on the radio interface, the BTS-TRAU interface and the TRAU-TRAU interface. + +The Active Codec Set is configured at call setup or reconfigured during the call. It consists of three or four Codec Modes at a given time, selected from the set of allowed Configurations. The selection of the Configuration may be constrained by the network to consider resources and radio conditions. The configurations (Active Codec Sets) in uplink and downlink are identical. + +First, at start up of Tandem Free Operation both Active Codec Sets are taken into account to determine the common Active Codec Set. The set of allowed AMR-WB configurations guarantees that WB-TFO is always possible. In a later phase the Codec Lists of both radio legs may be taken into account to find the optimum configuration. For exact details see 3GPP TS 28.062. All configuration data and update protocols are transmitted inband. + +The DTX scheme of the Adaptive Multi-Rate Wideband Codec Type marks with a specific SID\_FIRST frame the end of a speech burst. SID\_FIRST does not contain Comfort Noise parameters. This SID\_FIRST starts the comfort noise generation with parameters that are calculated at receiver side from the latest received seven speech frames. A DTX hangover period needs to be applied therefore at transmitter side before sending of this SID\_FIRST. + +Absolutely coded SID\_UPDATE frames follow about every eighth frame (160 ms) in speech pauses. SID\_UPDATE frames are sent independently of the cell's TDMA frame structure and are related only to the source signal. + +An ONSET frame (typically) precedes in uplink direction the beginning of a new speech burst. DTX on or off is defined by the network on a cell basis. The defined Tandem Free Operation allows the reception of FR\_AMR-WB DTX information for the downlink direction in all cases. + +Normative for UTRAN terminals for UMTS\_AMR-WB. + +If an UTRAN terminal offers Codec Type UMTS\_AMR-WB in the capability list, then all allowed AMR-WB Configurations shall be supported by this terminal. + +Normative for UTRAN infrastructures for UMTS\_AMR-WB. + +If an UTRAN infrastructure supports Codec Type UMTS\_AMR-WB, then at least AMR-WB Configuration 0 shall be supported. The other AMR-WB Configurations are not normative, but optional. + +For information on UMTS procedures for UMTS\_AMR-WB (for exact details see 3GPP TS 28.062 (TFO) and 3GPP TS 23.153 (TrFO)): + +The active Codec Mode is selected from the Active Codec Set (ACS) by the network. This Codec Mode Adaptation, also termed Rate Control, can be performed for the UMTS\_AMR-WB every 20 ms for the downlink traffic channel, but only every 40ms for the uplink traffic channel by going to another Codec Mode within the ACS. The UE selects at call setup one of the two possible phases for Codec Mode Adaptation (odd or even frames). During the call changes of the Codec Mode in uplink direction are only allowed in this selected phase. Rate Control commands received in downlink direction are considered at the next possible phase. By this definition the UMTS\_AMR-WB Codec Type is TFO and TrFO compatible to the FR\_AMR-WB, the OHR\_AMR-WB and OFR\_AMR-WB and the UMTS\_AMR-WB Codec Types. + +The Codec Modes in uplink and downlink at one radio leg may be different. In Tandem Free Operation or Transcoder Free Operation both radio legs (A and B) are considered for the optimal selection of the active Codec Mode in each direction (uplink A and then downlink B, respectively vice versa) by a "Distributed Rate Decision" algorithm. The worst of both radio legs determine the highest allowed Codec Mode, respectively the maximally allowed rate. All rate control commands are transmitted inband on the Iu and Nb interfaces and out of band on the radio interface. + +The Active Codec Set is selected at call setup or reselected during the call. It consists of three or four Codec Modes at a given time, selected from the allowed configurations. The selection of the configuration may be constrained by the network to consider resources and radio conditions. + +The Active Codec Sets in uplink and downlink are typically identical. + +At call setup with TrFO negotiation the Originating Side sends its preferred AMR-WB configuration and indicates whether it allows a change of this preferred configuration or not (included in the Codec List). The Terminating side then selects a suitable configuration from the given information and sends it back. In case the terminating side does not support TrFO a transcoder is allocated in the path at a suitable position, preferably as close as possible to the terminating side. This transcoder may by inband signalling install a Tandem Free Operation after call setup. The set of allowed AMR-WB configurations guarantees that WB-TFO is always possible. In a later phase the Codec Lists of both radio legs may be taken into account to find the optimum configuration. All configuration data and update protocols are transmitted inband on the TFO interface, but out of band within the UMTS network. For information on Tandem Free Operation see 3GPP TS 28.062 and on Transcoder Free Operation see 3GPP TS 23.153. + +The SCR scheme of the Adaptive Multi-Rate WideBand Codec Types mark with a specific SID\_FIRST frame the end of a speech burst. SID\_FIRST does not contain Comfort Noise parameters. This SID\_FIRST starts the comfort noise generation with parameters that are calculated at receiver side from the latest received seven speech frames. A DTX hangover period needs to be applied therefore at transmitter side before sending of this SID\_FIRST. + +Absolutely coded SID\_UPDATE frames follow about every eighth frame (160 ms) in speech pauses. SID\_UPDATE frames are sent independently of the cell's timing structure and are related only to the source signal. + +An ONSET frame does (typically) not exist in UMTS networks, but may be received in TFO from the distant partner. It marks the beginning of a speech burst. "SCR on" is always defined by the network. The defined Tandem Free Operation and Transcoder Free Operation allows the reception of AMR-WB SCR information for the downlink direction in all cases. + +The SCR scheme of the UMTS AMR-WB Codec Type is fully compatible to the DTX schemes of FR AMR-WB, OHR AMR-WB and OFR AMR-WB. + +The exact details of these Codec Types and their related procedures (DTX, Rate Control, etc) are described in the respective standard documentation. + +## 5.8 MuMe Dummy Codec (3G.324M) + +The Codec Identification (CoID) code is defined to be: MuMe\_CoID:= 0x1111.1111. + +The MuMe codec has one additional mandatory parameter: + +**B/W Multiplier, BWM:** eight bits. + +This defines the required bandwidth for the bearer; the value is a factor of 64K b/s when not equal to 0. When equal to zero then a 32k b/s. + +The "Single Codec" information element consists of 6 octets in case of the MuMe Dummy Codec (table 5.8): + +**Table 5.8: Coding of "Single Codec" for the MuMe Dummy Codec Type** + +| Octet | Parameter | MSB 8 | 7 | 6 | 5 | 4 | 3 | 2 | 1 LSB | +|-------|-------------------|----------------------------------|---|---|---|---|---|---|-------| +| 1 m | Single Codec | Single Codec (see ITU-T Q.765.5) | | | | | | | | +| 2 m | Length Indication | 4 | | | | | | | | +| 3 m | Compat. Info | Compatibility Information | | | | | | | | +| 4 m | OID | ETSI OID (See ITU-T Q.765.5 [6]) | | | | | | | | +| 5 m | CoID | MuMe_CoID | | | | | | | | +| 6 m | BWM | BandWidth Multiplier – see note1 | | | | | | | | + +with "m" = mandatory + +### Note 1: + +BWM == 0 => 32Kb/s + +BWM == 1-255 => factor n (multiplier of 64Kb/s) + +The procedures for use of this codec are defined in TS 23.172 [13]. + +This MuMe Dummy codec type is only for use in Core Network OoBTC procedures it shall NOT be used across the radio interface. + +The MuMe Dummy codec indicates that an Unrestricted multimedia path (UDI) is required, subsequent codec negotiation may occur within this path using MuMe protocols, e.g. H.324M. There are no encoding properties or codec specifications associated to this codec type; it is purely an indication for a MuMe pipe. + +## 5.9 MuMe2 Dummy Codec (3G.324M2) + +The Codec Identification (CoID) code is defined to be: MuMe2\_CoID:= 0x1111.1110. Otherwise, the Coding is identical to the MuME Dummy Codec described in Clause 5.8. + +The Procedural description provided for MuME Dummy Codec in Clause 5.8 is also applicable for the MuMe2 Dummy Codec. The MuMe2 Dummy Codec is used in core network procedures to indicate that a service change to multimedia was indicated by the network. The procedures for use of this codec are defined in TS 23.172 [13]. + +## 5.10 Codec Extension + +The Codec Identification (CoID) code is defined to be: Codec\_Extension\_CoID:= 0x0000.1111 in the "long form" and 0x1111 = 0xFh in the "short form". + +In TFO, see 3GPP TS 28.062 [7] and in AoIP, see 3GPP TS 48.008 [23] the Codec Lists use in general the short form (4 bits) for the Codec Identifier. In order to allow future extensions of this Codec Lists beyond 16 Codec Types the "Codec\_Extension" is defined. These Codec Lists may contain a certain CoID in the range [0x0h, 0xEh] or they may contain the so called "Codec\_Extension" (0xFh), in which case the real Codec Type follows in the next octet in its long form (8 bits). + +## 5.11 CSData Dummy Codec (AoIP) + +The Codec Identification (CoID) code is defined to be: CSData\_CoID:= 0x1111.1101. + +The CSData Dummy Codec has one mandatory parameter of one octet length, for details see TS 48.008 [23]. + +# --- 6 Codec List for the Call Control Protocol + +For call control on the air interface the Codec Lists need to be specified for each radio access technology separately, because it can not be expected that an UE supports the same Codec Types in different radio access technologies. + +*3GPP TS 24.008 [9] defines the call control signalling and how to use the "Supported Codec List Information Element" (IE).* It contains Codec Lists (in form of Codec Bitmaps) for each supported radio access technology (identified by a SysID). + +The coding of this IE is given here. It is also used for TFO in 3GPP TS 28.062 [7]. + +## 6.1 System Identifiers for GSM and UMTS + +The system identifiers for the radio access technologies supported by this specification are: + +SysID for GSM: 0x0000.0000 (bit 8 .. bit 1) + +SysID for UMTS: 0x0000.0100 (bit 8 .. bit 1) + +These values are selected in accordance with [7] (3GPP TS 28.062). + +## 6.2 Codec Bitmap + +The Codec Types are coded in the first and second octet of the Codec List Bitmap as follows: + +| 8 | 7 | 6 | 5 | 4 | 3 | 2 | bit 1 | | +|----------|------------|----------|--------|--------|---------|--------|--------|---------| +| TDMA EFR | UMTS AMR 2 | UMTS AMR | HR AMR | FR AMR | GSM EFR | GSM HR | GSM FR | Octet 1 | + +| bit 16 | 15 | 14 | 13 | 12 | 11 | 10 | bit 9 | | +|------------|------------|------------|------------|---------|-------------|-----------|---------|---------| +| (reserved) | (reserved) | OHR AMR-WB | OFR AMR-WB | OHR AMR | UMTS AMR-WB | FR AMR-WB | PDC EFR | Octet 2 | + +A Codec Type is supported, if the corresponding bit is set to "1". All reserved bits shall be set to "0". + +## 6.3 Selected Codec Type + +The Selected Codec Type in a BICC-based OoBTC negotiation is coded as shown in Table 6.3-1. The same coding is used also in 3GPP TS 28.062 [7]. + +**Table 6.3-1: Coding of the selected Codec\_Type (long form)** + +| Bit 8...Bit 1
CoID | Codec_Type | Name | +|-----------------------|-----------------------------------------------------------------------------------------------|---------------------------------------| +| 0000.0000 | GSM Full Rate (13.0 kBit/s) | GSM FR | +| 0000.0001 | GSM Half Rate (5.6 kBit/s) | GSM HR | +| 0000.0010 | GSM Enhanced Full Rate (12.2 kBit/s) | GSM EFR | +| 0000.0011 | Full Rate Adaptive Multi-Rate | FR AMR | +| 0000.0100 | Half Rate Adaptive Multi-Rate | HR AMR | +| 0000.0101 | UMTS Adaptive Multi-Rate | UMTS AMR | +| 0000.0110 | UMTS Adaptive Multi-Rate 2 | UMTS AMR 2 | +| 0000.0111 | TDMA Enhanced Full Rate (7.4 kBit/s) | TDMA EFR | +| 0000.1000 | PDC Enhanced Full Rate (6.7 kBit/s) | PDC EFR | +| 0000.1001 | Full Rate Adaptive Multi-Rate WideBand | FR AMR-WB | +| 0000.1010 | UMTS Adaptive Multi-Rate WideBand | UMTS AMR-WB | +| 0000.1011 | 8PSK Half Rate Adaptive Multi-Rate | OHR AMR | +| 0000.1100 | 8PSK Full Rate Adaptive Multi-Rate WideBand | OFR AMR-WB | +| 0000.1101 | 8PSK Half Rate Adaptive Multi-Rate WideBand | OHR AMR-WB | +| 0000.1110 | spare, for future use | | +| 0000.1111 | Reserved for Codec_Extension | for AoIP and
TFO,
not for OoBTC | +| Up to
1111.1100 | spare for future use | | +| 1111.1101 | Reserved for CSData dummy Codec Type | for AoIP,
not for OoBTC | +| 1111.1110 | Reserved for MuMe2 dummy Codec Type

NOTE: codec not to be used across radio interface. | MuMe2 | +| 1111.1111 | Reserved for MuMe dummy Codec Type

NOTE: codec not to be used across radio interface. | MuMe | + +# 7 3GPP Codecs for OoBTC in a SIP-I -based Circuit Switched Core Network + +## 7.1 Overview + +In a SIP-I -based Circuit Switched Core Network, as specified in 3GPP TS 23.231 [14], SDP (IETF RFC 4566 [19]) and SDP offer-answer procedures (IETF RFC 3264 [16]) are applied for Out of Band Transcoder Control as specified in 3GPP TS 23.153 [8]. + +Table 7.1.1 lists the supported 3GPP Speech Codecs for a SIP-I -based Circuit Switched Core Network. + +**Table 7.1.1 Supported 3GPP Codecs in a SIP-I -based Circuit Switched Core Network** + +| Payload Type Name | References | Remarks | Support | +|-----------------------|--------------------|----------------------------------------------------------------|-------------------------------------------------------------------------------------------------------------| +| audio/AMR | IETF RFC 4867 [21] | Applicable for FR_AMR, HR_AMR, OHR_AMR, UMTS_AMR and UMTS_AMR2 | Mandatory. Not all AMR configurations are mandatory. Some configurations are preferred, see below. | +| audio/AMR-WB | IETF RFC 4867 [21] | Applicable for FR_AMR-WB, OHR_AMR-WB, OFR_AMR-WB, UMTS_AMR-WB | Optional. AMR-WB is Mandatory if WB speech is supported. Not all WB configurations are mandatory, see below | +| audio/GSM-EFR | IETF RFC 3551 [17] | Useful if an A-interface over IP is attached or TFO is used. | Optional | +| audio/GSM-FR | IETF RFC 3551 [17] | Useful if an A-interface over IP is attached or TFO is used. | Optional | +| audio/GSM-HR | IETF RFC 5993 [22] | Useful if an A-interface over IP is attached, or TFO is used | Optional | +| audio/PCMA | IETF RFC 3551 [17] | ITU-TG.711, Alaw | Mandatory | +| audio/PCMU | IETF RFC 3551 [17] | ITU-T G.711, ulaw | Mandatory | +| audio/telephone-event | IETF RFC 4733 [20] | Used to transport DTMF | Mandatory | + +## 7.2 AMR + +AMR (FR\_AMR, HR\_AMR, OHR\_AMR, UMTS\_AMR and UMTS\_AMR2) shall be encoded in SDP according to the MIME registration in IETF RFC 4867 [21]. The SDP offer-answer related rules in this RFC apply. + +The bandwidth efficient mode of RFC 4867 shall be used. To offer the bandwidth-efficient mode, the octet-align parameter should be omitted in SDP. + +The AMR Codec Types can be used in conversational speech telephony services in a number of different configurations. Configuration related procedures in Clause 5.4 shall be applied also within a SIP-I based CS CN. The set of preferred configurations is defined in TS 28.062 [7], Table 7.11.3.1.3-2. The configuration is encoded in SDP in the mode-set parameter. + +One of these preferred configurations, **Config-NB-Code 1**, is recommended for TFO-TrFO harmonisation between GSM and UMTS networks. This configuration shall be supported in a SIP-I based circuit switched core network to ensure interoperability with an AoIP-based BSS. + +However, it is recommended that nodes in the core network (MSC-S and MGW) support all AMR modes for maximum interoperability. + +To offer the AMR codec in different configurations, the AMR codec may be included several times with different configurations in an SDP m-line. + +A core network node performing a transcoding free interworking towards an A-Interface (TFO towards any A-Interface or TrFO towards an IP-based A-Interface) shall provide the parameters "mode-change-period=2" and "mode-change-neighbour=1" in offer or answer. The parameter "mode-change-capability=2" shall be included by all other CS CN + +nodes in the offer to ensure interoperability unless they received an offer from other nodes without this parameter and do not transcode. + +## 7.3 AMR-WB + +AMR-WB (FR\_AMR-WB, OHR\_AMR-WB, OFR\_AMR-WB, UMTS\_AMR-WB) shall be encoded in SDP according to the MIME registration in IETF RFC 4867 [21]. The SDP offer-answer related rules in this RFC apply. + +The bandwidth efficient mode of RFC 4867 shall be used. To offer the bandwidth-efficient mode, the octet-align parameter should be omitted in SDP. + +The AMR-WB Codec Types can be used in conversational speech telephony services in a number of different configurations. Configuration related procedures in Clause 5.7 shall be applied also within a SIP-I based CS CN. The set of configurations is defined in Table 5.7-1. The configuration is encoded in SDP in the mode-set parameter. + +One of these configurations, **Config-WB-Code 0**, shall be supported by all nodes supporting the AMR-WB codec in a circuit switched core network to ensure interoperability. + +However, it is recommended that a node in the core network supports all AMR-WB modes for maximum interoperability. + +To offer the AMR-WB codec in different configurations, the AMR-WB codec may be included several times with different configurations in an SDP m-line. + +A core network node performing a transcoding free interworking towards an A-Interface (TFO towards any A-Interface or TrFO towards an IP-based A-Interface) shall provide the parameters "mode-change-period=2" and "mode-change-neighbour=1" in offer or answer. The parameter "mode-change-capability=2" shall be included by all other CS CN nodes in the offer to ensure interoperability unless they received an offer from other nodes without this parameter and do not transcode + +## 7.4 GSM\_EFR + +GSM\_EFR shall be encoded in SDP using either the fixed payload type assigned in IETF RFC 3551 [17] or a dynamic payload type described according to the MIME registration in IETF RFC 3551 [17] + +The GSM\_EFR standard comprises a DTX scheme with VAD, SID frames and Comfort Noise generation that is automatically included in this SDP negotiation. For User Plane details see 3GPP TS 26.102 [24]. No other DTX scheme shall be negotiated in SDP for GSM\_EFR.]. + +## 7.5 GSM\_FR + +GSM\_FR shall be encoded in SDP using either the fixed payload type assigned in IETF RFC 3551 [17] or a dynamic payload type described according to the MIME registration in IETF RFC 3551 [17] + +The GSM\_FR standard comprises a DTX scheme with VAD, SID frames and Comfort Noise generation that is automatically included in this SDP negotiation. For User Plane details see 3GPP TS 26.102 [24]. No other DTX scheme shall be negotiated in SDP for GSM\_FR.]. + +## 7.6 GSM\_HR + +GSM\_HR shall be encoded in SDP according to the MIME registration in [22]. GSM\_HR shall be encoded in SDP using a dynamic payload type described according to the MIME registration in [22]. The options specified in [22] are not applied inside the Circuit Switched Core Network and not across the A-Interface, but set to pre-defined values as follows: a single frame (Speech or SID) shall be included in one RTP packet, FEC and Interleaving (redundancy) shall not be used, Encryption shall not be used, a packetization time of 20ms shall be applied. + +The GSM\_HR standard comprises a DTX scheme with VAD, SID frames and Comfort Noise generation that is automatically included in this SDP negotiation. For User Plane details see [22] and 3GPP TS 26.102 [24]. No other DTX scheme shall be negotiated in SDP for GSM\_HR. + +## 7.7 PCM + +PCMU and PCMA shall be encoded in SDP using either the fixed payload type assigned in IETF RFC 3551 [ee] or a dynamic payload type described according to the MIME registration in IETF RFC 3551 [ee]. + +## 7.8 Telephone-Event + +Telephony-Event shall be encoded in SDP according to the MIME registration in IETF RFC 4733 [20]. + +The MIME type audio/telephone-event in IETF RFC 4733 [20] with default events and default rate shall be used to encode DTMF. Therefore, the rate and event parameters do not need to be supplied. + +# Annex A (informative): Example Supported Codec List for UMTS + +This Annex gives some informative examples how the Codec List for UMTS may look like for the OoBTC protocol in a BICC-based Circuit Switched Core Network. + +In this example the UMTS Circuit Switched Core Network does support: UMTS AMR2(set1), (GSM) FR AMR(set1) and (GSM) HR AMR(set1) and GSM EFR. It supports PCM, i.e. ITU-T G.711, here in the Alaw version, with transcoding. It may support also UMTS\_AMR(set7), GSM FR, and GSM\_HR (not included in the list). + +One "Supported Codec List" (with arbitrarily selected Codec Type preference) could look at Originating side like: + +| Octet | Parameter | MSB 8 | 7 | 6 | 5 | 4 | 3 | 2 | 1 LSB | | | | +|-------|------------------------|--------------------------------------|---------|---------|---------|---------|---------|---------|---------|--|--|--| +| 1 | Codec List | Codec List (see ITU-T Q.765.5) | | | | | | | | | | | +| 2 | Length Indication (LI) | 30 | | | | | | | | | | | +| 3 | Compat. Info | Compatibility Information | | | | | | | | | | | +| 4 | Single Codec | Single Codec (see ITU-T Q.765.5) | | | | | | | | | | | +| 5 | LI | 6 | | | | | | | | | | | +| 6 | Compat. Info | Compatibility Information | | | | | | | | | | | +| 7 | OID | ETSI OID (See ITU-T Q.765.5 [6]) | | | | | | | | | | | +| 8 | CoID | UMTS_AMR2 CoID | | | | | | | | | | | +| 9 | ACS (set1) | 12.2(1) | 10.2(0) | 7.95(0) | 7.40(1) | 6.70(0) | 5.90(1) | 5.15(0) | 4.75(1) | | | | +| 10 | SCS (set1) | 12.2(1) | 10.2(0) | 7.95(0) | 7.40(1) | 6.70(0) | 5.90(1) | 5.15(0) | 4.75(1) | | | | +| 11 | MACS | (spare) | (spare) | (spare) | (spare) | OM(0) | MACS(4) | | | | | | +| 12 | Single Codec | Single Codec (see ITU-T Q.765.5) | | | | | | | | | | | +| 13 | LI | 3 | | | | | | | | | | | +| 14 | Compat. Info | Compatibility Information | | | | | | | | | | | +| 15 | OID | ITU-T OID (See ITU-T Q.765.5 [6]) | | | | | | | | | | | +| 16 | CoID | Codec Identifier for PCM Alaw 64kbps | | | | | | | | | | | +| 17 | Single Codec | Single Codec (see ITU-T Q.765.5) | | | | | | | | | | | +| 18 | LI | 6 | | | | | | | | | | | +| 19 | Compat. Info | Compatibility Information | | | | | | | | | | | +| 20 | OID | ETSI OID (See ITU-T Q.765.5 [6]) | | | | | | | | | | | +| 21 | CoID | FR_AMR CoID | | | | | | | | | | | +| 22 | ACS (set1) | 12.2(1) | 10.2(0) | 7.95(0) | 7.40(1) | 6.70(0) | 5.90(1) | 5.15(0) | 4.75(1) | | | | +| 23 | SCS (set1) | 12.2(1) | 10.2(0) | 7.95(0) | 7.40(1) | 6.70(0) | 5.90(1) | 5.15(0) | 4.75(1) | | | | +| 24 | MACS | (spare) | (spare) | (spare) | (spare) | OM(0) | MACS(4) | | | | | | +| 25 | Single Codec | Single Codec (see ITU-T Q.765.5) | | | | | | | | | | | +| 26 | LI | 6 | | | | | | | | | | | +| 27 | Compat. Info | Compatibility Information | | | | | | | | | | | +| 28 | OID | ETSI OID (See ITU-T Q.765.5 [6]) | | | | | | | | | | | +| 29 | CoID | HR_AMR CoID | | | | | | | | | | | +| 30 | ACS (set1) | (spare) | (spare) | 7.95(0) | 7.40(1) | 6.70(0) | 5.90(1) | 5.15(0) | 4.75(1) | | | | +| 31 | SCS (set1) | (spare) | (spare) | 7.95(0) | 7.40(1) | 6.70(0) | 5.90(1) | 5.15(0) | 4.75(1) | | | | +| 32 | MACS | (spare) | (spare) | (spare) | (spare) | OM(0) | MACS(3) | | | | | | +| 33 | Single Codec | Single Codec (see ITU-T Q.765.5) | | | | | | | | | | | +| 34 | LI | 3 | | | | | | | | | | | +| 35 | Compat. Info | Compatibility Information | | | | | | | | | | | +| 36 | OID | ETSI OID (See ITU-T Q.765.5 [6]) | | | | | | | | | | | +| 37 | CoID | EFR_CoID | | | | | | | | | | | + +The Terminating Side selects one of the Codec Types and returns it, together with the selected codec attributes. + +The AMR Codec Types may have very similar, if not identical codec attributes at Originating side. The UMTS as Originating side can, however, already decide, which configuration would be preferred in case the Terminating side is + +UMTS or GSM. A GSM Circuit Switched Core Network as Originating side can not offer UMTS AMR (unless it provides local transcoding) and the Codec attributes for FR AMR and HR AMR may be quite different. + +# Annex B (informative): Change history + +| Change history | | | | | | | | +|----------------|---------|-----------|------|-----|------------------------------------------------------------------------|--------|--------| +| Date | TSG SA# | TSG Doc. | CR | Rev | Subject/Comment | Old | New | +| 12-2000 | 10 | SP-000576 | 004 | | Introduction of Codec Type Bit-Map for Codec Negotiation | 3.0.0 | 4.0.0 | +| 12-2000 | 10 | SP-000576 | 005 | | Introduction of Selected Codec Type for Codec Negotiation | 3.0.0 | 4.0.0 | +| 12-2000 | 10 | SP-000576 | 006 | | Clarification for the use of the Codec List Information Element | 3.0.0 | 4.0.0 | +| 03-2001 | 11 | SP-010104 | 007 | | Simplification of the Optimisation Mode Field | 4.0.0 | 4.1.0 | +| 03-2001 | 11 | SP-010199 | 008 | 3 | Introduction of UMTS_AMR_2 | 4.0.0 | 4.1.0 | +| 03-2001 | 11 | SP-010199 | 009 | | Introduction of AMR Wideband | 4.1.0 | 5.0.0 | +| 03-2002 | 15 | SP-020078 | 015 | | Introduction of GERAN-8PSK Codec Types into Codec List | 5.0.0 | 5.1.0 | +| 03-2002 | 15 | SP-020078 | 017 | | Introduction of codepoint for Dummy Codec for CS Multi Media (3G 324M) | 5.0.0 | 5.1.0 | +| 06-2002 | 16 | SP-020223 | 014 | 2 | UMTS_AMR2 is default Codec Type in all terminals of Rel-4 and onwards | 5.1.0 | 5.2.0 | +| 09-2002 | 17 | SP-020437 | 020 | 1 | TrFO-Signalling for allowed AMR-WB Configurations | 5.2.0 | 5.3.0 | +| 12-2002 | 18 | SP-020690 | 021 | 1 | Correction of uplink SCR activation for UMTS AMR | 5.3.0 | 5.4.0 | +| 12-2002 | 18 | SP-020690 | 022 | | Correction to the Codec ID Table | 5.3.0 | 5.4.0 | +| 09-2004 | 25 | SP-040646 | 028 | 1 | Correction of Size and Reference of MuMe Codec | 5.4.0 | 5.5.0 | +| 09-2004 | 25 | SP-040646 | 023 | 2 | Harmonisation of AMR Configurations | 5.5.0 | 6.0.0 | +| 09-2004 | 25 | SP-040646 | 025 | 1 | Error Fixes | 5.5.0 | 6.0.0 | +| 09-2004 | 25 | SP-040646 | 029 | 1 | Correction of Size and Reference of MuMe Codec | 5.5.0 | 6.0.0 | +| 12-2004 | 26 | SP-040845 | 032 | | TFO/TrFO Compatibility of UMTS_AMR and UMTS_AMR2 | 6.0.0 | 6.1.0 | +| 12-2004 | 26 | SP-040847 | 036 | 1 | Clarifications for AMR | 6.0.0 | 6.1.0 | +| 03-2006 | 31 | SP-060008 | 0037 | | 3G-324.M2 Codec for Indication of Network-Initiated Service Change | 6.1.0 | 6.2.0 | +| 06-2007 | 36 | | | | Version for Release 7 | 6.2.0 | 7.0.0 | +| 09-2008 | 41 | SP-080475 | 0038 | 2 | Addition of CS over IP User Plane | 7.0.0 | 8.0.0 | +| 12-2008 | 42 | SP-080678 | 0039 | 2 | Corrections to CS over IP User Plane | 8.0.0 | 8.1.0 | +| 12-2009 | 46 | | | | Version for Release 9 | 8.1.0 | 9.0.0 | +| 03-2011 | 51 | SP-110034 | 0041 | | Correction of reference for GSM-HR payload format | 9.0.0 | 9.1.0 | +| 03-2011 | 51 | | | | Version for Release 10 | 9.1.0 | 10.0.0 | +| 09-2012 | 57 | | | | Version for Release 11 | 10.0.0 | 11.0.0 | \ No newline at end of file diff --git a/marked/Rel-11/26_series/26104/raw.md b/marked/Rel-11/26_series/26104/raw.md new file mode 100644 index 0000000000000000000000000000000000000000..e95c115e3f198d8d781384eafc2734cd1ff30773 --- /dev/null +++ b/marked/Rel-11/26_series/26104/raw.md @@ -0,0 +1,1031 @@ + + + + + + +# --- Contents + +| | | +|--------------------------------------------------------------------|----| +| Foreword ..... | 4 | +| 1 Scope..... | 5 | +| 2 Normative references..... | 5 | +| 3 Definitions and abbreviations ..... | 6 | +| 3.1 Definitions..... | 6 | +| 3.2 Abbreviations ..... | 6 | +| 4 C code structure ..... | 6 | +| 4.1 Contents of the C source code..... | 6 | +| 4.2 Program execution ..... | 7 | +| 4.3 Coding style ..... | 7 | +| 4.4 Code hierarchy ..... | 7 | +| 4.5 Variables, constants and tables ..... | 10 | +| 4.5.1 Description of constants used in the C code..... | 11 | +| 4.5.2 Description of fixed tables used in the C code..... | 11 | +| 4.5.3 Static variables used in the C code..... | 13 | +| 5 Homing procedure ..... | 16 | +| 6 File formats ..... | 22 | +| 6.1 Speech file (encoder input / decoder output) ..... | 22 | +| 6.2 Mode control file (encoder input) ..... | 22 | +| 6.3 Parameter bitstream file (encoder output / decoder input)..... | 22 | +| Annex A (informative): Change History..... | 23 | + +# --- Foreword + +This Technical Specification (TS) has been produced by the 3rd Generation Partnership Project (3GPP). + +The contents of the present document are subject to continuing work within the TSG and may change following formal TSG approval. Should the TSG modify the contents of the present document, it will be re-released by the TSG with an identifying change of release date and an increase in version number as follows: + +Version x.y.z + +where: + +- x the first digit: + - 1 presented to TSG for information; + - 2 presented to TSG for approval; + - 3 or greater indicates TSG approved document under change control. +- y the second digit is incremented for all changes of substance, i.e. technical enhancements, corrections, updates, etc. +- z the third digit is incremented when editorial only changes have been incorporated in the document. + +# --- 1 Scope + +This Technical Standard (TS) contains an electronic copy of the ANSI-C code for a floating-point implementation of the Adaptive Multi-Rate codec. This floating-point codec specification is mainly targeted to be used in multimedia applications such as the 3G-324M terminal specified in 3GPP TS 26.110, or in packet-based (e.g., H.323) applications. The bit-exact fixed-point ANSI-C code in 3GPP TS 26.073 remains the preferred implementation for all applications, but the floating-point codec may be used instead of the fixed-point codec when the implementation platform is better suited for a floating-point implementation. It has been verified that the fixed-point and floating-point codecs interoperate with each other without any artefacts. + +The floating-point ANSI-C code in this specification is the only standard conforming non-bit-exact implementation of the Adaptive Multi Rate speech transcoder (3GPP TS 26.090 [2]), Voice Activity Detection (3GPP TS 26.094 [6]), comfort noise generation (3GPP TS 26.092 [4]), and source controlled rate operation (3GPP TS 26.093 [5]). The floating-point code also contains example solutions for substituting and muting of lost frames (3GPP TS 26.091 [3]). + +**The fixed-point specification in 26.073 shall remain the only allowed implementation for the 3G mandatory speech service and the use of the floating-point codec is strictly limited to other services.** + +The floating-point encoder in this specification is a non-bit-exact implementation of the fixed-point encoder producing quality indistinguishable from that of the fixed-point encoder. The decoder in this specification is functionally a bit-exact implementation of the fixed-point decoder, but the code has been optimized for speed and the standard fixed-point libraries are not used as such. + +# --- 2 Normative references + +The following documents contain provisions which, through reference in this text, constitute provisions of the present document. + +- References are either specific (identified by date of publication, edition number, version number, etc.) or non-specific. +- For a specific reference, subsequent revisions do not apply. +- For a non-specific reference, the latest version applies. In the case of a reference to a 3GPP document (including a GSM document), a non-specific reference implicitly refers to the latest version of that document *in the same Release as the present document*. + +- [1] 3GPP TS 26.074: "AMR Speech Codec; Test sequences". +- [2] 3GPP TS 26.090: "AMR Speech Codec; Speech transcoding". +- [3] 3GPP TS 26.091: "AMR Speech Codec; Substitution and muting of lost frames". +- [4] 3GPP TS 26.092: "AMR Speech Codec; Comfort noise aspects". +- [5] 3GPP TS 26.093: "AMR Speech Codec; Source controlled rate operation". +- [6] 3GPP TS 26.094: "AMR Speech Codec; Voice Activity Detection". +- [7] 3GPP TS 26.073: "ANSI-C code for the Adaptive Multi Rate speech codec". +- [8] 3GPP TS 26.101: "AMR Speech Codec Frame Structure". +- [9] RFC 3267: "A Real-Time Transport Protocol (RTP) Payload Format and File Storage Format for Adaptive Multi-Rate (AMR) and Adaptive Multi-Rate Wideband (AMR-WB) Audio Codecs", June 2002. + +# --- 3 Definitions and abbreviations + +## 3.1 Definitions + +Definition of terms used in the present document, can be found in 3GPP TS 26.090 [2], 3GPP TS 26.091 [3], 3GPP TS 26.092 [4], 3GPP TS 26.093 [5], and 3GPP TS 26.094 [6]. + +## 3.2 Abbreviations + +For the purpose of the present document, the following abbreviations apply: + +| | | +|------|-----------------------------------------| +| ANSI | American National Standards Institute | +| ETS | European Telecommunication Standard | +| GSM | Global System for Mobile communications | +| I/O | Input/Output | +| RAM | Random Access Memory | +| ROM | Read Only Memory | + +# --- 4 C code structure + +This clause gives an overview of the structure of the floating-point C code and provides an overview of the contents and organization of the C code attached to this document. The basic structure of the floating-point C code follows that of the bit-exact fixed-point code [7]. + +The C code has been verified on the following systems: + +- IBM PC/AT compatible computers with Windows NT40 and Microsoft Visual C++ v.5.0 compiler; +- HP workstations and GNU gcc compiler; +- IBM PC/AT compatible computers with Linux operating system and GNU gcc compiler; + +ANSI-C 9899 was selected as the programming language because portability was desirable + +## 4.1 Contents of the C source code + +The C code distribution has all files in the root level. + +The files with suffix "c" contain the source code and the files with suffix "h" are the header files. The ROM data is contained in "rom" files with suffix "h". + +The C code does not contain any speech coder installation verification data files. Verification for the bit-exact decoder is defined in specification 3GPP TS 26.073 [7]. + +Makefiles are provided for the platforms in which the C code has been verified (listed above). Once the software is installed, this directory will have a compiled version of encoder and decoder and all the object files. + +## 4.2 Program execution + +The Adaptive Multi-Rate codec is implemented in two programs: + +- (*encoder*) speech encoder; +- (*decoder*) speech decoder. + +The programs should be called like: + +``` +encoder [-dtx] mode speech_file bitstream_file +``` + +or + +``` +encoder [-dtx] -modefile=mode_file speech_file bitstream_file +``` + +``` +decoder +``` + +The speech files contain 16-bit linear encoded PCM speech samples and the parameter files contain encoded speech data and some additional flags. + +See the file *readme.txt* for more information on how to run the *encoder* and *decoder* programs. + +## 4.3 Coding style + +The C code has been written according to structuring conventions used in 3GPP TS 26.073 [7]. Encoder and decoder state structures are allocated and initialized with special initializing functions. There are no separate functions for each module, as opposed to the fixed-point implementation in 3GPP TS 26.073 [7]. + +## 4.4 Code hierarchy + +The code hierarchy follows the one specified in 3GPP TS 26.073 [7]. + +Figures 1 to 4 are call graphs that show the functions used in the speech codec, including the functions of VAD, DTX, and comfort noise generation. + +Each column represents a call level and each cell a function. The functions contain calls to the functions in rightwards neighbouring cells. The time order in the call graphs is from the top downwards as the processing of a frame advances. All standard C functions, such as *printf()*, *fwrite()*, etc., have been omitted. + +The encoder call graph is broken down into three separate call graphs, shown in Tables 1 to 3. + +**Table 1: Speech encoder call structure** + +| | | | | | | +|---------------------|-------------|---------------------|-------------------|---------------------------|---------------------------| +| Speech_Encode_Frame | Pre_Process | | | | | +| | cod_amr | vad | filter_bank | first_filter_stage | | +| | | | | filter5 | | +| | | | | filter3 | | +| | | | | level_calculation | | +| | | | vad_decision | complex_estimate_adapt | | +| | | | | complex_vad | | +| | | | | noise_estimate_update | update_cntrl | +| | | | | hangover_addition | | +| | | tx_dtx_handler | | | | +| | | lpc | Autocorr | | | +| | | | Levinson | | | +| | | lsp | Az_lsp | Chebps | | +| | | | Q_plsf_5 | Lsp_lsf | | +| | | | | Lsf_wt | | +| | | | | Vq_subvec | | +| | | | | Vq_subvec_s | | +| | | | | Reorder_lsf | | +| | | | | Lsf_lsp | | +| | | | Int_lpc_1and3_2 | Lsp_az | Get_lsp_pol | +| | | | Int_lpc_1and3 | Lsp_az | Get_lsp_pol | +| | | | Q_plsf_3 | Lsp_lsf | | +| | | | | Lsf_wt | | +| | | | | Vq_subvec3 | | +| | | | | Vq_subvec4 | | +| | | | | Reorder_lsf | | +| | | | | Lsf_lsp | | +| | | | Int_lpc_1to3_2 | Lsp_az | Get_lsp_pol | +| | | | Int_lpc_1to3 | Lsp_az | Get_lsp_pol | +| | | dtx_buffer | Dotproduct40 | | | +| | | dtx_enc | Lsp_lsf | | | +| | | | Reorder_lsf | | | +| | | | Lsf_lsp | | | +| | | | Q_plsf_3 | Lsp_lsf | | +| | | | | Lsf_wt | | +| | | | | Vq_subvec3 | | +| | | | | Vq_subvec4 | | +| | | | | Reorder_lsf | | +| | | | | Lsf_lsp | | +| | | check_lsp | | | | +| | | pre_big | Weight_Ai | | | +| | | | Residu | | | +| | | | Syn_filt | | | +| | | ol_ltp | Pitch_ol | vad_tone_detection_update | | +| | | | | Lag_max | vad_tone_detection | +| | | | | comp_corr | | +| | | | | hp_max | | +| | | | Pitch_ol_wgh | comp_corr | | +| | | | | Lag_max_wght | vad_tone_detection_update | +| | | | | gmed_n | vad_tone_detection | +| | | | | hp_max 2 | | +| | | vad_pitch_detection | | | | +| | | subframePreProc | Weight_Ai | | | +| | | | Syn_filt | | | +| | | | Residu | | | +| | | cl_ltp | Pitch_fr | getRange | | +| | | | | Norm_Corr | Dotproduct40 | +| | | | | searchFrac | Interpol_3or6 | +| | | | | Enc_lag3 | | +| | | | | Enc_lag6 | | +| | | | Pred_lt_3or6 | | | +| | | | G_pitch | Dotproduct40 | | +| | | | check_gp_clipping | | | +| | | | q_gain_pitch | | | +| | | cbsearch | see Table 2 | | | +| | | gainQuant | see Table 3 | | | +| | | update_gp_clipping | Copy | | | +| | | subframePostProc | Syn_filt | | | +| | | Pred_lt_3or6 | | | | +| | | Convolve | | | | + +**Table 2: cbsearch call structure** + +| | | | | +|----------|-------------------|-------------------------|--------------| +| cbsearch | code_2i40_9bits | cor h_x | Dotproduct40 | +| | | set_sign | | +| | | cor h | Dotproduct40 | +| | | search 2i40_9bits | | +| | | build_code 2i40_9bits | | +| | code_2i40_11bits | cor h_x | Dotproduct40 | +| | | set_sign | | +| | | cor h | Dotproduct40 | +| | | search 2i40_11bits | | +| | | build_code 2i40_11bits | | +| | code_3i40_14bits | cor h_x | Dotproduct40 | +| | | set_sign | | +| | | cor h | Dotproduct40 | +| | | search 3i40 | | +| | | build_code 3i40_14bits | | +| | code_4i40_17bits | cor h_x | Dotproduct40 | +| | | set_sign | | +| | | cor h | Dotproduct40 | +| | | search 4i40 | | +| | | build_code 4i40 | | +| | code_8i40_31bits | cor h_x | Dotproduct40 | +| | | set_sign12k2 | Dotproduct40 | +| | | cor h | Dotproduct40 | +| | | search 8i40 | | +| | | build_code 8i40_31bits | | +| | | compress_code | compress10 | +| | code_10i40_35bits | cor h_x | Dotproduct40 | +| | | set_sign12k2 | Dotproduct40 | +| | | cor h | Dotproduct40 | +| | | search 10i40 | | +| | | build_code 10i40_35bits | | +| | | g_p | | + +**Table 3: gainQuant call structure** + +| | | | | +|-----------|-----------------------|---------------------------|--------------| +| gainQuant | gc_pred | Dotproduct40 | | +| | calc_filt_energies | Dotproduct40 | | +| | Dotproduct40 | | | +| | MR475_update_unq_pred | | | +| | MR475_gain_quant | gc_pred | Dotproduct40 | +| | g_gain_code | | | +| | MR795_gain_quant | g_gain_pitch | | +| | | MR795_gain_code_quant3 | | +| | | calc_unfilt_energies | Dotproduct40 | +| | | gain_adapt | Gmed_n_f | +| | | MR795_gain_code_quant_mod | | +| | Qua_gain | | | + +**Table 4: Speech decoder call structure** + +| | | | | | | +|---------------------|-------------|-------------------------|-------------------------|--------------|--| +| Speech_Decode_Frame | Decoder_amr | rx_dtx_handler | | | | +| | | Decoder_amr_reset | | | | +| | | dtx_dec | Copy | | | +| | | | Lsf_lsp | | | +| | | | D_plsf_3 | Lsf_lsp | | +| | | | pseudonoise | | | +| | | | Lsp_lsf | | | +| | | | Reorder_lsf | | | +| | | | Lsp_Az | Get_lsp_pol | | +| | | | A_Refl | | | +| | | | Log2 | Log2_norm | | +| | | | Pow2 | | | +| | | | Build_CN_code | pseudonoise | | +| | | | Syn_filt | | | +| | | Lsf_lsp | | | | +| | | Lsp_avg | | | | +| | | Build_CN_param | | | | +| | | D_plsf_3 | Lsf_lsp | | | +| | | Int_lpc_1to3 | Lsp_Az | Get_lsp_pol | | +| | | D_plsf_5 | Reorder_lsf | | | +| | | | Lsf_lsp | | | +| | | Int_lpc_1and3 | Lsp_Az | Get_lsp_pol | | +| | | Dec_lag3 | | | | +| | | Pred_lt_3or6_40 | | | | +| | | Dec_lag6 | | | | +| | | decode_2140_9bits | | | | +| | | decode_2140_11bits | | | | +| | | decode_3140_14bits | | | | +| | | decode_4140_17bits | | | | +| | | decode_8140_31bits | decompress_codewords | decompress10 | | +| | | ec_gain_pitch | gmed_n | | | +| | | d_gain_pitch | | | | +| | | ec_gain_pitch_update | | | | +| | | decode_10140_35bits | | | | +| | | Dec_gain | Log2 | Log2_norm | | +| | | | gc_pred | Log2 | | +| | | | | Log2_norm | | +| | | | Pow2 | | | +| | | | gc_pred_update | | | +| | | ec_gain_code | gmed_n | | | +| | | | gc_pred_average_limited | | | +| | | | gc_pred_update | | | +| | | ec_gain_code_update | | | | +| | | d_gain_code | gc_pred | Log2 | | +| | | | | Log2_norm | | +| | | | Pow2 | | | +| | | | gc_pred_update | | | +| | Post_Filter | Int_lsf | | | | +| | | Cb_gain_average | | | | +| | | ph_disp | | | | +| | | sqrt_l_exp | | | | +| | | Ex_ctrl | gmed_n | | | +| | | agc2 | Inv_sqrt | | | +| | | Syn_filt | | | | +| | | Bgn_scd | gmed_n | | | +| Post_Process | | dtx_dec_activity_update | Copy | | | +| | | | Log2 | Log2_norm | | +| | | Lsp_avg | | | | +| | | Residu40 | | | | +| | | Syn_filt | | | | +| | | agc | energy_new | energy_old | | +| | | | Inv_sqrt | | | + +## 4.5 Variables, constants and tables + +The data types of variables and tables used in the floating-point implementation are signed integers in 2's complement representation, defined by: + +**Word8** 8 bit variable + +**UWord8** 8 bit unsigned variable + +**Word16** 16 bit variable + +**Word32** 32 bit variable + +Floating-point numbers use the IEEE (Institute of Electrical and Electronics Engineers) format: + +**Float32** 8 bit exponent, 23 bit mantissa, 1 bit sign + +**Float64** 11 bit exponent, 52 bit mantissa, 1 bit sign + +Furthermore some **enum** types are used, all possible to represent with one byte, and a Boolean **Flag**. + +### 4.5.1 Description of constants used in the C code + +Constants for the codec are defined in rom (h) files. + +### 4.5.2 Description of fixed tables used in the C code + +This section contains a listing of all fixed tables sorted by source file name and table name. + +**Table 5: Speech encoder fixed tables** + +| File | Table name | Type[Length] | Description | +|-----------|----------------------|----------------|-----------------------------------------------------------------------| +| rom_enc.h | trackTable | Word8[4*5] | track table for algebraic code book search (MR475, MR515) | +| rom_enc.h | gamma1 | Float32[10] | spectral expansion factors | +| rom_enc.h | gamma1_12k2 | Float32[10] | spectral expansion factors | +| rom_enc.h | gamma2 | Float32[10] | spectral expansion factors | +| rom_enc.h | b60 | Float32[61] | interpolation filter coefficients | +| rom_enc.h | startPos1 | Word16[2] | track start search position for first pulse | +| rom_enc.h | startPos2 | Word16[4] | track start search position for second pulse | +| rom_enc.h | startPos | Word16[16] | track start search position | +| rom_enc.h | corrweight | Float32[251] | weighting of the correlation function in open loop LTP search (MR102) | +| rom_enc.h | qua_gain_pitch | Float32[16] | adaptive codebook gain quantization table (MR795) | +| rom_enc.h | qua_gain_pitch_MR122 | Float32[16] | adaptive codebook gain quantization table (MR122) | +| rom_enc.h | qua_gain_code | Float32[64] | fixed codebook gain quantization table (MR122, MR795) | +| rom_enc.h | gray | Word8[8] | gray coding table | +| rom_enc.h | grid | Float32[61] | grid points at which Chebyshev polynomials are evaluated | +| rom_enc.h | b24 | Float32[25] | interpolation filter coefficients | +| rom_enc.h | lag_wind | Float32[10] | lag window table | +| rom_enc.h | lsp_init_data | Float32[10] | initialization table for lsp history in DTX | +| rom_enc.h | past_rq_init | Float32[80] | initialization table for the MA predictor in DTX | +| rom_enc.h | mean_lsf_3 | Float32[10] | LSF means (not in MR122) | +| rom_enc.h | mean_lsf_5 | Float32[10] | LSF means (MR122) | +| rom_enc.h | pred_fac | Float32[10] | LSF prediction factors (not in MR122) | +| rom_enc.h | dico1_lsf_3 | Float32[3*256] | 1 st LSF quantizer (not in MR122 and MR795) | +| rom_enc.h | dico2_lsf_3 | Float32[3*512] | 2 nd LSF quantizer (not in MR122) | +| rom_enc.h | dico3_lsf_3 | Float32[4*512] | 3 rd LSF quantizer (not in MR122, MR515 and MR475) | +| rom_enc.h | mr515_3_lsf | Float32[4*128] | 3 rd LSF quantizer (MR515 and MR475) | +| rom_enc.h | mr795_1_lsf | Float32[3*512] | 1 st LSF quantizer (MR795) | +| rom_enc.h | dico1_lsf_5 | Float32[4*128] | 1 st LSF quantizer (MR122) | +| rom_enc.h | dico2_lsf_5 | Float32[4*256] | 2 nd LSF quantizer (MR122) | +| rom_enc.h | dico3_lsf_5 | Float32[4*256] | 3 rd LSF quantizer (MR122) | +| rom_enc.h | dico4_lsf_5 | Float32[4*256] | 4 th LSF quantizer (MR122) | +| rom_enc.h | dico5_lsf_5 | Float32[4*64] | 5 th LSF quantizer (MR122) | +| rom_enc.h | table_gain_MR475 | Float32[4*256] | gain quantization table (MR475) | +| rom_enc.h | table_gain_highrates | Float32[128*3] | gain quantization table (MR67, MR74 and MR102) | +| rom_enc.h | table_gain_lowrates | Float32[64*3] | gain quantization table (MR515 and MR59) | +| rom_enc.h | window_200_40 | Float32[240] | LP analysis window (not in MR122) | +| rom_enc.h | window_160_80 | Float32[240] | 1 st LP analysis window (MR122) | +| rom_enc.h | window_232_8 | Float32[240] | 2 nd LP analysis window (MR122) | +| rom_enc.h | corrweight | Float32[251] | correlation weights | +| rom_enc.h | mode_dep_parm | Word8[8*9] | parameters defining the adaptive codebook search per mode | + +**Table 6: Speech decoder fixed tables** + +| File | Table name | Type[Length] | Description | +|-----------|----------------------|---------------|---------------------------------------------------------------| +| rom_dec.h | dtx_log_en_adjust | Word16[9] | level adjustments for each mode | +| rom_dec.h | cdown | Word32[7] | attenuation factors for codebook gain | +| rom_dec.h | pdown | Word32[7] | attenuation factors for adaptive codebook gain | +| rom_dec.h | pred | Word32[4] | algebraic code book gain MA predictor coefficients | +| rom_dec.h | pred_MR122 | Word32[4] | algebraic code book gain MA predictor coefficients (MR122) | +| rom_dec.h | gamma3_MR122 | Word32[10] | spectral expansion factors | +| rom_dec.h | gamma3 | Word32[10] | spectral expansion factors | +| rom_dec.h | gamma4_MR122 | Word32[10] | spectral expansion factors | +| rom_dec.h | gamma4 | Word32[10] | spectral expansion factors | +| rom_dec.h | bitno_MR475 | Word16[17] | number of bits per parameter to transmit (MR475) | +| rom_dec.h | bitno_MR515 | Word16[19] | number of bits per parameter to transmit (MR515) | +| rom_dec.h | bitno_MR59 | Word16[19] | number of bits per parameter to transmit (MR59) | +| rom_dec.h | bitno_MR67 | Word16[19] | number of bits per parameter to transmit (MR67) | +| rom_dec.h | bitno_MR74 | Word16[19] | number of bits per parameter to transmit (MR74) | +| rom_dec.h | bitno_MR795 | Word16[23] | number of bits per parameter to transmit (MR795) | +| rom_dec.h | bitno_MR102 | Word16[39] | number of bits per parameter to transmit (MR102) | +| rom_dec.h | bitno_MR122 | Word16[57] | number of bits per parameter to transmit (MR122) | +| rom_dec.h | bitno_MRDTX | Word16[5] | number of bits per parameter to transmit (MRDTX) | +| rom_dec.h | qua_gain_pitch | Word32[16] | adaptive codebook gain quantization table (MR122, MR795) | +| rom_dec.h | qua_gain_code | Word32[96] | fixed codebook gain quantization table (MR122, MR795) | +| rom_dec.h | gray | Word8[8] | gray coding table | +| rom_dec.h | dgray | Word8[8] | gray decoding table | +| rom_dec.h | sqrt_table | Word32[49] | table to compute sqrt(x) | +| rom_dec.h | inv_sqrt_table | Word32[49] | table used in inverse square root computation | +| rom_dec.h | log2_table | Word32[33] | table used in base 2 logarithm computation | +| rom_dec.h | pow2_table | Word32[33] | table used in 2 to the power computation | +| rom_dec.h | cos_table | Word32[65] | table to compute cos(x) in Lsf_lsp() | +| rom_dec.h | acos_slope | Word32[64] | table to compute acos(x) in Lsp_lsf() | +| rom_dec.h | ph_imp_low_MR795 | Word32[40] | phase dispersion impulse response (MR795) | +| rom_dec.h | ph_imp_mid_MR795 | Word32[40] | phase dispersion impulse response (MR795) | +| rom_dec.h | ph_imp_low | Word32[40] | phase dispersion impulse response (MR475 - MR67) | +| rom_dec.h | ph_imp_mid | Word32[40] | phase dispersion impulse response (MR475 - MR67) | +| rom_dec.h | past_rq_init | Word32[80] | initialization table for the MA predictor in DTX | +| rom_dec.h | mean_lsf_3 | Word32[10] | LSF means (not in MR122) | +| rom_dec.h | mean_lsf_5 | Word32[10] | LSF means (MR122) | +| rom_dec.h | pred_fac | Word32[10] | LSF prediction factors (not in MR122) | +| rom_dec.h | dico1_lsf_3 | Word32[3*256] | 1 st LSF quantizer (not in MR122 and MR795) | +| rom_dec.h | dico2_lsf_3 | Word32[3*512] | 2 nd LSF quantizer (not in MR122) | +| rom_dec.h | dico3_lsf_3 | Word32[4*512] | 3 rd LSF quantizer (not in MR122, MR515 and MR475) | +| rom_dec.h | mr515_3_lsf | Word32[4*128] | 3 rd LSF quantizer (MR515 and MR475) | +| rom_dec.h | mr795_1_lsf | Word32[3*512] | 1 st LSF quantizer (MR795) | +| rom_dec.h | dico1_lsf_5 | Word32[4*128] | 1 st LSF quantizer (MR122) | +| rom_dec.h | dico2_lsf_5 | Word32[4*256] | 2 nd LSF quantizer (MR122) | +| rom_dec.h | dico3_lsf_5 | Word32[4*256] | 3 rd LSF quantizer (MR122) | +| rom_dec.h | dico4_lsf_5 | Word32[4*256] | 4 th LSF quantizer (MR122) | +| rom_dec.h | dico5_lsf_5 | Word32[4*64] | 5 th LSF quantizer (MR122) | +| rom_dec.h | table_gain_MR475 | Word32[4*256] | gain quantization table (MR475) | +| rom_dec.h | table_gain_highrates | Word32[128*4] | gain quantization table (MR67, MR74 and MR102) | +| rom_dec.h | table_gain_lowrates | Word32[64*4] | gain quantization table (MR515 and MR59) | +| rom_dec.h | inter_6 | Word32[61] | interpolation filter coefficients | +| rom_dec.h | window_200_40 | Word32[240] | LP analysis window (not in MR122) | +| rom_dec.h | table_speech_bad | UWord8[9] | comparison optimisation table in DTX | +| rom_dec.h | table_SID | UWord8[9] | comparison optimisation table in DTX | +| rom_dec.h | table_DTX | UWord8[9] | comparison optimisation table in DTX | +| rom_dec.h | table_mute | UWord8[9] | comparison optimisation table in DTX | + +### 4.5.3 Static variables used in the C code + +In this section, two tables that specify the static variables for the speech encoder and decoder, respectively, are shown. All static variables are declared within a C **struct**. + +**Table 7: Speech encoder static variables** + +| Struct name | Variable | Type[Length] | Description | +|------------------------------|--------------------|------------------|------------------------------------------------------------------------| +| Speech_Encode_
FrameState | cod_amr_state | cod_amrState | see below in this table | +| | pre_state | Pre_ProcessState | see below in this table | +| | dtx | Word32 | Is set if DTX functionality is used | +| Pre_ProcessState | y2 | Float32 | filter state | +| | y1 | Word16 Float32 | filter state | +| | x0 | Float32 | filter state | +| | x1 | Float32 | filter state | +| cod_amrState | old_speech | Float32 [320] | speech buffer | +| | speech | Float32* | pointer to current frame in old_speech | +| | p_window | Float32* | pointer to LPC analysis window in old_speech | +| | p_window_12k2 | Float32* | pointer to LPC analysis window with no lookahead in old_speech (MR122) | +| | new_speech | Float32* | pointer to the last 160 speech samples in old_speech | +| | old_wsp | Float32 [303] | buffer holding spectral weighted speech | +| | wsp | Float32* | pointer to the current frame in old_wsp | +| | old_lags | Word32[5] | open loop LTP states | +| | ol_gain_flg | Float32 [2] | enables open loop pitch lag weighting (MR102) | +| | old_exc | Float32 [314] | excitation vector | +| | exc | Float32* | current excitation | +| | ai_zero | Float32 [51] | history of weighted synth. filter followed by zero vector | +| | zero | Float32* | zero vector | +| | h1 | Float32* | impulse response of weighted synthesis filter | +| | hvec | Float32 [80] | zero vector followed by impulse response | +| | lpcSt | lpcState | see below in this table | +| | lspSt | lspState | see below in this table | +| | clLtpSt | clLtpState | see below in this table | +| | gainQuantSt | gainQuantState | see below in this table | +| | pitchOLWghtSt | pitchOLWghtState | see below in this table | +| | tonStabSt | tonStabState | see below in this table | +| | vadSt | vadState | see below in this table | +| | vadSt2 | vadState2 | see below in this table | +| | dtx | Word32 | is set if DTX functionality is used | +| | dtx_encSt | dtx_encState | see below in this table | +| | mem_syn | Float32 [10] | synthesis filter memory | +| | mem_w0 | Float32 [10] | weighting filter memory (applied to error signal) | +| | mem_w | Float32 [10] | weighting filter memory (applied to input signal) | +| | mem_err | Float32 [50] | filter memory for production of error vector | +| | error | Float32* | error signal (input minus synthesized speech) | +| | sharp | Float32 | pitch sharpening gain | +| vadState | bckr_est | Float32 [9] | background noise estimate | +| | ave_level | Float32 [9] | averaged input components for stationary estimation | +| | old_level | Float32 [9] | input levels of the previous frame | +| | sub_level | Float32 [9] | input levels calculated at the end of a frame (lookahead) | +| | a_data5 | Float32 [6] | memory for the filter bank | +| | a_data3 | Float32 [5] | memory for the filter bank | +| | burst_count | Word16 | counts length of a speech burst | +| | hang_count | Word16 | hangover counter | +| | stat_count | Word16 | stationary counter | +| | vadreg | Word32 | 15 flags for intermediate VAD decisions | +| | pitch | Word32 | 15 flags for pitch detection | +| | tone | Word16 | 15 flags for tone detection | +| | complex_high | Word16 | flags for complex detection | +| | complex_low | Word16 | flags for complex detection | +| | oldlag_count | Word32 | variables for pitch detection | +| | oldlag | Word32 | variables for pitch detection | +| | complex_hang_count | Word16 | complex hangover counter, used by VAD | +| | complex_hang_timer | Word16 | hangover initiator, used by CAD | + +| Struct name | Variable | Type[Length] | Description | +|------------------|---------------------|----------------|--------------------------------------------------| +| | best_corr_hp | Float32 | filtered value | +| | speech_vad_decision | Word16 | final decision | +| | complex_warning | Word16 | complex background warning | +| | sp_burst_count | Word16 | counts length of a speech burst incl HO addition | +| | corr_hp_fast | Word16 | filtered value | +| dtx_encState | lsp_hist | Float32[80] | LSP history (8 frames) | +| | log_en_hist | Float32 [8] | logarithmic frame energy history (8 frames) | +| | hist_ptr | Word16 | pointer to the cyclic history vectors | +| | log_en_index | Word16 | Index for logarithmic energy | +| | init_lsf_vq_index | Word32 | initial index for lsf predictor | +| | lsp_index | Word16[3] | lsp indecies to the three code books | +| | dtxHangoverCount | Word16 | is decreased in DTX hangover period | +| | decAnaElapsedCount | Word16 | counter for elapsed speech frames in DTX | +| lpcState | LevinsonSt | LevinsonState | see below | +| LevinsonState | old_A | Float32[11] | last frames direct form coefficients | +| lspState | lsp_old | Float32 [10] | old LSP vector | +| | lsp_old_q | Float32 [10] | old quantized LSP vector | +| | qSt | Q_plsfState | see below in this table | +| Q_plsfState | past_rq | Float32[10] | past quantized LSF prediction error | +| clLtpState | pitchSt | Pitch_frState | see below in this table | +| tonStabState | count | Word16 | count consecutive (potential) resonance frames | +| | gp | Float32[7] | pitch gain history | +| Pitch_frState | T0_prev_subframe | Word32 | integer. pitch lag of previous subframe | +| gainQuantState | sf0_gcode0 | Float32 | subframe 0/2 codebook gain | +| | sf0_target_en | Float32 | subframe 0/2 target energy | +| | sf0_coeff | Float32 [5] | subframe 0/2 energy coefficient | +| | gain_idx_ptr | Word16* | pointer to gain index value in parameter frame | +| | gc_predSt | gc_predState | see below in this table | +| | gc_predUncSt | gc_predState | see below in this table | +| | adaptSt | GainAdaptState | see below in this table | +| gc_predState | past_qua_en | Float32[4] | MA predictor memory (20*log10(pred. error)) | +| GainAdaptState | onset | Word16 | onset counter | +| | prev_alpha | Float32 | previous adaptor output | +| | prev_gc | Float32 | previous codebook gain | +| | ltpg_mem | Float32 [5] | pitch gain history | +| pitchOLWghtState | old_T0_med | Word32 | weighted open loop pitch lag | +| | ada_w | Float32 | weigthing level depeding on open loop pitch gain | +| | wght_flg | Word16 | switches lag weighting on and off | + +**Table 8: Speech decoder static variables** + +| Struct name | Variable | Type[Length] | Description | +|--------------------------|---------------------|----------------------|-------------------------------------------------| +| Speech_Decode_FrameState | decoder_amrState | Decoder_amrState | see below in this table | +| | post_state | Post_FilterState | see below in this table | +| | postHP_state | Post_ProcessState | see below in this table | +| Decoder_amrState | old_exc | Word32[194] | excitation vector | +| | exc | Word32* | current excitation | +| | lsp_old | Word32[10] | LSP vector of previous frame | +| | mem_syn | Word32[10] | synthesis filter memory | +| | sharp | Word32 | pitch sharpening gain | +| | old_T0 | Word32 | pitch sharpening lag | +| | prev_bf | Word16 | previous value of "bad frame" flag | +| | prev_pdf | Word16 | previous value of "pot. dangerous frame" flag | +| | state | Word16 | ECU state (0..6) | +| | excEnergyHist | Word32[9] | excitation energy history | +| | T0_lagBuff | Word32 | received pitch lag for ECU | +| | inBackgroundNoise | Word32 | background noise flag | +| | voicedHangover | Word32 | hangover flag | +| | ltpGainHistory | Word32[9] | pitch gain history | +| | background_state | Bgn_scdState | see below in this table | +| | Cb_gain_averState | Cb_gain_averageState | see below in this table | +| | lsp_avg_st | lsp_avgState | see below in this table | +| | lsfState | D_plsfState | see below in this table | +| | ec_gain_p_st | ec_gain_pitchState | see below in this table | +| | ec_gain_c_st | ec_gain_codeState | see below in this table | +| | pred_state | gc_predState | see table 7 | +| | nodataSeed | Word16 | seed for CN generator | +| | ph_disp_st | ph_dispState | see below in this table | +| | dtxDecoderState | dtx_decState | see below in this table | +| dtx_decState | since_last_sid | Word16 | number of frames since last SID frame | +| | true_sid_period_inv | Word16 | inverse of true SID update rate | +| | log_en | Word32 | logarithmic frame energy | +| | old_log_en | Word32 | previous value of log_en | +| | pn_seed_rx | Word32 | random number generator seed | +| | lsp | Word32[10] | LSP vector | +| | lsp_old | Word32[10] | previous LSP vector | +| | lsf_hist | Word32[80] | LSF vector history (8 frames) | +| | lsf_hist_ptr | Word16 | index to beginning of LSF history | +| | lsf_hist_mean | Word32[80] | mean-removed LSF history (8 frames) | +| | log_pg_mean | Word16 | mean-removed logarithmic prediction gain | +| | log_en_hist | Word32[8] | logarithmic frame energy history | +| | log_en_hist_ptr | Word16 | index to beginning of log. frame energy history | +| | log_en_adjust | Word16 | mode-dependent frame energy adjustment | +| | dtxHangoverCount | Word16 | counts down in hangover period | +| | decAnaElapsedCount | Word16 | counts elapsed speech frames after DTX | +| | sid_frame | Word16 | flags SID frames | +| | valid_data | Word16 | flags SID frames containing valid data | +| | dtxHangoverAdded | Word16 | flags hangover period at end of speech | +| | dtxGlobalState | enum DTXStateType | DTX state flags | +| | data_updated | Word16 | flags CNI updates | +| Bgn_scdState | frameEnergyHist | Word32[60] | history of synthesis frame energy | +| | bgHangover | Word16 | number of frames since last speech frame | +| Cb_gain_averageState | cbGainHistory | Word32[7] | codebook gain history | +| | hangVar | Word16 | counts length of talkspurt in subframes | +| | hangCount | Word16 | number of subframes since last talkspurt | +| lsp_avgState | lsp_meanSave | Word32[10] | averaged LSP vector | +| D_plsfState | past_r_q | Word32[10] | past quantized LSF prediction vector | +| | past_lsf_q | Word32[10] | past dequantized LSF vector | +| ec_gain_pitchState | pbuf | Word32[5] | pitch gain history | +| | past_gain_pit | Word32 | previous pitch gain (limited to 1.0) | +| | prev_gp | Word32 | previous good pitch gain | +| ec_gain_codeState | gbuf | Word32[5] | codebook gain history | +| | past_gain_code | Word32 | previous codebook gain | +| | prev_gc | Word32 | previous good codebook gain | +| ph_dispState | gainMem | Word32[5] | pitch gain history | +| | prevState | Word32 | previously used impulse response | +| | prevCbGain | Word32 | previous codebook gain | +| | lockFull | Word16 | force maximum phase dispersion | +| | onset | Word16 | onset counter | +| | res2 | Word32[40] | LP residual | +| Post_FilterState | mem_syn_pst | Word32[10] | synthesis filter memory | +| | synth_buf | Word16[170] | synthesis filter work area | +| | agc_state | agcState | see below in this table | +| | preemph_state | preemphasisState | see below in this table | +| agcState | past_gain | Word16 | past agc gain | +| preemphasisState | mem_pre | Word16 | filter state | + +| Struct name | Variable | Type[Length] | Description | +|-------------------|----------|--------------|--------------------------| +| Post_ProcessState | y2_hi | Word32 | filter state, upper word | +| | y2_lo | Word32 | filter state, lower word | +| | y1_hi | Word32 | filter state, upper word | +| | y1_lo | Word32 | filter state, lower word | +| | x0 | Word32 | filter state | +| | x1 | Word32 | filter state | + +# 5 Homing procedure + +The principles of the homing procedures are described in 3GPP TS 06.090 [2]. This specification only includes a detailed description of the 8 decoder homing frames. For each AMR codec mode, the corresponding decoder homing frame has a fixed set of speech parameters shown in table 9a-9h. The bit allocation within these parameters is identical to the corresponding bit allocation of the source encoder output parameters given in 3GPP TS 06.090 [2]. + +In the following tables, the following naming convention is used for the individual parameters. Letters in *italics* indicate numbers. + +- LPC *n* index of *n*th LSF submatrix +- LTP-LAG *m* adaptive codebook index for subframe *m* +- LTP-GAIN *m* adaptive codebook gain index in subframe *m* +- FCB-GAIN *m* fixed codebook gain index in subframe *m* +- GAIN\_VQ *m* codebook gain VQ index in subframe *m* (subframe *m* and *m+1* for MR475) +- POS *m\_n* position index of *n*th pulse in subframe *m* +- POS *m\_n\_k* position index of *n*th and *k*th pulse in subframe *m* +- POS *m\_n\_k\_l\_j* position index of *n*th, *k*th, *l*th, and *j*th pulse in subframe *m* +- SIGN *m\_n\_k* sign information for *n*th and *k*th pulse in subframe *m* +- SIGN *m\_n\_k\_l\_j* sign information for *n*th, *k*th, *l*th, and *j*th pulse in subframe *m* +- SIGN *m\_n\_k\_POS\_m\_n* sign information for *n*th and *k*th pulse and position index for *n*th pulse in subframe *m* + +**Table 9a: Parameter values for the decoder homing frame (MR475)** + +| Parameter | Value (LSB=b0) | +|------------|----------------| +| LPC 1 | 0x00F8 | +| LPC 2 | 0x009D | +| LPC 3 | 0x001C | +| LTP-LAG 1 | 0x0066 | +| POS 1_1_2 | 0x0000 | +| SIGN_1_1_2 | 0x0003 | +| GAIN-VQ 1 | 0x0028 | +| LTP-LAG 2 | 0x000F | +| POS 2_1_2 | 0x0038 | +| SIGN_2_1_2 | 0x0001 | +| LTP-LAG 3 | 0x000F | +| POS 3_1_2 | 0x0031 | +| SIGN_3_1_2 | 0x0002 | +| GAIN-VQ 3 | 0x0008 | +| LTP-LAG 4 | 0x000F | +| POS 4_1_2 | 0x0026 | +| SIGN_4_1_2 | 0x0003 | + +**Table 9b: Parameter values for the decoder homing frame (MR515)** + +| Parameter | Value (LSB=b0) | +|------------------|-----------------------| +| LPC 1 | 0x00F8 | +| LPC 2 | 0x009D | +| LPC 3 | 0x001C | +| LTP-LAG 1 | 0x0066 | +| POS 1_1_2 | 0x0000 | +| SIGN_1_1_2 | 0x0003 | +| GAIN-VQ 1 | 0x0037 | +| LTP-LAG 2 | 0x000F | +| POS 2_1_2 | 0x0000 | +| SIGN_2_1_2 | 0x0003 | +| GAIN-VQ 2 | 0x0005 | +| LTP-LAG 3 | 0x000F | +| POS 3_1_2 | 0x0037 | +| SIGN_3_1_2 | 0x0003 | +| GAIN-VQ 3 | 0x0037 | +| LTP-LAG 4 | 0x000F | +| POS 4_1_2 | 0x0023 | +| SIGN_4_1_2 | 0x0003 | +| GAIN-VQ 4 | 0x001F | + +**Table 9c: Parameter values for the decoder homing frame (MR59)** + +| Parameter | Value (LSB=b0) | +|------------------|-----------------------| +| LPC 1 | 0x00F8 | +| LPC 2 | 0x00E3 | +| LPC 3 | 0x002F | +| LTP-LAG 1 | 0x00BD | +| POS 1_1_2 | 0x0000 | +| SIGN_1_1_2 | 0x0003 | +| GAIN-VQ 1 | 0x0037 | +| LTP-LAG 2 | 0x000F | +| POS 2_1_2 | 0x0001 | +| SIGN_2_1_2 | 0x0003 | +| GAIN-VQ 2 | 0x000F | +| LTP-LAG 3 | 0x0060 | +| POS 3_1_2 | 0x00F9 | +| SIGN_3_1_2 | 0x0003 | +| GAIN-VQ 3 | 0x0037 | +| LTP-LAG 4 | 0x000F | +| POS 4_1_2 | 0x0000 | +| SIGN_4_1_2 | 0x0003 | +| GAIN-VQ 4 | 0x0037 | + +**Table 9d: Parameter values for the decoder homing frame (MR67)** + +| Parameter | Value (LSB=b0) | +|------------------|-----------------------| +| LPC 1 | 0x00F8 | +| LPC 2 | 0x00E3 | +| LPC 3 | 0x002F | +| LTP-LAG 1 | 0x00BD | +| POS 1_1_2_3 | 0x0002 | +| SIGN_1_1_2_3 | 0x0007 | +| GAIN-VQ 1 | 0x0000 | +| LTP-LAG 2 | 0x000F | +| POS 2_1_2_3 | 0x0098 | +| SIGN_2_1_2_3 | 0x0007 | +| GAIN-VQ 2 | 0x0061 | +| LTP-LAG 3 | 0x0060 | +| POS 3_1_2_3 | 0x05C5 | +| SIGN_3_1_2_3 | 0x0007 | +| GAIN-VQ 3 | 0x0000 | +| LTP-LAG 4 | 0x000F | +| POS 4_1_2_3 | 0x0318 | +| SIGN_4_1_2_3 | 0x0007 | +| GAIN-VQ 4 | 0x0000 | + +**Table 9e: Parameter values for the decoder homing frame (MR74)** + +| Parameter | Value (LSB=b0) | +|------------------|-----------------------| +| LPC 1 | 0x00F8 | +| LPC 2 | 0x00E3 | +| LPC 3 | 0x002F | +| LTP-LAG 1 | 0x00BD | +| POS 1_1_2_3_4 | 0x0006 | +| SIGN_1_1_2_3_4 | 0x000F | +| GAIN-VQ 1 | 0x0000 | +| LTP-LAG 2 | 0x001B | +| POS 2_1_2_3_4 | 0x0208 | +| SIGN_2_1_2_3_4 | 0x000F | +| GAIN-VQ 2 | 0x0062 | +| LTP-LAG 3 | 0x0060 | +| POS 3_1_2_3_4 | 0x1BA6 | +| SIGN_3_1_2_3_4 | 0x000F | +| GAIN-VQ 3 | 0x0000 | +| LTP-LAG 4 | 0x001B | +| POS 4_1_2_3_4 | 0x0006 | +| SIGN_4_1_2_3_4 | 0x000F | +| GAIN-VQ 4 | 0x0000 | + +**Table 9f: Parameter values for the decoder homing frame (MR795)** + +| Parameter | Value (LSB=b0) | +|----------------|----------------| +| LPC 1 | 0x00C2 | +| LPC 2 | 0x00E3 | +| LPC 3 | 0x002F | +| LTP-LAG 1 | 0x00BD | +| POS_1_1_2_3_4 | 0x0006 | +| SIGN_1_1_2_3_4 | 0x000F | +| LTP-GAIN 1 | 0x000A | +| FCB-GAIN 1 | 0x0000 | +| LTP-LAG 2 | 0x0039 | +| POS_2_1_2_3_4 | 0x1C08 | +| SIGN_2_1_2_3_4 | 0x0007 | +| LTP-GAIN 2 | 0x000A | +| FCB-GAIN 2 | 0x000B | +| LTP-LAG 3 | 0x0063 | +| POS_3_1_2_3_4 | 0x11A6 | +| SIGN_3_1_2_3_4 | 0x000F | +| LTP-GAIN 3 | 0x0001 | +| FCB-GAIN 3 | 0x0000 | +| LTP-LAG 4 | 0x0039 | +| POS_4_1_2_3_4 | 0x09A0 | +| SIGN_4_1_2_3_4 | 0x000F | +| LTP-GAIN 4 | 0x0002 | +| FCB-GAIN 4 | 0x0001 | + +**Table 9g: Parameter values for the decoder homing frame (MR102)** + +| Parameter | Value (LSB=b0) | +|------------------|-----------------------| +| LPC 1 | 0x00F8 | +| LPC 2 | 0x00E3 | +| LPC 3 | 0x002F | +| LTP-LAG 1 | 0x0045 | +| SIGN_1_1_5 | 0x0000 | +| SIGN_1_2_6 | 0x0000 | +| SIGN_1_3_7 | 0x0000 | +| SIGN_1_4_8 | 0x0000 | +| POS_1_1_2_5 | 0x0000 | +| POS_1_3_6_7 | 0x0000 | +| POS_1_4_8 | 0x0000 | +| GAIN-VQ_1 | 0x0000 | +| LTP-LAG 2 | 0x001B | +| SIGN_2_1_5 | 0x0000 | +| SIGN_2_2_6 | 0x0001 | +| SIGN_2_3_7 | 0x0000 | +| SIGN_2_4_8 | 0x0001 | +| POS_2_1_2_5 | 0x0326 | +| POS_2_3_6_7 | 0x00CE | +| POS_2_4_8 | 0x007E | +| GAIN-VQ_2 | 0x0051 | +| LTP-LAG 3 | 0x0062 | +| SIGN_3_1_5 | 0x0000 | +| SIGN_3_2_6 | 0x0000 | +| SIGN_3_3_7 | 0x0000 | +| SIGN_3_4_8 | 0x0000 | +| POS_3_1_2_5 | 0x015A | +| POS_3_3_6_7 | 0x0359 | +| POS_3_4_8 | 0x0076 | +| GAIN-VQ_3 | 0x0000 | +| LTP-LAG 4 | 0x001B | +| SIGN_4_1_5 | 0x0000 | +| SIGN_4_2_6 | 0x0000 | +| SIGN_4_3_7 | 0x0000 | +| SIGN_4_4_8 | 0x0000 | +| POS_4_1_2_5 | 0x017C | +| POS_4_3_6_7 | 0x0215 | +| POS_4_4_8 | 0x0038 | +| GAIN-VQ_4 | 0x0030 | + +**Table 9h: Parameter values for the decoder homing frame (MR122)** + +| Parameter | Value (LSB=b0) | +|---------------------|-----------------------| +| LPC1 | 0x0004 | +| LPC2 | 0x002A | +| LPC3 | 0x00DB | +| LPC4 | 0x0096 | +| LPC5 | 0x002A | +| LTP-LAG 1 | 0x0156 | +| LTP-GAIN 1 | 0x000B | +| SIGN_1_1_6_POS_1_1 | 0x0000 | +| SIGN_1_2_7_POS_1_2 | 0x0000 | +| SIGN_1_3_8_POS_1_3 | 0x0000 | +| SIGN_1_4_9_POS_1_4 | 0x0000 | +| SIGN_1_5_10_POS_1_5 | 0x0000 | +| POS 1_6 | 0x0000 | +| POS 1_7 | 0x0000 | +| POS 1_8 | 0x0000 | +| POS 1_9 | 0x0000 | +| POS 1_10 | 0x0000 | +| FCB-GAIN 1 | 0x0000 | +| LTP-LAG 2 | 0x0036 | +| LTP-GAIN 2 | 0x000B | +| SIGN_2_1_6_POS_2_1 | 0x0000 | +| SIGN_2_2_7_POS_2_2 | 0x000F | +| SIGN_2_3_8_POS_2_3 | 0x000E | +| SIGN_2_4_9_POS_2_4 | 0x000C | +| SIGN_2_5_10_POS_2_5 | 0x000D | +| POS 2_6 | 0x0000 | +| POS 2_7 | 0x0001 | +| POS 2_8 | 0x0005 | +| POS 2_9 | 0x0007 | +| POS 2_10 | 0x0001 | +| FCB-GAIN 2 | 0x0008 | +| LTP-LAG 3 | 0x0024 | +| LTP-GAIN 3 | 0x0000 | +| SIGN_3_1_6_POS_3_1 | 0x0001 | +| SIGN_3_2_7_POS_3_2 | 0x0000 | +| SIGN_3_3_8_POS_3_3 | 0x0005 | +| SIGN_3_4_9_POS_3_4 | 0x0006 | +| SIGN_3_5_10_POS_3_5 | 0x0001 | +| POS 3_6 | 0x0002 | +| POS 3_7 | 0x0004 | +| POS 3_8 | 0x0007 | +| POS 3_9 | 0x0004 | +| POS 3_10 | 0x0002 | +| FCB-GAIN 3 | 0x0003 | +| LTP-LAG 4 | 0x0036 | +| LTP-GAIN 4 | 0x000B | +| SIGN_4_1_6_POS_4_1 | 0x0000 | +| SIGN_4_2_7_POS_4_2 | 0x0002 | +| SIGN_4_3_8_POS_4_3 | 0x0004 | +| SIGN_4_4_9_POS_4_4 | 0x0000 | +| SIGN_4_5_10_POS_4_5 | 0x0003 | +| POS 4_6 | 0x0006 | +| POS 4_7 | 0x0001 | +| POS 4_8 | 0x0007 | +| POS 4_9 | 0x0006 | +| POS 4_10 | 0x0005 | +| FCB-GAIN 4 | 0x0000 | + +# 6 File formats + +This section describes the file formats used by the encoder and decoder programs. The test sequences defined in [2] also use the file formats described here. + +## 6.1 Speech file (encoder input / decoder output) + +Speech files read by the encoder and written by the decoder consist of 16-bit words where each word contains a 13-bit, left aligned speech sample. The byte order depends on the host architecture (e.g. MSByte first on SUN workstations, LSByte first on PCs etc.). Both the encoder and the decoder program process complete frames (of 160 samples) only. + +This means that the encoder will only process $n$ frames if the length of the input file is $n*160 + k$ words, while the files produced by the decoder will always have a length of $n*160$ words. + +## 6.2 Mode control file (encoder input) + +The encoder program can optionally read in a mode control file which specifies the encoding mode for each frame of speech processed. The file is a text file containing one line per speech frame. Each line contains one of the mode names from the list {MR475, MR515, MR59, MR67, MR74, MR795, MR102, MR122}. + +## 6.3 Parameter bitstream file (encoder output / decoder input) + +The files produced by the speech encoder/expected by the speech decoder contain an arbitrary number of frames in the format described in RFC 3267 [9], sections 5.1 and 5.3. + +By using preprocessor definition encoder/decoder can optionally use AMR Interface Format 2. The format is described in TS 26.101 [8] Annex A. + +By using another preprocessor definition encoder/decoder can optionally use format compatible with the existing AMR fixed-point C-code. Frame format is following. + +| | | | | | | | | | +|------------|----|----|-----|------|-----------|----------------|-----|----------------| +| FRAME_TYPE | B1 | B2 | ... | B244 | MODE_INFO | unused1 | ... | unused4 | +|------------|----|----|-----|------|-----------|----------------|-----|----------------| + +Each box corresponds to one `Word16` value in the bitstream file, for a total of 250 words or 500 bytes per frame. The fields have the following meaning: + +**FRAME\_TYPE** transmit frame type, which is one of + +- TX\_SPEECH (0x0000)** +- TX\_SID\_FIRST (0x0001)** +- TX\_SID\_UPDATE (0x0002)** +- TX\_NO\_DATA (0x0003)** + +**B0...B244** speech encoder parameter bits (i.e. the bitstream itself). Each B $x$ either has the value 0x0000 or 0x0001. Only mode MR122 really uses all 244 bits; for the other modes, only the first $n$ bits are used ( $35 \le n \le 204$ ). The remaining bits are unused (written as 0x0000) + +**MODE\_INFO** encoding mode information, which is one of + +- MR475 (0x0000)** +- MR515 (0x0001)** +- MR59 (0x0002)** +- MR67 (0x0003)** +- MR74 (0x0004)** +- MR795 (0x0005)** +- MR102 (0x0006)** +- MR122 (0x0007)** + +***unused1...4*** unused, written as 0x0000 + +As indicated in section 6.1 above, the byte order depends on the host architecture. + +# Annex A (informative): Change History + +| TSG
SA# | Tdoc | CR | Rev | Cat | PH | Vers | New
Vers | Subject | +|------------|-----------|------|-----|-----|--------|-------|-------------|------------------------------------------------------------------------| +| 10 | SP-000577 | 002 | | A | Rel-4 | 3.0.0 | 4.0.0 | AMR Core Frame bit ordering (AMR speech Codec; Floating point C-Code | +| 12 | SP-010306 | 004 | 1 | A | Rel-4 | 4.0.0 | 4.1.0 | Limiting predicted codebook gain computing in encoder | +| 12 | SP-010306 | 006 | 1 | A | Rel-4 | 4.0.0 | 4.1.0 | Correction of decoder operation in error concealment of lost frames | +| 12 | SP-010306 | 008 | 1 | A | Rel-4 | 4.0.0 | 4.1.0 | Correction of mode state bug in AMR decoder | +| 12 | SP-010306 | 012 | 1 | A | Rel-4 | 4.0.0 | 4.1.0 | Correction of decoder Reset | +| 12 | SP-010306 | 014 | 1 | A | Rel-4 | 4.0.0 | 4.1.0 | Correction of comfort noise parameter interpolation bug of AMR decoder | +| 12 | SP-010306 | 016 | 1 | A | Rel-4 | 4.0.0 | 4.1.0 | Correction of the TX_TYPE and RX_TYPE identifiers | +| | MCC | | | | Rel-4 | 4.1.0 | 4.1.1 | Correction of bugs in code | +| 13 | SP-010452 | 010 | 1 | A | Rel-4 | 4.1.1 | 4.2.0 | Correction to make encoder and decoder memories independent | +| 13 | SP-010452 | 018 | | A | Rel-4 | 4.1.1 | 4.2.0 | Correction of decoder operation in error concealment of lost frames | +| 15 | SP-020079 | 019 | | A | Rel-4 | 4.2.0 | 4.3.0 | Maintaining bit-exactness with TS 26.073 | +| 16 | | | | | | | 5.0.0 | Version for Release 5 | +| 19 | SP-030088 | 21 | 1 | F | Rel-5 | 5.0.0 | 5.1.0 | MMS compatible i/o format option | +| 19 | SP-030088 | 24 | | A | Rel-5 | 5.0.0 | 5.1.0 | Correction to floating-point implementation of sp_dec.c | +| 20 | SP-030214 | 26 | | A | Rel-5 | 5.1.0 | 5.2.0 | Correction on codec mode handling during DTX | +| 22 | SP-030681 | 29 | 1 | F | Rel-5 | 5.2.0 | 5.3.0 | Correction on the implementation of the interface of decoder.c | +| 22 | SP-030682 | 30 | 1 | D | Rel-6 | 5.3.0 | 6.0.0 | Correction on the default behaviour of the unix makefile | +| 23 | SP-040198 | 32 | | A | Rel-6 | 6.0.0 | 6.1.0 | Correction of floating point AMR DTX functionality | +| 36 | SP-070321 | 0033 | 1 | F | Rel-7 | 6.1.0 | 7.0.0 | Bit order of Mode Indication in AMR comfort noise frames | +| 42 | | | | | Rel-8 | | 8.0.0 | Version for Release 8 | +| 46 | | | | | Rel-9 | | 9.0.0 | Version for Release 9 | +| 51 | | | | | Rel-10 | | 10.0.0 | Version for Release 10 | +| 57 | | | | | Rel-11 | | 11.0.0 | Version for Release 11 | \ No newline at end of file diff --git a/marked/Rel-11/26_series/26110/ca4d4ff86cf319ed7cc36a1ecda29101_img.jpg b/marked/Rel-11/26_series/26110/ca4d4ff86cf319ed7cc36a1ecda29101_img.jpg new file mode 100644 index 0000000000000000000000000000000000000000..065f2a910b9715e24eaba3f34f82a4aa880faec8 --- /dev/null +++ b/marked/Rel-11/26_series/26110/ca4d4ff86cf319ed7cc36a1ecda29101_img.jpg @@ -0,0 +1,3 @@ +version https://git-lfs.github.com/spec/v1 +oid sha256:797df5345aae7ebb9c76712569954a1550ae9e749be233bdf4b798d9b2eb293e +size 98565 diff --git a/marked/Rel-11/26_series/26110/raw.md b/marked/Rel-11/26_series/26110/raw.md new file mode 100644 index 0000000000000000000000000000000000000000..548d28ce6590e75e68d3cc39d2de4eab2ac69cfc --- /dev/null +++ b/marked/Rel-11/26_series/26110/raw.md @@ -0,0 +1,355 @@ + + + + + + +# --- Contents + +| | | +|------------------------------------------------------------|-----------| +| Foreword ..... | 4 | +| Introduction ..... | 4 | +| 1 Scope..... | 5 | +| 2 References..... | 5 | +| 3 Definitions and abbreviations ..... | 6 | +| 3.1 Definitions..... | 6 | +| 3.2 Abbreviations ..... | 6 | +| 4 General..... | 7 | +| 5 ITU-T H.324 ..... | 8 | +| 6 Modifications to H.324 (3GPP TS 26.111)..... | 8 | +| 7 Call set-up requirements ..... | 8 | +| 8 Terminal implementor's guide (3GPP TR 26.911)..... | 8 | +| Annex A (informative): Background information ..... | 9 | +| A.1 Video I/O Equipment..... | 9 | +| A.2 Video Codec..... | 9 | +| A.2.1 H.261 ..... | 10 | +| A.2.2 H.263 ..... | 10 | +| A.2.3 MPEG-4 ..... | 10 | +| A.3 Audio I/O Codec ..... | 10 | +| A.4 Speech Codec..... | 10 | +| A.4.1 3GPP AMR ..... | 10 | +| A.4.2 G.723.1 ..... | 11 | +| A.5 User Data Applications ..... | 11 | +| A.5.1 Data conferencing – T.120..... | 11 | +| A.5.2 Text conversation – T.140..... | 12 | +| A.6 Data Protocols..... | 12 | +| A.7 System Control..... | 12 | +| A.8 Call Set-up..... | 12 | +| A.9 H.245..... | 12 | +| A.10 H.223..... | 12 | +| A.10.1 Level 0..... | 13 | +| A.10.2 Level 1..... | 13 | +| A.10.3 Level 2..... | 13 | +| A.10.4 Level 3..... | 13 | +| Annex B (informative): Bibliography ..... | 13 | +| Annex C (informative): Change history..... | 14 | + +# --- Foreword + +This Technical Specification has been produced by the 3GPP. + +The present document introduces the set of specifications which apply to 3G-324M multimedia terminals within the 3GPP system. + +The contents of the present document are subject to continuing work within the TSG and may change following formal TSG approval. Should the TSG modify the contents of this TS, it will be re-released by the TSG with an identifying change of release date and an increase in version number as follows: + +Version 3.y.z + +where: + +- x the first digit: + - 1 presented to TSG for information; + - 2 presented to TSG for approval; + - 3 Indicates TSG approved document under change control. +- y the second digit is incremented for all changes of substance, i.e. technical enhancements, corrections, updates, etc. +- z the third digit is incremented when editorial only changes have been incorporated in the specification; + +# --- Introduction + +This document contains a specification for H.324 based multimedia codecs for circuit switched 3GPP networks. The term codec is usually associated with a single media type. However, many multimedia services require a close integration of disparate media types. In this sense, the representations of these media types (in the form of media streams) are at least logically bound into a single multimedia stream. As such, a H.324 based multimedia codec must handle multiplexing/de-multiplexing and skew. It will also have to provide codecs for each of the derived media streams. End-to-end, in-band control is also required for the purposes of configuration and establishing individual media streams. Finally, since 3GPP networks are inherently error prone, error detection and/or correction must also be provided by the multimedia codec since it has a comprehensive view of the bit stream it produces and therefore can apply the most efficient form of error detection and/or correction. + +# --- 1 Scope + +This specification introduces the set of specifications which apply to 3G-324M multimedia terminals. + +# --- 2 References + +The following documents contain provisions which, through reference in this text, constitute provisions of the present document. + +- References are either specific (identified by date of publication, edition number, version number, etc.) or non-specific. + - For a specific reference, subsequent revisions do not apply. + - For a non-specific reference, the latest version applies. In the case of a reference to a 3GPP document (including a GSM document), a non-specific reference implicitly refers to the latest version of that document *in the same Release as the present document*. +- [1] ITU-T Recommendation H.223: "Multiplexing protocol for low bitrate multimedia communication" +- [2] ITU-T Recommendation H.223 — Annex A: "Multiplexing protocol for low bitrate multimedia communication over low error-prone channels" +- [3] ITU-T Recommendation H.223 — Annex B: "Multiplexing protocol for low bitrate multimedia communication over moderate error-prone channels" +- [4] ITU-T Recommendation H.223 — Annex C: "Multiplexing protocol for low bitrate multimedia communication over highly error-prone channels" +- [5] ITU-T Recommendation H.223 — Annex D: "Optional multiplexing protocol for low bitrate multimedia communication over highly error-prone channels" +- [6] ITU-T Recommendation H.245: "Control protocol for multimedia communication" +- [7] ITU-T Recommendation G.723.1: "Dual rate speech coder for multimedia communication transmitting at 5.3 & 6.3 kbit/s" +- [8] ITU-T Recommendation H.263: "Video coding for low bitrate communication" +- [9] ITU-T Recommendation H.261: "Video CODEC for audiovisual services at p X 64 kbit/s" +- [10] ITU-T Recommendation H.324: "Terminal for low bitrate multimedia communication" +- [11] 3GPP TS 26.111: "Modifications to H.324" +- [12] 3GPP TR 26.911: "Terminal Implementor's Guide" +- [13] ITU-T Recommendation X.691: "Information Technology - ASN.1 Encoding Rules - Specification of Packed Encoding Rules (PER)" +- [14] International Standard ISO/IEC 14494-2: "Information technology — Generic coding of audiovisual object — Part 2: Visual, 1999" +- [15] 3GPP TS 26.071: "Mandatory Speech Codec; General Description" +- [16] 3GPP TS 26.090: "Mandatory Speech Codec; Speech Transcoding Functions" +- [17] 3GPP TS 26.073: "Mandatory Speech Codec; ANSI C-Code" +- [18] ITU-T T.140 (1998) Presentation protocol for text conversation application. +- [19] 3GPP TS 22.226: "Global Text Telephony; Stage 1" +- [20] 3GPP TS 24.008: "Mobile Radio Interface - Layer 3 MM/CC Specification". + +- [21] 3GPP TS 27.001: "General on Terminal Adaptation Functions (TAF) for Mobile Stations (MS)". +- [22] 3GPP TS 29.007: "General requirements on interworking between the Public Land Mobile Network (PLMN) and the Integrated Services Digital Network (ISDN) or Public Switched Telephone Network (PSTN)". +- [23] 3GPP TS 23.108: "Mobile radio interface layer 3 specification; Core Network protocols; stage 2". +- [24] ITU-T Recommendation G.712: "Transmission performance characteristics of pulse code modulation channels". +- [25] ITU-T Recommendation T.120: "Data protocols for multimedia conferencing". + +# --- 3 Definitions and abbreviations + +## 3.1 Definitions + +For the purposes of the present document, the following terms and definitions apply. + +**H.324 terminal:** ITU-T H.324 recommendation, including Annex C + +**3G-324M terminal:** Based on ITU-T H.324 recommendation modified by 3GPP for purposes of 3GPP circuit switched network based video telephony + +## 3.2 Abbreviations + +For the purposes of the present document, the following abbreviations apply: + +| | | +|----------|---------------------------------------------------| +| ACELP | Algebraic-Code-Excited Linear-Prediction | +| ADC | Analogue Digital Converter | +| AEC | Acoustic Echo Cancellation | +| AL | Adaptation Layer | +| CCSRL | Control Channel Segmentation and Reassembly Layer | +| CELP | Code-Excited Linear-Prediction | +| CT | Correlation Threshold | +| DAC | Digital Analogue Converter | +| DCT | Discrete Cosine Transformation | +| EI | Error Indication | +| EOB | End Of Block | +| FEC | Forward Error Correction | +| GOB | Group Of Blocks | +| GQUANT | Group Quantizer information | +| GTT | Global Text Telephony | +| HDLC | High-Level Data Link Control | +| HEC | Header Error Control | +| ISDN | Integrated Services Digital Network | +| LAPM | Link Access Procedure for Modems | +| LC | Logical Channel | +| MC | Multiplex Code | +| MCU | Multipoint Communication Unit | +| MP-MLQ | Multipulse Maximum Likelihood Quantization | +| MPL | Multiplex Payload Length | +| MR-ACELP | Multi-rate ACELP | +| PC | Personal Computer | +| MCU | Multipoint Conference Unit | +| MUX | H.223 Multiplex layer | +| PDU | Protocol Data Unit | +| SN | Sequence Number | +| VLC | Variable Length Code | + +# 4 General + +3G-324M terminals provide real-time video, audio, or data, in any combination, including none, over 3GPP circuit-switched, radio networks. They are based on ITU-T H.324 with Annex C, and Annex H when mobile multilink operation is supported. Communication may be either 1-way or 2-way. Such terminals may be part of a portable device or integrated into an automobile or other non fixed location device. They may also be fixed, stand-alone devices; for example, a video telephone or kiosk. 3G-324M terminals may also be integrated into PCs and workstations. + +In addition to 3G-324M to 3G-324M communication, interoperation with other types of multimedia telephone terminals is possible, however a gateway may be required. + +Multipoint communication between more than two 3G-324M terminals is possible using a Multipoint Communication Unit (MCU). MCU functionality is for further study. + +3G-324M terminals are based on ITU-T H.324 with Annex C, and Annex H when mobile multilink operation is supported. For performance reasons and to reference the call set-up procedures, some modifications to H.324 were made. These are described in 3GPP TS 26.111, except call set-up procedures are described in 3GPP TS 24.008, 27.001, 29.007 and 23.108. 3G-324M terminals shall conform to these specifications. Because of the many options in H.324, an implementor's guide, 3GPP TR 26.911, provides preferred options for 3G-324M implementations. + +Figure 1 below shows the functional components of a generic 3GPP multimedia terminal. The video, speech, data and multilink components are optional. If a media type is supported, the standards indicated are mandatory except those enclosed in square brackets are optional. + +![Figure 1: Scope of circuit switched multimedia 3GPP specification. The diagram shows the functional components of a generic 3GPP multimedia terminal. A central dashed box labeled '3GPP TS 26.111' contains several functional blocks. On the left, four input blocks are shown: 'Video I/O Equipment', 'Audio I/O Equipment', 'User Data Applications [T.120, ...]', and 'System Control'. The 'Video I/O Equipment' connects to a 'Video Codec H.263, [MPEG-4, H.261 ...]'. The 'Audio I/O Equipment' connects to a 'Speech Codec 3GPP-AMR, [G.723.1 ...]'. The 'Speech Codec' connects to an 'Optional Receive Path Delay' block, which then connects to a 'Multiplex/Demultiplex H.223, H.223 Annex A, H.223 Annex B, [H.223 Annex C, H.223 Annex D]' block. The 'User Data Applications' connect to 'Data Protocols [V.14, LAPM, ...]', which also connect to the 'Multiplex/Demultiplex' block. The 'System Control' connects to an 'H.245' block, which connects to 'CCSRL', which connects to 'NSRP[LAP M/V.42]', which then connects to the 'Multiplex/Demultiplex' block. The 'Multiplex/Demultiplex' block connects to an 'Optional Multilink H.324 Annex H' block, which in turn connects to the '3GPP Network'. The 'System Control' also connects to a 'Call Set-up' block, which connects to the '3GPP Network'.](ca4d4ff86cf319ed7cc36a1ecda29101_img.jpg) + +Figure 1: Scope of circuit switched multimedia 3GPP specification. The diagram shows the functional components of a generic 3GPP multimedia terminal. A central dashed box labeled '3GPP TS 26.111' contains several functional blocks. On the left, four input blocks are shown: 'Video I/O Equipment', 'Audio I/O Equipment', 'User Data Applications [T.120, ...]', and 'System Control'. The 'Video I/O Equipment' connects to a 'Video Codec H.263, [MPEG-4, H.261 ...]'. The 'Audio I/O Equipment' connects to a 'Speech Codec 3GPP-AMR, [G.723.1 ...]'. The 'Speech Codec' connects to an 'Optional Receive Path Delay' block, which then connects to a 'Multiplex/Demultiplex H.223, H.223 Annex A, H.223 Annex B, [H.223 Annex C, H.223 Annex D]' block. The 'User Data Applications' connect to 'Data Protocols [V.14, LAPM, ...]', which also connect to the 'Multiplex/Demultiplex' block. The 'System Control' connects to an 'H.245' block, which connects to 'CCSRL', which connects to 'NSRP[LAP M/V.42]', which then connects to the 'Multiplex/Demultiplex' block. The 'Multiplex/Demultiplex' block connects to an 'Optional Multilink H.324 Annex H' block, which in turn connects to the '3GPP Network'. The 'System Control' also connects to a 'Call Set-up' block, which connects to the '3GPP Network'. + +**Figure 1 Scope of circuit switched multimedia 3GPP specification. Items in [brackets] are optional.** + +Short descriptions of ITU-T H.324, 3GPP TS 26.111, and 3GPP TR 26.911 are given below. + +# 5 ITU-T H.324 + +ITU-T H.324 describes terminals for low bitrate multimedia communication. That ITU-T recommendation contains “ANNEX C, Multimedia Telephone Terminals Over Error Prone Channels” (sometimes referred to as H.324/M) and “ANNEX H, Mobile Multilink Operation”. These annexes are considered an integral part of the recommendation. + +Therefore, herewith H.324 shall mean ITU-T H.324 with Annex C. When multilink operation is utilized, H.324 shall also mean to include H.324 Annex H. + +Originally designed for V.34 modems, H.324 now supports ISDN and wireless networks. Therefore, it is well suited as a basis for 3GPP multimedia codecs. Relevant to wireless networks, H.324 describes the overall system architecture and introduces control (H.245), mux (H.223), video (H.261 and H.263), text (T.140), and audio (G.723.1). + +Annex A provides a short overview of H.324 and multimedia codecs. + +# --- 6 Modifications to H.324 (3GPP TS 26.111) + +To enable cost-effective, high-quality H.324 terminals for 3GPP networks, some modifications were made to H.324. These modifications are described in 3GPP TS 26.111. Terminals adhering to this specification are herewith known as 3G-324M terminals. 3G-324M terminals shall conform to 3GPP TS 26.111. + +# --- 7 Call set-up requirements + +H.324 does not describe call set-up procedures for 3GPP networks. These are described in 3GPP TS 24.008, 27.001, 29.007, 23.108 and shall be used for 3G-324M terminals. + +# --- 8 Terminal implementor's guide (3GPP TR 26.911) + +A successful 3G-324M terminal will have to function well at bandwidths as low as 32 KBPS and in potentially high error rate environments. 3G-324M contains many options that may be employed by an implementor. To help choose which options and combinations of options are useful, an implementor's guide is provided in 3GPP TR 26.911. + +# --- Annex A (informative): Background information + +The section is intended for informational purposes only. This is not an integral part of this specification. Each section below relates to the functional components in figure 1. + +--- + +## A.1 Video I/O Equipment + +For a video telephone this would most likely consist of a video camera and display monitor. Other possible input sources could be a VCR or disk drive. While most applicable I/O equipment relies on a standard format for the video signal or bit stream, this format is likely to differ from that mandated by the video codec. In such cases, circuitry or software is used to transcode between the two formats. + +--- + +## A.2 Video Codec + +ITU-R 601 (NTSC or PAL) is a typical video input signal and represents a bit stream of 20.7 Mbyte/s for the actual image (excluding blanking intervals). The first order of compression occurs by reducing the resolution of the input signal.1 For example, CIF resolution at 30 fps produces a bit stream 4.6 Mbyte/s. Additional savings occur by dropping frames. In a videoconference, where motion is relatively slow, 10 fps is considered adequate. Thus, the original signal of 20.7 Mbytes/s could be reduced to 1.5 Mbyte/s with just these techniques. However, this is still 188 times greater than can be transmitted on, for example, a 64 KBPS channel. Substantial compression is still required, especially considering that framing, control, and audio would as well require a portion of the available bandwidth. + +To achieve the degree of compression required for video telephony, all of the video codecs that can currently be employed in a 3GPP multimedia codec use a combination of spatial and temporal redundancy reduction to reduce the bandwidth required by the video media stream. Spatial redundancy can be reduced by converting the input signal from the time domain to the frequency domain using a DCT. This produces a DC value and other coefficients, where most of the scene energy is concentrated in the coefficients corresponding to the lower frequencies. Next, a coarse quantizer is applied (which, in this domain, has little effect on image quality). This results in many of the coefficients being encoded to 0. The significant coefficients are encoded to a much smaller range of values. The coefficients are then reordered so that, typically, the larger magnitude values will occur first followed by 0 value coefficients. Finally, the coefficients are replaced with a count of the number of zero value coefficients followed by the value of a nonzero coefficient. This combination is translated into a VLC. Applying this type of compression to the entire video frame produces an intra frame. + +Despite the efficiency of intra coding, significantly more compression is required. In addition to removing spatial redundancy, all video codecs apply temporal reduction as well. This is achieved by comparing the current frame to the previous and estimating the set of vectors which when applied to their respective areas of the scene would create the new, current frame based on the old, previous frame. The match is usually not perfect, so an error component is transmitted as well. The error component is also transformed to the frequency domain, so the same compression efficiency achieved in the intra frame is achieved here as well — enhanced by the fact the range of error coefficients is less than intra coefficients. Since generally only a few areas of a scene change from frame to frame, high compression can be achieved by sending a series of inter frames. If the error component for a particular block is too large, it can be encoded as an intra block. + +Since, by their nature, VLCs are not fixed length, a single bit error can make it impossible to decode an entire frame. Unfortunately, each inter coded frame relies on its previous frame to be decoded. Thus, a single bit error can destroy the entire remaining bit stream. Video codecs have various ways of handling errors. The simplest is to use error detection to determine if a frame contains an error. The transmitter is then signalled that an error occurred. It then sends an intra coded frame, which does not depend on any previous frames. This approach consumes considerable bandwidth and is only practical for very low error networks. Other, more sophisticated schemes are available using the video codecs available to 3G-324M terminals. + +--- + +1 Note that ITU-R 601 represents 16 bit precision colour, whereas true colour is usually considered to require 24 bit precision. Also, the spatial resolution of ITU-R 601 is substantially less than can be achieved with normal human vision. + +### A.2.1 H.261 + +H.261 supports CIF and QCIF images as input. It provides good video quality at 64 kbit/s or higher. It uses BCH codes for Forward Error Correction (FEC). However, this is not recommended for H.324. + +### A.2.2 H.263 + +H.263 is an extension of H.261. It allows sub-QCIF, 4CIF and 16CIF as additional input formats. H.263, in its original version, provides four annexes that describe optional modes for enhanced coding. + +- Advanced prediction mode (Annex F) provides half-pel motion estimation, median-based motion vector prediction, 4 motion vectors per macroblock (one per block), and overlapped block motion compensation +- Unrestricted motion vectors (Annex D) work in conjunction with advanced prediction mode and allow motion vectors to point outside the picture area +- Arithmetic coding (Annex E) can be used instead of variable length coding +- PB-frames (Annex G) allow bi-directional prediction similar to MPEG + +Other significant differences exist, but require a level of detail to explain that renders them outside the scope of this document. + +A second version of H.263 (known as H.263+) adds annexes I through T, some of which address error prone environments and are therefore of special interest to 3GPP multimedia codecs. + +### A.2.3 MPEG-4 + +MPEG-4 Visual (ISO/IEC 14496-2) is a generic video codec. One of its target areas is mobile communications. Error resiliency and high efficiency make this codec particularly well suited for 3G-324M. + +MPEG-4 Visual is organised into Profiles. Within a Profile, various Levels are defined. Profiles define subsets of tool sets. Levels are related to computational complexity. Among these Profiles, Simple Visual Profile provides error resilience (through data partitioning, RVLC, resynchronization marker and header extension code) and low complexity. + +MPEG-4 allows various input formats, including general formats such as QCIF and CIF. It is also baseline compatible with H.263. + +## --- A.3 Audio I/O Codec + +Generally, a video telephone would require a handset, headset, or microphone and speaker. Often, integrated circuits are employed that convert the typically analogue input signal to a PCM format bit stream (ADC) and convert PCM to an analogue signal for acoustic output (DAC). This is helpful since many speech codecs use PCM for input and output. Video telephones often use a separate microphone and speaker. This allows the user to be seen without a handset or headset. However, if this is so, AEC will be required. + +## --- A.4 Speech Codec + +### A.4.1 3GPP AMR + +The AMR codec uses eight source codecs with bit-rates of 12.2, 10.2, 7.95, 7.40, 6.70, 5.90, 5.15 and 4.75 kbit/s. The coder operates on speech frames of 20 ms corresponding to 160 samples at the sampling frequency of 8000 sample/s. It performs the mapping from input blocks of 160 speech samples in 13-bit uniform PCM format to encoded blocks of 95, 103, 118, 134, 148, 159, 204, and 244 bits and from encoded blocks of 95, 103, 118, 134, 148, 159, 204, and 244 bits to output blocks of 160 reconstructed speech samples. The coding scheme for the multi-rate coding modes is the so-called Algebraic Code Excited Linear Prediction Coder (ACELP). The multi-rate ACELP coder is referred to as MR-ACELP. At each 160 speech samples, the speech signal is analysed to extract the parameters of the CELP model (LP filter coefficients, adaptive and fixed codebooks' indices and gains). These parameters are encoded and transmitted. At the decoder, these parameters are decoded and speech is synthesised by filtering the reconstructed excitation signal through the LP synthesis filter. + +The adaptive multi-rate speech codec is described in a bit-exact arithmetic in form of a fixed-point ANSI-C code to allow for easy type approval as well as general testing purposes of the adaptive multi-rate speech codec. + +The DTX mechanism includes a Voice Activity Detector (VAD) on the TX side; evaluation of the background acoustic noise on the TX side, in order to transmit characteristic parameters to the RX side; and generation of comfort noise on the RX side during periods where the radio transmission is turned off. + +The AMR specification contains error concealment. The purpose of frame substitution is to conceal the effect of lost AMR speech frames. The purpose of muting the output in the case of several lost frames is to indicate the breakdown of the channel to the user and to avoid generating possible annoying sounds as a result from the frame substitution procedure. + +### A.4.2 G.723.1 + +G.723.1 can be used for compressing the speech or other audio signal component of multimedia services at a very low bitrate as part of H.324. This coder has two bit-rates associated with it, 5.3 and 6.3 kbit/s. The higher bitrate has greater quality. The lower bit-rate gives good quality and provides system designers with additional flexibility. Both rates are a mandatory part of the encoder and decoder. It is possible to switch between the two rates at any frame boundary. An option for variable rate operation using discontinuous transmission and noise fill during non-speech intervals is also possible using a series of silence frames or a single silence frame followed by no frames until speech is detected. + +G.723.1 encodes speech or other audio signals in frames using linear predictive analysis-by-synthesis coding. The excitation signal for the high rate coder is Multipulse Maximum Likelihood Quantization (MP-MLQ) and for the low rate coder is Algebraic-Code-Excited Linear-Prediction (ACELP). The frame size is 30 ms and there is an additional look ahead of 7.5 msec,. This coder is designed to operate with a digital signal obtained by first performing telephone bandwidth filtering (ITU-T Recommendation G.712) of the analogue input, then sampling at 8000 Hz and then converting to 16-bit linear PCM for the input to the encoder. The output of the decoder is converted back to analogue by similar means. + +G.723.1 has been designed to be robust for indicated frame erasures. An error concealment strategy for frame erasures has been included in the decoder. However, this strategy must be triggered by an external indication that the bit stream for the current frame has been erased. This can be achieved in H.324 using the AL2 Error Indication (EI) flag and the optional AL2 Sequence Number (SN). Because the coder was designed for burst errors, there is no error correction mechanism provided for random bit errors. If a frame erasure has occurred, the decoder switches from regular decoding to frame erasure concealment mode. + +G.723.1 contains three annexes. Annex A describes the silence compression system designed for the G.723.1 speech coder (mentioned above). Annex B describes an alternative implementation of G.723.1 contained in floating point C source code. Annex C specifies a channel coding scheme which can be used with the triple rate speech codec G.723.1. The channel codec is scalable in bit-rate and is designed for mobile multimedia applications as a part of the overall H.324 family of standards. + +## --- A.5 User Data Applications + +### A.5.1 Data conferencing – T.120 + +An example of a User Data Application is T.120. This protocol allows multipoint data conferencing that includes data and image transferral. Other functions, such as shared whiteboards and applications, are possible. + +### A.5.2 Text conversation – T.140 + +The real time text conversation application, is supported by the presentation protocol ITU-T T.140 [19]. The Global Text Telephony feature is implemented in the CS Multimedia environment by applying T.140, as specified in H.324. The text stream may be opened simultaneously with voice, video and other data applications. Text-only sessions are also possible. Further requirements applicable to the Global Text Telephony feature are specified in TS 22.226 [20]. + +The data protocol for T.140 is specified in H.324 to be AL1. + +## --- A.6 Data Protocols + +Various data protocols can be supported. These always support data applications (see A.5 User Data Applications). A specific protocol or set of protocols is often stipulated by the data application. Each protocol provides varying degrees of error detection and/or correction. + +## --- A.7 System Control + +In general, system control constitutes the overall state machine for the terminal. It usually has to be aware of when a connection has been established. At that point it can begin H.245 procedures such as master/slave determination, capabilities exchange, and opening logical channels. Upon call termination, either initiated at the near or far ends, system control generally initiates H.245 end session procedures. + +## --- A.8 Call Set-up + +All out-of-band network signalling for the purpose of call control is handled by call set-up, which is usually implemented as a state machine. This includes initiating, answering, and tearing down calls. + +## --- A.9 H.245 + +H.245 specifies the syntax and semantics for in-band, terminal-to-terminal control messages and the procedures for their use. Most importantly, H.245 is used for master/slave determination, capabilities exchange, H.223 mux table transmission, and opening and closing logical channels. There is also a large array of general control and indication messages. H.245 addresses a wide range of terminals and applications. Therefore, only a subset of the messages listed in H.245 pertain to 3G-324M terminals. Messages fall into one of four categories: Request (requires a Response), Response (in response to a Request), Command (requires an action), and Indication (informative only). + +H.245 messages are carried on a single logical channel within the H.223 mux. This channel is labelled LC 0 and is considered to be open upon establishing digital communications end-to-end and survives until digital communication is terminated. Due to the characteristics of the H.223 mux, bandwidth for H.245 messages is allocated on an as-needed basis. Since most H.245 traffic occurs at the beginning and end of the session, this conserves much needed bandwidth for video and audio. Error control is not provided for within H.245 and is specified elsewhere. + +## --- A.10 H.223 + +H.223 describes the multiplexing protocol used between H.324 terminals. It is packet oriented and each packet can contain a subset of a maximum of 65536 LCs. Each LC represents a single media, information, or control channel. The H.223 protocol is split into two layers, the lowest being the Multiplex Layer. + +The Multiplex Layer exchanges data with the end terminal via MUX-PDUs. Multiplex table entries, of which there are 16 (and can be changed during a session), describe which octets from within the PDU are allocated to which logical channels. The multiplex table entry employed for a particular PDU is indicated by the 4 bit MC field in the MUX-PDU header. MUX-PDUs contain an integer number of octets. Errors within the MUX-PDU header are controlled using the HEC field in the MUX-PDU header. H.324 terminals utilising the V.34 transmission protocol frame MUX-PDUs with HDLC. Bit stuffing is used for data transparency in this case. + +Above the Multiplex Layer is the Adaptation Layer, of which there are three different types. + +- 1) AL1 is designed primarily for control information and data protocols. It can be either framed or unframed and does not provide any error control. +- 2) AL2 is designed primarily for the transfer of digital audio. AL2 PDUs contain 1 octet for an 8-bit CRC and an optional octet for a sequence number. +- 3) AL3 is designed primarily for the transfer of digital video. AL3 PDUs contain 2 octets for a 16-bit CRC. There is also optionally 1 or 2 octets for control. AL3 also allows limited retransmission. + +For purposes of video telecommunications over wireless networks, four annexes to H.223 have been created. These create four levels of error detection and error correction. + +### A.10.1 Level 0 + +Level 0 applies to H.223 as described above. + +### A.10.2 Level 1 + +Level 1, described in Annex A, replaces HDLC framing with 1 or 2 16 bit flags. Unlike HDLC, Level 1 does not guarantee data transparency. However, if the MUX-PDU header is constructed in such a way as to make emulating the Level 1 framing flags impossible, data transparency can be achieved by correctly decoding the MUX-PDU. Should there be an error in the MUX-PDU header, resynchronization techniques will have to be applied. + +### A.10.3 Level 2 + +Level 2, described in Annex B, uses the same framing as Level 1, but utilises a 3 octet header. This header starts with a 4 bit MC, which is the same as in Level 0. This is followed by an 8-bit MPL-field, with a range of values 0 – 254. Lastly, a 12 bit extended Golay code is used for parity bits. The PM in Level 2 is signalled through the polarity of the MUX-PDU flag. If the output of the correlator is greater than or equal to CT, the PM is 0. If it is less than or equal to -CT, the PM equals 1. After the parity bits, there can be an optional MUX-PDU header for the previous (corrupted) MUX-PDU. This 1 octet field uses the format described in Level 0. Level 2 also offers enhanced packet resynchronization. + +### A.10.4 Level 3 + +Level 3, described in Annexes C and D, provides error correction capabilities at the mux level. + +# --- Annex B (informative): Bibliography + +(void) + +# Annex C (informative): Change history + +| Change history | | | | | | | | +|----------------|-------|-----------|-----|-----|-------------------------------------------------------------------|--------|--------| +| Date | TSG # | TSG Doc. | CR | Rev | Subject/Comment | Old | New | +| 06-1999 | 04 | | | | Approved at TSG-SA#4 | | 3.0.0 | +| 09-2000 | 09 | SP-000395 | 001 | | CS Multimedia Codec specification for real time text conversation | 3.0.1 | 4.0.0 | +| 03-2001 | 011 | SP-010105 | 002 | 1 | Support of mobile multi-link operation in 3G-324M | 4.0.0 | 4.1.0 | +| 03-2001 | 011 | SP-010105 | 004 | 1 | Correction of incorrect reference | 4.0.0 | 4.1.0 | +| 06-2002 | 016 | | | | Version for Release 5 | 4.1.0 | 5.0.0 | +| 12-2004 | 026 | | | | Version for Release 6 | 5.0.0 | 6.0.0 | +| 06-2007 | 036 | | | | Version for Release 7 | 6.0.0 | 7.0.0 | +| 12-2008 | 042 | | | | Version for Release 8 | 7.0.0 | 8.0.0 | +| 12-2009 | 046 | | | | Version for Release 9 | 8.0.0 | 9.0.0 | +| 03-2011 | 051 | | | | Version for Release 10 | 9.0.0 | 10.0.0 | +| 09-2012 | 057 | | | | Version for Release 11 | 10.0.0 | 11.0.0 | \ No newline at end of file diff --git a/marked/Rel-11/26_series/26111/raw.md b/marked/Rel-11/26_series/26111/raw.md new file mode 100644 index 0000000000000000000000000000000000000000..2a04af4f85c10e9b85dc3122fb90a890f3c162ad --- /dev/null +++ b/marked/Rel-11/26_series/26111/raw.md @@ -0,0 +1,324 @@ + + + + + + +# --- Contents + +| | | +|-------------------------------------------------------|----| +| Foreword ..... | 4 | +| Introduction ..... | 4 | +| 1 Scope..... | 5 | +| 2 References..... | 5 | +| 3 Definitions and abbreviations ..... | 6 | +| 3.1 Definitions..... | 6 | +| 3.2 Abbreviations ..... | 6 | +| 4 General ..... | 6 | +| 5 Document structure ..... | 6 | +| 6 Functional requirements..... | 7 | +| 6.1 Required elements ..... | 7 | +| 6.2 Information streams..... | 7 | +| 6.3 Modem ..... | 7 | +| 6.4 Multiplex ..... | 7 | +| 6.5 Control channel ..... | 7 | +| 6.6 Video channels ..... | 7 | +| 6.6.1 MPEG-4 interface to multiplex ..... | 10 | +| 6.6.2 H.264 (MPEG-4 AVC) interface to multiplex ..... | 10 | +| 6.7 Audio channels..... | 11 | +| 6.8 Data channels ..... | 11 | +| 7 Terminal procedures ..... | 11 | +| 8 Optional enhancements ..... | 11 | +| 9 Interoperation with other terminals..... | 11 | +| 10 Multipoint considerations ..... | 11 | +| 11 Maintenance..... | 12 | +| Annex A (informative): Change History..... | 13 | + +# --- Foreword + +This Technical Specification (TS) has been produced by the 3rd Generation Partnership Project (3GPP). + +The contents of the present document are subject to continuing work within the TSG and may change following formal TSG approval. Should the TSG modify the contents of the present document, it will be re-released by the TSG with an identifying change of release date and an increase in version number as follows: + +Version x.y.z + +where: + +- x the first digit: + - 1 presented to TSG for information; + - 2 presented to TSG for approval; + - 3 or greater indicates TSG approved document under change control. +- y the second digit is incremented for all changes of substance, i.e. technical enhancements, corrections, updates, etc. +- z the third digit is incremented when editorial only changes have been incorporated in the document. + +# --- Introduction + +In the present document is described additions, deletions, and changes made to ITU-T Recommendation H.324 [10] with annex C for the purpose of using that recommendation as a basis for the technical specification for circuit switched multimedia service in 3GPP networks. The present document does not address call setup procedures, but references to the specifications which cover call setup are found in 3GPP TS 26.110 [11]. + +# --- 1 Scope + +In ITU-T Recommendation H.324 [10] with annex C describes a generic multimedia codec for use in error-prone, wireless networks. The scope of the present document are the changes, deletions, and additions to those texts necessary to fully specify a multimedia codec for use in 3GPP networks. Note that this implicitly excludes the network interface and call setup procedures. Also excluded are any general introductions to the system components. + +# --- 2 References + +The following documents contain provisions which, through reference in this text, constitute provisions of the present document. + +- References are either specific (identified by date of publication, edition number, version number, etc.) or non-specific. + - For a specific reference, subsequent revisions do not apply. + - For a non-specific reference, the latest version applies. In the case of a reference to a 3GPP document (including a GSM document), a non-specific reference implicitly refers to the latest version of that document *in the same Release as the present document*. +- [1] ITU-T Recommendation H.223: "Multiplexing protocol for low bit rate multimedia communication". +- [2] ITU-T Recommendation H.223 - Annex A: "Multiplexing protocol for low bit rate multimedia mobile communication over low error-prone channels". +- [3] ITU-T Recommendation H.223 - Annex B: "Multiplexing protocol for low bit rate multimedia mobile communication over moderate error-prone channels". +- [4] ITU-T Recommendation H.223 - Annex C: "Multiplexing protocol for low bit rate multimedia mobile communication over highly error-prone channels". +- [5] ITU-T Recommendation H.223 - Annex D: "Optional multiplexing protocol for low bit rate multimedia mobile communication over highly error-prone channels". +- [6] ITU-T Recommendation H.245: "Control protocol for multimedia communication". +- [7] ITU-T Recommendation G.723.1: "Dual rate speech coder for multimedia communication transmitting at 5,3 and 6,3 kbit/s". +- [8] ITU-T Recommendation H.263: "Video coding for low bitrate communication". +- [9] ITU-T Recommendation H.261: "Video CODEC for audiovisual services at px64 kbit/s". +- [10] ITU-T Recommendation H.324: "Terminal for low bitrate multimedia communication". +- [11] 3GPP TS 26.110: "Codec for Circuit Switched Multimedia Telephony Service; General description". +- [12] 3GPP TR 26.911: "Codec for circuit switched multimedia telephony service; terminal implementor's Guide (Release 4)". +- [13] ITU-T Recommendation X.691: "Information Technology - ASN.1 Encoding Rules - Specification of Packed Encoding Rules (PER)". +- [14] ISO/IEC 14496-2: "Information technology - Coding of audio-visual objects - Part 2: Visual". +- [15] 3GPP TS 26.071: "General description". +- [16] 3GPP TS 26.090: "Transcoding functions". +- [17] 3GPP TS 26.073: "Adaptive Multi-Rate (AMR); ANSI C source code". + +- [18] 3GPP TS 26.171: "AMR Wideband Speech codec; General Description". +- [19] ITU-T Recommendation H.264 (2003): "Advanced video coding for generic audiovisual services" | ISO/IEC 14496-10:2003: "Information technology – Coding of audio-visual objects – Part 10: Advanced Video Coding". +- [20] ITU-T Recommendation H.241 (2003): "Extended video procedures and control signals for H.300 series terminals". +- [21] ITU-T Recommendation G.722.2 Annex F (2002): "AMR-WB usage in H.245". +- [22] 3GPP TS 26.201 : "Adaptive Multi-Rate – Wideband (AMR-WB) speech codec ; Frame Structure." + +# --- 3 Definitions and abbreviations + +## 3.1 Definitions + +For the purposes of the present document, the following terms and definitions apply: + +**H.324:** ITU-T H.324 [10] with annex C + +**3G-324M terminal:** based on ITU-T H.324 [10] recommendation modified by 3GPP for purposes of 3GPP circuit switched network based video telephony + +## 3.2 Abbreviations + +For the purposes of the present document, the following abbreviations apply: + +| | | +|--------|------------------------------| +| AMR | Adaptive Multi-Rate | +| AMR-WB | AMR Wide-Band | +| AVC | Advanced Video Codec | +| FLC | Fixed Length Code | +| RVLC | Reverse Variable Length Code | +| DP | Data Partitioning | +| RM | Resynchronization Marker | +| MCU | Multipoint Control Unit | + +# --- 4 General + +The present document contains any deviations to ITU-T H.324 [10] required for the specification of 3G-324M Terminals. + +# --- 5 Document structure + +The structure of H.324 [10] is followed in the present document. Where there are no differences in a specific section, that section is skipped. Where differences are minor, only the differences are described. Where major differences exist, the section is rewritten in the present document. It is important to note that for wireless terminals, Annex C of H.324 [10] supersedes respective portions of the main body of H.324 [10] For the present document, these modifications are treated as if they are part of the main body of H.324 [10] Therefore, a reader must keep in mind both the main body and Annex C of H.324 [10] when reading the present document. + +# --- 6 Functional requirements + +## 6.1 Required elements + +3G-324M implementations are not required to have each functional element except a wireless interface, H.223 [1] with Annex A and B multiplex, and H.245 [6] version 3 or later versions for system control protocol. + +3G-324M terminals offering audio communication shall support the AMR audio codec. Support for G.723.1 [7] is not mandatory, but recommended. + +3G-324M terminals offering video communication shall support the H.263 [8] video codec. Support for MPEG-4 simple profile and H.261 [9] is optional. + +3G-324M terminals shall support H.223 [1] with annex A and annex B. + +3G-324M terminals shall support at least 32 kbit/s minimum bit rate at the mux to wireless network interface. + +## 6.2 Information streams + +V.25ter discussion does not apply. + +## 6.3 Modem + +Does not apply. + +## 6.4 Multiplex + +3G-324M terminals shall support H.223 [1] with annex A and annex B. All other aspects shall follow H.324 [10] with annex C. H.223 [1] Annex C and D are optional. + +## 6.5 Control channel + +No differences with H.324 [10]. + +Should it not be possible to signal an element of the 3G-324M terminal using a published version of H.245 [6], a procedure will be defined here. + +## 6.6 Video channels + +Support for H.261 [9] is optional. + +Support for MPEG-4 Visual is optional. When supported, MPEG-4 Visual codecs shall support Simple Profile @ Level 0. The FLC code 0000 1000 in Table G-1 – "FLC table for profile\_and\_level\_indication" in ISO/IEC 14496-2 [14] is assigned to it. Additional information can be found in [14]. + +MPEG-4 Visual Simple Profile @ level 0 provides error concealment as part of the simple profile through Data Partitioning (DP), Reversible Variable Length Coding (RVLC), Resynchronization Marker (RM) and header extension code. MPEG-4 Visual is baseline compatible with H.263 [8]. + +When opening a logical channel for MPEG-4 Visual, configuration information (Visual Object Sequence Header, Visual Object Header, and Video Object Layer Header) shall be sent in the decoderConfigurationInformation parameter. The same information shall also be sent in the MPEG-4 video bitstream. If the operational mode of MPEG-4 Visual encoder needs to be changed, the existing MPEG-4 video logical channel shall be closed and H.245 [6] procedures for opening a new MPEG-4 video logical channel shall be started. The new operational mode shall be indicated in the parameters of the new logical channel. + +Support for H.264 (MPEG-4 AVC) [19] is optional. When supported, H.264 codecs shall support Baseline level 1, without requirements on output timing conformance (Annex C of [19]). + +Support for H.264 [19] shall be signalled according to H.241 chapter 8 "Capability Exchange signalling" [20]. + +When opening a logical channel for H.264 [19], initial sequence parameter set(s) and picture parameter set(s) should be sent in a H.264 DecoderConfigurationInformation (DCI) defined in Table 1 below, amending H.241 parameters [20]. Additionally, decoder capabilities may be sent in a H.264 AcceptRedundantSlices and a H.264 ProfileIOP defined in Table 2 and 3 below, amending H.241 parameters [20]. + +NOTE: The H.264 DCI parameter can also be used when either party signals a H.245 [6] MasterSlaveDetermination terminalType parameter greater than 128, such as e.g. a Multipoint Conference Unit (MCU). + +A sequence parameter set or a picture parameter set with a particular value of seq\_parameter\_set\_id or pic\_parameter\_set\_id, respectively, sent in the H.264 [19] DCI shall be identical to the earliest occurrence of the sequence parameter set or picture parameter set with the same value of seq\_parameter\_set\_id or pic\_parameter\_set\_id, respectively, sent in the H.264 bitstream. + +If DCI was used when a H.264 [19] logical channel was opened and H.264 sequence parameter sets need to be changed or new sets need to be added during the session, the existing H.264 logical channel shall be closed and H.245 [6] procedures for opening a new H.264 logical channel shall be started, in which sequence parameter set(s) and picture parameter set(s) shall be sent in a DCI. Each sequence parameter set of H.264 [19] shall contain the vui\_parameters syntax structure including the num\_reorder\_frames syntax element set equal to 0. + +If H.264 picture parameter sets need to be changed or new sets need to be added during a session, it may be done either by opening a new logical channel using the same procedure as described above or within the current channel, by including picture parameter set NAL units directly in the bitstream. + +**Table 1 / TS 26.111 – H.264 Capability Parameter – DecoderConfigurationInformation (DCI)** + +| | | +|----------------------------|-------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------| +| Parameter name | DecoderConfigurationInformation | +| Parameter description | This is a nonCollapsing GenericParameter.

DecoderConfigurationInformation indicates how to configure the decoder for a particular H.264 video sequence [19]. It contains sequence parameter set NAL units, picture parameter set NAL units, or both, using the byte stream format specified in Annex B/H.264, separating NAL units with a start code. The use of a start code before the first parameter set NAL unit is optional. | +| Parameter identifier value | 43 | +| Parameter status | Optional. Shall not be present for Capability Exchange and Mode Request. May be present exactly once for Logical Channel Signalling. | +| Parameter type | OctetString | +| Supersedes | - | + +A decoder may indicate its' capability to make use of H.264 redundant slices by the following parameter. + +**Table 2 / TS 26.111 – H.264 Capability Parameter – AcceptRedundantSlices** + +| | | +|----------------------------|------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------| +| Parameter name | AcceptRedundantSlices | +| Parameter description |

This is a collapsing GenericParameter.

AcceptRedundantSlices indicates the capability to use H.264 redundant slices and corresponds to the MIME video/H264 parameter “redundant-pic-cap”.

When False or when the parameter is not present, it indicates that the receiver makes no attempt to use redundant coded pictures to correct incorrectly decoded primary coded pictures and a sender should not send redundant slices.

When True, it indicates that the receiver is capable of decoding any such redundant slice that covers a corrupted area in a primary decoded picture (at least partly), and a sender may send redundant slices.

When using a H.264 profile (or subset of a profile as indicated by the H.264 ProfileIOP parameter defined in Table 3) and level that disallows the use of redundant slices, this parameter shall be ignored.

| +| Parameter identifier value | 44 | +| Parameter status | Optional. May be present exactly once for Capability Exchange Signalling. | +| Parameter type | Logical | +| Supersedes | - | + +NOTE: An encoder should only code redundant slices if it knows that the far-end decoder makes use of this feature. Encoders should also pay attention to potential implications on end-to-end delay. + +A decoder may indicate additional limitations that only the common subset of the algorithmic features and limitations of the Baseline level 1 are supported by the following parameter. + +**Table 3 / TS 26.111 – H.264 Capability Parameter – ProfileIOP** + +| | | +|----------------------------|-----------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------| +| Parameter name | ProfileIOP | +| Parameter description |

ProfileIOP indicates that the capability to decode H.264 streams is limited to a common subset of the algorithmic features included in the indicated profile and level.

This parameter is a Boolean array.

bit 1 (value 128) is constraint_set0_flag.

bit 2 (value 64) is constraint_set1_flag.

bit 3 (value 32) is constraint_set2_flag.

All other bits are reserved, shall be set to 0, and shall be ignored by receivers.

constraint_set0_flag, constraint_set1_flag and constraint_set2_flag are defined in [18].

As an example, a receiver indicating decoding support of the intersection of the baseline and main profile will signal value 11000000 (constraint_set0_flag = 1, constraint_set1_flag = 1, constraint_set2_flag = 0).

| +| Parameter identifier value | 46 | +| Parameter status | Optional. May be present exactly once for Capability Exchange Signalling. | +| Parameter type | BooleanArray | +| Supersedes | - | + +A terminal supporting H.264 encoding should respond to all `videoFastUpdatePicture` commands received via the H.245 control channel. If an H.264 encoder responds to `videoFastUpdatePicture`, it shall use the procedure specified in subclause 6.2.2 of H.241. + +A terminal supporting H.264 shall start decoding immediately when it receives data (even if the stream does not start with an IDR access unit) or alternatively no later than it receives the next IDR access unit or the next recovery point SEI message, whichever is earlier in decoding order. The decoding process for a stream not starting with an IDR access unit shall be the same as for a valid H.264 bitstream. However, the client shall be aware that such a stream may contain references to picture not available in the decoded picture buffer. The display behaviour of the client is out of scope of this specification. + +NOTE: Terminals may use full-frame freeze and full-frame freeze release SEI messages of H.264 to control the display process. + +### 6.6.1 MPEG-4 interface to multiplex + +As H.263 [8] encoders align picture start codes with the start of an AL-SDU, the same concept applies to MPEG-4 encoders. The following are the requirements of the MPEG-4 interface to the H.223 [1] multiplex. + +- Each 3G-324M MPEG-4 encoder shall align each `visual_object_sequence_start_code` with the start of an AL-SDU. +- Each 3G-324M MPEG-4 encoder shall align each `group_of_vop_start_code` (the beginning of a GOV field) with the start of an AL-SDU unless the GOV field immediately follows configuration information. +- Each 3G-324M MPEG-4 encoder shall align each `vop_start_code` with the start of an AL-SDU unless the `vop_start_code` immediately follows configuration information or a GOV field. + +In these requirements, GOV stands for `Group_of_VideoObjectPlane()` and Configuration information consists of Visual Object Sequence Header, Visual Object Header, and Video Object Layer Header. + +### 6.6.2 H.264 (MPEG-4 AVC) interface to multiplex + +Shall conform to the byte stream format according to H.241 chapter 7.1.5 "Transport of H.264 streams in H.324 systems" [20]. + +More strict alignment of AL-SDU and NAL units may optionally be used. To signal capability for and use of this mode, the generic parameter described in Table 3 shall be used, amending the H.264 Generic Capability in H.241 [20]. + +**Table 3 / TS 26.111 – H.264 Capability Parameter – NalAlignedMode** + +| | | +|----------------------------|----------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------| +| Parameter name | NalAlignedMode | +| Parameter description | This is a collapsing GenericParameter.

NalAlignedMode indicates that every AL-SDU carrying H.264 shall contain an integer number of NAL units and that the start of the AL-SDU shall be aligned with the start of a NAL. Individual NAL units within the AL-SDU shall be separated by start codes as described in Annex B/H.264. The use of a start code before the first NAL in an AL-SDU is optional. | +| Parameter identifier value | 45 | +| Parameter status | Optional. May be present exactly once for Capability Exchange, Logical Channel, or Mode Request Signalling. | +| Parameter type | Logical | +| Supersedes | - | + +## 6.7 Audio channels + +AMR is the mandatory speech codec. Support for G.723.1 [7] is not mandatory, but recommended. Support for AMR-WB [18] is also recommended. + +When AMR-WB is supported, the AMR-WB speech data shall be carried in IF2 frame format as defined in Annex A of 3GPP TS 26.201 [22], and the signalling for AMR-WB shall be according to G.722.2 Annex F [21] with the following additional restrictions: + +- octetAlign shall be present +- The following parameters are not compatible with IF2 frame format, and so shall not be used: + - crc + - robustSorting + - interleaving + +When both the receiving and transmitting terminals support multiple codecs in common, the use of AMR and AMR-WB is preferred: + +- If both the receiving and transmitting terminals support AMR and other codecs (e.g. G.723.1) but not AMR-WB, then AMR shall be used. +- If both the receiving and transmitting terminals support AMR and other codecs including AMR-WB, either AMR or AMR-WB shall be used. + +Asymmetric configurations with one codec in one direction and another one in the other direction (e.g. AMR in one direction and AMR-WB in the other direction) are allowed, if supported by both terminals. + +This applies to connections without an Multipoint Control Unit (MCU). + +## 6.8 Data channels + +No differences with H.324 [10]. + +# --- 7 Terminal procedures + +See 3GPP TS 26.110 [11]. + +# --- 8 Optional enhancements + +No differences with H.324 [10]. + +# --- 9 Interoperation with other terminals + +For further study. + +# --- 10 Multipoint considerations + +For further study. + +# --- 11 Maintenance + +No differences with H.324 [10]. + +# Annex A (informative): Change History + +| History | | | | | | | | +|---------|-----------|--------|------------------|--------------|------|-----|------------------------------------------------------------------| +| TSG_# | TSG_DOC | SPEC | VERS_CURRE
NT | VERS_
NEW | CR | REV | SUBJECT | +| SP-05 | SP-99359 | 26.111 | 3.0.1 | 3.0.2 | 001 | | Changes to editorial notes. | +| SP-06 | SP-99434 | 26.111 | 3.0.2 | 3.1.0 | 002 | 2 | Specification of coding parameters for MPEG-4 video codec | +| SP-06 | SP-99514 | 26.111 | 3.0.2 | 3.1.0 | 003 | | Transmission of MPEG-4 configuration information in 3G-324M | +| SP-08 | SP-00263 | 26.111 | 3.1.0 | 3.2.0 | 004 | | Changes to editorial notes | +| SP-09 | SP-000396 | 26.111 | 3.2.0 | 3.3.0 | 006 | | MPEG-4 interface to multiplex | +| SP-10 | SP-000653 | 26.111 | 3.3.0 | 3.4.0 | 005 | 1 | MPEG4 visual simple profile @ level 0 | +| SP-11 | | | | 4.0.0 | | | Version for Release 4 | +| SP-16 | | | 4.0.0 | 5.0.0 | | | Version for Release 5 | +| SP-20 | SP-030215 | 26.111 | 5.0.0 | 5.1.0 | 009 | 1 | Removal of Reference to TS 26.112 | +| | | | | | | | | +| SP-25 | SP-040659 | 26.111 | 5.1.0 | 6.0.0 | 010 | 3 | 3G-324M Improvements | +| SP-25 | SP-040648 | 26.111 | 5.1.0 | 6.0.0 | 011 | 1 | 3G-324M Improvements: Addition of optional AMR-WB support | +| SP-26 | SP-040842 | 26.111 | 6.0.0 | 6.1.0 | 012 | 1 | Addition of the missing signalling of H.264 decoder capabilities | +| SP-26 | SP-040842 | 26.111 | 6.0.0 | 6.1.0 | 013 | 1 | Reference Corrections | +| SP-36 | | | 6.1.0 | 7.0.0 | | | Version for Release 7 | +| SP-40 | SP-080247 | 26.111 | 7.0.0 | 7.1.0 | 0015 | 1 | Use of AMR-WB in 3G-324M | +| SP-42 | | | 7.1.0 | 8.0.0 | | | Version for Release 8 | +| SP-46 | | | 8.0.0 | 9.0.0 | | | Version for Release 9 | +| SP-51 | | | 9.0.0 | 10.0.0 | | | Version for Release 10 | +| SP-57 | | | 10.0.0 | 11.0.0 | | | Version for Release 11 | \ No newline at end of file diff --git a/marked/Rel-11/26_series/26115/raw.md b/marked/Rel-11/26_series/26115/raw.md new file mode 100644 index 0000000000000000000000000000000000000000..64e994dd131ba53ef7b588c6c123d850228183dc --- /dev/null +++ b/marked/Rel-11/26_series/26115/raw.md @@ -0,0 +1,106 @@ + + + + + + +# --- Contents + +- Foreword ..... 4 +- Introduction ..... 4 +- 1 Scope..... 5 +- 2 References..... 5 +- 3 Abbreviations ..... 5 +- 4 Interfaces..... 5 +- 5 Narrow Band Speech Telephony Network Echo Control ..... 6 +- 5.1 GSTN Network Echo Cancellation ..... 6 +- Annex A (informative): Change history..... 7 + +# --- Foreword + +This Technical Specification has been produced by the 3rd Generation Partnership Project (3GPP). + +The contents of the present document are subject to continuing work within the TSG and may change following formal TSG approval. Should the TSG modify the contents of the present document, it will be re-released by the TSG with an identifying change of release date and an increase in version number as follows: + +Version x.y.z + +where: + +- x the first digit: + - 1 presented to TSG for information; + - 2 presented to TSG for approval; + - 3 or greater indicates TSG approved document under change control. +- y the second digit is incremented for all changes of substance, i.e. technical enhancements, corrections, updates, etc. +- z the third digit is incremented when editorial only changes have been incorporated in the document. + +# --- Introduction + +The present document specifies minimum performance requirements for the transmission planning aspects of 3G speech and multi-media services. + +The objective is to reach a quality as close as possible to ITU-T standards for PSTN circuits. However, due to technical and economic factors, there cannot be full compliance with the general characteristics of international telephone connections and circuits recommended by the ITU-T. + +The performance requirements are specified the main body of the text; the test methods and considerations are described in [tbd]. + +# --- 1 Scope + +The present document specifies minimum performance requirements for the gateway echo control of 3G speech and multi-media services. The present document is applicable to any narrow band speech telephony or multimedia service. + +# --- 2 References + +The following documents contain provisions which, through reference in this text, constitute provisions of the present document. + +- References are either specific (identified by date of publication, edition number, version number, etc.) or non-specific. + - For a specific reference, subsequent revisions do not apply. + - For a non-specific reference, the latest version applies. In the case of a reference to a 3GPP document (including a GSM document), a non-specific reference implicitly refers to the latest version of that document *in the same Release as the present document*. +- [1] ITU-T Recommendation G.114 (2000): "One-way transmission time". +- [2] ITU-T Recommendation G.168 (2000): "Digital network echo cancellers". +- [3] ITU-T Recommendation G.131 (1996): "Control of talker echo". +- [4] ITU-T Recommendation G.703 (1998): "Physical/electrical characteristics of hierarchical digital interfaces". +- [5] ITU-T Recommendation G.711 (1988): "Pulse code modulation (PCM) of voice frequencies". + +# --- 3 Abbreviations + +For the purposes of the present document, the following abbreviations apply: + +| | | +|------|--------------------------------------| +| ADC | Analogue to Digital Converter | +| DAC | Digital to Analogue Converter | +| DTX | Discontinuous Transmission | +| EC | Echo Canceller | +| EEC | Electrical Echo Control | +| EL | Echo Loss | +| ERL | Echo Return Loss | +| ERLE | Echo Return Loss Enhancement | +| PCM | Pulse Code Modulation | +| POI | Point of Interconnection (with PSTN) | +| PSTN | Public Switched Telephone Network | +| TCL | Terminal Coupling Loss | +| TX | Transmission | + +# --- 4 Interfaces + +The POI with the public switched telephone network (PSTN) will generally be at the 2 048 kbits/ level at an interface in accordance with ITU-T Recommendation G.703 [4]/G.704 or STM1 155Mbit/s. At this point, which is considered to have a relative level of 0 dBr, the analogue signals will be represented by 8-bit A-law or $\mu$ -law according to ITU-T Recommendation G.711 [5]. Analogue measurements may be made at this point using a standard send and receive side, as defined in ITU-T Recommendations. + +# --- 5 Narrow Band Speech Telephony Network Echo Control + +## 5.1 GSTN Network Echo Cancellation + +Narrow band speech calls from the 3G mobile system to the public GSTN are terminated on local switch line cards where two to four wire conversion takes place. The hybrid used to carry out this function is never perfect and echo is generated which degrades the speech call quality for the 3G mobile user. To overcome this situation an echo cancellation device should be used at the gateway from the 3G mobile network to the GSTN. This echo control device shall conform to ITU-T G.168 [2]. + +NOTE: Acoustic Echo Control: Narrow band speech calls from the 3G mobile network to the public GSTN involve a high delay. The only echo path that is audible to the GSTN user is the acoustic echo path in the UE. To overcome this echo a Terminal Coupling Loss (TCL) of 46dB should be achieved by the terminal. This provides adequate echo protection for calls up to a delay of 300ms as defined by ITU-T Recommendation G.131 [3]. + +# --- Annex A (informative): Change history + +| Change history | | | | | | | | +|----------------|-------|-----------|----|-----|-------------------------------------------------------|--------|--------| +| Date | TSG # | TSG Doc. | CR | Rev | Subject/Comment | Old | New | +| - | 07 | SP-000020 | - | - | Approved at TSG SA #7 and placed under Change Control | - | 3.0.0 | +| 03-2001 | 11 | - | - | - | Version for Release 4 | 3.0.0 | 4.0.0 | +| 06-2002 | 16 | - | - | - | Version for Release 5 | 4.0.0 | 5.0.0 | +| 12-2004 | 26 | - | - | - | Version for Release 6 | 5.0.0 | 6.0.0 | +| 06-2007 | 36 | - | - | - | Version for Release 7 | 6.0.0 | 7.0.0 | +| 12-2008 | 42 | - | - | - | Version for Release 8 | 7.0.0 | 8.0.0 | +| 12-2009 | 46 | | | | Version for Release 9 | 8.0.0 | 9.0.0 | +| 03-2011 | 51 | | | | Version for Release 10 | 9.0.0 | 10.0.0 | +| 09-2012 | 57 | | | | 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@@ -0,0 +1,3059 @@ + + + + + + +# Contents + +| | | +|----------------------------------------------------------------------------------|----| +| Foreword ..... | 6 | +| Introduction ..... | 6 | +| 1 Scope..... | 7 | +| 2 References..... | 7 | +| 3 Definitions, symbols and abbreviations ..... | 9 | +| 3.1 Definitions..... | 9 | +| 3.2 Abbreviations ..... | 9 | +| 4 Interfaces..... | 9 | +| 5 Test configurations..... | 10 | +| 5.1 Setup for terminals ..... | 10 | +| 5.1.1 Setup for handset terminals ..... | 11 | +| 5.1.2 Setup for headset terminals ..... | 12 | +| 5.1.3 Setup for hands-free terminals..... | 12 | +| 5.1.3.1 Vehicle-mounted hands-free..... | 12 | +| 5.1.3.2 Desktop hands-free ..... | 13 | +| 5.1.3.3 Hand-held hands-free..... | 13 | +| 5.1.3.4 Softphone including speakers and microphone ..... | 15 | +| 5.1.3.5 Softphone with separate speakers ..... | 17 | +| 5.1.4 Position and calibration of HATS ..... | 21 | +| 5.1.5 Test setup for quality in the presence of ambient noise measurements ..... | 21 | +| 5.2 Setup of the electrical interfaces ..... | 22 | +| 5.2.1 Codec approach and specification..... | 22 | +| 5.2.2 Direct digital processing approach..... | 22 | +| 5.3 Accuracy of test equipment..... | 22 | +| 5.4 Test signals..... | 23 | +| 5.5 Void..... | 24 | +| 5.5.1 Void..... | 24 | +| 5.5.2 Void..... | 24 | +| 6 Test conditions ..... | 24 | +| 6.1 Environmental conditions ..... | 24 | +| 6.1.1 Handset and headset terminals ..... | 24 | +| 6.1.2 Hands-free terminals ..... | 24 | +| 6.2 System simulator conditions ..... | 25 | +| 7 Narrowband telephony transmission performance test methods ..... | 25 | +| 7.1 Applicability..... | 25 | +| 7.2 Overall loss/loudness ratings..... | 25 | +| 7.2.1 General ..... | 25 | +| 7.2.2 Connections with handset UE ..... | 26 | +| 7.2.2.1 Sending loudness rating (SLR)..... | 26 | +| 7.2.2.2 Receiving loudness rating (RLR) ..... | 26 | +| 7.2.3 Connections with desktop and vehicle-mounted hands-free UE..... | 26 | +| 7.2.3.1 Sending loudness rating (SLR)..... | 26 | +| 7.2.3.2 Receiving Loudness Rating (RLR)..... | 27 | +| 7.2.4 Connections with hand-held hands-free UE..... | 27 | +| 7.2.4.1 Sending loudness rating (SLR)..... | 27 | +| 7.2.4.2 Receiving loudness rating (RLR) ..... | 27 | +| 7.2.5 Connections with headset UE..... | 28 | +| 7.3 Idle channel noise (handset and headset UE)..... | 28 | +| 7.3.1 Sending ..... | 28 | +| 7.3.2 Receiving..... | 29 | +| 7.4 Sensitivity/frequency characteristics..... | 29 | +| 7.4.1 Handset and headset UE sending ..... | 29 | +| 7.4.2 Handset and headset UE receiving..... | 29 | + +| | | | +|--------------|--------------------------------------------------------------------------------------|----| +| 7.4.3 | Desktop and vehicle-mounted hands-free UE sending..... | 30 | +| 7.4.4 | Desktop and vehicle-mounted hands-free UE receiving ..... | 30 | +| 7.4.5 | Hand-held hands-free UE sending..... | 30 | +| 7.4.6 | Hand-held hands-free UE receiving ..... | 30 | +| 7.5 | Sidetone characteristics..... | 31 | +| 7.5.1 | Connections with handset UE ..... | 31 | +| 7.5.1.1 void | | 31 | +| 7.5.1.2 | Connections with handset UE – HATS method ..... | 31 | +| 7.5.2 | Headset UE..... | 31 | +| 7.5.3 | Hands-free UE (all categories) ..... | 32 | +| 7.5.4 | Sidetone delay for handset or headset ..... | 32 | +| 7.6 | Stability loss ..... | 32 | +| 7.7 | Acoustic echo control..... | 34 | +| 7.7.1 | General ..... | 34 | +| 7.7.2 | Acoustic echo control in a hands-free UE..... | 34 | +| 7.7.3 | Acoustic echo control in handset UE ..... | 34 | +| 7.7.4 | Acoustic echo control in a headset UE..... | 34 | +| 7.8 | Distortion..... | 34 | +| 7.8.1 | Sending distortion..... | 34 | +| 7.8.2 | Receiving..... | 36 | +| 7.9 | Void..... | 37 | +| 7.10 | Delay ..... | 37 | +| 7.10.0 | UE Delay Measurement Methodologies ..... | 37 | +| 7.10.1 | Delay in sending direction (Handset UE)..... | 37 | +| 7.10.1a | Delay in sending direction (headset UE)..... | 38 | +| 7.10.2 | Delay in receiving direction (handset UE) ..... | 39 | +| 7.10.2a | Delay in receiving direction (headset UE) ..... | 40 | +| 7.10.3 | Delay in sending + receiving direction using “echo” method (handset UE)..... | 40 | +| 7.10.3a | Delay in sending + receiving direction using “echo” method (headset UE)..... | 41 | +| 7.11 | Echo control characteristics ..... | 42 | +| 7.11.1 | Test set-up and test signals ..... | 42 | +| 7.11.2 | Test method ..... | 42 | +| 7.11.2.1 | Signal alignment ..... | 43 | +| 7.11.2.2 | Signal level computation and frame classification ..... | 43 | +| 7.11.2.3 | Classification into categories..... | 44 | +| 7.12 | Quality (speech quality, noise intrusiveness) in the presence of ambient noise ..... | 44 | +| 8 | Wideband telephony transmission performance test methods ..... | 46 | +| 8.1 | Applicability..... | 46 | +| 8.2 | Overall loss/loudness ratings..... | 46 | +| 8.2.1 | General ..... | 46 | +| 8.2.2 | Connections with handset UE ..... | 46 | +| 8.2.2.1 | Sending loudness rating (SLR)..... | 46 | +| 8.2.2.2 | Receiving loudness rating (RLR) ..... | 46 | +| 8.2.3 | Connections with desktop and vehicle-mounted hands-free UE..... | 47 | +| 8.2.3.1 | Sending loudness rating (SLR)..... | 47 | +| 8.2.3.2 | Receiving loudness rating (RLR) ..... | 47 | +| 8.2.4 | Connections with hand-held hands-free UE..... | 47 | +| 8.2.4.1 | Sending loudness rating (SLR)..... | 47 | +| 8.2.4.2 | Receiving loudness rating (RLR) ..... | 48 | +| 8.2.5 | Connections with headset UE..... | 48 | +| 8.3 | Idle channel noise (handset and headset UE)..... | 48 | +| 8.3.1 | Sending ..... | 48 | +| 8.3.2 | Receiving..... | 49 | +| 8.4 | Sensitivity/frequency characteristics..... | 49 | +| 8.4.1 | Handset and headset UE sending ..... | 49 | +| 8.4.2 | Handset and headset UE receiving..... | 50 | +| 8.4.3 | Desktop and vehicle-mounted hands-free UE sending..... | 50 | +| 8.4.4 | Desktop and vehicle-mounted hands-free UE receiving ..... | 50 | +| 8.4.5 | Hand-held hands-free UE sending..... | 51 | +| 8.4.6 | Hand-held hands-free UE receiving ..... | 51 | + +| | | | +|-------------------------------|--------------------------------------------------------------------------------------|-----------| +| 8.5 | Sidetone characteristics..... | 51 | +| 8.5.1 | Connections with handset UE ..... | 51 | +| 8.5.2 | Headset UE..... | 52 | +| 8.5.3 | Hands-free UE (all categories) ..... | 52 | +| 8.5.4 | Sidetone delay for handset or headset ..... | 52 | +| 8.6 | Stability loss ..... | 53 | +| 8.7 | Acoustic echo control..... | 54 | +| 8.7.1 | General ..... | 54 | +| 8.7.2 | Acoustic echo control in a hands-free UE..... | 54 | +| 8.7.3 | Acoustic echo control in a handset UE..... | 55 | +| 8.7.4 | Acoustic echo control in a headset UE..... | 55 | +| 8.8 | Distortion..... | 55 | +| 8.8.1 | Sending distortion..... | 55 | +| 8.8.2 | Receiving..... | 56 | +| 8.9 | Void..... | 58 | +| 8.10 | Delay ..... | 58 | +| 8.10.0 | UE Delay Measurement Methodologies ..... | 58 | +| 8.10.1 | Delay in sending direction (handset UE)..... | 58 | +| 8.10.1a | Delay in sending direction (headset UE)..... | 59 | +| 8.10.2 | Delay in receiving direction (handset UE) ..... | 59 | +| 8.10.2a | Delay in receiving direction (headset UE) ..... | 61 | +| 8.10.3 | Delay in sending + receiving direction using “echo” method (handset UE)..... | 61 | +| 8.10.3a | Delay in sending + receiving direction using “echo” method (headset UE) ..... | 62 | +| 8.11 | Echo control characteristics ..... | 63 | +| 8.11.1 | Test set-up and test signals ..... | 63 | +| 8.11.2 | Test method ..... | 63 | +| 8.11.2.1 | Signal alignment ..... | 64 | +| 8.11.2.2 | Signal level computation and frame classification ..... | 64 | +| 8.11.2.3 | Classification into categories..... | 65 | +| 8.12 | Quality (speech quality, noise intrusiveness) in the presence of ambient noise ..... | 65 | +| Annex A: | Interpolation method for diffuse-field correction (normative)..... | 67 | +| Annex B (informative): | Reference algorithm for echo control characteristics evaluation. .... | 69 | +| B.1 | General ..... | 69 | +| B.2 | Test script..... | 70 | +| B.3 | Reference algorithm..... | 71 | +| B.3.1 | Main algorithm ..... | 71 | +| B.3.2 | Delay compensation ..... | 72 | +| B.3.3 | Signal level computation and frame classification..... | 73 | +| B.3.4 | Level vs time computation ..... | 75 | +| B.3.5 | Categorization ..... | 77 | +| B.3.6 | Auxiliary functions for reporting data..... | 79 | +| B.3.7 | Other helper functions ..... | 80 | +| Annex C (informative): | Change history..... | 81 | + +# --- Foreword + +This Technical Specification has been produced by the 3GPP. + +The contents of the present document are subject to continuing work within the TSG and may change following formal TSG approval. Should the TSG modify the contents of this TS, it will be re-released by the TSG with an identifying change of release date and an increase in version number as follows: + +Version x.y.z + +where: + +- x the first digit: + - 1 presented to TSG for information; + - 2 presented to TSG for approval; + - 3 or greater indicates TSG approved document under change control. +- y the second digit is incremented for all changes of substance, i.e. technical enhancements, corrections, updates, etc. +- z the third digit is incremented when editorial only changes have been incorporated in the specification. + +# --- Introduction + +The present document specifies test methods to allow the minimum performance requirements for the acoustic characteristics of GSM and 3G terminals when used to provide narrowband or wideband telephony to be assessed. + +The objective for narrowband services is to reach a quality as close as possible to ITU-T standards for PSTN circuits. However, due to technical and economic factors, there cannot be full compliance with the general characteristics of international telephone connections and circuits recommended by the ITU-T. + +The performance requirements are specified in TS 26.131; the test methods and considerations are specified in the main body of the text. + +# --- 1 Scope + +The present document is applicable to any terminal capable of supporting narrowband or wideband telephony, either as a stand-alone service or as the telephony component of a multimedia service. The present document specifies test methods to allow the minimum performance requirements for the acoustic characteristics of GSM and 3G terminals when used to provide narrowband or wideband telephony to be assessed. + +# --- 2 References + +The following documents contain provisions which, through reference in this text, constitute provisions of the present document. + +- References are either specific (identified by date of publication, edition number, version number, etc.) or non-specific. + - For a specific reference, subsequent revisions do not apply. + - For a non-specific reference, the latest version applies. In the case of a reference to a 3GPP document (including a GSM document), a non-specific reference implicitly refers to the latest version of that document *in the same Release as the present document*. +- [1] 3GPP TS 26.131: "Terminal Acoustic Characteristics for Telephony; Requirements". +- [2] ITU-T Recommendation B.12 (1988): "Use of the decibel and the neper in telecommunications". +- [3] ITU-T Recommendation G.103 (1998): "Hypothetical reference connections". +- [4] ITU-T Recommendation G.111 (1993): "Loudness ratings (LRs) in an international connection". +- [5] ITU-T Recommendation G.121 (1993): "Loudness ratings (LRs) of national systems". +- [6] ITU-T Recommendation G.122 (1993): "Influence of national systems on stability and talker echo in international connections". +- [7] ITU-T Recommendation G.711 (1988): "Pulse code modulation (PCM) of voice frequencies". +- [8] ITU-T Recommendation P.11 (1993): "Effect of transmission impairments". +- [9] ITU-T Recommendation P.38 (1993): "Transmission characteristics of operator telephone systems (OTS)". +- [10] ITU-T Recommendation P.50 (1993): "Artificial voices". +- [11] 3GPP TS 03.58 (Release 1997): "Digital Cellular Telecommunications System (Phase 2+) Characterization test methods and quality assessment for hands-free mobiles". +- [12] IEC Publication 60651: "Sound Level Meters". +- [13] ITU-T Recommendation P.51 (1996): "Artificial mouth". +- [14] ITU-T Recommendation P.57 (2005): "Artificial ears". +- [15] ITU-T Recommendation P.58 (1996): "Head and torso simulator for telephonometry." +- [16] ITU-T Recommendation P.79 (2007) with Annex A: "Calculation of loudness ratings for telephone sets." +- [17] 3GPP TS 46.077 : "Minimum Performance Requirements for Noise Suppressor Application to the AMR Speech Encoder". +- [18] ITU-T Recommendation P.64: "Determination of sensitivity/frequency characteristics of local telephone systems". + +- [19] ITU-T Recommendation P.581: "Use of head and torso simulator (HATS) for hands-free and handset terminal testing". +- [20] ITU-T Recommendation P.340: "Transmission characteristics and speech quality parameters of hands-free terminals". +- [21] ITU-T Recommendation G.712: "Transmission performance characteristics of pulse code modulation channels". +- [22] ITU-T Recommendation P.501: "Test signals for use in telephonometry". +- [23] ITU-T Recommendation O.41: "Psophometer for use on telephone-type circuits". +- [24] ITU-T Recommendation O.131: "Quantizing distortion measuring equipment using a pseudo-random noise test signal". +- [25] Void. +- [26] ISO 3745: "Acoustics - Determination of sound power levels of noise sources using sound pressure - Precision methods for anechoic and hemi-anechoic rooms". +- [27] ITU-T Recommendation O.132: "Quantizing distortion measuring equipment using a sinusoidal test signal". +- [28] ETSI TS 103 737 (2010-08) V1.1.2: "Transmission requirements for narrowband wireless terminals (handset and headset) from a QoS perspective as perceived by the user". +- [29] ETSI TS 103 738 (2010-09) V1.1.2: "Transmission requirements for narrowband wireless terminals (handsfree) from a QoS perspective as perceived by the user". +- [30] ETSI TS 103 739 (2010-09) V1.1.2: "Transmission requirements for wideband wireless terminals (handset and headset) from a QoS perspective as perceived by the user". +- [31] ETSI TS 103 740 (2010-09) V1.1.2: "Transmission requirements for wideband wireless terminals (handsfree) from a QoS perspective as perceived by the user". +- [32] ITU-T Recommendation P.380: "Electro-acoustic measurements on headsets". +- [33] ITU-T Recommendation P.501 Amendment 1 (2012): "Test signals for use in telephonometry". +- [34] ETSI TS 103 106 (2013-03) V1.2.1: "Speech Quality performance in the presence of background noise: Background noise transmission of mobile terminals-Objective test methods". +- [35] ETSI ES 202 396-1 (2012-10) V1.4.1: "Speech quality performance in the presence of background noise; Part 1: Background noise simulation technique and background noise database". +- [36] ETSI EG 202 396-3 (2011-02) V1.3.1: "Speech quality performance in the presence of background noise; Part 3: *Background noise transmission – objective test methods*: Background noise simulation technique and background noise database". +- [37] ITU-T Recommendation P.56: "Objective measurement of active speech level". +- [38] IEC 61672: "Electroacoustics – sound level meters - part 1: specifications". + +# --- 3 Definitions, symbols and abbreviations + +## 3.1 Definitions + +For the purposes of the present document the term *narrowband* refers to signals sampled at 8 kHz; *wideband* refers to signals sampled at 16 kHz. + +For the purposes of the present document, the terms dB, dBr, dBm0, dBm0p and dBA, shall be interpreted as defined in ITU-T Recommendation B.12 [2]; the term dBPa shall be interpreted as the sound pressure level relative to 1 pascal expressed in dB (0 dBPa is equivalent to 94 dB SPL). + +A 3GPP softphone is a telephony system running on a general purpose computer or PDA complying with the 3GPP terminal acoustic requirements (TS 26.131 and 26.132). + +## 3.2 Abbreviations + +For the purposes of the present document, the following abbreviations apply: + +| | | +|------|--------------------------------------| +| ADC | Analogue to Digital Converter | +| CSS | Composite Source Signal | +| DAC | Digital to Analogue Converter | +| DRP | Eardrum Reference Point | +| DTX | Discontinuous Transmission | +| EEC | Electrical Echo Control | +| EEP | Ear Entrance Point | +| EL | Echo Loss | +| ERP | Ear Reference Point | +| FFT | Fast Fourier Transform | +| HATS | Head and Torso Simulator | +| LSTR | Listener Sidetone Rating | +| MRP | Mouth Reference Point | +| MS | Mobile Station | +| OLR | Overall Loudness Rating | +| PCM | Pulse Code Modulation | +| PDA | Personal Digital Assistant | +| POI | Point of Interconnection (with PSTN) | +| PSTN | Public Switched Telephone Network | +| RLR | Receive Loudness Rating | +| RMS | Root Mean Squared | +| SLR | Send Loudness Rating | +| SS | System Simulator | +| STMR | Sidetone Masking Rating | +| SS | System Simulator | +| TX | Transmission | +| UE | User Equipment | + +# --- 4 Interfaces + +Access to terminals for acoustic testing is always made via the acoustic or air interfaces. The Air Interface is specified by the GSM 05, GSM 45 and 3G 25 series specifications and is required to achieve user equipment (UE) transportability. Measurements can be made at this point using a system simulator (SS) comprising the appropriate radio terminal equipment and speech transcoder. The losses and gains introduced by the test speech transcoder will need to be specified. + +The POI with the public switched telephone network (PSTN) is considered to have a relative level of 0 dBr, where signals will be represented by 8-bit A-law, according to ITU-T Recommendation G.711 [7]. Measurements may be made at this point using a standard send and receive side, as defined in ITU-T Recommendations. + +Five classes of acoustic interface are considered in this specification: + +- Handset UE including softphone UE used as a handset; +- Headset UE including softphone UE used with headset; +- Vehicle Mounted Hands-free UE including softphone UE mounted in a vehicle; +- Desktop-mounted hands-free UE including softphone UE with external loudspeaker(s) used in hands-free mode; + +- Hand-held hands-free UE including softphone UE with internal loudspeaker(s) used in hands-free mode. + +(See definition of softphone in Clause 3.1) + +NOTE: The test setup for a softphone UE shall be derived according to the following rules: + +- When using a softphone UE as a handset: the test setup shall correspond to handset mode. +- When using a softphone UE with headset: the test setup shall correspond to headset mode. +- When a softphone UE is mounted in a vehicle: the test setup shall correspond to vehicle-mounted hands-free mode. +- When using a softphone UE in hands-free mode: + - When using internal loudspeaker(s), the test setup shall correspond to hand-held hands-free. + - When using external loudspeaker(s), the test setup shall correspond to desktop-mounted hands-free. + +# --- 5 Test configurations + +This section describes the test setups for terminal acoustic testing. + +NOTE: If the terminal has several mechanical configurations (e.g., sliding design open or closed), all manufacturer-defined configurations shall be tested. + +## 5.1 Setup for terminals + +The general access to terminals is described in figure 1. The preferred acoustic access to GSM and 3G terminals is the most realistic simulation of the “average” subscriber. This can be made by using HATS (head and torso simulator), with appropriate ear simulation and appropriate mountings of handset terminals to the HATS in a realistic but reproducible way. Hands-free terminals shall use the HATS or free field microphone techniques in a realistic but reproducible way. + +HATS is described in ITU-T Recommendation P.58 [15], appropriate ears are described in ITU-T Recommendation P.57 [14] (Type 3.3), proper positioning of handsets in realistic conditions is found in ITU-T Recommendation P.64, and the test setups for various types of hands-free terminals can be found in ITU-T Recommendation P.581. + +Unless stated otherwise, if a volume control is provided, the setting is chosen such that the nominal RLR is met as close as possible. + +The preferred way of testing is the connection of a terminal to the system simulator with exact defined settings and access points. The test sequences are fed in either electrically using a reference codec, using the direct signal processing approach, or acoustically using ITU-T specified devices. + +![Diagram of GSM/3G interfaces for specification and testing of terminal acoustic characteristics. The diagram shows a flow from left to right: Artificial Mouth (MRP) connected via a Connection cable to Headset Microphone, which is connected to UE Signal Processing Entities. This is followed by RF Transmission & Speech coder, which has an upward arrow. To the right is a Test equipment block, also with an upward arrow. A bracket labeled T_s spans from the Artificial Mouth to the RF Transmission & Speech coder. Another bracket labeled T_TES spans under the Test equipment block.](5a4e62bead259c258d069fd3663ea670_img.jpg) + +The diagram illustrates the acoustic interfaces for testing terminal characteristics. On the far left is an 'Artificial Mouth' (MRP) represented by a trapezoidal shape. A 'Connection cable' (indicated by a thick horizontal line) connects it to a 'Headset Microphone' (represented by an oval). The headset microphone is connected to a block labeled 'UE Signal Processing Entities'. This block is connected to another block labeled 'RF Transmission & Speech coder', which has an upward-pointing arrow. To the right of the RF block is a 'Test equipment' block, also with an upward-pointing arrow. A large bracket labeled $T_s$ spans from the Artificial Mouth to the RF Transmission & Speech coder. Another bracket labeled $T_{TES}$ is positioned under the Test equipment block. + +Diagram of GSM/3G interfaces for specification and testing of terminal acoustic characteristics. The diagram shows a flow from left to right: Artificial Mouth (MRP) connected via a Connection cable to Headset Microphone, which is connected to UE Signal Processing Entities. This is followed by RF Transmission & Speech coder, which has an upward arrow. To the right is a Test equipment block, also with an upward arrow. A bracket labeled T\_s spans from the Artificial Mouth to the RF Transmission & Speech coder. Another bracket labeled T\_TES spans under the Test equipment block. + +NOTE 1: Includes DTX functionality. + +NOTE 2: Connection to PSTN should include electrical echo control (EEC). + +**Figure 1: GSM/3G Interfaces for specification and testing of terminal acoustic characteristics** + +### 5.1.1 Setup for handset terminals + +When using a handset UE, the handset is placed on HATS as described in ITU-T Recommendation P.64 Annex E [18]. A suitable position shall be defined for each handset UE and documented in the test report. The artificial mouth shall conform to ITU-T Recommendation P.58 [15]. The artificial ear shall conform to ITU-T Recommendation P.57 [14]. Type 3.3 ear shall be used and positioned on HATS according to ITU-T Recommendation P.58 [15]. + +#### Position and calibration of HATS + +The sending and receiving characteristics shall be tested with the HATS. It shall be indicated what application force was used. If not stated otherwise in TS 26.131, an application force of $8 \pm 2$ N shall be used. + +The horizontal positioning of the HATS reference plane shall be guaranteed within $\pm 2^\circ$ . + +### 5.1.2 Setup for headset terminals + +Recommendations for the setup and positioning of headsets are given in ITU-T Recommendation P.380. If not stated otherwise, headsets shall be placed in their recommended wearing position. Some insert earphones might not fit properly in Type 3.3 ear simulators. For such insert type headsets, an ITU-T Recommendation P.57 [14] Type 2 ear simulator may be used in conjunction with the HATS mouth simulator. The HATS should be equipped with two artificial ears as specified in ITU-T Recommendation P.57 [14]. For binaural headsets two artificial ears are required. + +### 5.1.3 Setup for hands-free terminals + +#### 5.1.3.1 Vehicle-mounted hands-free + +If not stated otherwise, the artificial head (HATS – head and torso simulator, according to ITU-T Recommendation P.58 [15]) is positioned in the driver's seat for the measurement as shown in figure 3a. The position has to be in line with the average users' position; therefore, all positions and sizes of users have to be taken into account. Typically, all except the tallest 5% and the shortest 5% of the driving population have to be considered. The size of these persons can be derived, e.g., from the 'anthropometric data set' for the corresponding year (e.g., based on data used by car manufacturers). The position of the HATS (mouth/ears) within the positioning arrangement is given individually by each car manufacturer. The position used has to be reported in detail in the test report. If no requirements for positioning are given the distance from the microphone to the MRP is defined by the test lab. + +By using suitable measures (e.g., marks in the car, relative position to A-pillar, B-pillar, height from the floor, etc.) an exact reproduction of the artificial head position must be possible at any later time. + +NOTE – Different positions of the artificial head may greatly influence the test results. Depending on the application, different positions of the artificial head may be chosen for the tests. It is recommended to check the worst-case position, e.g., those positions where the SNR and/or the speech quality in send may be worst. + +**Figure 2: void** + +**Figure 3: void** + +![Figure 3a: Test Configuration for vehicle mounted hands-free, using HATS. The diagram shows a signal flow from left to right: Artificial Ear (DRP) connected to UE Loudspeaker, which is connected to UE Signal Processing Entities. This is followed by RF Reception & Speech Decoder, and finally Test equipment. The Artificial Ear and UE Loudspeaker are grouped by a bracket labeled T_T. The RF Reception & Speech Decoder and Test equipment are grouped by a bracket labeled T_TER.](ff0952ef692c9d960ce5f6708bcc9711_img.jpg) + +The diagram illustrates the test configuration for vehicle-mounted hands-free using HATS. It shows a sequence of components: an Artificial Ear (DRP) connected to a UE Loudspeaker, which is connected to UE Signal Processing Entities. This is followed by RF Reception & Speech Decoder, and finally Test equipment. The Artificial Ear and UE Loudspeaker are grouped by a bracket labeled $T_T$ . The RF Reception & Speech Decoder and Test equipment are grouped by a bracket labeled $T_{TER}$ . + +Figure 3a: Test Configuration for vehicle mounted hands-free, using HATS. The diagram shows a signal flow from left to right: Artificial Ear (DRP) connected to UE Loudspeaker, which is connected to UE Signal Processing Entities. This is followed by RF Reception & Speech Decoder, and finally Test equipment. The Artificial Ear and UE Loudspeaker are grouped by a bracket labeled T\_T. The RF Reception & Speech Decoder and Test equipment are grouped by a bracket labeled T\_TER. + +**Figure 3a: Test Configuration for vehicle mounted hands-free, using HATS** + +#### 5.1.3.2 Desktop hands-free + +For HATS test equipment, the definition of hands-free terminals and setup for desktop hands-free terminals can be found in ITU-T Recommendation P.581. Measurement setup using a free-field microphone and a discrete P.51 [13] artificial mouth for desktop hands-free terminals can be found in ITU-T Recommendation P.340. The positioning for different types of desktop hands-free terminals is given in ETSI TS 103 738 and ETSI TS 103 740. + +#### 5.1.3.3 Hand-held hands-free + +Either HATS or a free-field microphone with a discrete P.51 [13] artificial mouth may be used to measure a hand-held hands-free type UE. + +If HATS measurement equipment is used, it shall be configured to the hand-held hands-free UE according to figure 4. The HATS should be positioned so that the HATS Reference Point is at a distance $d_{HF}$ from the centre point of the visual display of the Mobile Station. The distance $d_{HF}$ is specified by the manufacturer. A vertical angle $\theta_{HF}$ may be specified by the manufacturer. Where it is not specified, the nominal distance $d_{HF}$ shall be 42 cm and $\theta_{HF}$ shall be $0^\circ$ . + +**NOTE:** The nominal distance of 42 cm corresponds to the distance between the HATS reference point and lip-plane (12 cm) with an additional 30 cm giving a realistic figure as a reference usage of hand-held terminals. + +![Figure 4: Configuration of hand-held hands-free UE relative to the HATS. The diagram shows a profile of a head (HATS) with a reference point at the ear. A mobile phone is held in front of the face. A horizontal dashed line extends from the HATS reference point to the front of the phone. The distance between the reference point and the phone is labeled d_{HF}. The angle between the horizontal dashed line and the line of sight to the phone is labeled +\theta_{HF}. A vertical dashed line at the front of the phone indicates the normal vector from the front of the phone.](a1d3651b1300f3670e3a9547bafc4db6_img.jpg) + +Figure 4: Configuration of hand-held hands-free UE relative to the HATS. The diagram shows a profile of a head (HATS) with a reference point at the ear. A mobile phone is held in front of the face. A horizontal dashed line extends from the HATS reference point to the front of the phone. The distance between the reference point and the phone is labeled d\_{HF}. The angle between the horizontal dashed line and the line of sight to the phone is labeled +\theta\_{HF}. A vertical dashed line at the front of the phone indicates the normal vector from the front of the phone. + +**Figure 4: Configuration of hand-held hands-free UE relative to the HATS** + +If a free-field microphone and a discrete P.51 [13] mouth are used, they shall be configured to the hand-held hands-free UE according to figure 5 for receiving measurements and figure 6 for sending measurements. The microphone should be located at a distance $d_{HF}$ from the centre of the visual display of the UE. The mouth simulator should be located at a distance $d_{HF}-12$ cm from the centre of the visual display of the UE. The distance $d_{HF}$ is specified by the manufacturer. Where it is not specified the nominal distance $d_{HF}$ shall be 42 cm. + +![Figure 5: Configuration of hand-held hands-free UE; free-field microphone for receiving measurements. The diagram shows a free-field measurement microphone on the left and a mobile phone on the right. A horizontal dashed line connects the microphone to the front of the phone. The distance between them is labeled d_{HF}. A vertical dashed line at the front of the phone indicates the normal vector from the front of the phone.](3293245c6893d9d49c2c878828423ecd_img.jpg) + +Figure 5: Configuration of hand-held hands-free UE; free-field microphone for receiving measurements. The diagram shows a free-field measurement microphone on the left and a mobile phone on the right. A horizontal dashed line connects the microphone to the front of the phone. The distance between them is labeled d\_{HF}. A vertical dashed line at the front of the phone indicates the normal vector from the front of the phone. + +**Figure 5: Configuration of hand-held hands-free UE; free-field microphone for receiving measurements** + +![Figure 6: Configuration of hand-held hands-free UE; discrete P.51 artificial mouth for sending measurements. The diagram shows a discrete P.51 artificial mouth (lip ring position) on the left and a mobile phone on the right. A horizontal dashed line connects the lip ring position to the front of the phone. The distance between them is labeled d_{HF} - 12 cm. A vertical dashed line at the front of the phone indicates the normal vector from the front of the phone.](bf9297824aec2a021ecbad6f70536914_img.jpg) + +Figure 6: Configuration of hand-held hands-free UE; discrete P.51 artificial mouth for sending measurements. The diagram shows a discrete P.51 artificial mouth (lip ring position) on the left and a mobile phone on the right. A horizontal dashed line connects the lip ring position to the front of the phone. The distance between them is labeled d\_{HF} - 12 cm. A vertical dashed line at the front of the phone indicates the normal vector from the front of the phone. + +**Figure 6: Configuration of hand-held hands-free UE; discrete P.51 artificial mouth for sending measurements** + +#### 5.1.3.4 Softphone including speakers and microphone + +This test setup is applicable to laptop computers or similar devices as seen in figure 7 through figure 11. + +Where the manufacturer gives conditions of use, these will apply for testing. If the manufacturer gives no other requirement, the softphone will be positioned according the following conditions: + +Measurement with artificial ear and microphone: + +Artificial mouth (for sending tests) + +![Diagram showing the configuration of a softphone relative to an artificial mouth in a side view. The artificial mouth is on the left, tilted at an angle, with a 'Lip Ring' indicated by a dot and line. The softphone is on the right, resting on a horizontal surface. Dimensions are shown: 30 cm from the base of the artificial mouth to the front of the softphone, and 20 cm from the base of the artificial mouth to the front edge of the softphone.](ace13edeb79bdfa129ed84fbb4ac44e5_img.jpg) + +Diagram showing the configuration of a softphone relative to an artificial mouth in a side view. The artificial mouth is on the left, tilted at an angle, with a 'Lip Ring' indicated by a dot and line. The softphone is on the right, resting on a horizontal surface. Dimensions are shown: 30 cm from the base of the artificial mouth to the front of the softphone, and 20 cm from the base of the artificial mouth to the front edge of the softphone. + +**Figure 7: Configuration of a softphone relative to the artificial mouth side view** + +Free field microphone (for receiving): + +![Diagram showing the configuration of a softphone relative to a free field microphone in a side view. The free field microphone is on the left, positioned above the softphone. The softphone is on the right, resting on a horizontal surface. Dimensions are shown: 30 cm from the base of the microphone to the front of the softphone, and 20 cm from the base of the microphone to the front edge of the softphone.](349cffebeefaae56d9034d3fe65bf7c6_img.jpg) + +Diagram showing the configuration of a softphone relative to a free field microphone in a side view. The free field microphone is on the left, positioned above the softphone. The softphone is on the right, resting on a horizontal surface. Dimensions are shown: 30 cm from the base of the microphone to the front of the softphone, and 20 cm from the base of the microphone to the front edge of the softphone. + +**Figure 8: Configuration of a softphone relative to the free field microphone side view** + +Position of a softphone on the table: + +![Diagram of a softphone relative to a microphone or artificial mouth.](555df5c0300cb1fca5dc028fec5ec6be_img.jpg) + +A top-down view diagram showing a 'Hands free softphone' represented by a rectangle with a shaded area and two small circles. Below it, a 'Microphone (or artificial mouth)' is shown as a narrow vertical rectangle. A double-headed vertical arrow between them is labeled '20 cm'. To the left of the softphone, the word 'Test' is written. A curved line extends from the bottom of the microphone. + +Diagram of a softphone relative to a microphone or artificial mouth. + +**Figure 9: Configuration of a softphone relative to the free-field microphone or artificial mouth viewed from above** + +Measurement with HATS: + +![Side view diagram of a softphone relative to a HATS head.](a8e5c2ac336eb43cda4e333ea9c73237_img.jpg) + +A side view diagram showing a 'HATS' head model on the left. A 'Lip Ring' is indicated by a dot on the lips. To the right, a 'Softphone' is shown as a horizontal bar. A horizontal double-headed arrow between the lip ring and the softphone is labeled '30 cm'. Another horizontal double-headed arrow, starting from the vertical plane of the lip ring and extending further right, is labeled '20 cm'. + +Side view diagram of a softphone relative to a HATS head. + +**Figure 10: Configuration of a softphone relative to the HATS side view** + +![Diagram showing the configuration of a softphone relative to the HATS viewed from above. The HATS is on the test table, and the handsfree softphone is positioned 20 cm above it.](6cc4a2d5ea0462e4825d57bd689bd2b3_img.jpg) + +The diagram illustrates a top-down view of a testing setup. A horizontal line represents the 'Test table'. Below this line, a circular object labeled 'HATS' is shown. Above the test table line, a rectangular object labeled 'Handsfree softphone' is positioned. A vertical double-headed arrow between the HATS and the softphone indicates a distance of '20 cm'. + +Diagram showing the configuration of a softphone relative to the HATS viewed from above. The HATS is on the test table, and the handsfree softphone is positioned 20 cm above it. + +**Figure 11: Configuration of a softphone relative to the HATS viewed from above** + +#### 5.1.3.5 Softphone with separate speakers + +This test setup is applicable to laptop computers or similar devices as seen in figure 12 through figure 15. + +Where the manufacturer gives conditions of use, these will apply for testing. If the manufacturer gives no other requirement, the softphone will be positioned according to the following conditions: + +Where separate loudspeakers are used, the system will be positioned as in figure 12 or figure 13. + +![Diagram showing the configuration of a softphone using external speakers relative to a microphone or artificial mouth. The diagram includes labels for 'Loudspeakers', 'Hands free softphone', 'Test table', and 'Microphone or artificial mouth', with dimensions: 80 cm between speakers, 40 cm from table to speakers, 40 cm from center line to each speaker, and 20 cm from center line to microphone.](9b9d2abd741ed4bafe7f78f89961c663_img.jpg) + +The diagram illustrates the setup for testing a hands-free softphone using external loudspeakers. A horizontal line represents the 'Test table'. On the table, a 'Hands free softphone' is positioned centrally. Two 'Loudspeakers' are placed on the table, one on each side of the softphone. The distance between the centers of the two loudspeakers is 80 cm. The distance from the test table surface to the center of each loudspeaker is 40 cm. The distance from the central vertical axis of the softphone to the center of each loudspeaker is 40 cm. Below the test table, a 'Microphone or artificial mouth' is positioned, with its center line aligned with the softphone's center line. The distance from the test table surface to the microphone or artificial mouth is 20 cm. + +Diagram showing the configuration of a softphone using external speakers relative to a microphone or artificial mouth. The diagram includes labels for 'Loudspeakers', 'Hands free softphone', 'Test table', and 'Microphone or artificial mouth', with dimensions: 80 cm between speakers, 40 cm from table to speakers, 40 cm from center line to each speaker, and 20 cm from center line to microphone. + +**Figure 12: Configuration of a softphone using external speakers relative to microphone or artificial mouth viewed from above** + +![Diagram showing the configuration of a softphone using external speakers relative to the HATS, viewed from above. The diagram includes a 'Test table' with a 'Hands free softphone' and two 'Loudspeaker' units. Dimensions are given: 80 cm between speakers, 40 cm from each speaker to the center line, 40 cm from the table edge to the speakers, and 20 cm from the center line to the HATS.](8fa679f79a1bb1f527cba9f29e784e89_img.jpg) + +The diagram illustrates the spatial arrangement of a 'Hands free softphone' and two 'Loudspeaker' units on a 'Test table', relative to a 'HATS' (Head and Torso Simulator) positioned below the table. The view is from above. + +- The distance between the centers of the two 'Loudspeaker' units is 80 cm. +- Each 'Loudspeaker' is positioned 40 cm from the central vertical axis of the 'Hands free softphone'. +- The 'Hands free softphone' is centered on the 'Test table'. +- The distance from the edge of the 'Test table' to the 'Loudspeaker' units is 40 cm. +- The 'HATS' is positioned 20 cm directly below the center of the 'Hands free softphone'. + +Diagram showing the configuration of a softphone using external speakers relative to the HATS, viewed from above. The diagram includes a 'Test table' with a 'Hands free softphone' and two 'Loudspeaker' units. Dimensions are given: 80 cm between speakers, 40 cm from each speaker to the center line, 40 cm from the table edge to the speakers, and 20 cm from the center line to the HATS. + +**Figure 13: Configuration of a softphone using external speakers relative to the HATS viewed from above** + +Where an external microphone and speakers are used, the system will be positioned as in figure 14 or figure 15. + +![Diagram showing the configuration of a softphone using external speakers and a microphone relative to microphone or artificial mouth viewed from above. The diagram shows a test table with an artificial mouth or microphone at the center. Two loudspeakers are positioned on either side, 40 cm away from the center. The distance between the two loudspeakers is 80 cm. The distance from the center to each loudspeaker is 40 cm. The distance from the artificial mouth or microphone to each loudspeaker is 40 cm.](523ab7b925beb555f88b2e1e1336974f_img.jpg) + +The diagram illustrates the spatial arrangement of equipment for a softphone test. A horizontal line represents the 'Test table'. At the center of this line is a vertical component labeled 'Artificial mouth or microphone'. Directly above this component, also on the central vertical axis, is a 'Microphone'. Flanking the central axis are two 'loudspeaker' units, each represented by a rectangle with a circle inside. The distance from the central vertical axis to each loudspeaker is marked as 40 cm. The total distance between the two loudspeakers is marked as 80 cm. The vertical distance from the test table to each loudspeaker is marked as 40 cm. The vertical distance from the test table to the microphone is also marked as 40 cm. A cable is shown extending from the bottom of the 'Artificial mouth or microphone' component. + +Diagram showing the configuration of a softphone using external speakers and a microphone relative to microphone or artificial mouth viewed from above. The diagram shows a test table with an artificial mouth or microphone at the center. Two loudspeakers are positioned on either side, 40 cm away from the center. The distance between the two loudspeakers is 80 cm. The distance from the center to each loudspeaker is 40 cm. The distance from the artificial mouth or microphone to each loudspeaker is 40 cm. + +**Figure 14: Configuration of a softphone using external speakers and a microphone relative to microphone or artificial mouth viewed from above** + +![Figure 15: Configuration of a softphone using external speakers and a microphone relative to the HATS viewed from above. The diagram shows a top-down view of a test setup. A central 'Microphone' is positioned 40 cm above a 'HATS' (Head and Torso Simulator) on a 'Test table'. Two 'loudspeaker' units are positioned on either side of the microphone, each 40 cm horizontally away from the central axis and 40 cm vertically away from the HATS. The total width between the two loudspeakers is 80 cm. The HATS is represented by a stylized head and shoulders shape.](79e1709a7317ead45379cbb8ff3ba802_img.jpg) + +Figure 15: Configuration of a softphone using external speakers and a microphone relative to the HATS viewed from above. The diagram shows a top-down view of a test setup. A central 'Microphone' is positioned 40 cm above a 'HATS' (Head and Torso Simulator) on a 'Test table'. Two 'loudspeaker' units are positioned on either side of the microphone, each 40 cm horizontally away from the central axis and 40 cm vertically away from the HATS. The total width between the two loudspeakers is 80 cm. The HATS is represented by a stylized head and shoulders shape. + +**Figure 15: Configuration of a softphone using external speakers and a microphone relative to the HATS viewed from above** + +### 5.1.4 Position and calibration of HATS + +The horizontal positioning of the HATS reference plane shall be guaranteed within $\pm 2^\circ$ for testing hands-free equipment. + +The HATS shall be equipped with a Type 3.3 Artificial Ear. For hands-free measurements the HATS shall be equipped with two artificial ears. The pinnae are specified in Recommendation P.57 [14] for Type 3.3 artificial ears. The pinnae shall be positioned on HATS according to ITU-T Recommendation P.58 [15]. + +The exact calibration and equalization procedures as well as how to combine the two ear signals for the purpose of measurements can be found in ITU-T Recommendation P.581. If not stated otherwise, the HATS shall be diffuse-field equalized. The reverse nominal diffuse field curve as found in table 3 of ITU-T Recommendation P.58 [15] shall be used. For measurements requiring diffuse-field correction values for closer frequency spacing than that which is specified in ITU-T Recommendation P.58 [15], the interpolation method found in annex A shall be used. + +For hand-held hands-free UE, the setup corresponding to 'portable hands-free' in ITU-T Recommendation P.581 should be used. + +### 5.1.5 Test setup for quality in the presence of ambient noise measurements + +The setup for simulating realistic ambient noises and the positioning of the HATS in a lab-type environment is described in ETSI EG 202 396-1 [35]. + +ETSI EG 202 396-1 [35] contains a description of the recording arrangement for realistic ambient noises, a description of the setup for a loudspeaker arrangement suitable to simulate an ambient noise field in a lab-type environment and a database of realistic ambient noises, part of which is used for testing the terminal performance with a variety of conditions. + +The equalization and calibration procedure for the test setup are given in detail in ETSI EG 202 396-1 [35]. + +## 5.2 Setup of the electrical interfaces + +### 5.2.1 Codec approach and specification + +In this approach, a codec is used to convert the digital input/output bit-stream of the system simulator to the equivalent analogue values. With this approach a system simulator simulating the radio link to the terminal under controlled and + +error-free conditions is required. The system simulator has to be equipped with a high-quality codec with characteristics as close as possible to ideal. + +Definition of 0 dBr point: + +- D/A converter - a Digital Test Sequence (DTS) representing the codec equivalent of an analogue sinusoidal signal with an RMS value of 3,14 dB below the maximum full-load capacity of the codec shall generate 0 dBm across a 600 ohm load; +- A/D converter - a 0 dBm signal generated from a 600 ohm source shall give the digital test sequence (DTS) representing the codec equivalent of an analogue sinusoidal signal with an RMS value of 3,14 dB below the maximum full-load capacity of the codec. + +#### **Narrowband telephony testing** + +For testing of a GSM or 3G terminal supporting narrowband telephony, the system simulator shall use the AMR speech codec as defined in the 3GPP TS 26 series of specifications, at the source coding bit-rate of 12,2 kbit/s. + +#### **Wideband telephony testing** + +For testing of a GSM or 3G terminal supporting wideband telephony, the system simulator shall use the AMR-WB speech codec as defined in 3GPP TS 26 series of specifications, at the source coding bit-rate of 12,65 kbit/s. + +### **5.2.2 Direct digital processing approach** + +In this approach, the digital input/output bit-stream of the terminal connected through the radio link to the system simulator is operated upon directly. + +#### **Narrowband telephony testing** + +For testing of a GSM or 3G terminal supporting narrowband telephony, the system simulator shall use the AMR speech codec as defined in the 3GPP TS 26 series of specifications, at the source coding bit-rate of 12,2 kbit/s. + +#### **Wideband telephony testing** + +For testing of a GSM or 3G terminal supporting wideband telephony, the system simulator shall use the AMR-WB speech codec as defined in the 3GPP TS 26 series of specifications, at the source coding bit rate of 12,65 kbit/s. + +## **5.3 Accuracy of test equipment** + +Unless specified otherwise, the accuracy of measurements made by test equipment shall exceed the requirements defined in table 1a. + +**Table 1a: Test equipment measurement accuracy** + +| Item | Accuracy | +|-------------------------|----------------------------------------| +| Electrical Signal Power | $\pm 0,2$ dB for levels $\geq -50$ dBm | +| | $\pm 0,4$ dB for levels $< -50$ dBm | +| Sound pressure | $\pm 0,7$ dB | +| Time | $\pm 5\%$ | +| Frequency | $\pm 0,2\%$ | + +Unless specified otherwise, the accuracy of the signals generated by the test equipment shall exceed the requirements defined in table 1b. + +**Table 1b: Test equipment signal generation accuracy** + +| Quantity | Accuracy | +|-----------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------|---------------------------------| +| Sound pressure level at MRP | $\pm 1$ dB for 200 Hz to 4 kHz | +| | $\pm 3$ dB for 100 Hz to 200 Hz | +| | $\pm 3$ dB for 4 kHz to 8 kHz | +| Electrical excitation levels | $\pm 0,4$ dB (see note 1) | +| Frequency generation | $\pm 2\%$ (see note 2) | +| NOTE 1: Across the whole frequency range.
NOTE 2: When measuring sampled systems, it is advisable to avoid measuring at sub-multiples of the sampling frequency. There is a tolerance of $\pm 2\%$ on the generated frequencies, which may be used to avoid this problem, except for 4 kHz where only the -2% tolerance may be used. | | + +The measurements' results shall be corrected for the measured deviations from the nominal level. +The sound level measurement equipment shall conform to IEC 60651 Type 1. + +## 5.4 Test signals + +Unless stated otherwise, appropriate test signals for GSM/3G acoustic tests are generally described and defined in ITU-T Recommendation P.501 [22]. + +More information can be found in the test procedures described below. + +For testing the narrowband telephony service provided by the UE, the test signal used shall be band limited between 100 Hz and 4 kHz with a bandpass filter providing a minimum of 24 dB/oct. filter roll-off, when feeding into the receiving direction. + +For testing the wideband telephony service provided by the UE, the test signal used shall be band limited between 100 Hz and 8 kHz with a bandpass filter providing a minimum of 24 dB/oct. filter roll-off, when feeding into the receiving direction. + +The test signal levels are referred to the average level of the (band limited in receiving direction) test signal, averaged over the complete test sequence, unless specified otherwise. For real speech, the test signal levels are referred to the ITU-T P.56 [37] active speech level of the (band limited in receiving direction) test signal, calculated over the complete test sequence. + +## 5.5 Void + +### 5.5.1 Void + +### 5.5.2 Void + +# 6 Test conditions + +## 6.1 Environmental conditions + +### 6.1.1 Handset and headset terminals + +For handset and headset measurements the test room shall be practically free-field down to a lowest frequency of 275 Hz; the handset or headset, including the HATS, shall be totally within this free-field volume. + +Qualification of the test room may be performed using the method and limits for deviation from ideal free-field conditions described in either ISO 3745 Annex A (Table A.2), or ITU-T P. 340 §5.4 (Table 1). + +Alternatively, a test room may be used which meets the following two criteria: + +1. The relationship between the pressure at the mouth opening and that at 5,0 cm, 7,5 cm and 10 cm in front of the centre of the lip ring is within $\pm 0.5$ dB of that which exists in a known acoustic free-field. +2. The relationship between the pressure at the mouth opening and that at the Ear canal Entrance Point (EEP) at both the left and right ears of the HATS does not differ by more than $\pm 1$ dB from that which exists in a known free-field. + +The ambient noise level shall be less than -30 dBPa(A); for idle channel noise measurements the ambient noise level shall be less than -64dBPa(A). + +Echo measurements shall be conducted in realistic rooms with an ambient noise level $\leq -64$ dBPa(A). + +### 6.1.2 Hands-free terminals + +Hands-free terminals should generally be tested in their typical environment of application. Care must be taken that, e.g., noise levels are sufficiently low in order not to interfere with the measurements. + +For desktop hands-free terminals the appropriate requirements shall be taken from ITU-T Recommendation P.340. + +The broadband noise level shall not exceed -70 dBPa(A). The octave band noise level shall not exceed the values specified in Table 2. + +**Table 2: P.340 Noise level** + +| Center frequency (Hz) | Octave band pressure level (dBPa) | +|-----------------------|-----------------------------------| +| 63 | -45 | +| 125 | -60 | +| 250 | -65 | +| 500 | -65 | +| 1 000 | -65 | +| 2 000 | -65 | +| 4 000 | -65 | +| 8 000 | -65 | + +Echo measurements shall be conducted in realistic rooms with an ambient noise level $\leq -70$ dBPa(A). + +## 6.2 System simulator conditions + +The system simulator should provide an error-free radio connection to the UE under test. The default speech codec in narrowband, the AMR speech codec, shall be used at its highest bit-rate of 12,2 kbit/s. The default speech codec in wideband, the AMR-WB speech codec, shall be used at 12,65 kbit/s. Discontinuous Transmission (DTX) silence suppression shall be disabled for the purposes of GSM/3G acoustic testing. + +# --- 7 Narrowband telephony transmission performance test methods + +## 7.1 Applicability + +The test methods in this clause shall apply when testing a UE that is used to provide narrowband or wideband telephony, either as a stand-alone service, or as part of a multimedia service. + +## 7.2 Overall loss/loudness ratings + +### 7.2.1 General + +The SLR and RLR values for GSM or 3G networks apply up to the POI. However, the main determining factors are the characteristics of the UE, including the analogue to digital conversion (ADC) and digital to analogue conversion (DAC). In practice, it is convenient to specify loudness ratings to the Air Interface. For the normal case, where the GSM or 3G network introduce no additional loss between the Air Interface and the POI, the loudness ratings to the PSTN boundary (POI) will be the same as the loudness ratings measured at the Air Interface. + +### 7.2.2 Connections with handset UE + +#### 7.2.2.1 Sending loudness rating (SLR) + +- The test signal to be used for the measurements shall be the British-English single talk sequence described in ITU-T Recommendation P.501 [22]. The spectrum of the acoustic signal produced by the artificial mouth is calibrated under free-field conditions at the MRP. The test signal level shall be -4,7 dBPa measured at the MRP. The test signal level is calculated over the complete test signal sequence. +- The handset terminal is setup as described in clause 5. The sending sensitivity shall be calculated from each band of the 14 frequencies given in table 1 of ITU-T Recommendation P.79 [16], bands 4 to 17. For the calculation, the averaged measured level at the electrical reference point for each frequency band is referred to the averaged test signal level measured in each frequency band at the MRP. +- The sensitivity is expressed in terms of dBV/Pa and the SLR shall be calculated according to ITU-T Recommendation P.79 [16], formula (A-23b), over bands 4 to 17, using $m = 0,175$ and the sending weighting factors from ITU-T Recommendation P.79 [16], table 1. + +#### 7.2.2.2 Receiving loudness rating (RLR) + +- The test signal to be used for the measurements shall be the British-English single talk sequence described in ITU-T Recommendation P.501 [22]. The test signal level shall be -16 dBm0 measured at the digital reference point or the equivalent analogue point. The test signal level is calculated over the complete test signal sequence. +- The handset terminal is setup as described in clause 5. The receiving sensitivity shall be calculated from each band of the 14 frequencies given in table 1 of ITU-T Recommendation P.79 [16], bands 4 to 17. For the calculation, the averaged measured level at each frequency band is referred to the averaged test signal level measured in each frequency band. +- The sensitivity is expressed in terms of dBPa/V and the RLR shall be calculated according to ITU-T Recommendation P.79 [16], formula (A-23c), over bands 4 to 17, using $m = 0,175$ and the receiving weighting factors from table 1 of ITU-T Recommendation P.79 [16]. +- DRP-ERP correction is used. No leakage correction shall be applied. + +### 7.2.3 Connections with desktop and vehicle-mounted hands-free UE + +Vehicle-mounted hands-free UE should be tested within the vehicle (for totally integrated vehicle hands-free systems) or in a vehicle simulator, as described in 3GPP TS 03.58 [11]. + +Free-field measurements for vehicle-mounted hands-free are for further study. + +#### 7.2.3.1 Sending loudness rating (SLR) + +- a) The test signal to be used for the measurements shall be the British-English single talk sequence described in ITU-T Recommendation P.501 [22]. The spectrum of the acoustic signal produced by the artificial mouth is calibrated under free-field conditions at the MRP. The test signal level shall be -4,7 dBPa measured at the MRP. The test signal level is calculated over the complete test signal sequence. The broadband signal level is then adjusted to -28,7 dBPa at the HFRP or the HATS HFRP (as defined in ITU-T Recommendation P.581) and the spectrum is not altered. + +The spectrum at the MRP and the actual level at the MRP (measured in 1/3-octaves) are used as references to determine the sending sensitivity $S_{mj}$ . + +- b) The hands-free terminal is setup as described in clause 5. The sending sensitivity shall be calculated from each band of the 14 frequencies given in table 1 of ITU-T Recommendation P.79 [16], bands 4 to 17. For the calculation, the averaged measured level at the electrical reference point for each frequency band is referred to the averaged test signal level measured in each frequency band at the MRP. +- c) The sensitivity is expressed in terms of dBV/Pa and the SLR shall be calculated according to ITU-T Recommendation P.79 [16], formula (A-23b), over bands 4 to 17, using $m = 0,175$ and the sending weighting factors from ITU-T Recommendation P.79 [16], table 1. + +#### 7.2.3.2 Receiving Loudness Rating (RLR) + +- a) The test signal to be used for the measurements shall be the British-English single talk sequence described in ITU-T Recommendation P.501 [22]. The test signal level shall be -16 dBm0 measured at the digital reference point or the equivalent analogue point. The test signal level is calculated over the complete test signal sequence. +- b) The hands-free terminal is setup as described in clause 5. If a HATS is used, then it is free-field equalized as described in ITU-T Recommendation P.581. The equalized output signal of each artificial ear is power-averaged over the total duration of the analysis; the right and left artificial ear signals are voltage-summed for each 1/3-octave frequency band; these 1/3-octave band data are considered as the input signal to be used for calculations or measurements. The receiving sensitivity shall be calculated from each band of the 14 frequencies given in table 1 of ITU-T Recommendation P.79 [16], bands 4 to 17. + +For the calculation, the averaged measured level at each frequency band is referred to the averaged test signal level measured in each frequency band. + +- c) The sensitivity is expressed in terms of dBPa/V and the RLR shall be calculated according to ITU-T Recommendation P.79 [16], formula (A-23c), over bands 4 to 17, using $m = 0,175$ and the receiving weighting factors from table 1 of ITU-T Recommendation P.79 [16]. +- d) No leakage correction shall be applied. The hands-free correction, as described in ITU-T Recommendation P.340 shall be applied. To compute the receiving loudness rating (RLR) for a hands-free terminal (see also ITU-T Recommendation P.340), when using the combination of left and right artificial ear signals from the HATS, the $HFL_E$ has to be 8 dB instead of 14 dB. For further information see ITU-T Recommendation P.581. + +### 7.2.4 Connections with hand-held hands-free UE + +#### 7.2.4.1 Sending loudness rating (SLR) + +- a) The test signal to be used for the measurements shall be the British-English single talk sequence described in ITU-T Recommendation P.501 [22]. The spectrum of the acoustic signal produced by the artificial mouth is calibrated under free-field conditions at the MRP. The test signal level shall be -4,7 dBPa measured at the MRP. The test signal level is calculated over the complete test signal sequence. The broadband signal level is then + +adjusted to -28,7 dBPa at the HFRP or the HATS HFRP (as defined in ITU-T Recommendation P.581) and the spectrum is not altered. + +The spectrum at the MRP and the actual level at the MRP (measured in 1/3-octaves) are used as references to determine the sending sensitivity $S_{mj}$ . + +- b) The hands-free terminal is setup as described in clause 5. The sending sensitivity shall be calculated from each band of the 14 frequencies given in table 1 of ITU-T Recommendation P.79 [16], bands 4 to 17. For the calculation, the averaged measured level at the electrical reference point for each frequency band is referred to the averaged test signal level measured in each frequency band at the MRP. +- c) The sensitivity is expressed in terms of dBV/Pa and the SLR shall be calculated according to ITU-T Recommendation P.79 [16], formula (A-23b), over bands 4 to 17, using $m = 0,175$ and the sending weighting factors from ITU-T Recommendation P.79 [16], table 1. + +#### 7.2.4.2 Receiving loudness rating (RLR) + +- a) The test signal to be used for the measurements shall be the British-English single talk sequence described in ITU-T Recommendation P.501 [22]. The test signal level shall be -16 dBm0 measured at the digital reference point or the equivalent analogue point. The test signal level is calculated over the complete test signal sequence. +- b) The hands-free terminal is setup as described in clause 5. If a HATS is used, then it is free-field equalized as described in ITU-T Recommendation P.581. The equalized output signal of each artificial ear is power-averaged over the total duration of the analysis; the right and left artificial ear signals are voltage-summed for each 1/3-octave frequency band; these 1/3-octave band data are considered as the input signal to be used for calculations or measurements. The receiving sensitivity shall be calculated from each band of the 14 frequencies given in table 1 of ITU-T Recommendation P.79 [16], bands 4 to 17. + +For the calculation, the averaged measured level at each frequency band is referred to the averaged test signal level measured in each frequency band. + +- c) The sensitivity is expressed in terms of dBPa/V and the RLR shall be calculated according to ITU-T Recommendation P.79 [16], formula (A-23c), over bands 4 to 17, using $m = 0,175$ and the receiving weighting factors from table 1 of ITU-T Recommendation P.79 [16]. +- d) No leakage correction shall be applied. The hands-free correction as described in ITU-T Recommendation P.340 shall be applied. To compute the receiving loudness rating (RLR) for hands-free terminals (see also ITU-T Recommendation P.340), when using the combination of left and right artificial ear signals from the HATS, the $HFL_E$ has to be 8 dB instead of 14 dB. For further information see ITU-T Recommendation P.581. + +### 7.2.5 Connections with headset UE + +Same as for handset. + +## 7.3 Idle channel noise (handset and headset UE) + +For idle noise measurements in sending and receiving directions, care should be taken that only the noise is windowed out by the analysis and the result is not impaired by any remaining reverberation or by noise and/or interference from various other sources. Some examples are air-conducted or vibration-conducted noise from sources inside or outside the test chamber, disturbances from lights and regulators, mains supply induced noise including grounding issues, test system and system simulator inherent noise as well as radio interference from the UE to test equipment such as ear simulators, microphone amplifiers, etc. + +### 7.3.1 Sending + +The terminal should be configured to the test equipment as described in subclause 5.1. + +The environment shall comply with the conditions described in subclause 6.1. + +The noise level at the output of the SS is measured with psophometric weighting. The psophometric weighting filter is described in ITU-T Recommendation O.41. + +A test signal may have to be intermittently applied to prevent 'silent mode' operation of the MS. This is for further study. + +The measured part of the noise shall be 170,667 ms (which equals 8192 samples in a 48 kHz sample rate test system). The spectral distribution of the noise is analyzed with an 8k FFT using windowing with $\leq 0,1$ dB leakage for non bin-centered signals. This can be achieved with a window function commonly known as a "flat top window". Within the specified frequency range, the FFT bin that has the highest level is searched for; the level of this bin is the maximum level of a single frequency disturbance. + +To improve repeatability, the test sequence (optional activation followed by the noise level measurement) may be contiguously repeated one or more times. + +The total noise powers obtained from such repeats shall be averaged. The total result shall be $10 \cdot \log_{10}$ of this average in dB. + +The single frequency maximum powers obtained from such repeats shall be averaged. The total result shall be $10 \cdot \log_{10}$ of this average in dB. + +### 7.3.2 Receiving + +The terminal should be configured to the test equipment as described in subclause 5.1. + +The environment shall comply with the conditions described in subclause 6.1. + +A test signal may have to be intermittently applied to prevent 'silent mode' operation of the MS. This is for further study. + +The noise level shall be measured with A-weighting at the DRP with diffuse-field correction. The A-weighting filter is described in IEC 60651. + +The measured part of the noise shall be 170,667 ms (which equals 8192 samples in a 48 kHz sample rate test system). The spectral distribution of the noise is analyzed with an 8k FFT using windowing with $\leq 0,1$ dB leakage for non bin-centred signals. This can be achieved with a window function commonly known as a "flat top window". Within the specified frequency range, the FFT bin that has the highest level is searched for; the level of this bin is the maximum level of a single frequency disturbance. + +To improve repeatability, considering the test sequence (optional activation followed by the noise level measurement) may be contiguously repeated one or more times. + +The total noise powers obtained from such repeats shall be averaged. The total result shall be $10 \cdot \log_{10}$ of this average in dB. + +The single frequency maximum powers obtained from such repeats shall be averaged. The total result shall be $10 \cdot \log_{10}$ of this average in dB. + +## 7.4 Sensitivity/frequency characteristics + +### 7.4.1 Handset and headset UE sending + +- a) The test signal to be used for the measurements shall be the British-English single talk sequence described in ITU-T Recommendation P.501 [22]. The spectrum of the acoustic signal produced by the artificial mouth is calibrated under free-field conditions at the MRP. The test signal level shall be -4,7 dBPa measured at the MRP. The test signal level is calculated over the complete test signal sequence. +- b) The handset terminal is setup as described in clause 5. Measurements shall be made at 1/12-octave intervals as given by the R.40 series of preferred numbers in ISO 3 for frequencies from 100 Hz to 4 kHz inclusive. For the calculation, the averaged measured level at the electrical reference point for each frequency band is referred to the averaged test signal level measured in each frequency band at the MRP. +- c) The sensitivity is expressed in terms of dBV/Pa. + +### 7.4.2 Handset and headset UE receiving + +- The test signal to be used for the measurements shall be the British-English single talk sequence described in ITU-T Recommendation P.501 [22]. The test signal level shall be -16 dBm0 measured at the digital reference point or the equivalent analogue point. The test signal level is calculated over the complete test signal sequence. +- The handset terminal is setup as described in clause 5. Measurements shall be made at 1/12-octave intervals as given by the R.40 series of preferred numbers in ISO 3 for frequencies from 100 Hz to 4 kHz inclusive. For the calculation, the averaged measured level at each frequency band is referred to the averaged test signal level measured in each frequency band. +- The HATS is diffuse-field equalized. The sensitivity is expressed in terms of dBPa/V. Information about correction factors is available in ITU-T Recommendation P.57 [14]. + +Optionally, the measurements may be repeated with a 2 N and 13 N application force. For these test cases no normative values apply. + +### 7.4.3 Desktop and vehicle-mounted hands-free UE sending + +- The test signal to be used for the measurements shall be the British-English single talk sequence described in ITU-T Recommendation P.501 [22]. The spectrum of the acoustic signal produced by the artificial mouth is calibrated under free-field conditions at the MRP. The test signal level shall be -4,7 dBPa measured at the MRP. The test signal level is calculated over the complete test signal sequence. The broadband signal level is then adjusted to -28,7 dBPa at the HFRP or the HATS HFRP (as defined in ITU-T Recommendation P.581) and the spectrum is not altered. + +The spectrum at the MRP and the actual level at the MRP (measured in 1/3-octaves) are used as references to determine the sending sensitivity $S_{mj}$ . + +- The hands-free terminal is setup as described in clause 5. Measurements shall be made at 1/3-octave intervals as given by the R.40 series of preferred numbers in ISO 3 for frequencies from 100 Hz to 4 kHz inclusive. For the calculation, the averaged measured level at each frequency band is referred to the averaged test signal level measured in each frequency band. +- The sensitivity is expressed in terms of dBV/Pa. + +### 7.4.4 Desktop and vehicle-mounted hands-free UE receiving + +- The test signal to be used for the measurements shall be the British-English single talk sequence described in ITU-T Recommendation P.501 [22]. The test signal level shall be -16 dBm0 measured at the digital reference point or the equivalent analogue point. The test signal level is calculated over the complete test signal sequence. +- The hands-free terminal is setup as described in clause 5. If a HATS is used, then it is free-field equalized as described in ITU-T Recommendation P.581. The equalized output signal of each artificial ear is power-averaged over the total duration of the analysis; the right and left artificial ear signals are voltage-summed for each 1/3-octave frequency band; these 1/3-octave band data are considered as the input signal to be used for calculations or measurements. Measurements shall be made at 1/3-octave intervals as given by the R.40 series of preferred numbers in ISO 3 for frequencies from 100 Hz to 4 kHz inclusive. For the calculation the averaged measured level at each frequency band is referred to the averaged test signal level measured in each frequency band. +- The sensitivity is expressed in terms of dBPa/V. + +### 7.4.5 Hand-held hands-free UE sending + +- The test signal to be used for the measurements shall be the British-English single talk sequence described in ITU-T Recommendation P.501 [22]. The spectrum of the acoustic signal produced by the artificial mouth is calibrated under free-field conditions at the MRP. The test signal level shall be -4,7 dBPa measured at the MRP. The test signal level is calculated over the complete test signal sequence. The broadband signal level then is adjusted to -28,7 dBPa at the HFRP or the HATS HFRP (as defined in ITU-T Recommendation P.581) and the spectrum is not altered. + +The spectrum at the MRP and the actual level at the MRP (measured in 1/3-octaves) are used as reference to determine the sending sensitivity $S_{mJ}$ . + +- b) The hands-free terminal is setup as described in clause 5. Measurements shall be made at 1/3-octave intervals as given by the R.40 series of preferred numbers in ISO 3 for frequencies from 100 Hz to 4 kHz inclusive. For the calculation, the averaged measured level at each frequency band is referred to the averaged test signal level measured in each frequency band. +- c) The sensitivity is expressed in terms of dBV/Pa. + +### 7.4.6 Hand-held hands-free UE receiving + +- a) The test signal to be used for the measurements shall be the British-English single talk sequence described in ITU-T Recommendation P.501 [22]. The test signal level shall be -16 dBm0 measured at the digital reference point or the equivalent analogue point. The test signal level is calculated over the complete test signal sequence. +- b) The hands-free terminal is setup as described in clause 5. If a HATS is used, then it is free-field equalized as described in ITU-T Recommendation P.581. The equalized output signal of each artificial ear is power-averaged over the total duration of the analysis; the right and left artificial ear signals are voltage-summed for each 1/3-octave band frequency band; these 1/3-octave band data are considered as the input signal to be used for calculations or measurements. Measurements shall be made at 1/3-octave intervals as given by the R.40 series of preferred numbers in ISO 3 for frequencies from 100 Hz to 4 kHz inclusive. For the calculation, the averaged measured level at each frequency band is referred to the averaged test signal level measured in each frequency band. +- c) The sensitivity is expressed in terms of dBPa/V. + +## 7.5 Sidetone characteristics + +### 7.5.1 Connections with handset UE + +The test signal to be used for the measurements shall be the British-English single talk sequence described in ITU-T Recommendation P.501 [22]. The spectrum of the acoustic signal produced by the artificial mouth is calibrated under free-field conditions at the MRP. The test signal level shall be -4,7 dBPa measured at the MRP. The test signal level is calculated over the complete test signal sequence. + +#### 7.5.1.1 void + +#### 7.5.1.2 Connections with handset UE – HATS method + +The handset UE is setup as described in clause 5. The application force shall be 13 N on the Type 3.3 artificial ear. + +Where a user-operated volume control is provided, the measurements shall be carried out at the nominal setting of the volume control. In addition, the measurement is repeated at the maximum volume control setting. + +Measurements shall be made at 1/12-octave intervals as given by the R.40 series of preferred numbers in ISO 3 for frequencies from 100 Hz to 8 kHz inclusive. For the calculation, the averaged measured level at each frequency band (ITU-T Recommendation P.79 [16], table 4, bands 4 to 17) is referred to the averaged test signal level measured in each frequency band. + +The sidetone path loss ( $L_{meST}$ ), as expressed in dB, and the Sidetone Masking Rating (STMR), expressed in dB, shall be calculated from formula 5-1 of ITU-T Recommendation P.79 [16], using $m = 0.225$ and the weighting factors in table B.2 (unsealed condition) of ITU-T Recommendation P.79 [16]. No leakage correction ( $L_E$ ) shall be applied. DRP-ERP correction is used. + +In case the STMR is below the limit, the measurement shall be repeated with the electrical sidetone path disabled and both sets of results shall be reported. In case the STMR is below the limit also with the electrical sidetone path disabled, the result shall not be regarded as a failure. Disconnecting the call is normally disabling the electrical sidetone path; otherwise the UE can be switched off to enter the wanted state. + +### 7.5.2 Headset UE + +The test signal to be used for the measurements shall be the British-English single talk sequence described in ITU-T Recommendation P.501 [22]. The spectrum of the acoustic signal produced by the artificial mouth is calibrated under free-field conditions at the MRP. The test signal level shall be -4,7 dBPa measured at the MRP. The test signal level is calculated over the complete test signal sequence. + +Measurements shall be made at 1/12-octave intervals as given by the R.10 series of preferred numbers in ISO 3 for frequencies from 100 Hz to 8 kHz inclusive. For the calculation, the averaged measured level at each frequency band (ITU-T Recommendation P.79 [16], table 4, bands 4 to 17) is referred to the averaged test signal level measured in each frequency band. + +The sidetone path loss ( $L_{\text{meST}}$ ), as expressed in dB, shall be calculated from each band of the 14 frequencies given in table 1 of ITU-T Recommendation P.79 [16], bands 4 to 17. The STMR (in dB) shall be calculated from formula B-4 of ITU-T Recommendation P.79 [16], using $m = 0.225$ and the weighting factors in table B.2 (unsealed condition) of ITU-T Recommendation P.79 [16]. No leakage correction ( $L_E$ ) shall be applied. DRP-ERP correction is used. + +In case the STMR is below the limit, the measurement shall be repeated with the electrical sidetone path disabled and both sets of results shall be reported. In case the STMR is below the limit also with the electrical sidetone path disabled, the result shall not be regarded as a failure. Disconnecting the call is normally disabling the electrical sidetone path; otherwise the UE can be switched off to enter the wanted state. + +### 7.5.3 Hands-free UE (all categories) + +No requirement other than echo control. + +### 7.5.4 Sidetone delay for handset or headset + +The handset or headset terminal is setup as described in clause 5. + +The test signal is a CS-signal complying with ITU-T Recommendation P.501 using a PN-sequence with a length, $T$ , of 4 096 points (for a 48 kHz sample rate test system). The duration of the complete test signal is as specified in ITU-T Recommendation P.501. The level of the signal shall be -4,7 dBPa at the MRP. + +The cross-correlation function $\Phi_{xy}(\tau)$ between the input signal $S_x(t)$ generated by the test system in send direction and the output signal $S_y(t)$ measured at the artificial ear is calculated in the time domain: + +$$\Phi_{xy}(\tau) = \frac{1}{T} \int_{t=-\frac{T}{2}}^{\frac{T}{2}} S_x(t) \cdot S_y(t + \tau)$$ + +The measurement window, $T$ , shall be identical to the test signal period, $T$ , with the measurement window synchronized to the PN-sequence of the test signal. + +The sidetone delay is calculated from the envelope $E(\tau)$ of the cross-correlation function $\Phi_{xy}(\tau)$ . The first maximum of the envelope function occurs in correspondence with the direct sound produced by the artificial mouth; the second one occurs with a possible delayed sidetone signal. The difference between the two maxima corresponds to the sidetone delay. The envelope $E(\tau)$ is calculated by the Hilbert transformation $H\{xy(\tau)\}$ of the cross-correlation: + +$$H\{xy(\tau)\} = \sum_{u=-\infty}^{+\infty} \frac{\Phi_{xy}(u)}{\pi(\tau-u)}$$ + +$$E(\tau) = \sqrt{[\Phi_{xy}(\tau)]^2 + [H\{xy(\tau)\}]^2}$$ + +It is assumed that the measured sidetone delay is less than $T/2$ . + +## 7.6 Stability loss + +Where a user-controlled volume control is provided it is set to maximum. + +**Handset UE:** The handset is placed on a hard plane surface with the earpiece facing the surface. + +**Headset UE:** The requirement applies for the closest possible position between microphone and headset receiver within the intended wearing position. + +NOTE: Depending on the type of headset it may be necessary to repeat the measurement in different positions. + +**Hands-free UE (all categories):** No requirement other than echo loss. + +Before the actual test a training sequence consisting of the British-English single talk sequence described in ITU-T Recommendation P.501 [22] is applied. The training sequence level shall be -16 dBm0 in order to not overload the codec. + +The test signal is a PN-sequence complying with ITU-T Recommendation P.501 with a length of 4 096 points (for a 48 kHz sampling rate system) and a crest factor of 6 dB instead of 11 dB. The PN-sequence is generated as described in P.501 with $W(k)$ constant within the frequency range 200-4000 Hz and zero outside this range. The duration of the test signal is 250 ms. With an input signal of -3 dBm0, the attenuation from input to output of the system simulator shall be measured under the following conditions: + +- a) The handset or the headset, with the transmission circuit fully active, shall be positioned on a hard plane surface with at least 400 mm free space in all directions; the earpiece shall face towards the surface as shown in figure 15c; +- b) The headset microphone is positioned as close as possible to the receiver(s) within the intended wearing position; +- c) For a binaural headset, the receivers are placed symmetrically around the microphone. + +![Figure 15c: Test configuration for stability loss measurement on handset or headset UE. The diagram shows a top-down perspective view of a rectangular 'Clear Area' with dimensions 'min 500 mm' for width and 'min 400 mm' for depth. In the center of this area is a dotted oval labeled 'Area of Test Setup'. Below this, a side view shows a handset or headset resting on a hatched 'Surface min 500 mm'. A 'Clear Area' of 'min 400 mm' is indicated to the left of the device. The device itself is shown with an earpiece facing the surface and a microphone positioned near it.](e6fbffa8f0a33d829216b3e99c9e1103_img.jpg) + +The diagram illustrates the physical setup for testing. The top part shows a perspective view of a rectangular surface labeled "Clear Area". It has a width dimension of "min 500 mm" and a depth dimension of "min 400 mm". In the center is a dotted elliptical region labeled "Area of Test Setup". The bottom part shows a side profile of a handset resting face-down on a hatched surface. The surface width is labeled "Surface min 500 mm". To the left of the handset, a horizontal arrow indicates a "Clear Area" of "min 400 mm". + +Figure 15c: Test configuration for stability loss measurement on handset or headset UE. The diagram shows a top-down perspective view of a rectangular 'Clear Area' with dimensions 'min 500 mm' for width and 'min 400 mm' for depth. In the center of this area is a dotted oval labeled 'Area of Test Setup'. Below this, a side view shows a handset or headset resting on a hatched 'Surface min 500 mm'. A 'Clear Area' of 'min 400 mm' is indicated to the left of the device. The device itself is shown with an earpiece facing the surface and a microphone positioned near it. + +NOTE: All dimensions in mm. + +**Figure 15c. Test configuration for stability loss measurement on handset or headset UE** + +The attenuation from input to output of the system simulator shall be measured in the frequency range from 200 Hz to 4 kHz. The spectral distribution of the output signal is analysed with a 4k FFT (for a 48 kHz sample rate test system), + +thus the measured part of the output signal is 85.333 ms. To avoid leakage effects, the frequency resolution of the FFT must be the same as the frequency spacing of the PN-sequence. + +## 7.7 Acoustic echo control + +### 7.7.1 General + +The echo loss (EL) presented by the GSM or 3G networks at the POI should be at least 46 dB during single talk. This value takes into account the fact that UE is likely to be used in a wide range of noise environments. + +### 7.7.2 Acoustic echo control in a hands-free UE + +The hands-free UE is setup in a room with acoustic properties similar to a typical “office-type” room; a vehicle-mounted hands-free UE should be tested in a vehicle or vehicle simulator, as specified by the UE manufacturer (see also 3GPP TS 03.58 [11]). The ambient noise level $\leq 70$ dBPa(A). The attenuation from reference point input to reference point output shall be measured using the compressed real speech signal described in clause 7.3.3 of ITU-T P.501 Amendment 1 [33]. + +The TCLw is calculated according to ITU-T Recommendation G.122 [8], annex B, clause B.4 (trapezoidal rule). For the calculation, the averaged measured echo level at each frequency band is referred to the averaged test signal level measured in each frequency band. The first 17,0 s of the test signal (6 sentences) are discarded from the analysis to allow for convergence of the acoustic echo canceller. The analysis is performed over the remaining length of the test sequence (last 6 sentences). + +The test signal level shall be -10 dBm0. + +### 7.7.3 Acoustic echo control in handset UE + +The handset is set up according to clause 5. The ambient noise level shall be $\leq -64$ dBPa(A). The attenuation from the reference point input to reference point output shall be measured using the compressed real speech signal described in clause 7.3.3 of ITU-T P.501 Amendment 1 [33]. + +The TCLw is calculated according to ITU-T Recommendation G.122 [8], annex B, clause B.4 (trapezoidal rule). For the calculation, the averaged measured echo level at each frequency band is referred to the averaged test signal level measured in each frequency band. The first 17,0 s of the test signal (6 sentences) are discarded from the analysis to allow for convergence of the acoustic echo canceller. The analysis is performed over the remaining length of the test sequence (last 6 sentences). + +The test signal level shall be -10 dBm0. + +### 7.7.4 Acoustic echo control in a headset UE + +The headset is set up according to clause 5. The ambient noise level shall be $\leq -64$ dBPa(A). The attenuation from reference point input to reference point output shall be measured using the compressed real speech signal described in clause 7.3.3 of ITU-T P.501 Amendment 1 [33]. + +The TCLw is calculated according to ITU-T Recommendation G.122 [8], annex B, clause B.4 (trapezoidal rule). For the calculation, the averaged measured echo level at each frequency band is referred to the averaged test signal level measured in each frequency band. The first 17,0 s of the test signal (6 sentences) are discarded from the analysis to allow for convergence of the acoustic echo canceller. The analysis is performed over the remaining length of the test sequence (last 6 sentences). + +The test signal level shall be -10 dBm0. + +## 7.8 Distortion + +### 7.8.1 Sending distortion + +The handset, headset, or hands-free UE is setup as described in clause 5. + +The signal used is a sine-wave signal with a frequency of 1020 Hz. The sine-wave signal level shall be calibrated to the following RMS levels at the MRP: 5, 0, -4, 7, -10, -15, -20 dBPa. The test signals have to be applied in this sequence, i.e., from high levels down to low levels. + +The duration of the sine-wave signal is recommended to be 360 ms. The manufacturer shall be allowed to request tone lengths up to 1 s. The measured part of the signal shall be 170.667 ms (which equals $2 \cdot 4096$ samples in a 48 kHz sample rate test system). The times are selected to be relatively short in order to reduce the risk that the test tone is treated as a stationary signal. + +It is recommended that an optional activation signal be presented immediately preceding each test signal to ensure that the UE is in a typical state during measurement. An appropriate speech or speech-like activation signal shall be chosen from ITU-T Recommendations P.501 or P.50 [10]. A recommendation for the use of an activation signal as part of the measurement is defined in figure 16. The RMS level of the active parts of this activation signal is recommended to be equal to the subsequent test tone RMS level. In practice, certain types of processing may be impacted due to the introduction of the activation signal. The manufacturer shall be allowed to specify disabling of the activation signal. It shall be reported whether an activation signal was used or not, along with the characteristics of the activation signal, as specified by the manufacturer. + +The ratio of the signal to total distortion power of the signal output of the SS shall be measured with the psophometric noise weighting (see ITU-T Recommendations G.712, O.41 and O.132). The psophometric filter shall be normalized (0 dB gain) at 800 Hz as specified in ITU-T Recommendation O.41. The weighting function shall be applied to the total distortion component only (not to the signal component). + +For measurement of the total distortion component an octave-wide band-stop filter shall be applied to the signal to suppress the sine-wave signal and associated coding artefacts. The filter shall have a lower passband ending at $0.7071 \cdot f_s$ , and an upper passband starting at $1.4142 \cdot f_s$ , where $f_s$ is the frequency of the sine-wave signal. The passband ripple of the filter shall be $\leq 0.2$ dB. The attenuation of the band-stop filter at the sine-wave frequency shall be $\geq 60$ dB. Alternatively, the described characteristics can be implemented by an appropriate weighting on the spectrum obtained from an FFT. The total distortion component is defined as the measured signal within the frequency range 200 Hz to 4 kHz, after applying psophometric and stop filters (hence no correction for the lost power due to the stop filter, known as “bandwidth correction”, shall be applied). + +To improve repeatability, considering the variability introduced by speech coding and voice processing, the test sequence (activation signal followed by the test signal) may be contiguously repeated one or more times.. The single signal-to-total-distortion power ratios obtained from such repeats shall be averaged. The total result shall be $10 \cdot \log_{10}$ of this average in dB. + +![Figure 16: Recommended activation sequence and test signal. The figure shows a waveform plot with three distinct segments labeled 1, 2, and 3. Segment 1 is a voiced part of the activation signal. Segment 2 is an unvoiced part. Segment 3 is a pause. Below these, segments 4, 5, and 6 show the same signal but polarity inverted. Segment 7 shows the voiced part only. Segment 8 is a settling time. Segment 9 is the analysis period for the test tone.](3788d43ff8c1f359e46e9373a533432f_img.jpg) + +Figure 16: Recommended activation sequence and test signal. The figure shows a waveform plot with three distinct segments labeled 1, 2, and 3. Segment 1 is a voiced part of the activation signal. Segment 2 is an unvoiced part. Segment 3 is a pause. Below these, segments 4, 5, and 6 show the same signal but polarity inverted. Segment 7 shows the voiced part only. Segment 8 is a settling time. Segment 9 is the analysis period for the test tone. + +**Figure 16: Recommended activation sequence and test signal.** + +The activation signal consists of a “Bandlimited composite source signal with speech-like power density spectrum” signal according to ITU-T Recommendation P.501 with 48,62 ms voiced part (1), 200 ms unvoiced part (2) and 101,38 ms pause (3), followed by the same signal but polarity inverted (4, 5, 6), followed by the voiced part only (7). The pure test tone is applied and after 50 ms settling time (8), the analysis is made over the following 170,667 ms (9). + +NOTE 1: Void. + +NOTE 2: In order to ensure that the correct part of the signal is analyzed, the total delay of the terminal and SS may have to be determined prior to the measurement. + +NOTE 3: For hands-free terminals tested in environments defined in subclause 6.1.2, care should be taken that the reverberation in the test room, caused by the activation signal, does not affect the test results to an unacceptable degree, referring to subclause 5.3. + +### 7.8.2 Receiving + +The handset, headset, or hands-free UE is setup as described in clause 5. + +The signal used is a sine-wave signal with frequency of 1020 Hz. The signal shall be applied at the signal input of the SS at the following levels: 0, -3, -10, -16, -20, -30, -40, -45 dBm0. The test signals have to be applied in this sequence, i.e., from high levels down to low levels. + +The duration of the sine-wave signal is recommended to be 360 ms. The manufacturer shall be allowed to request tone lengths up to 1 s. The measured part of the signal shall be 170.667 ms (which equals $2 \cdot 4096$ samples in a 48 kHz sample rate test system). The times are selected to be relatively short in order to reduce the risk that the test tone is treated as a stationary signal. + +It is recommended that an optional activation signal be presented immediately preceding each test signal to ensure that the UE is in a typical state during measurement. An appropriate speech or speech-like activation signal shall be chosen from ITU-T Recommendations P.501 or P.50 [10]. A recommendation for the use of an activation signal as part of the measurement is defined in figure 17. The RMS level of the active parts of this activation signal is recommended to be equal to the subsequent test tone RMS level for low and medium test levels. To avoid saturation of the SS speech encoder, it is recommended for high test levels that the activation signal level be adjusted such that its peak level equals the peak level of the test tone. In practice, certain types of processing may be impacted due to the introduction of the activation signal. The manufacturer shall be allowed to specify disabling of the activation signal. It shall be reported whether an activation signal was used or not, along with the characteristics of the activation signal, as specified by the manufacturer. + +The ratio of the signal to total distortion power shall be measured at the applicable acoustic measurement point (DRP with diffuse-field correction for handset and headset modes; free field for hands-free modes) with psophometric noise weighting (see ITU-T Recommendations G.712, O.41 and 0.132). The psophometric filter shall be normalized to have 0 dB gain at 800 Hz as specified in ITU-T Recommendation O.41. The weighting function shall be applied to the total distortion component only (not to the signal component). + +For measurement of the total distortion component an octave-wide band-stop filter shall be applied to the signal to suppress the sine-wave signal and associated coding artefacts. The filter shall have a lower passband ending at $0,7071 \cdot f_s$ , and an upper passband starting at $1,4142 \cdot f_s$ , where $f_s$ is the frequency of the sine-wave signal. The passband ripple of the filter shall be $\leq 0.2$ dB. The attenuation of the band-stop filter at the sine-wave frequency shall be $\geq 60$ dB. Alternatively, the described characteristics can be implemented by an appropriate weighting on the spectrum obtained from an FFT. The total distortion component is defined as the measured signal within the frequency range 200 Hz to 4 kHz, after applying psophometric and stop filters (hence no correction for the lost power due to the stop filter, known as “bandwidth correction”, shall be applied). + +To improve repeatability, considering the variability introduced by speech coding and voice processing, the test sequence (activation signal followed by the test signal) may be contiguously repeated one or more times. The single signal-to-total-distortion power ratios obtained from such repeats shall be averaged. The total result shall be $10 \cdot \log_{10}$ of this average in dB. + +![Figure 17: Recommended activation sequence and test signal. The graph shows three distinct signal segments over time, labeled 1, 2, 3, 4, 5, 6, 7, 8, 9 on the x-axis. Segment 1 (activation signal) is a burst of speech-like noise. Segment 2 (test signal) is a continuous sine wave. Segment 3 (activation signal) is another burst of speech-like noise. The y-axis represents amplitude.](861fb7e32583067b653cee4688d49793_img.jpg) + +Figure 17: Recommended activation sequence and test signal. The graph shows three distinct signal segments over time, labeled 1, 2, 3, 4, 5, 6, 7, 8, 9 on the x-axis. Segment 1 (activation signal) is a burst of speech-like noise. Segment 2 (test signal) is a continuous sine wave. Segment 3 (activation signal) is another burst of speech-like noise. The y-axis represents amplitude. + +Figure 17: Recommended activation sequence and test signal. + +The activation signal consists of a “Bandlimited composite source signal with speech-like power density spectrum” signal according to ITU-T Recommendation P.501 with 48,62 ms voiced part (1), 200 ms unvoiced part (2) and 101,38 ms pause (3), followed by the same signal but polarity inverted (4, 5, 6), followed by the voiced part only (7). The pure test tone is applied and after 50 ms settling time (8), the analysis is made over the following 170,667 ms (9). + +**NOTE 1:** Void. + +**NOTE 2:** In order to ensure that the correct part of the signal is analyzed, the total delay of the terminal and SS may have to be determined prior to the measurement. + +**NOTE 3:** For hands-free terminals tested in environments defined in subclause 6.1.2, care should be taken that the reverberation in the test room, caused by the activation signal, does not affect the test results to an unacceptable degree, referring to subclause 5.3. + +## 7.9 Void + +## 7.10 Delay + +### 7.10.0 UE Delay Measurement Methodologies + +The sum of the UE delays in the sending and receiving directions ( $T_S+T_R$ ) shall be measured according to the methods described in clauses 7.10.1 and 7.10.2. In the event that the system simulator delays in send and/or receive directions are not stable between calls or cannot be accurately determined, the alternative method described in clause 7.10.3 may be used to obtain ( $T_S+T_R$ ) and the measured instability or inaccuracy observed when the methods described in 7.10.1 and 7.10.2 were performed shall be recorded in the test report. The test method(s) used and all results obtained shall also be recorded in the test report. + +### 7.10.1 Delay in sending direction (Handset UE) + +The handset terminal is setup as described in clause 5.1.1. + +The delay shall include all entities in sending direction from MRP to the POI, but shall exclude the delays introduced by the test equipment. + +![Diagram of entities contributing to delay in sending direction. It shows a signal path from Artificial Mouth (MRP) to UE Microphone, then to UE Signal Processing Entities, then to RF Transmission & Speech coder (with an antenna arrow), and finally to Test equipment (with an antenna arrow). A brace labeled T_s spans from the MRP to the point before Test equipment. A brace labeled T_TES spans the Test equipment block.](e038bf4fcea08f51944b4a2dd6e197a5_img.jpg) + +``` + +[Artificial Mouth (MRP)] --> [UE Microphone] --> [UE Signal Processing Entities] --> [RF Transmission & Speech coder] --> [Test equipment] +|____________________________________________________________________________________| |____________| + T_s T_TES + +``` + +Diagram of entities contributing to delay in sending direction. It shows a signal path from Artificial Mouth (MRP) to UE Microphone, then to UE Signal Processing Entities, then to RF Transmission & Speech coder (with an antenna arrow), and finally to Test equipment (with an antenna arrow). A brace labeled T\_s spans from the MRP to the point before Test equipment. A brace labeled T\_TES spans the Test equipment block. + +**Figure 17b1: Different entities contributing to the delay in sending direction** + +The delay in sending direction, measured from MRP to POI, is $T_s + T_{TES}$ . + +All test equipment delays, for the network type, codec type and bitrate used according to clause 5, (including radio access, speech codec, A/D and D/A conversions etc.) are included in $T_{TES}$ . The values used for testing (typical value considering variations due to interleaving etc.) as declared by the test equipment manufacturers shall be reported along with the measurement results. + +1. For the measurements, a Composite Source Signal (CSS) according to ITU-T Recommendation P.501 [22] is used. The pseudo random noise (pn)-part of the CSS has to be longer than the maximum expected delay. It is + +recommended to use a pn sequence of 32 k samples (with 48 kHz sampling rate). The test signal level is -4,7 dBPa at the MRP. + +- 2 The reference signal is the original signal (test signal). The setup of the handset/headset terminal is made corresponding to clause 5.1. +3. The delay is determined by cross-correlation analysis between the measured signal at the electrical access point and the original signal. The measurement is corrected by subtracting the test equipment delay $T_{TES}$ . +4. The delay is measured in ms and the maximum of the cross-correlation envelope is used for the determination. + +#### 7.10.1a Delay in sending direction (headset UE) + +The delay shall include all entities in sending direction from MRP to the POI, but shall exclude the delays introduced by the test equipment. + +![Diagram of the test setup for delay measurement in the sending direction with a headset connected via cable. The setup includes an Artificial Mouth (MRP) connected to a Headset Microphone via a Connection cable. The Headset Microphone is connected to a UE Signal Processing Entities block, which is connected to an RF Transmission & Speech coder block. The RF Transmission & Speech coder block is connected to a Test equipment block. The delay T_s is indicated by a bracket under the MRP, Headset Microphone, Connection cable, UE Signal Processing Entities, and RF Transmission & Speech coder blocks. The delay T_TES is indicated by a bracket under the Test equipment block.](6be06b7dc72bb42afcb3465394667c3b_img.jpg) + +``` + +graph LR + AM[Artificial Mouth] -- MRP --> HM[Headset Microphone] + HM -- Connection cable --> UE[UE Signal Processing Entities] + UE --> RF[RF Transmission & Speech coder] + RF --> TE[Test equipment] + + subgraph Ts [Delay Ts] + AM + HM + UE + RF + end + + subgraph TTES [Delay TTES] + TE + end + +``` + +Diagram of the test setup for delay measurement in the sending direction with a headset connected via cable. The setup includes an Artificial Mouth (MRP) connected to a Headset Microphone via a Connection cable. The Headset Microphone is connected to a UE Signal Processing Entities block, which is connected to an RF Transmission & Speech coder block. The RF Transmission & Speech coder block is connected to a Test equipment block. The delay T\_s is indicated by a bracket under the MRP, Headset Microphone, Connection cable, UE Signal Processing Entities, and RF Transmission & Speech coder blocks. The delay T\_TES is indicated by a bracket under the Test equipment block. + +**Figure 17b2: Different entities contributing to the delay in sending direction with a headset connected via cable** + +Note: The test setup only applies to headsets connected by wire. Wireless headsets (e.g. connected by Bluetooth) are currently out of scope. + +The test method is the same as for handset UE (clause 7.10.1). + +### 7.10.2 Delay in receiving direction (handset UE) + +The handset terminal is setup as described in clause 5. + +The delay shall include all entities in receiving direction from the POI to the DRP, but shall exclude the delays introduced by the test equipment. + +![Figure 17b3: Different entities contributing to the delay in receiving direction. The diagram shows a signal flow from left to right: Artificial Ear (DRP) -> UE Loudspeaker -> UE Signal Processing Entities -> RF Reception & Speech Decoder -> Test equipment. A bracket labeled T_r spans from the DRP to the RF Reception & Speech Decoder. Another bracket labeled T_TER spans from the Test equipment to the RF Reception & Speech Decoder.](6629e8a87e7552e2454b7c3e9f6d73a0_img.jpg) + +``` + + graph LR + AE[Artificial Ear +DRP] --- UEL[UE Loudspeaker] + UEL --- UESPE[UE Signal Processing Entities] + UESPE --- RFRSD[RF Reception & +Speech Decoder] + RFRSD --- TE[Test equipment] + +``` + +Figure 17b3: Different entities contributing to the delay in receiving direction. The diagram shows a signal flow from left to right: Artificial Ear (DRP) -> UE Loudspeaker -> UE Signal Processing Entities -> RF Reception & Speech Decoder -> Test equipment. A bracket labeled T\_r spans from the DRP to the RF Reception & Speech Decoder. Another bracket labeled T\_TER spans from the Test equipment to the RF Reception & Speech Decoder. + +**Figure 17b3: Different entities contributing to the delay in receiving direction** + +The delay in receiving direction, measured from POI to DRP, is $T_r + T_{TER}$ . + +All test equipment delays, for the network type, codec type and bitrate used according to clause 5, (including radio access, speech codec, A/D and D/A conversions etc.) are included in $T_{TER}$ . The values used for testing (typical value considering variations due to interleaving etc.) as declared by the test equipment manufacturers shall be reported along with the measurement results. + +1. For the measurements a Composite Source Signal (CSS) according to ITU-T Recommendation P.501 [22] is used. The pseudo random noise (pn)-part of the CSS has to be longer than the maximum expected delay. It is recommended to use a pn sequence of 32 k samples (with 48 kHz sampling rate). The test signal level is -16 dBm0 measured at the digital reference point or the equivalent analogue point. +2. The reference signal is the original signal (test signal). The setup of the handset/headset terminal is in correspondence to clause 5.1. +3. The delay is determined by cross-correlation analysis between the measured signal at the electrical access point and the original signal. The measurement is corrected by subtracting the test equipment delay $T_{TER}$ . +4. The delay is measured in ms and the maximum of the cross-correlation envelope is used for the determination. + +#### 7.10.2a Delay in receiving direction (headset UE) + +The delay shall include all entities in receiving direction from the POI to the DRP, but shall exclude the delays introduced by the test equipment. + +![Diagram for headset UE delay measurement. The signal flow is: Artificial Ear (DRP) -> Headset Loudspeaker -> Connection cable -> UE Signal Processing Entities -> RF Transmission & Speech coder -> Test equipment. A bracket labeled T_r spans from the DRP to the RF Transmission & Speech coder. Another bracket labeled T_TES spans from the Test equipment to the RF Transmission & Speech coder.](257c8341b41f1f4a287f27d33227974c_img.jpg) + +``` + + graph LR + AE[Artificial Ear +DRP] --- HL[Headset Loudspeaker] + HL --- CC[Connection cable] + CC --- UESPE[UE Signal Processing Entities] + UESPE --- RFT[RF Transmission & +Speech coder] + RFT --- TE[Test equipment] + +``` + +Diagram for headset UE delay measurement. The signal flow is: Artificial Ear (DRP) -> Headset Loudspeaker -> Connection cable -> UE Signal Processing Entities -> RF Transmission & Speech coder -> Test equipment. A bracket labeled T\_r spans from the DRP to the RF Transmission & Speech coder. Another bracket labeled T\_TES spans from the Test equipment to the RF Transmission & Speech coder. + +**Figure 17b4: Different entities contributing to the delay in receiving direction with a headset connected via cable** + +Note: The test setup only applies to headsets connected by wire. Wireless headsets (e.g. connected by Bluetooth) are currently out of scope. + +The test method is the same as for handset UE (clause 7.10.2). + +### 7.10.3 Delay in sending + receiving direction using “echo” method (handset UE) + +The mobile station delay shall include all entities from MRP to DRP (mouth-to-ear), but shall exclude the delays introduced by the test equipment and system simulator. + +![Diagram illustrating the delay components in the receiving direction for a handset UE using the 'echo' method. The diagram shows two parallel signal paths. The top path (sending direction) starts at the Artificial Mouth (MRP), passes through the Mobile Station Microphone, Mobile Station SND Signal Processing Entities, and RF Transmission & Speech coder. The bottom path (receiving direction) starts at the Artificial Ear (DRP), passes through the Mobile Station Loudspeaker, Mobile Station RCV Signal Processing Entities, and RF Reception & Speech Decoder. Both paths then enter the System Simulator. The top path goes through System Simulator RF Reception, and the bottom path goes through System Simulator RF Transmission. The delay from MRP to the start of the System Simulator is labeled T_S. The delay from the start of the System Simulator to the DRP is labeled T_R. The total delay through the System Simulator is labeled T_Ss.](f1091147d93cee4dfa88498610e395a7_img.jpg) + +Diagram illustrating the delay components in the receiving direction for a handset UE using the 'echo' method. The diagram shows two parallel signal paths. The top path (sending direction) starts at the Artificial Mouth (MRP), passes through the Mobile Station Microphone, Mobile Station SND Signal Processing Entities, and RF Transmission & Speech coder. The bottom path (receiving direction) starts at the Artificial Ear (DRP), passes through the Mobile Station Loudspeaker, Mobile Station RCV Signal Processing Entities, and RF Reception & Speech Decoder. Both paths then enter the System Simulator. The top path goes through System Simulator RF Reception, and the bottom path goes through System Simulator RF Transmission. The delay from MRP to the start of the System Simulator is labeled T\_S. The delay from the start of the System Simulator to the DRP is labeled T\_R. The total delay through the System Simulator is labeled T\_Ss. + +The delay measured from MRP to DRP is $(T_R + T_S + T_{Ss})$ . + +All system simulator delays, for the used network type, codec type and bitrate, (including radio access, speech codec, A/D and D/A conversions etc., added echo delay) are included in $T_{Ss}$ . The values used for testing (typical value considering variations due to interleaving etc.) as declared by the test equipment manufacturers shall be reported along with the measurement results. + +#### Method of measurement + +1. For the measurements a Composite Source Signal (CSS) according to ITU-T Recommendation P.501 [22] is used. It is recommended to use a pn sequence of 32 k samples (with 48 kHz sampling rate). The test signal level is -4.7 dBPa at the MRP. +2. The system simulator is configured for “loopback” or “echo” operation. In “loopback” or “echo” operation, the packets in the sending direction are routed to the receiving direction by the system simulator. +3. The reference signal is the original signal (test signal). The setup of the mobile station is in correspondence to clause 5.1. +4. The mouth-to-ear delay is determined by cross-correlation analysis between the measured signal at DRP and the original signal. The analysis window for the cross-correlation shall start at an instant $T > 50\text{ms}$ in order to discard the cross-correlation peaks corresponding to the direct acoustic path from mouth to ear and possible + +delayed sidetone signal. The measurement is corrected by subtracting the system simulator delay $T_{SS}$ to obtain the $T_R + T_S$ delay. + +5. The delay is measured in ms and the maximum of the cross-correlation envelope is used for the determination. + +#### 7.10.3a Delay in sending + receiving direction using “echo” method (headset UE) + +The mobile station delay shall include all entities from MRP to DRP (mouth-to-ear), but shall exclude the delays introduced by the test equipment and system simulator. + +The test method is the same as for handset UE (clause 7.10.3) + +## 7.11 Echo control characteristics + +### 7.11.1 Test set-up and test signals + +The device is set up according to clause 5. The ambient noise level shall be $\leq -64$ dBPa(A). + +The test shall be performed with the British-English “long” double-talk and conditioning speech sequences from ITU-T Recommendation P.501 [22], with the signals in the receiving direction band limited according to clause 5.4. + +A description of the test stimuli is presented in Table 2a and Table 2b. The test sequence is composed of an initial conditioning sequence of 23,5 s and a double talk sequence of 35 s. For the analysis, the double talk sequence is divided into two segments, a first double-talk sequence with single short near-end words (0 – 20 s), and a second double-talk sequence with continuous double talk (20 – 35 s). + +The sending speech during double-talk and the “near-end speech only” are recorded individually, with the “near-end speech only” sequence recorded with silence in the receiving direction. The time-alignment of the two recorded sequences is performed off-line during the analysis. + +**Table 2a: Test stimuli for recording of Echo Canceller operation** + +| | Conditioning | Single words (segment 1) and full sentence (segment 2) double talk | +|--------------------------------|-------------------------------------|--------------------------------------------------------------------| +| Far-end signal | FB_female_conditioning_seq_long.wav | FB_male_female_single-talk_seq.wav | +| Artificial mouth signal | FB_male_conditioning_seq_long.wav | FB_male_female_double-talk_seq.wav | + +**Table 2b: Test stimuli for reference "near-end speech only" recording.** + +| | Conditioning | Single words (segment 1) and full sentence (segment 2) double talk | +|--------------------------------|-------------------------------------|--------------------------------------------------------------------| +| Far-end signal | FB_female_conditioning_seq_long.wav | silence | +| Artificial mouth signal | FB_male_conditioning_seq_long.wav | FB_male_female_double-talk_seq.wav | + +The level of the signal of the artificial mouth shall be -4,7 dBPa measured at the MRP. In order to obtain a reproducible time alignment as seen by the UE, the artificial mouth signal shall be delayed by the amount of the receiving direction delay. For the purpose of this alignment, the receiving direction delay for handset and headset modes is defined from the system simulator input to the artificial ear. For hands-free modes, the downlink delay is defined from the system simulator input to the acoustic output from the UE loudspeaker. + +The level of the downlink signal shall be -16 dBm0 measured at the digital reference point or the equivalent analogue point. + +### 7.11.2 Test method + +The test method measures the duration of any level difference between the sending signal of a double-talk sequence (where the echo canceller has been exposed to simultaneous echo and near-end speech) and the sending signal of the + +same near-end speech only. The level difference is classified into eight categories according to Figure 17b5 and Table 2c, representing various degrees of “Full duplex operation”, “Near-end clipping”, and “Residual echo”. + +NOTE 1: The limits for specifying the categories in Figure 17b5 and Table 2c are provisional pending further analysis and validation. + +NOTE 2: The categories in Figure 17b5 and Table 2c are labelled in a functional order and the subjective impression of the respective categories is for further study. + +NOTE 3: To reduce potential issues associated with low-frequency test room noise, a [4th]-order high-pass filter with a cut-off frequency of [100] Hz can be applied before the level computation. + +![Figure 17b5: Classification of echo canceller performance. A graph showing Level difference [dB] on the y-axis and Duration [ms] on the x-axis. The y-axis has markers at 4, -4, and -15. The x-axis has markers at 25 and 150. The graph is divided into regions: E (top left), F (top middle), G (top right), A1 (middle left), A2 (middle middle), B (bottom left), C (bottom middle), and D (bottom right).](7a0663ba11ddcae06cc5a490f81a7243_img.jpg) + +Figure 17b5: Classification of echo canceller performance. A graph showing Level difference [dB] on the y-axis and Duration [ms] on the x-axis. The y-axis has markers at 4, -4, and -15. The x-axis has markers at 25 and 150. The graph is divided into regions: E (top left), F (top middle), G (top right), A1 (middle left), A2 (middle middle), B (bottom left), C (bottom middle), and D (bottom right). + +Figure 17b5: Classification of echo canceller performance + +Table 2c: Categories for echo canceller performance classification + +| Category | Level difference ( $\Delta L$ ) | Duration (D) | Description | +|----------|------------------------------------------------|-----------------------------------------|-----------------------------------------------| +| A1 | $-4 \text{ dB} \leq \Delta L < 4 \text{ dB}$ | | Full-duplex and full transparency | +| A2 | $-15 \text{ dB} \leq \Delta L < -4 \text{ dB}$ | | Full-duplex with level loss in Tx | +| B | $\Delta L < -15 \text{ dB}$ | $D < 25 \text{ ms}$ | Very short clipping | +| C | $\Delta L < -15 \text{ dB}$ | $25 \text{ ms} \leq D < 150 \text{ ms}$ | Short clipping resulting in loss of syllables | +| D | $\Delta L < -15 \text{ dB}$ | $D \geq 150 \text{ ms}$ | Clipping resulting in loss of words | +| E | $\Delta L \geq 4 \text{ dB}$ | $D < 25 \text{ ms}$ | Very short residual echo | +| F | $\Delta L \geq 4 \text{ dB}$ | $25 \text{ ms} \leq D < 150 \text{ ms}$ | Echo bursts | +| G | $\Delta L \geq 4 \text{ dB}$ | $D \geq 150 \text{ ms}$ | Continuous echo | + +A pseudo-code reference of the test method including test scripts and test-vectors is presented in clause C.3 and outlined in the following sub clauses. + +#### 7.11.2.1 Signal alignment + +For the analysis of the signal level difference, the send signal during double-talk and the near-end only signal are aligned using a correlation analysis as described in clause C.3.2. + +#### 7.11.2.2 Signal level computation and frame classification + +The analysis is based on the digital level measured with a meter according to IEC 61672 [38] with a time constant of 12,5 ms, sampled at 5 ms intervals corresponding to the evaluated frames. + +The “double-talk” frames are defined as the frames where both the far-end (receiving direction) signal includes active speech (extended with a hang-over period of 200 ms) and the near-end signal is composed of active speech. Active speech is defined to be detected using a speech level meter according to ITU-T P.56, and frames within -15.9 dB from the active speech level are classified as active speech frames. + +The “far-end single-talk adjacent to double-talk” frames are similarly defined using a speech level meter according to ITU-T P.56 as the frames with active far-end speech (extended with a hang-over period of 200 ms) and no active near-end speech (extended with a hang-over period of 200 ms). + +A reference implementation of the signal level computation and frame classification is presented in clause C.3.3. + +#### 7.11.2.3 Classification into categories + +The analysis and classification into the categories according to Figure 17b5 and Table 2c is performed according to the reference implementation described in clause C.3.4 and C.3.4. + +The frames are first categorized according to the level categories defined in Table 2c. To determine the durations, the amount of adjacent frames falling into the same level category is determined. + +The classification is then performed individually for the following situations: + +- frames classified as “double-talk” from segment 1 of the double-talk sequence (see clause 7.11.1) +- frames classified as “far-end single-talk adjacent to double-talk” from segment 1 of the double-talk sequence +- frames classified as “double-talk” from segment 2 of the double-talk sequence +- frames classified as “far-end single-talk adjacent to double-talk” from segment 2 of the double-talk sequence + +To determine the percentage values for each category (A1, A2, B, C, D, E, F, and G) within each situation, the number of frames falling into the respective category is divided by the total number of frames within the situation in question. + +To determine the averaged level difference of the frames for each category (A1, A2, B, C, D, E, F, and G) within each situation, the sum of the level difference (in dB) of the frames falling into the respective category is divided by the total number of frames within the situation in question. + +## 7.12 Quality (speech quality, noise intrusiveness) in the presence of ambient noise + +The speech quality in sending for narrowband systems is tested based on ETSI TS 103 106 [34]. This test method leads to three MOS-LQOn quality numbers: + +N-MOS-LQOn: Transmission quality of the background noise + +S-MOS-LQOn: Transmission quality of the speech + +G-MOS-LQOn: Overall transmission quality + +The test arrangement is given in clause 5.1.5. The measurement is conducted for 8 noise conditions as described in Table 2d. The measurements should be made in the same unique and dedicated call. The noise types shall be presented according to the order specified in Table 2d. + +**Table 2d: Noise conditions used for ambient noise simulation** + +| Description | File name | Duration | Level | Type | +|-----------------------------------|---------------------------------------|-----------------|--------------------------------|-------------| +| Recording in pub | Pub_Noise_binaural_V2 | 30 s | L: 75,0 dB(A)
R: 73,0 dB(A) | Binaural | +| Recording at pavement | Outside_Traffic_Road_binaural | 30 s | L: 74,9 dB(A)
R: 73,9 dB(A) | Binaural | +| Recording at pavement | Outside_Traffic_Crossroads_binaural | 20 s | L: 69,1 dB(A)
R: 69,6 dB(A) | Binaural | +| Recording at departure platform | Train_Station_binaural | 30 s | L: 68,2 dB(A)
R: 69,8 dB(A) | Binaural | +| Recording at the drivers position | Fullsize_Car1_130Kmh_binaural | 30 s | L: 69,1 dB(A)
R: 68,1 dB(A) | Binaural | +| Recording at sales counter | Cafeteria_Noise_binaural | 30 s | L: 68,4 dB(A)
R: 67,3 dB(A) | Binaural | +| Recording in a cafeteria | Mensa_binaural | 22 s | L: 63,4 dB(A)
R: 61,9 dB(A) | Binaural | +| Recording in business office | Work_Noise_Office_Callcenter_binaural | 30 s | L: 56,6 dB(A)
R: 57,8 dB(A) | Binaural | + +- 1) Before starting the measurements a proper conditioning sequence shall be used. The conditioning sequence shall be comprised of the four additional sentences 1- 4 described in ETSI TS 103 106 [34], applied to the beginning of the 16-sentence test sequence. The conditioning signal level is -1,7 dBPa at the MRP, measured as the active speech level according to ITU-T P.56 [37]. + +NOTE: The sequence of speech samples concatenated for the test signal, consisting of alternating talkers in the sending direction, reduces the overall test time but may represent an unrealistic behaviour for certain voice enhancement technologies. Alternative concatenations are for further study. + +- 2) The send speech signal consists of the 16 sentences of speech as described in ETSI TS 103 106 [34]. The test signal level is -1,7 dBPa at the MRP, measured as the active speech level according to ITU-T P.56 [37]. Three signals are required for the tests: + - The clean speech signal is used as the undisturbed reference (see ETSI TS 103 106 [34], ETSI EG 202 396-3 [36]). + - The speech plus undisturbed background noise signal is recorded at the terminal's microphone position using an omnidirectional measurement microphone with a linear frequency response between 50 Hz and 12 kHz. + - The send signal is recorded at the POI. +- 3) N-MOS-LQOn, S-MOS-LQOn and G-MOS-LQOn are calculated as described in ETSI TS 103 106 [34] on a per sentence basis and averaged over all 16 sentences. The results shall be reported as average and standard deviation. +- 4) The measurement is repeated for each ambient noise condition described in Table 2d. +- 5) The average of the results derived from all ambient noise types is calculated. + +# 8 Wideband telephony transmission performance test methods + +## 8.1 Applicability + +The test methods in this clause shall apply when testing a UE that is used to provide narrowband or wideband telephony, either as a stand-alone service, or as part of a multimedia service. + +The application force used to apply the handset against the artificial ear shall be $8 \pm 2$ N. For the headset case, the application of the headset shall comply with ITU-T Recommendation P.57 [14]. + +## 8.2 Overall loss/loudness ratings + +### 8.2.1 General + +The SLR and RLR values for GSM or 3G networks apply up to the POI. However, the main determining factors are the characteristics of the UE, including the analogue to digital conversion (ADC) and digital to analogue conversion (DAC). In practice, it is convenient to specify loudness ratings to the Air Interface. For the normal case, where the GSM or 3G network introduce no additional loss between the Air Interface and the POI, the loudness ratings to the PSTN boundary (POI) will be the same as the loudness ratings measured at the Air Interface. + +### 8.2.2 Connections with handset UE + +#### 8.2.2.1 Sending loudness rating (SLR) + +- The test signal to be used for the measurements shall be the British-English single talk sequence described in ITU-T Recommendation P.501 [22]. The spectrum of the acoustic signal produced by the artificial mouth is calibrated under free-field conditions at the MRP. The test signal level shall be $-4,7$ dBPa measured at the MRP. The test signal level is calculated over the complete test signal sequence. +- The handset terminal is setup as described in clause 5. The sending sensitivity shall be calculated from each band of the 20 frequencies given in table G.1 of ITU-T Recommendation P.79 Annex A [16], bands 1 to 20. For the calculation, the averaged measured level at the electrical reference point for each frequency band is referred to the averaged test signal level measured in each frequency band at the MRP. +- The sensitivity is expressed in terms of dBV/Pa and the SLR shall be calculated according to ITU-T Recommendation P.79 [16], formula (A-23b), over bands 1 to 20, using $m = 0,175$ and the sending weighting factors from ITU-T Recommendation P.79 Annex A [16], table A2. + +#### 8.2.2.2 Receiving loudness rating (RLR) + +- The test signal to be used for the measurements shall be the British-English single talk sequence described ITU-T Recommendation P.501 [22]. The test signal level shall be $-16$ dBm0 measured at the digital reference point or the equivalent analogue point. The test signal level is calculated over the complete test signal sequence. +- The handset terminal is setup as described in clause 5. The receiving sensitivity shall be calculated from each band of the 20 frequencies given in table A.2 of ITU-T Recommendation P.79 Annex A [16], bands 1 to 20. For the calculation, the averaged measured level at each frequency band is referred to the averaged test signal level measured in each frequency band. +- The sensitivity is expressed in terms of dBPa/V and the RLR shall be calculated according to ITU-T Recommendation P.79 [16], formula (A-23c), over bands 1 to 20, using $m = 0,175$ and the receiving weighting factors from table A.2 of ITU-T Recommendation P.79 Annex A [16]. +- DRP-ERP correction is applied. No leakage correction shall be applied. + +### 8.2.3 Connections with desktop and vehicle-mounted hands-free UE + +Vehicle-mounted hands-free UE should be tested within the vehicle (for the totally integrated vehicle hands-free systems) or in a vehicle simulator, as described in 3GPP TS 03.58 [11]. + +Free-field measurements for vehicle-mounted hands-free are for further study. + +#### 8.2.3.1 Sending loudness rating (SLR) + +- a) The test signal to be used for the measurements shall be the British-English single talk sequence described in ITU-T Recommendation P.501 [22]. The spectrum of the acoustic signal produced by the artificial mouth is calibrated under free-field conditions at the MRP. The test signal level shall be -4,7 dBPa measured at the MRP. The test signal level is calculated over the complete test signal sequence. The broadband signal level then is adjusted to -28,7 dBPa at the HFRP or the HATS HFRP (as defined in ITU-T Recommendation P.581) and the spectrum is not altered. + +The spectrum at the MRP and the actual level at the MRP (measured in 1/3-octaves) are used as references to determine the sending sensitivity $S_{mj}$ . + +- b) The hands-free terminal is setup as described in clause 5. The sending sensitivity shall be calculated from each band of the 20 frequencies given in table A.2 of ITU-T Recommendation P.79 Annex A [16], bands 1 to 20. For the calculation, the averaged measured level at the electrical reference point for each frequency band is referred to the averaged test signal level measured in each frequency band at the MRP. +- c) The sensitivity is expressed in terms of dBV/Pa and the SLR shall be calculated according to ITU-T Recommendation P.79 [16], formula (A-23b), over bands 1 to 20, using $m = 0,175$ and the sending weighting factors from ITU-T Recommendation P.79 Annex A [16], table A.2. + +#### 8.2.3.2 Receiving loudness rating (RLR) + +- a) The test signal to be used for the measurements shall be the British-English single talk sequence described in ITU-T Recommendation P.501 [22]. The test signal level shall be -16 dBm0 measured at the digital reference point or the equivalent analogue point. The test signal level is calculated over the complete test signal sequence. +- b) The hands-free terminal is setup as described in clause 5. If a HATS is used, then it is free-field equalized as described in ITU-T Recommendation P.581. The equalized output signal of each artificial ear is power-averaged over the total duration of the analysis; the right and left artificial ear signals are voltage-summed for each 1/3-octave frequency band; these 1/3-octave band data are considered as the input signal to be used for calculations or measurements. The receiving sensitivity shall be calculated from each band of the 20 frequencies given in table A.2 of ITU-T Recommendation P.79 Annex A [16], bands 1 to 20. + +For the calculation, the averaged measured level at each frequency band is referred to the averaged test signal level measured in each frequency band. + +- c) The sensitivity is expressed in terms of dBPa/V and the RLR shall be calculated according to ITU-T Recommendation P.79 [16], formula (A-23c), over bands 1 to 20, using $m = 0,175$ and the receiving weighting factors from table A.2 of ITU-T Recommendation P.79 Annex A [16]. +- d) No leakage correction shall be applied. The hands-free correction as described in ITU-T Recommendation P.340 shall be applied. To compute the receiving loudness rating (RLR) for a hands-free terminal (see also ITU-T Recommendation P.340), when using the combination of left and right artificial ear signals from the HATS, the $HFL_E$ has to be 8 dB instead of 14 dB. For further information see ITU-T Recommendation P.581. + +### 8.2.4 Connections with hand-held hands-free UE + +#### 8.2.4.1 Sending loudness rating (SLR) + +- a) The test signal to be used for the measurements shall be the British-English single talk sequence described in ITU-T Recommendation P.501 [22]. The spectrum of the acoustic signal produced by the artificial mouth is calibrated under free-field conditions at the MRP. The test signal level shall be -4,7 dBPa measured at the MRP. The test signal level is calculated over the complete test signal sequence. The broadband signal level then is adjusted to -28,7 dBPa at the HFRP or the HATS HFRP (as defined in P.581) and the spectrum is not altered. + +The spectrum at the MRP and the actual level at the MRP (measured in 1/3-octaves) are used as reference to determine the sending sensitivity $S_{mj}$ . + +- b) The hands-free terminal is setup as described in clause 5. The sending sensitivity shall be calculated from each band of the 20 frequencies given in table A.2 of ITU-T Recommendation P.79 Annex A [16], bands 1 to 20. For + +the calculation the averaged measured level at the electrical reference point for each frequency band is referred to the averaged test signal level measured in each frequency band at the MRP. + +- c) The sensitivity is expressed in terms of dBV/Pa and the SLR shall be calculated according to ITU-T Recommendation P.79 [16], formula (A-23b), over bands 1 to 20, using $m = 0,175$ and the sending weighting factors from ITU-T Recommendation P.79 Annex A [16], table A.2. + +#### 8.2.4.2 Receiving loudness rating (RLR) + +- a) The test signal to be used for the measurements shall be the British-English single talk sequence described in ITU-T Recommendation P.501 [22]. The test signal level shall be -16 dBm0 measured at the digital reference point or the equivalent analogue point. The test signal level is calculated over the complete test signal sequence. +- b) The hands-free terminal is setup as described in clause 5. If a HATS is used, then it is free-field equalized as described in ITU-T Recommendation P.581. The equalized output signal of each artificial ear is power-averaged over the total duration of the analysis; the right and left artificial ear signals are voltage-summed for each 1/3-octave frequency band; these 1/3-octave band data are considered as the input signal to be used for calculations or measurements. The receiving sensitivity shall be calculated from each band of the 20 frequencies given in table A.2 of ITU-T Recommendation P.79 Annex A [16], bands 1 to 20. + +For the calculation, the averaged measured level at each frequency band is referred to the averaged test signal level measured in each frequency band. + +- c) The sensitivity is expressed in terms of dBPa/V and the RLR shall be calculated according to ITU-T Recommendation P.79 [16], formula (A-23c), over bands 1 to 20, using $m = 0,175$ and the receiving weighting factors from table A.2 of ITU-T Recommendation P.79 Annex A [16]. +- d) No leakage correction shall be applied. The hands-free correction as described in ITU-T Recommendation P.340 shall be applied. To compute the receiving loudness rating (RLR) for hands-free terminals (see also ITU-T Recommendation P.340) when using the combination of left and right artificial ear signals from the HATS the $HFL_E$ has to be 8 dB, instead of 14 dB. For further information see ITU-T Recommendation P.581. + +### 8.2.5 Connections with headset UE + +Same as for handset. + +## 8.3 Idle channel noise (handset and headset UE) + +For idle noise measurements in sending and receiving directions, care should be taken that only the noise is windowed out by the analysis and the result is not impaired by any remaining reverberation or by noise and/or interference from various other sources. Some examples are air-conducted or vibration-conducted noise from sources inside or outside the test chamber, disturbances from lights and regulators, mains supply induced noise including grounding issues, test system and system simulator inherent noise as well as radio interference from the UE to test equipment such as ear simulators, microphone amplifiers, etc. + +### 8.3.1 Sending + +The terminal should be configured to the test equipment as described in subclause 5.1. + +The environment shall comply with the conditions described in subclause 6.1. + +The noise level at the output of the SS is measured with A-weighting. The A-weighting filter is described in IEC 60651. + +A test signal may have to be intermittently applied to prevent 'silent mode' operation of the MS. This is for further study. + +The measured part of the noise shall be 170,667 ms (which equals 8192 samples in a 48 kHz sample rate test system). The spectral distribution of the noise is analyzed with an 8k FFT using windowing with $\leq 0,1$ dB leakage for non bin-centered signals. This can be achieved with a window function commonly known as a "flat top window". Within the specified frequency range, the FFT bin that has the highest level is searched for; the level of this bin is the maximum level of a single frequency disturbance. + +To improve repeatability, the test sequence (optional activation followed by the noise level measurement) may be contiguously repeated one or more times. + +The total noise powers obtained from such repeats shall be averaged. The total result shall be $10 \cdot \log_{10}$ of this average in dB. + +The single frequency maximum powers obtained from such repeats shall be averaged. The total result shall be $10 \cdot \log_{10}$ of this average in dB. + +### 8.3.2 Receiving + +The terminal should be configured to the test equipment as described in subclause 5.1. + +The environment shall comply with the conditions described in subclause 6.1. + +A test signal may have to be intermittently applied to prevent 'silent mode' operation of the MS. This is for further study. + +The noise shall be measured with A-weighting at the DRP with diffuse-field correction. The A-weighting filter is described in IEC 60651. + +The measured part of the noise shall be 170,667 ms (which equals 8192 samples in a 48 kHz sample rate test system). The spectral distribution of the noise is analyzed with an 8k FFT using windowing with $\leq 0,1$ dB leakage for non bin-centered signals. This can be achieved with a window function commonly known as a "flat top window". Within the specified frequency range the FFT bin that has the highest level is searched for; the level of this bin is the maximum level of a single frequency disturbance. + +To improve repeatability, the test sequence (optional activation followed by the noise level measurement) may be contiguously repeated one or more times. + +The total noise powers obtained from such repeats shall be averaged. The total result shall be $10 \cdot \log_{10}$ of this average in dB. + +The single frequency maximum powers obtained from such repeats shall be averaged. The total result shall be $10 \cdot \log_{10}$ of this average in dB. + +## 8.4 Sensitivity/frequency characteristics + +### 8.4.1 Handset and headset UE sending + +The headset case is similar to the handset one, except for the application force. + +- The test signal to be used for the measurements shall be the British-English single talk sequence described in ITU-T Recommendation P.501 [22]. The spectrum of the acoustic signal produced by the artificial mouth is calibrated under free-field conditions at the MRP. The test signal level shall be $-4,7$ dBPa measured at the MRP. The test signal level is calculated over the complete test signal sequence. +- The handset terminal is setup as described in clause 5. Measurements shall be made at 1/12-octave intervals as given by the R.40 series of preferred numbers in ISO 3 for frequencies from 100 Hz to 8 kHz inclusive. For the calculation, the averaged measured level at the electrical reference point for each frequency band is referred to the averaged test signal level measured in each frequency band at the MRP. +- The sensitivity is expressed in terms of dBV/Pa. + +### 8.4.2 Handset and headset UE receiving + +- The test signal to be used for the measurements shall be the British-English single talk sequence described in ITU-T Recommendation P.501 [22]. The test signal level shall be $-16$ dBm0 measured at the digital reference point or the equivalent analogue point. The test signal level is calculated over the complete test signal sequence. +- The handset terminal is setup as described in clause 5. Measurements shall be made at 1/12-octave intervals as given by the R.40 series of preferred numbers in ISO 3 for frequencies from 100 Hz to 8 kHz inclusive. For the + +calculation, the averaged measured level at each frequency band is referred to the averaged test signal level measured in each frequency band. + +- c) The HATS is diffuse-field equalized. The sensitivity is expressed in terms of dBPa/V. Information about correction factors is available in ITU-T Recommendation P.57 [14]. + +Optionally, the measurements may be repeated with 2 N and 13 N application force. For these test cases no normative values apply. + +### 8.4.3 Desktop and vehicle-mounted hands-free UE sending + +- a) The test signal to be used for the measurements shall be the British-English single talk sequence described in ITU-T Recommendation P.501 [22]. The spectrum of the acoustic signal produced by the artificial mouth is calibrated under free-field conditions at the MRP. The test signal level shall be $-4,7$ dBPa measured at the MRP. The test signal level is calculated over the complete test signal sequence. The broadband signal level is then adjusted to $-28,7$ dBPa at the HFRP or the HATS HFRP (as defined in ITU-T Recommendation P.581) and the spectrum is not altered. + +The spectrum at the MRP and the actual level at the MRP (measured in 1/3-octaves) are used as references to determine the sending sensitivity $S_{mj}$ . + +- b) The hands-free terminal is setup as described in clause 5. Measurements shall be made at 1/3-octave intervals as given by the R.40 series of preferred numbers in ISO 3 for frequencies from 100 Hz to 8 kHz inclusive. For the calculation the averaged measured level at each frequency band is referred to the averaged test signal level measured in each frequency band. +- c) The sensitivity is expressed in terms of dBV/Pa. + +### 8.4.4 Desktop and vehicle-mounted hands-free UE receiving + +- a) The test signal to be used for the measurements shall be the British-English single talk sequence described in ITU-T Recommendation P.501 [22]. The test signal level shall be $-16$ dBm0 measured at the digital reference point or the equivalent analogue point. The test signal level is calculated over the complete test signal sequence. +- b) The hands-free terminal is setup as described in clause 5. If a HATS is used, then it is free-field equalized as described in ITU-T Recommendation P.581. The equalized output signal of each artificial ear is power-averaged over the total duration of the analysis; the right and left artificial ear signals are voltage-summed for each 1/3-octave frequency band; these 1/3-octave band data are considered as the input signal to be used for calculations or measurements. Measurements shall be made at 1/3-octave intervals as given by the R.40 series of preferred numbers in ISO 3 for frequencies from 100 Hz to 8 kHz inclusive. For the calculation, the averaged measured level at each frequency band is referred to the averaged test signal level measured in each frequency band. +- c) The sensitivity is expressed in terms of dBPa/V. + +### 8.4.5 Hand-held hands-free UE sending + +- a) The test signal to be used for the measurements shall be the British-English single talk sequence described in ITU-T Recommendation P.501 [22]. The spectrum of the acoustic signal produced by the artificial mouth is calibrated under free-field conditions at the MRP. The test signal level shall be $-4,7$ dBPa measured at the MRP. The test signal level is calculated over the complete test signal sequence. The broadband signal level is then adjusted to $-28,7$ dBPa at the HFRP or the HATS HFRP (as defined in ITU-T Recommendation P.581) and the spectrum is not altered. + +The spectrum at the MRP and the actual level at the MRP (measured in 1/3-octaves) are used as reference to determine the sending sensitivity $S_{mj}$ . + +- b) The hands-free terminal is setup as described in clause 5.1.3.3. Measurements shall be made at 1/3-octave intervals as given by the R.40 series of preferred numbers in ISO 3 for frequencies from 100 Hz to 8 kHz inclusive. For the calculation, the averaged measured level at each frequency band is referred to the averaged test signal level measured in each frequency band. +- c) The sensitivity is expressed in terms of dBV/Pa. + +### 8.4.6 Hand-held hands-free UE receiving + +- a) The test signal to be used for the measurements shall be the British-English single talk sequence described in ITU-T Recommendation P.501 [22]. The test signal level shall be $-16$ dBm0 measured at the digital reference point or the equivalent analogue point. The test signal level is calculated over the complete test signal sequence. +- b) The hands-free terminal is setup as described in clause 5. If a HATS is used, then it is free-field equalized as described in ITU-T Recommendation P.581. The equalized output signal of each artificial ear is power-averaged over the total duration of the analysis; the right and left artificial ear signals are voltage-summed for each $1/3$ -octave band frequency band; these $1/3$ -octave band data are considered as the input signal to be used for calculations or measurements. Measurements shall be made at $1/3$ -octave intervals as given by the R.40 series of preferred numbers in ISO 3 for frequencies from 100 Hz to 8 kHz inclusive. For the calculation, the averaged measured level at each frequency band is referred to the averaged test signal level measured in each frequency band. +- c) The sensitivity is expressed in terms of dBPa/V. + +## 8.5 Sidetone characteristics + +### 8.5.1 Connections with handset UE + +The test signal to be used for the measurements shall be the British-English single talk sequence described in ITU-T Recommendation P.501 [22]. The spectrum of the acoustic signal shall be produced by the HATS. The test signal level shall be $-4,7$ dBPa measured at the MRP. The test signal level is calculated over the complete test signal sequence. + +The handset UE is set up as described in clause 5. The application force shall be 13 N on the Type 3.3 artificial ear. + +Where a user-operated volume control is provided, the measurements shall be carried out at the nominal setting of the volume control. In addition the measurement is repeated at the maximum volume control setting. + +Measurements shall be made at $1/12$ -octave intervals as given by the R.40 series of preferred numbers in ISO 3 for frequencies from 100 Hz to 8 kHz inclusive. For the calculation, the averaged measured level at each frequency band (ITU-T Recommendation P.79 [16], table 4, bands 1 to 20) is referred to the averaged test signal level measured in each frequency band. + +The sidetone path loss ( $L_{meST}$ ), as expressed in dB, and the Sidetone Masking Rating (STMR), expressed in dB, shall be calculated from formula 5-1 of ITU-T Recommendation P.79 [16], using $m = 0.225$ and the weighting factors in table B2 (unsealed condition) of ITU-T Recommendation P.79 [16]. No leakage correction ( $L_E$ ) shall be applied. DRP-ERP correction is used. + +In case the STMR is below the limit, the measurement shall be repeated with the electrical sidetone path disabled and both sets of results shall be reported. In case the STMR is below the limit also with the electrical sidetone path disabled, the result shall not be regarded as a failure. Disconnecting the call is normally disabling the electrical sidetone path; otherwise the UE can be switched off to enter the wanted state. + +### 8.5.2 Headset UE + +The test signal to be used for the measurements shall be the British-English single talk sequence described in ITU-T Recommendation P.501 [22]. The spectrum of the acoustic signal produced by the artificial mouth is calibrated under free-field conditions at the MRP. The test signal level shall be $-4,7$ dBPa measured at the MRP. The test signal level is calculated over the complete test signal sequence. + +Measurements shall be made at $1/12$ -octave intervals as given by the R.40 series of preferred numbers in ISO 3 for frequencies from 100 Hz to 8 kHz inclusive. For the calculation, the averaged measured level at each frequency band (ITU-T Recommendation P.79 [16], table 4, bands 1 to 20) is referred to the averaged test signal level measured in each frequency band. + +The sidetone path loss ( $L_{meST}$ ), as expressed in dB, shall be calculated from each band of the 20 frequencies given in table G.1 of ITU-T Recommendation P.79 Annex A [16], bands 1 to 20. The STMR (in dB) shall be calculated from formula B-4 of ITU-T Recommendation P.79 [16], using $m = 0.225$ and the weighting factors in table B.2 (unsealed condition) of ITU-T Recommendation P.79 [16]. No leakage correction ( $L_E$ ) shall be applied. DRP-ERP correction is used. + +In case the STMR is below the limit, the measurement shall be repeated with the electrical sidetone path disabled and both sets of results shall be reported. In case the STMR is below the limit also with the electrical sidetone path disabled, the result shall not be regarded as a failure. Disconnecting the call is normally disabling the electrical sidetone path; otherwise the UE can be switched off to enter the wanted state. + +### 8.5.3 Hands-free UE (all categories) + +No requirement other than echo control. + +### 8.5.4 Sidetone delay for handset or headset + +The handset or headset terminal is setup as described in clause 5. + +The test signal is a CS-signal complying with ITU-T Recommendation P.501 using a PN-sequence with a length, $T$ , of 4 096 points (for a 48 kHz sample rate test system). The duration of the complete test signal is as specified in ITU-T Recommendation P.501. The level of the signal shall be -4,7 dBPa at the MRP. + +The cross-correlation function $\Phi_{xy}(\tau)$ between the input signal $S_x(t)$ generated by the test system in send direction and the output signal $S_y(t)$ measured at the artificial ear is calculated in the time domain: + +$$\Phi_{xy}(\tau) = \frac{1}{T} \int_{t=-\frac{T}{2}}^{\frac{T}{2}} S_x(t) \cdot S_y(t + \tau) \quad (1)$$ + +The measurement window, $T$ , shall be identical to the test signal period, $T$ , with the measurement window synchronized to the PN-sequence of the test signal. + +The sidetone delay is calculated from the envelope $E(\tau)$ of the cross-correlation function $\Phi_{xy}(\tau)$ . The first maximum of the envelope function occurs in correspondence with the direct sound produced by the artificial mouth; the second one occurs with a possible delayed sidetone signal. The difference between the two maxima corresponds to the sidetone delay. The envelope $E(\tau)$ is calculated by the Hilbert transformation $H\{\Phi_{xy}(\tau)\}$ of the cross-correlation: + +$$H\{\Phi_{xy}(\tau)\} = \sum_{u=-\infty}^{+\infty} \frac{\Phi_{xy}(u)}{\pi(\tau-u)} \quad (2)$$ + +$$E(\tau) = \sqrt{[\Phi_{xy}(\tau)]^2 + [H\{\Phi_{xy}(\tau)\}]^2} \quad (3)$$ + +It is assumed that the measured sidetone delay is less than $T/2$ . + +## 8.6 Stability loss + +Where a user-controlled volume control is provided it is set to maximum. + +**Handset UE:** The handset is placed on a hard plane surface with the earpiece facing the surface. + +**Headset UE:** The requirement applies for the closest possible position between microphone and headset receiver within the intended wearing position. + +NOTE: Depending on the type of headset it may be necessary to repeat the measurement in different positions. + +**Hands-free UE (all categories):** No requirement other than echo loss. + +Before the actual test a training sequence consisting of the British-English single talk sequence described in ITU-T Recommendation P.501 [22] is applied. The training sequence level shall be -16 dBm0 in order to not overload the codec. + +The test signal is a PN-sequence complying with ITU-T Recommendation P.501 with a length of 4 096 points (for a 48 kHz sampling rate system) and a crest factor of 6 dB instead of 11 dB. The PN-sequence is generated as described in P.501 with $W(k)$ constant within the frequency range 100-8000 Hz and zero outside this range. The duration of the test + +signal is 250 ms. With an input signal of -3 dBm0, the attenuation from input to output of the system simulator shall be measured under the following conditions: + +- a) The handset or the headset, with the transmission circuit fully active, shall be positioned on a hard plane surface with at least 400 mm free space in all directions. The earpiece shall face towards the surface as shown in figure 17c; +- b) The headset microphone is positioned as close as possible to the receiver(s) within the intended wearing position; +- c) For a binaural headset, the receivers are placed symmetrically around the microphone. + +![Figure 17c: Test configuration for stability loss measurement on handset or headset UE. The diagram consists of two parts. The top part shows a perspective view of a rectangular 'Clear Area' on a surface. The length of the clear area is labeled 'min 500 mm' and its width is labeled 'min 400 mm'. In the center of this area is a dotted oval labeled 'Area of Test Setup'. The bottom part shows a side view of a handset or headset resting on a hatched surface. The handset's earpiece is facing the surface. A horizontal dimension line indicates a 'Clear Area' of 'min 400 mm' from the handset to the edge of the surface. The surface itself is labeled 'Surface min 500 mm'.](c1a9218b182545d694b285ed2800b1b7_img.jpg) + +Figure 17c: Test configuration for stability loss measurement on handset or headset UE. The diagram consists of two parts. The top part shows a perspective view of a rectangular 'Clear Area' on a surface. The length of the clear area is labeled 'min 500 mm' and its width is labeled 'min 400 mm'. In the center of this area is a dotted oval labeled 'Area of Test Setup'. The bottom part shows a side view of a handset or headset resting on a hatched surface. The handset's earpiece is facing the surface. A horizontal dimension line indicates a 'Clear Area' of 'min 400 mm' from the handset to the edge of the surface. The surface itself is labeled 'Surface min 500 mm'. + +NOTE: All dimensions in mm. + +**Figure 17c. Test configuration for stability loss measurement on handset or headset UE** + +The attenuation from input to output shall be measured in the frequency range from 100 Hz to 8 kHz. The spectral distribution of the output signal is analysed with a 4k FFT (for a 48 kHz sample rate test system), thus the measured part of the output signal is 85,333 ms. To avoid leakage effects the frequency resolution of the FFT must be the same as the frequency spacing of the PN-sequence. + +## 8.7 Acoustic echo control + +### 8.7.1 General + +The echo loss (EL) presented by the GSM or 3G networks at the POI should be at least 46 dB during single talk. This value takes into account the fact that UE is likely to be used in a wide range of noise environments. + +### 8.7.2 Acoustic echo control in a hands-free UE + +The hands-free UE is setup in a room with acoustic properties similar to a typical “office-type” room; a vehicle-mounted hands-free UE should be tested in a vehicle or vehicle simulator, as specified by the UE manufacturer (see also 3GPP TS 03.58 [11]). The ambient noise level shall be $\leq -70$ dBPa(A). The attenuation from reference point input to reference point output shall be measured using the compressed real speech signal described in clause 7.3.3 of ITU-T P.501 Amendment 1 [33]. + +The TCLw is calculated according to ITU-T Recommendation G.122 [8], annex B, clause B.4 (trapezoidal rule) but using the frequency range of 300 Hz to 6 700 Hz (instead of 300 Hz to 3 400 Hz). For the calculation, the averaged measured echo level at each frequency band is referred to the averaged test signal level measured in each frequency band. The first 17,0 s of the test signal (6 sentences) are discarded from the analysis to allow for convergence of the acoustic echo canceller. The analysis is performed over the remaining length of the test sequence (last 6 sentences). + +The test signal level shall be -10 dBm0. + +### 8.7.3 Acoustic echo control in a handset UE + +The handset is set up according to clause 5. The ambient noise level shall be $\leq -64$ dBPa(A). The attenuation from the reference point input to reference point output shall be measured using the compressed real speech signal described in clause 7.3.3 of ITU-T P.501 Amendment 1 [33]. + +The TCLw is calculated according to ITU-T Recommendation G.122 [8], annex B, clause B.4 (trapezoidal rule) but using the frequency range of 300 Hz to 6 700 Hz (instead of 300 Hz to 3 400 Hz). For the calculation, the averaged measured echo level at each frequency band is referred to the averaged test signal level measured in each frequency band. The first 17,0 s of the test signal (6 sentences) are discarded from the analysis to allow for convergence of the acoustic echo canceller. The analysis is performed over the remaining length of the test sequence (last 6 sentences). + +The test signal level shall be -10 dBm0. + +### 8.7.4 Acoustic echo control in a headset UE + +The headset is set up according to clause 5. The ambient noise level shall be $\leq -64$ dBPa(A). The attenuation from the reference point input to reference point output shall be measured using the compressed real speech signal described in clause 7.3.3 of ITU-T P.501 Amendment 1 [33]. + +The TCLw is calculated according to ITU-T Recommendation G.122 [8], annex B, clause B.4 (trapezoidal rule) but using the frequency range of 300 Hz to 6 700 Hz (instead of 300 Hz to 3 400 Hz). For the calculation, the averaged measured echo level at each frequency band is referred to the averaged test signal level measured in each frequency band. The first 17,0 s of the test signal (6 sentences) are discarded from the analysis to allow for convergence of the acoustic echo canceller. The analysis is performed over the remaining length of the test sequence (last 6 sentences). + +The test signal level shall be -10 dBm0. + +## 8.8 Distortion + +### 8.8.1 Sending distortion + +The handset, headset, or hands-free UE is setup as described in clause 5. + +The signal used is a sine-wave signal with frequencies specified in clause 6.8 of 3GPP TS 26.131. The sine-wave signal level shall be calibrated to -4,7 dBPa at the MRP for all frequencies, except for the sine-wave with a frequency 1020 Hz which shall be applied at the following levels at the MRP: 5, 0, -4,7, -10, -15, -20 dBPa. The test signals have to be applied in this sequence, i.e., from high levels down to low levels. + +The duration of the sine-wave signal is recommended to be 360 ms. The manufacturer shall be allowed to request tone lengths up to 1 s. The measured part of the signal shall be 170,667 ms (which equals $2 \times 4096$ samples in a 48 kHz sample rate test system). The times are selected to be relatively short in order to reduce the risk that the test tone is treated as a stationary signal. + +It is recommended that an optional activation signal be presented immediately preceding each test signal to ensure that the UE is in a typical state during measurement (see Note 1.). An appropriate speech or speech-like activation signal + +shall be chosen from ITU-T Recommendations P.501 or P.50 [10]. A recommendation for the use of an activation signal as part of the measurement is defined in figure 18. The RMS level of the active parts of this activation signal is recommended to be equal to the subsequent test tone RMS level. In practice, certain types of processing may be impacted due to the introduction of the activation signal. The manufacturer shall be allowed to specify disabling of the activation signal. It shall be reported whether an activation signal was used or not, along with the characteristics of the activation signal, as specified by the manufacturer. + +The ratio of the signal to total distortion power of the signal output of the SS shall be measured with the psophometric noise weighting (see ITU-T Recommendations G.712, O.41 and O.132). The psophometric filter shall be normalized (0 dB gain) at 800 Hz as specified in ITU-T Recommendation O.41. The weighting function shall be applied to the total distortion component only (not to the signal component). + +For measurement of the total distortion component an octave-wide band-stop filter shall be applied to the signal to suppress the sine-wave signal and associated coding artefacts. The filter shall have a lower passband ending at $0.7071 \cdot f_s$ , and an upper passband starting at $1.4142 \cdot f_s$ , where $f_s$ is the frequency of the sine-wave signal. The passband ripple of the filter shall be $\leq 0,2$ dB. The attenuation of the band-stop filter at the sine-wave frequency shall be $\geq 60$ dB. Alternatively, the described characteristics can be implemented by an appropriate weighting on the spectrum obtained from an FFT. The total distortion component is defined as the measured signal within the frequency range 100 Hz to 6 kHz, after applying psophometric and stop filters (hence no correction for the lost power due to the stop filter, known as “bandwidth correction”, shall be applied). + +To improve repeatability, considering the variability introduced by speech coding and voice processing, the test sequence (activation signal followed by the test signal) may be contiguously repeated one or more times. The single signal-to-total-distortion power ratios obtained from such repeats shall be averaged. The total result shall be $10 \cdot \log_{10}$ of this average in dB. + +![Figure 18: Recommended activation sequence and test signal. The figure shows a time-domain waveform plot. The x-axis is marked with numbers 1 through 9 corresponding to different segments of the signal. Segments 1-3 show a burst of activity followed by a lower amplitude section and a pause. Segments 4-6 repeat this pattern. Segment 7 shows a shorter burst. Segment 8 is a flat line representing settling time. Segment 9 shows a steady-state signal (test tone) enclosed in a rectangular box indicating the analysis window.](9abed3db2278fa60c7f31626150ad973_img.jpg) + +Figure 18: Recommended activation sequence and test signal. The figure shows a time-domain waveform plot. The x-axis is marked with numbers 1 through 9 corresponding to different segments of the signal. Segments 1-3 show a burst of activity followed by a lower amplitude section and a pause. Segments 4-6 repeat this pattern. Segment 7 shows a shorter burst. Segment 8 is a flat line representing settling time. Segment 9 shows a steady-state signal (test tone) enclosed in a rectangular box indicating the analysis window. + +**Figure 18: Recommended activation sequence and test signal.** + +The activation signal consists of a “Bandlimited composite source signal with speech-like power density spectrum” signal according to ITU-T Recommendation P.501 with 48,62 ms voiced part (1), 200 ms unvoiced part (2) and 101,38 ms pause (3), followed by the same signal but polarity inverted (4, 5, 6), followed by the voiced part only (7). The pure test tone is applied and after 50 ms settling time (8), the analysis is made over the following 170,667 ms (9). + +NOTE 1: Depending on the type of codec the test signal used may need to be adapted. If a sine-wave is not usable, an alternative test signal could be a band-limited noise signal centered on the above frequencies. + +NOTE 2: Void. + +NOTE 3: Void. + +NOTE 4: In order to ensure that the correct part of the signal is analyzed, the total delay of the terminal and SS may have to be determined prior to the measurement. + +NOTE 5: For hands-free terminals tested in environments defined in subclause 6.1.2, care should be taken that the reverberation in the test room, caused by the activation signal, does not affect the test results to an unacceptable degree, referring to subclause 5.3. + +### 8.8.2 Receiving + +The handset, headset, or hands-free UE is setup as described in clause 5. + +The signal used is a sine-wave signal with frequencies specified in clause 6.8 of 3GPP TS 26.131. The signal level shall be -16 dBm0, except for the sine-wave signal with a frequency 1020 Hz that shall be applied at the signal input of the SS at the following levels: 0, -3, -10, -16, -20, -30, -40, -45 dBm0. The test signals have to be applied in this sequence, i.e., from high levels down to low levels. + +The duration of the sine-wave signal is recommended to be 360 ms. The manufacturer shall be allowed to request tone lengths up to 1 s. The measured part of the signal shall be 170,667 ms (which equals $2 * 4096$ samples in a 48 kHz sample rate test system). The times are selected to be relatively short in order to reduce the risk that the test tone is treated as a stationary signal. + +It is recommended that an optional activation signal be presented immediately preceding each test signal to ensure that the UE is in a typical state during measurement (see Note 1.). An appropriate speech or speech-like activation signal shall be chosen from ITU-T Recommendations P.501 or P.50 [10]. A recommendation for the use of an activation signal as part of the measurement is defined in figure 19. The RMS level of the active parts of this activation signal is recommended to be equal to the subsequent test tone RMS level for low and medium test levels. To avoid saturation of the SS speech encoder, it is recommended for high test levels that the activation signal level is adjusted so that its peak level equals the peak level of the test tone. In practice, certain types of processing may be impacted due to the introduction of the activation signal. The manufacturer shall be allowed to specify disabling of the activation signal. It shall be reported whether an activation signal was used or not, along with the characteristics of the activation signal, as specified by the manufacturer. + +The ratio of the signal to total distortion power shall be measured at the applicable acoustic measurement point (DRP with diffuse-field correction for handset and headset modes; free field for hands-free modes) with the psophometric noise weighting (see ITU-T Recommendations G.712, O.41 and O.132). The psophometric filter shall be normalized to have 0 dB gain at 800 Hz as specified in ITU-T Recommendation O.41. The weighting function shall be applied to the total distortion component only (not to the signal component). + +For measurement of the total distortion component an octave-wide band-stop filter shall be applied to the signal to suppress the sine-wave signal and associated coding artefacts. The filter shall have a lower passband ending at $0,7071 * f_s$ , and an upper passband starting at $1,4142 * f_s$ , where $f_s$ is the frequency of the sine-wave signal. The passband ripple of the filter shall be $\leq 0,2$ dB. The attenuation of the band stop filter at the sine-wave frequency shall be $\geq 60$ dB. Alternatively the described characteristics can be implemented by an appropriate weighting on the spectrum obtained from an FFT. The total distortion component is defined as the measured signal within the frequency range 100 Hz to 6 kHz, after applying psophometric and stop filters (hence no correction for the lost power due to the stop filter, known as “bandwidth correction”, shall be applied). + +To improve repeatability, considering the variability introduced by speech coding and voice processing, the test sequence (activation signal followed by the test signal) may be contiguously repeated one or more times. The single signal-to-total-distortion power ratios obtained from such repeats shall be averaged. The total result shall be $10 * \log_{10}$ of this average in dB. + +![Figure 19: Recommended activation sequence and test signal. The figure shows a waveform plot with three distinct segments labeled 1, 2, and 3. Segment 1 is a voiced part of the activation signal, segment 2 is an unvoiced part, and segment 3 is the test signal. The plot shows amplitude over time, with the test signal (3) being a continuous sine wave.](0ad96be66991e4f5b9bfab0768ae7160_img.jpg) + +Figure 19: Recommended activation sequence and test signal. The figure shows a waveform plot with three distinct segments labeled 1, 2, and 3. Segment 1 is a voiced part of the activation signal, segment 2 is an unvoiced part, and segment 3 is the test signal. The plot shows amplitude over time, with the test signal (3) being a continuous sine wave. + +**Figure 19: Recommended activation sequence and test signal.** + +The activation signal consists of a “Bandlimited composite source signal with speech-like power density spectrum” signal according to ITU-T Recommendation P.501 with 48,62 ms voiced part (1), 200 ms unvoiced part (2) and + +101,38 ms pause (3), followed by the same signal but polarity inverted (4, 5, 6), followed by the voiced part only (7). The pure test tone is applied and after 50 ms settling time (8), the analysis is made over the following 170,667 ms (9). + +**NOTE 1:** Void. + +**NOTE 2:** Void. + +**NOTE 3:** In order to ensure that the correct part of the signal is analyzed, the total delay of the terminal and SS may have to be determined prior to the measurement. + +**NOTE 4:** For hands-free terminals tested in environments defined in subclause 6.1.2, care should be taken that the reverberation in the test room, caused by the activation signal, does not affect the test results to an unacceptable degree, referring to subclause 5.3. + +## 8.9 Void + +## 8.10 Delay + +### 8.10.0 UE Delay Measurement Methodologies + +The sum of the UE delays in the sending and receiving directions ( $T_S+T_R$ ) shall be measured according to the methods described in clauses 8.10.1 and 8.10.2. In the event that the system simulator delays in send and/or receive directions are not stable between calls or cannot be accurately determined, the alternative method described in clause 8.10.3 may be used to obtain ( $T_S+T_R$ ) and the measured instability or inaccuracy observed when the methods described in 8.10.1 and 8.10.2 were performed shall be recorded in the test report. The test method(s) used and all results obtained shall also be recorded in the test report. + +### 8.10.1 Delay in sending direction (handset UE) + +The handset terminal is setup as described in clause 5.1.1. + +The delay shall include all entities in sending direction from MRP to the POI, but shall exclude the delays introduced by the test equipment. + +![Diagram of delay entities in sending direction](0ed8d001e745e9b2bb07fc63eb8525d5_img.jpg) + +The diagram illustrates the signal path for delay measurement. From left to right: An 'Artificial Mouth' with an arrow pointing to 'MRP'. This is followed by a 'UE Microphone' symbol. Next is a box for 'UE Signal Processing Entities', followed by a box for 'RF Transmission & Speech coder' which has an upward arrow. A bracket underneath these components from MRP to the RF block is labeled $T_s$ . To the right, a separate box labeled 'Test equipment' has an upward arrow and a bracket underneath labeled $T_{TES}$ . + +Diagram of delay entities in sending direction + +**Figure 19b1: Different entities contributing to the delay in sending direction** + +The delay in sending direction, measured from MRP to POI, is $T_s + T_{TES}$ . + +All test equipment delays, for the network type, codec type and bitrate used according to clause 5, (including radio access, speech codec, A/D and D/A conversions etc.) are included in $T_{TES}$ . The values used for testing (typical value considering variations due to interleaving etc.) as declared by the test equipment manufacturers shall be reported along with the measurement results. + +1. For the measurements, a Composite Source Signal (CSS) according to ITU-T Recommendation P.501 [22] is used. The pseudo random noise (pn)-part of the CSS has to be longer than the maximum expected delay. It is + +recommended to use a pn sequence of 32 k samples (with 48 kHz sampling rate). The test signal level is -4,7 dBPa at the MRP. + +- 2 The reference signal is the original signal (test signal). The setup of the handset/headset terminal is made corresponding to clause 5.1. +3. The delay is determined by cross-correlation analysis between the measured signal at the electrical access point and the original signal. The measurement is corrected by subtracting the test equipment delay $T_{TES}$ . +4. The delay is measured in ms and the maximum of the cross-correlation function is used for the determination. + +#### 8.10.1a Delay in sending direction (headset UE) + +The delay shall include all entities in sending direction from MRP to the POI, but shall exclude the delays introduced by the test equipment. + +![Diagram illustrating the delay components in the sending direction for a headset UE. The diagram shows a sequence of components: Artificial Mouth (MRP), Headset Microphone, UE Signal Processing Entities, RF Transmission & Speech coder, and Test equipment. The delay T_s is indicated by a bracket under the first four components, and T_TES is indicated by a bracket under the Test equipment component.](0997dbaa9dfecedd60029d70b53327b8_img.jpg) + +``` + + graph LR + AM[Artificial Mouth] -- MRP --> HM[Headset Microphone] + HM -- Connection cable --> UE[UE Signal Processing Entities] + UE --> RF[RF Transmission & Speech coder] + RF -- POI --> TE[Test equipment] + +``` + +The diagram shows the signal flow from left to right: Artificial Mouth (MRP) → Headset Microphone → Connection cable → UE Signal Processing Entities → RF Transmission & Speech coder → Test equipment. A bracket labeled $T_s$ spans from the MRP to the output of the RF Transmission & Speech coder. Another bracket labeled $T_{TES}$ spans the Test equipment block. + +Diagram illustrating the delay components in the sending direction for a headset UE. The diagram shows a sequence of components: Artificial Mouth (MRP), Headset Microphone, UE Signal Processing Entities, RF Transmission & Speech coder, and Test equipment. The delay T\_s is indicated by a bracket under the first four components, and T\_TES is indicated by a bracket under the Test equipment component. + +**Figure 19b2: Different entities contributing to the delay in sending direction with a headset connected via cable** + +Note: The test setup only applies to headsets connected by wire. Wireless headsets (e.g. connected by Bluetooth) are currently out of scope. + +The test method is the same as for handset UE (clause 8.10.1). + +### 8.10.2 Delay in receiving direction (handset UE) + +The handset terminal is setup as described in clause 5. + +The delay shall include all entities in receiving direction from the POI to the DRP, but shall exclude the delays introduced by the test equipment. + +![Figure 19b3: Diagram showing the signal flow and delays in the receiving direction. The signal path includes an Artificial Ear at the DRP, a UE Loudspeaker, UE Signal Processing Entities, RF Reception & Speech Decoder, and Test equipment. The delay from the RF/Speech decoder input to the DRP is labeled T_r. The delay within the Test equipment is labeled T_TER.](7ed5d5770331f31ade15439a21c31425_img.jpg) + +``` + +graph LR + subgraph UE_Path [T_r] + AE[Artificial Ear +DRP] --- LS[UE Loudspeaker] + LS --- SPE[UE Signal Processing Entities] + SPE --- RFD[RF Reception & Speech Decoder] + end + RFD --- TE[Test equipment] + subgraph TE_Path [T_TER] + TE --- Out[ ] + end + +``` + +Figure 19b3: Diagram showing the signal flow and delays in the receiving direction. The signal path includes an Artificial Ear at the DRP, a UE Loudspeaker, UE Signal Processing Entities, RF Reception & Speech Decoder, and Test equipment. The delay from the RF/Speech decoder input to the DRP is labeled T\_r. The delay within the Test equipment is labeled T\_TER. + +**Figure 19b3: Different entities contributing to the delay in receiving direction** + +The delay in receiving direction, measured from POI to DRP, is $T_r + T_{TER}$ . + +All test equipment delays, for the network type, codec type and bitrate used according to clause 5, (including radio access, speech codec, A/D and D/A conversions etc.) are included in $T_{TER}$ . The values used for testing (typical value considering variations due to interleaving etc.) as declared by the test equipment manufacturers shall be reported along with the measurement results. + +1. For the measurements, a Composite Source Signal (CSS) according to ITU-T Recommendation P.501 [22] is used. The pseudo random noise (pn)-part of the CSS has to be longer than the maximum expected delay. It is recommended to use a pn sequence of 32 k samples (with 48 kHz sampling rate). The test signal level is -16 dBm0 measured at the digital reference point or the equivalent analogue point. +2. The reference signal is the original signal (test signal). The setup of the handset/headset terminal is in correspondence to clause 5.1. +3. The delay is determined by cross-correlation analysis between the measured signal at the electrical access point and the original signal. The measurement is corrected by subtracting the test equipment delay $T_{TER}$ . +4. The delay is measured in ms and the maximum of the cross-correlation function is used for the determination. + +#### 8.10.2a Delay in receiving direction (headset UE) + +The delay shall include all entities in receiving direction from the POI to the DRP, but shall exclude the delays introduced by the test equipment. + +![Diagram illustrating the delay components in the receiving direction for a headset UE. The diagram shows a sequence of components: Artificial Ear (DRP), Headset Loudspeaker, Connection cable, UE Signal Processing Entities, RF Transmission & Speech coder, and Test equipment. The delay T_r is indicated by a bracket under the first four components, and T_TES is indicated by a bracket under the Test equipment component.](1033dc9fde75540d224c907681b1b7aa_img.jpg) + +The diagram shows the following components and delay measurements: + +- Artificial Ear (DRP)**: Represented by a circle with a vertical line on its right side. +- Headset Loudspeaker**: Represented by a trapezoidal shape. +- Connection cable**: A horizontal line connecting the Headset Loudspeaker to the UE Signal Processing Entities. +- UE Signal Processing Entities**: A rectangular block. +- RF Transmission & Speech coder**: A rectangular block connected to the UE Signal Processing Entities. It has an upward-pointing arrow from its top. +- Test equipment**: A rectangular block connected to the RF Transmission & Speech coder. It also has an upward-pointing arrow from its top. +- Delay $T_r$** : Indicated by a bracket underneath the Artificial Ear, Headset Loudspeaker, Connection cable, and UE Signal Processing Entities. +- Delay $T_{TES}$** : Indicated by a bracket underneath the Test equipment. + +Diagram illustrating the delay components in the receiving direction for a headset UE. The diagram shows a sequence of components: Artificial Ear (DRP), Headset Loudspeaker, Connection cable, UE Signal Processing Entities, RF Transmission & Speech coder, and Test equipment. The delay T\_r is indicated by a bracket under the first four components, and T\_TES is indicated by a bracket under the Test equipment component. + +**Figure 19b4: Different entities contributing to the delay in receiving direction with a headset connected via cable** + +Note: The test setup only applies to headsets connected by wire. Wireless headsets (e.g. connected by Bluetooth) are currently out of scope. + +The test method is the same as for handset UE (subclause 8.10.2). + +### 8.10.3 Delay in sending + receiving direction using “echo” method (handset UE) + +The mobile station delay shall include all entities from MRP to DRP (mouth-to-ear), but shall exclude the delays introduced by the test equipment and system simulator. + +![A graph showing Level difference [dB] on the y-axis versus Duration [ms] on the x-axis. The y-axis has labels 4, -4, and -15. The x-axis has labels 25 and 150. The graph is divided into regions E, F, G, A1, A2, B, C, and D by dashed lines. Region E is above 4 dB for duration < 25 ms. Region F is between 4 dB and -4 dB for duration < 25 ms. Region G is above 4 dB for duration > 150 ms. Region A1 is between 4 dB and -4 dB for duration < 150 ms. Region A2 is between -4 dB and -15 dB for duration < 150 ms. Region B is below -15 dB for duration < 25 ms. Region C is between -4 dB and -15 dB for duration < 150 ms. Region D is below -15 dB for duration > 150 ms.](730b6615db6d402580db1024a7f4e163_img.jpg) + +A graph showing Level difference [dB] on the y-axis versus Duration [ms] on the x-axis. The y-axis has labels 4, -4, and -15. The x-axis has labels 25 and 150. The graph is divided into regions E, F, G, A1, A2, B, C, and D by dashed lines. Region E is above 4 dB for duration < 25 ms. Region F is between 4 dB and -4 dB for duration < 25 ms. Region G is above 4 dB for duration > 150 ms. Region A1 is between 4 dB and -4 dB for duration < 150 ms. Region A2 is between -4 dB and -15 dB for duration < 150 ms. Region B is below -15 dB for duration < 25 ms. Region C is between -4 dB and -15 dB for duration < 150 ms. Region D is below -15 dB for duration > 150 ms. + +The delay measured from MRP to DRP is $(T_R + T_S + T_{SS})$ . + +All system simulator delays, for the used network type, codec type and bitrate, (including radio access, speech codec, A/D and D/A conversions etc., added echo delay) are included in $T_{SS}$ . The values used for testing (typical value considering variations due to interleaving etc.) as declared by the test equipment manufacturers shall be reported along with the measurement results. + +#### Method of measurement + +1. For the measurements a Composite Source Signal (CSS) according to ITU-T Recommendation P.501 [22] is used. It is recommended to use a pn sequence of 32 k samples (with 48 kHz sampling rate). The test signal level is -4.7 dBPa at the MRP. +2. The system simulator is configured for “loopback” or “echo” operation. In “loopback” or “echo” operation, the packets in the sending direction are routed to the receiving direction by the system simulator. +3. The reference signal is the original signal (test signal). The setup of the mobile station is in correspondence to clause 5.1. +4. The mouth-to-ear delay is determined by cross-correlation analysis between the measured signal at DRP and the original signal. The analysis window for the cross-correlation shall start at an instant $T > 50\text{ms}$ in order to discard the cross-correlation peaks corresponding to the direct acoustic path from mouth to ear and possible delayed sidetone signal. The measurement is corrected by subtracting the system simulator delay $T_{SS}$ to obtain the $T_R + T_S$ delay. +5. The delay is measured in ms and the maximum of the cross-correlation envelope is used for the determination. + +#### 8.10.3a Delay in sending + receiving direction using “echo” method (headset UE) + +The mobile station delay shall include all entities from MRP to DRP (mouth-to-ear), but shall exclude the delays introduced by the test equipment and system simulator. + +The test method is the same as for handset UE (clause 8.10.3). + +## 8.11 Echo control characteristics + +### 8.11.1 Test set-up and test signals + +The device is set up according to clause 5. The ambient noise level shall be $\leq -64$ dBPa(A). + +The test shall be performed with the British-English “long” double-talk and conditioning speech sequences from ITU-T Recommendation P.501 [22], with the signals in the receiving direction band limited according to clause 5.4. + +A description of the test stimuli is presented in Table 2e and Table 2f. The test sequence is composed of an initial conditioning sequence of 23,5 s and a double talk sequence of 35 s. For the analysis, the double talk sequence is divided into two segments, a first double-talk sequence with single short near-end words (0 – 20 s), and a second double-talk sequence with continuous double talk (20-35 s). + +The sending speech during double-talk and the “near-end speech only” are recorded individually, with the “near-end speech only” sequence recorded with silence in the receiving direction. The time-alignment of the two recorded sequences is performed off-line during the analysis. + +**Table 2e: Test stimuli for recording of Echo Canceller operation** + +| | Conditioning | Single words (segment 1) and full sentence (segment 2) double talk | +|--------------------------------|-------------------------------------|--------------------------------------------------------------------| +| Far-end signal | FB_female_conditioning_seq_long.wav | FB_male_female_single-talk_seq.wav | +| Artificial mouth signal | FB_male_conditioning_seq_long.wav | FB_male_female_double-talk_seq.wav | + +**Table 2f: Test stimuli for reference "near-end speech only" recording.** + +| | Conditioning | Single words (segment 1) and full sentence (segment 2) double talk | +|--------------------------------|-------------------------------------|--------------------------------------------------------------------| +| Far-end signal | FB_female_conditioning_seq_long.wav | silence | +| Artificial mouth signal | FB_male_conditioning_seq_long.wav | FB_male_female_double-talk_seq.wav | + +The level of the signal of the artificial mouth shall be - 4.7 dBPa measured at the MRP. In order to obtain a reproducible time alignment as seen by the UE, the artificial mouth signal shall be delayed by the amount of the receiving direction delay. For the purpose of this alignment, the receiving direction delay for handset and headset modes is defined from the system simulator input to the artificial ear. For handsfree modes, the downlink delay is defined from the system simulator input to the acoustic output from the UE loudspeaker. + +The level of the downlink signal shall be -16 dBm0 measured at the digital reference point or the equivalent analogue point. + +### 8.11.2 Test method + +The test method measures the duration of any level difference between the sending signal of a double-talk sequence (where the echo canceller has been exposed to simultaneous echo and near-end speech) and the sending signal of the same near-end speech only. The level difference is classified into eight categories according to Figure 19b5 and Table 2g, representing various degrees of “Full duplex operation”, “Near-end clipping”, and “Residual echo”. + +NOTE: The limits for specifying the categories in Figure 19b5 and Table 2g are provisional pending further analysis and validation. + +NOTE: The categories in Figure 19b5 and Table 2g are labelled in a functional order and the subjective impression of the respective categories is for further study. + +NOTE: To reduce potential issues associated with low-frequency test room noise, a [4th]-order high-pass filter with a cut-off frequency of [100] Hz can be applied before the level computation. + +![Figure 19b5: Classification of echo canceller performance. A graph showing Level difference [dB] on the y-axis (values: 4, -4, -15) versus Duration [ms] on the x-axis (values: 25, 150). The graph is divided into regions labeled A1, A2, B, C, D, E, F, and G based on these thresholds.](9c888dd6588358989047de6ced8b2bdb_img.jpg) + +The figure is a graph illustrating the classification of echo canceller performance based on Level difference [dB] (y-axis) and Duration [ms] (x-axis). The y-axis has markers at 4, -4, and -15 dB. The x-axis has markers at 25 and 150 ms. The graph is divided into several regions by dashed lines and solid lines: + +- Region **E**: Top-left area, $\Delta L \geq 4$ dB and $D < 25$ ms. +- Region **F**: Top-middle area, $\Delta L \geq 4$ dB and $25 \leq D < 150$ ms. +- Region **G**: Top-right area, $\Delta L \geq 4$ dB and $D \geq 150$ ms. +- Region **A1**: Middle band, $-4 \leq \Delta L < 4$ dB. +- Region **A2**: Middle band below A1, $-15 \leq \Delta L < -4$ dB. +- Region **B**: Bottom-left area, $\Delta L < -15$ dB and $D < 25$ ms. +- Region **C**: Bottom-middle area, $\Delta L < -15$ dB and $25 \leq D < 150$ ms. +- Region **D**: Bottom-right area, $\Delta L < -15$ dB and $D \geq 150$ ms. + +Figure 19b5: Classification of echo canceller performance. A graph showing Level difference [dB] on the y-axis (values: 4, -4, -15) versus Duration [ms] on the x-axis (values: 25, 150). The graph is divided into regions labeled A1, A2, B, C, D, E, F, and G based on these thresholds. + +Figure 19b5: Classification of echo canceller performance + +Table 2g: Categories for echo canceller performance classification + +| Category | Level difference ( $\Delta L$ ) | Duration (D) | Description | +|-----------|------------------------------------------------|-----------------------------------------|-----------------------------------------------| +| A1 | $-4 \text{ dB} \leq \Delta L < 4 \text{ dB}$ | | Full-duplex and full transparency | +| A2 | $-15 \text{ dB} \leq \Delta L < -4 \text{ dB}$ | | Full-duplex with level loss in Tx | +| B | $\Delta L < -15 \text{ dB}$ | $D < 25 \text{ ms}$ | Very short clipping | +| C | $\Delta L < -15 \text{ dB}$ | $25 \text{ ms} \leq D < 150 \text{ ms}$ | Short clipping resulting in loss of syllables | +| D | $\Delta L < -15 \text{ dB}$ | $D \geq 150 \text{ ms}$ | Clipping resulting in loss of words | +| E | $\Delta L \geq 4 \text{ dB}$ | $D < 25 \text{ ms}$ | Very short residual echo | +| F | $\Delta L \geq 4 \text{ dB}$ | $25 \text{ ms} \leq D < 150 \text{ ms}$ | Echo bursts | +| G | $\Delta L \geq 4 \text{ dB}$ | $D \geq 150 \text{ ms}$ | Continuous echo | + +A pseudo-code reference of the test method including test scripts and test-vectors is presented in Clause C.3 and outlined in the following sub clauses. + +#### 8.11.2.1 Signal alignment + +For the analysis of the signal level difference, the send signal during double-talk and the near-end only signal are aligned using a correlation analysis as described in Clause C.3.2. + +#### 8.11.2.2 Signal level computation and frame classification + +The analysis is based on the digital level measured with a meter according to IEC 61672 [38] with a time constant of 12.5 ms, sampled at 5 ms intervals corresponding to the evaluated frames. + +The “double-talk” frames are defined as the frames where both the far-end (receiving direction) signal includes active speech (extended with a hang-over period of 200 ms) and the near-end signal is composed of active speech. Active speech is defined to be detected using a speech level meter according to ITU-T P.56, and frames within -15.9 dB from the active speech level are classified as active speech frames. + +The “far-end single-talk adjacent to double-talk” frames are similarly defined using a speech level meter according to ITU-T P.56 as the frames with active far-end speech (extended with a hang-over period of 200 ms) and no active near-end speech (extended with a hang-over period of 200 ms). + +A reference implementation of the signal level computation and frame classification is presented in Clause C.3.3. + +#### 8.11.2.3 Classification into categories + +The analysis and classification into the categories according to Figure 19b5 and Table 2g is performed according to the reference implementation described in Clause C.3.4 and C.3.4. + +The frames are first categorized according to the level categories defined in Table 2g. To determine the durations, the amount of adjacent frames falling into the same level category is determined. + +The classification is then performed individually for the following situations: + +- frames classified as “double-talk” from segment 1 of the double-talk sequence (see 8.11.1) +- frames classified as “far-end single-talk adjacent to double-talk” from segment 1 of the double-talk sequence +- frames classified as “double-talk” from segment 2 of the double-talk sequence +- frames classified as “far-end single-talk adjacent to double-talk” from segment 2 of the double-talk sequence + +To determine the percentage values for each category (A1, A2, B, C, D, E, F, and G) within each situation, the number of frames falling into the respective category is divided by the total number of frames within the situation in question. + +To determine the averaged level difference of the frames for each category (A1, A2, B, C, D, E, F, and G) within each situation, the sum of the level difference (in dB) of the frames falling into the respective category is divided by the total number of frames within the situation in question. + +## 8.12 Quality (speech quality, noise intrusiveness) in the presence of ambient noise + +The speech quality in sending for narrowband systems is tested based on ETSI TS 103 106 [34]. This test method leads to three MOS-LQOw quality numbers: + +N-MOS-LQOw: Transmission quality of the background noise + +S-MOS-LQOw: Transmission quality of the speech + +G-MOS-LQOw: Overall transmission quality + +The test arrangement is given in clause 5.1.5. The measurement is conducted for 8 noise conditions as described in Table 2h. The measurements should be made in the same unique and dedicated call. The noise types shall be presented according to the order specified in Table 2h. + +**Table 2h: Noise conditions used for ambient noise simulation** + +| Description | File name | Duration | Level | Type | +|-----------------------------------|---------------------------------------|-----------------|--------------------------------|-------------| +| Recording in pub | Pub_Noise_binaural_V2 | 30 s | L: 75,0 dB(A)
R: 73,0 dB(A) | Binaural | +| Recording at pavement | Outside_Traffic_Road_binaural | 30 s | L: 74,9 dB(A)
R: 73,9 dB(A) | Binaural | +| Recording at pavement | Outside_Traffic_Crossroads_binaural | 20 s | L: 69,1 dB(A)
R: 69,6 dB(A) | Binaural | +| Recording at departure platform | Train_Station_binaural | 30 s | L: 68,2 dB(A)
R: 69,8 dB(A) | Binaural | +| Recording at the drivers position | Fullsize_Car1_130Kmh_binaural | 30 s | L: 69,1 dB(A)
R: 68,1 dB(A) | Binaural | +| Recording at sales counter | Cafeteria_Noise_binaural | 30 s | L: 68,4 dB(A)
R: 67,3 dB(A) | Binaural | +| Recording in a cafeteria | Mensa_binaural | 22 s | L: 63,4 dB(A)
R: 61,9 dB(A) | Binaural | +| Recording in business office | Work_Noise_Office_Callcenter_binaural | 30 s | L: 56,6 dB(A)
R: 57,8 dB(A) | Binaural | + +- 1) Before starting the measurements a proper conditioning sequence shall be used. The conditioning sequence shall be comprised of the four additional sentences 1-4 described in ETSI TS 103 106 [34], applied to the beginning of the 16-sentence test sequence. The conditioning signal level is - 1.7 dBPa at the MRP, measured as active speech level according to ITU-T P.56 [37]. + +NOTE: The sequence of speech samples concatenated for the test signal, consisting of alternating talkers in the sending direction, reduces the overall test time but may represent an unrealistic behaviour for certain voice enhancement technologies. Alternative concatenations are for further study. + +- 2) The send speech signal consists of the 16 sentences of speech as described in ETSI TS 103 106 [34]. The test signal level is - 1.7 dBPa at the MRP, measured as active speech level according to ITU-T P.56 [37]. Three signals are required for the tests: + - The clean speech signal is used as the undisturbed reference (see ETSI TS 103 106 [34], ETSI EG 202 396-3 [36]). + - The speech plus undisturbed background noise signal is recorded at the terminal's microphone position using an omnidirectional measurement microphone with a linear frequency response between 50 Hz and 12 kHz. + - The send signal is recorded at the POI. +- 3) N-MOS-LQOW, S-MOS-LQOW and G-MOS-LQOW are calculated as described in ETSI TS 103 106 [34] on a per sentence basis and averaged over all 16 sentences. The results shall be reported as average and standard deviation. +- 4) The measurement is repeated for each ambient noise condition described in Table 2h. +- 5) The average of the results derived from all ambient noise types is calculated. + +# Annex A: Interpolation method for diffuse-field correction (normative) + +Interpolated values for 1/12-octave bands shall be calculated from 1/3-octave band values using table 3. + +For measurements requiring diffuse-field correction values for closer frequency spacing than 1/12-octave bands, linear interpolation on a log scale from the 1/12-octave band interpolated values in table 3 shall be used. + +**Table 3: Interpolation parameters on 1/12-octave bands.** + +| Frequency (Hz) | Interpolated value (dB) | Frequency (Hz) | Interpolated value (dB) | +|----------------|-------------------------|----------------|-------------------------| +| 95 | 0,000 | 1 000 | 5,000 | +| 100 | 0,000 | 1 060 | 5,375 | +| 106 | 0,000 | 1 120 | 5,750 | +| 112 | 0,000 | 1 180 | 6,125 | +| 118 | 0,000 | 1 250 | 6,500 | +| 125 | 0,000 | 1 320 | 6,800 | +| 132 | 0,000 | 1 400 | 7,150 | +| 140 | 0,000 | 1 500 | 7,550 | +| 150 | 0,000 | 1 600 | 8,000 | +| 160 | 0,000 | 1 700 | 8,550 | +| 170 | 0,000 | 1 800 | 9,175 | +| 180 | 0,000 | 1 900 | 9,850 | +| 190 | 0,000 | 2 000 | 10,500 | +| 200 | 0,000 | 2 120 | 11,500 | +| 212 | 0,125 | 2 240 | 12,550 | +| 224 | 0,250 | 2 360 | 13,500 | +| 236 | 0,390 | 2 500 | 14,050 | +| 250 | 0,500 | 2 650 | 13,850 | +| 265 | 0,525 | 2 800 | 13,250 | +| 280 | 0,500 | 3 000 | 12,400 | +| 300 | 0,480 | 3 150 | 12,000 | +| 315 | 0,500 | 3 350 | 11,750 | +| 335 | 0,600 | 3 550 | 11,650 | +| 355 | 0,725 | 3 750 | 11,600 | +| 375 | 0,875 | 4 000 | 11,500 | +| 400 | 1,000 | 4 250 | 11,425 | +| 425 | 1,135 | 4 500 | 11,375 | +| 450 | 1,275 | 4 750 | 11,275 | +| 475 | 1,375 | 5 000 | 11,000 | +| 500 | 1,500 | 5 300 | 10,400 | +| 530 | 1,625 | 5 600 | 9,550 | +| 560 | 1,650 | 6 000 | 8,600 | +| 600 | 1,800 | 6 300 | 8,000 | +| 630 | 2,000 | 6 700 | 7,375 | +| 670 | 2,450 | 7 100 | 6,800 | +| 710 | 3,000 | 7 500 | 6,450 | +| 750 | 3,500 | 8 000 | 6,500 | +| 800 | 4,000 | 8 500 | 7,150 | +| 850 | 4,325 | 9 000 | 8,250 | +| 900 | 4,550 | 9 500 | 9,450 | +| 950 | 4,750 | 10 000 | 10,450 | + +Interpolated values for 1/12-octave bands can be also calculated from 1/3-octave band values using table 4 when frequencies are defined according to IEC 1260. + +**Table 4: Interpolation parameters on 1/12-octave bands with frequencies according to IEC 1260 (informative).** + +| Frequency (Hz) | Interpolated value (dB) | Frequency (Hz) | Interpolated value (dB) | +|----------------|-------------------------|----------------|-------------------------| +| 92 | 0,000 | 972 | 4,850 | +| 97 | 0,000 | 1029 | 5,180 | +| 103 | 0,000 | 1090 | 5,555 | +| 109 | 0,000 | 1155 | 5,969 | +| 115 | 0,000 | 1223 | 6,353 | +| 122 | 0,000 | 1296 | 6,720 | +| 130 | 0,000 | 1372 | 7,025 | +| 137 | 0,000 | 1454 | 7,345 | +| 145 | 0,000 | 1540 | 7,720 | +| 154 | 0,000 | 1631 | 8,165 | +| 163 | 0,000 | 1728 | 8,740 | +| 173 | 0,000 | 1830 | 9,370 | +| 183 | 0,000 | 1939 | 10,100 | +| 194 | 0,000 | 2054 | 10,900 | +| 205 | 0,055 | 2175 | 12,000 | +| 218 | 0,193 | 2304 | 13,080 | +| 230 | 0,330 | 2441 | 13,860 | +| 244 | 0,470 | 2585 | 13,985 | +| 259 | 0,520 | 2738 | 13,525 | +| 274 | 0,520 | 2901 | 12,810 | +| 290 | 0,490 | 3073 | 12,175 | +| 307 | 0,490 | 3255 | 11,850 | +| 325 | 0,550 | 3447 | 11,700 | +| 345 | 0,650 | 3652 | 11,625 | +| 365 | 0,790 | 3868 | 11,560 | +| 387 | 0,931 | 4097 | 11,460 | +| 410 | 1,055 | 4340 | 11,420 | +| 434 | 1,183 | 4597 | 11,375 | +| 460 | 1,313 | 4870 | 11,170 | +| 487 | 1,441 | 5158 | 10,700 | +| 516 | 1,560 | 5464 | 9,950 | +| 546 | 1,635 | 5788 | 9,070 | +| 579 | 1,720 | 6131 | 8,300 | +| 613 | 1,875 | 6494 | 7,700 | +| 649 | 2,180 | 6879 | 7,100 | +| 688 | 2,675 | 7286 | 6,610 | +| 729 | 3,222 | 7718 | 6,410 | +| 772 | 3,750 | 8175 | 6,655 | +| 818 | 4,140 | 8660 | 7,477 | +| 866 | 4,400 | 9173 | 8,680 | +| 917 | 4,620 | 9716 | 9,950 | + +# Annex B (informative): Reference algorithm for echo control characteristics evaluation. + +## B.1 General + +In this annex, a reference algorithm for evaluation of the echo control characteristics is described in pseudo code. The output of an implementation of the test method with the stimuli from the file “echo\_control\_reference\_files.zip” should equal the results presented in Table 3a and Table 3b. To run the verification, the additional file named “p501-downlink\_WB.pcm” in the pseudo code shall be created from the concatenated full band speech samples FB\_female\_conditioning\_seq\_long.wav and FB\_male\_female\_single-talk\_seq.wav from ITU-T Recommendation P.501, and processed with the following set of commands based on ITU-T Recommendation G.191: + +``` +filter -down HQ3 far_end_signal_48k.pcm far_end_signal_16k.pcm +filter P341 far_end_signal_16k.pcm p501-downlink_WB.pcm +``` + +**Table 3a: Characterization of segment 1.** + +| Category | Double talk | | Single talk | | +|-----------|-------------|----------------|-------------|----------------| +| | Activity | Av. Level [dB] | Activity | Av. Level [dB] | +| A1 | 60,8% | -1,2 | 95,1% | 0,1 | +| A2 | 39,2% | -5,1 | 1,4% | -4,8 | +| B | 0,0% | 0 | 0,0% | 0 | +| C | 0,0% | 0 | 0,0% | 0 | +| D | 0,0% | 0 | 0,0% | 0 | +| E | 0,0% | 0 | 0,3% | 9,4 | +| F | 0,0% | 0 | 3,2% | 8,7 | +| G | 0,0% | 0 | 0,0% | 0 | + +**Table 3b: Characterization of segment 2.** + +| Category | Double talk | | Single talk | | +|-----------|-------------|----------------|-------------|----------------| +| | Activity | Av. Level [dB] | Activity | Av. Level [dB] | +| A1 | 50.2% | -1.1 | 93,8% | 0,2 | +| A2 | 40.8% | -7.3 | 0,3% | -5.6 | +| B | 1.2% | -16,9 | 0,0% | 0 | +| C | 7.1% | -17,2 | 0,0% | 0 | +| D | 0,0% | 0 | 0,0% | 0 | +| E | 0,0% | 0 | 0,5% | 9,5 | +| F | 0,7% | 4.0 | 5.5% | 6,2 | +| G | 0,0% | 0 | 0,0% | 0 | + +The pseudo-code reference algorithm produces a text file output, and the implementation of the test method may be tested with the test script on the data in the file “echo\_control\_reference\_files.zip” for which the result shall equal + +``` +ms01-rec2; segm. 1; Processed signal; +active speech level [dBovl]; -45.8; RMS level [dBovl]; -51.5; speech activity; 0.269 +ms01-rec2; segm. 1; Near end signal; +active speech level [dBovl]; -42.6; RMS level [dBovl]; -49.1; speech activity; 0.225 +ms01-rec2; segm. 1; Downlink signal; +active speech level [dBovl]; -26.6; RMS level [dBovl]; -27.4; speech activity; 0.823 +ms01-rec2; segm. 1; delay 0; DL delay 0; +DT activity 0.100; 0.608; 0.392; 0.000; 0.000; 0.000; 0.000; 0.000; 0.000; +ms01-rec2; segm. 1; delay 0; DL delay 0; +DT level diff; -1.2; -5.1; 0.0; 0.0; 0.0; 0.0; 0.0; 0.0; +ms01-rec2; segm. 1; delay 0; DL delay 0; +ST activity 0.664; 0.951; 0.014; 0.000; 0.000; 0.000; 0.003; 0.032; 0.000; +ms01-rec2; segm. 1; delay 0; DL delay 0; +ST level diff; 0.1; -4.8; 0.0; 0.0; 0.0; 0.0; 9.4; 8.7; 0.0; +ms01-rec2; segm. 2; Processed signal; +``` + +``` + +active speech level [dBov1]; -42.0; RMS level [dBov1]; -44.4; speech activity; 0.581 +ms01-rec2; segm. 2; Near end signal; +active speech level [dBov1]; -40.6; RMS level [dBov1]; -42.7; speech activity; 0.625 +ms01-rec2; segm. 2; Downlink signal; +active speech level [dBov1]; -26.5; RMS level [dBov1]; -27.2; speech activity; 0.841 +ms01-rec2; segm. 2; delay -1; DL delay 0; +DT activity 0.348; 0.502; 0.408; 0.012; 0.071; 0.000; 0.000; 0.007; 0.000; +ms01-rec2; segm. 2; delay -1; DL delay 0; +DT level diff; -1.1; -7.3; -16.9; -17.2; 0.0; 0.0; 4.0; 0.0; +ms01-rec2; segm. 2; delay -1; DL delay 0; +ST activity 0.362; 0.938; 0.003; 0.000; 0.000; 0.000; 0.005; 0.055; 0.000; +ms01-rec2; segm. 2; delay -1; DL delay 0; +ST level diff; 0.2; -5.6; 0.0; 0.0; 0.0; 9.5; 6.2; 0.0; + +``` + +## B.2 Test script + +``` + +% +% Set data format +% +fs = 16000; +conditioningTime = 23.5; +downlinkSystemDelay = 0; + +% +% Segment the data +% +offsetDoubleTalk = conditioningTime; +offsetNearEnd = conditioningTime; + +segmentDoubleTalkIndex(1) = {[0, 20]}; +segmentNearEndIndex(1) = {[0, 20]}; + +segmentDoubleTalkIndex(2) = {[20, 35]}; +segmentNearEndIndex(2) = {[20, 35]}; + +lengthDoubleTalk = max(cell2mat(segmentDoubleTalkIndex(end))); +lengthNearEnd = max(cell2mat(segmentNearEndIndex(end))); + +firstSampleDoubleTalk = round(fs*offsetDoubleTalk) + 1; +firstSampleNearEnd = round(fs*offsetNearEnd) + 1; + +lastSampleDoubleTalk = round(fs*(offsetDoubleTalk+lengthDoubleTalk)); +lastSampleNearEnd = round(fs*(offsetNearEnd+lengthNearEnd)); + +indexDoubleTalk = [firstSampleDoubleTalk, lastSampleDoubleTalk]; +indexNearEnd = [firstSampleNearEnd, lastSampleNearEnd]; + +% +% Read data from file +% +fid = fopen('ms01_WB_rec2.pcm', 'r'); +fseek(fid, 2*round(fs*offsetDoubleTalk), 'bof'); +processedData = fread(fid, round(fs*lengthDoubleTalk), 'int16'); +fclose(fid); + +fid = fopen('ms01_WB_ref.pcm', 'r'); +fseek(fid, 2*round(fs*offsetNearEnd), 'bof'); +nearendData = fread(fid, round(fs*lengthNearEnd), 'int16'); +fclose(fid); + +fid = fopen('p501-downlink_WB.pcm', 'r'); +fseek(fid, 2*round(fs*offsetDoubleTalk), 'bof'); +downlinkData = fread(fid, round(fs*lengthDoubleTalk), 'int16'); +fclose(fid); + +% +% Evaluate +% +ecEvaluation(processedData, nearendData, downlinkData, ... + segmentDoubleTalkIndex, segmentNearEndIndex, ... + 'ms01-rec2', downlinkSystemDelay, ... + fs, 'bitExactTest.txt'); + +``` + +## B.3 Reference algorithm + +### B.3.1 Main algorithm + +``` + +%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% +% +% processedData: processed samples +% originalData: near-end-only samples +% downlinkData: down-link (loudspeaker) samples +% processedSegmentSet: set of indices to processed data segments +% originalSegmentSet: set of indices to original data segments +% PROC_FILE: name shown in diagrams +% downlinkSystemDelayInMs: delay in DL signal from data to acoustic out +% sampleRate: sampling frequency of the data +% resultsFile: output file +% +%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% +%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% + +function ecEvaluation(... + processedData, ... + nearendData, ... + downlinkData, ... + indexProcessed, ... + indexNearend, ... + PROC_FILE, ... + downlinkSystemDelayInMs, ... + sampleRate, ... + resultFile) + +fid = fopen(resultFile, 'a'); + +% Define the categories +global D1 D2 D3 D4 L1 L2 L3; +D1 = 25; +D2 = 150; +D3 = 25; +D4 = 150; +L1 = 4; +L2 = -4; +L3 = -15; + +global FRAME_LENGTH_MS ... + MAX_DURATION_MS ... + MAX_DURATION_FRAMES ... + MAX_LEVEL_DIFFERENCE ... + MIN_LEVEL_DIFFERENCE ... + HISTOGRAM_RESOLUTION_MS + +FRAME_LENGTH_MS = 5; +MAX_DURATION_MS = 200; +MAX_DURATION_FRAMES = MAX_DURATION_MS/FRAME_LENGTH_MS; +MAX_LEVEL_DIFFERENCE = 40; +MIN_LEVEL_DIFFERENCE = -40; +HISTOGRAM_RESOLUTION_MS = FRAME_LENGTH_MS; + +% Main processing loop +frameLengthInSamples = FRAME_LENGTH_MS*sampleRate/1000; % 5ms frames + +for segment = 1:length(indexProcessed) + % Get the data samples for the segment + segmentDataProcessed = cell2mat(indexProcessed(segment)); + segmentDataNearend = cell2mat(indexNearend(segment)); + + index = (sampleRate*segmentDataProcessed(1)+1):sampleRate*segmentDataProcessed(2); + x = processedData(index); + z = downlinkData(index); + index = (sampleRate*segmentDataNearend(1)+1):sampleRate*segmentDataNearend(2); + y = nearendData(index); + + % Estimate and compensate for delay between processed and near end + [x, y, z, delay] = compensateDelay(x, y, z, 0.5*sampleRate); + +``` + +``` +% Compute the signal levels and classify the frames +[Rx, Ry, Rz, doubleTalkFrames, singleTalkFrames] = ... +computeSignalLevels(x, y, z, ... + sampleRate, frameLengthInSamples, ... + downlinkSystemDelayInMs, ... + PROC_FILE, segment, fid); + +% Evaluate double-talk performance +numberOfDoubleTalkFrames = 0; +% Iterate over blocks of consecutive indices +H_dt = []; +doubleTalkFramesBlocks = findConsecutiveBlocks(doubleTalkFrames); +for i = 1:size(doubleTalkFramesBlocks,1) + IdxFrom = doubleTalkFramesBlocks(i,1); + IdxTo = doubleTalkFramesBlocks(i,2); + currentBlockSize = IdxTo - IdxFrom; + if currentBlockSize > 1 + [H_dt_Tmp, ld_ax_dt, dur_ax_dt] = levelTimeStatistics(Rx(IdxFrom:IdxTo), Ry(IdxFrom:IdxTo)); + if isempty(H_dt) + H_dt = H_dt_Tmp; + else + H_dt = H_dt + H_dt_Tmp; + end + numberOfDoubleTalkFrames = numberOfDoubleTalkFrames + currentBlockSize; + end +end + +[C_dt, L_dt] = evaluateHistogram(H_dt, ld_ax_dt, dur_ax_dt, ... + numberOfDoubleTalkFrames); +activityFactorDoubleTalk = numberOfDoubleTalkFrames/length(Rx); + +% Evaluate single-talk performance +numberOfSingleTalkFrames = 0; +% Iterate over blocks of consecutive indices +H_st = []; +singleTalkFramesBlocks = findConsecutiveBlocks(singleTalkFrames); +for i = 1:size(singleTalkFramesBlocks,1) + IdxFrom = singleTalkFramesBlocks(i,1); + IdxTo = singleTalkFramesBlocks(i,2); + currentBlockSize = IdxTo - IdxFrom; + if currentBlockSize > 1 + [H_st_Tmp, ld_ax_st, dur_ax_st] = levelTimeStatistics(Rx(IdxFrom:IdxTo), Ry(IdxFrom:IdxTo)); + if isempty(H_st) + H_st = H_st_Tmp; + else + H_st = H_st + H_st_Tmp; + end + numberOfSingleTalkFrames = numberOfSingleTalkFrames + currentBlockSize; + end +end + +[C_st, L_st] = evaluateHistogram(H_st, ld_ax_st, dur_ax_st, ... + numberOfSingleTalkFrames); +activityFactorSingleTalk = numberOfSingleTalkFrames/length(Rx); + +% Save to result file +writeResultsToFile(fid, ... + PROC_FILE, ... + segment, ... + delay, ... + round(downlinkSystemDelayInMs), ... + activityFactorDoubleTalk, ... + activityFactorSingleTalk, ... + C_dt, ... + C_st, ... + L_dt, ... + L_st); + +end + +fclose(fid); +``` + +### B.3.2 Delay compensation + +``` + +%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% +%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% +% +% Compensate for delay in processed file +% +%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% +%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% +function [x, y, z, delay] = ... +compensateDelay(... + x, ... + y, ... + z, ... + maxLag) + +ii = 1:min(1000000, length(x)); + +r = xcorr(x(ii), y(ii), maxLag); + [~, delay] = max(abs(r)); +delay = delay-maxLag-1; + +if (delay > 0) + x = x((delay+1):end); + z = z((delay+1):end); + y = y(1:(end-delay)); +elseif (delay < 0) + y = y((-delay+1):end); + x = x(1:(end+delay)); + z = z(1:(end+delay)); +end; + +``` + +### B.3.3 Signal level computation and frame classification + +``` + +%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% +%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% +% +% Determine speech activity and signal levels +% +%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% +%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% +function [Rx, Ry, Rz, doubleTalkFrames, singleTalkFrames] = ... +computeSignalLevels(x, y, z, ... + sampleRate, frameLengthInSamples, ... + downlinkSystemDelayInMs, ... + PROC_FILE, segment, fid) + +LEVEL_METER_INIT_TIME_MS = 100; +DOWNLINK_HANGOVER_FRAMES = 40; +NEAREND_HANGOVER_FRAMES = 40; + +levelMeterInitTime = LEVEL_METER_INIT_TIME_MS*sampleRate/1000; + +% Level according to IEC61672 +Rx = IEC61672(x, sampleRate, 12.5); +Ry = IEC61672(y, sampleRate, 12.5); +Rz = IEC61672(z, sampleRate, 12.5); + +% Correct for system delay +nRz = length(Rz); +minRz = min(Rz(levelMeterInitTime:end)); +Rz = [minRz*ones(floor(downlinkSystemDelayInMs*sampleRate/1000), 1); Rz]; +Rz = Rz(1:nRz); + +% Sub-sample and avoid initialization period of level meter +Rx = Rx(levelMeterInitTime:frameLengthInSamples:end); +Ry = Ry(levelMeterInitTime:frameLengthInSamples:end); +Rz = Rz(levelMeterInitTime:frameLengthInSamples:end); + +% Active speech level according to P.56 +[activeSpeechLevelProcessed, ... + +``` + +``` + +longTermLevelProcessed, ... +activityFactorProcessed] = ... +speechLevelMeter(x, sampleRate); + +[activeSpeechLevelNearend, ... +longTermLevelNearend, ... +activityFactorNearend] = ... +speechLevelMeter(y, sampleRate); + +[activeSpeechLevelDownlink, ... +longTermLevelDownlink, ... +activityFactorDownlink] = ... +speechLevelMeter(z, sampleRate); + +% Write active speech levels to file +writeSpeechLevelsToFile(PROC_FILE, segment, fid, ... + activeSpeechLevelProcessed, ... + activeSpeechLevelNearend, ... + activeSpeechLevelDownlink, ... + longTermLevelProcessed, ... + longTermLevelNearend, ... + longTermLevelDownlink, ... + activityFactorProcessed, ... + activityFactorNearend, ... + activityFactorDownlink); + +% +% Only evaluate for active downlink/near-end speech including hang-over +% +activeRyFrames = find(Ry > activeSpeechLevelNearend-15.9); +activeRzFrames = find(Rz > activeSpeechLevelDownlink-15.9); + +% Downlink with added hangover +activeDownlinkSpeechFrames = zeros(size(Rz)); +activeDownlinkSpeechFrames(activeRzFrames) = ones(size(activeRzFrames)); + +activeDownlinkSpeechFrames = conv(activeDownlinkSpeechFrames, ... + ones(DOWNLINK_HANGOVER_FRAMES, 1)); +activeDownlinkSpeechFrames = activeDownlinkSpeechFrames(1:length(Rz)); + +% Near-end +activeNearEndSpeechFrames = zeros(size(Ry)); +activeNearEndSpeechFrames(activeRyFrames) = ones(size(activeRyFrames)); +activeNearEndSpeechHtFrames = conv(activeNearEndSpeechFrames, ... + ones(NEAREND_HANGOVER_FRAMES, 1)); +activeNearEndSpeechHtFrames = activeNearEndSpeechHtFrames(1:length(Rz)); + +% Only evaluate double talk when both rx+hangover and near-end +doubleTalkSpeechFrames = (activeDownlinkSpeechFrames & ... + activeNearEndSpeechFrames); +doubleTalkFrames = find(doubleTalkSpeechFrames > 0); + +% Single talk defined as rx and no near-end including 200 ms hangover +singleTalkSpeechFrames = (activeDownlinkSpeechFrames & ... + ~activeNearEndSpeechHtFrames); +singleTalkFrames = find(singleTalkSpeechFrames > 0); + +%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% +%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% +% +% Average speech and noise levels +% +%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% +%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% +function [... + activeSpeechLevel, ... + longTermLevel, ... + activityFactor ... +] = ... +speechLevelMeter(x, sampleRate) + +SPEECH_LEVEL_HANGOVER_TIME_IN_MS = 200; + +% Filter data +g = exp(-1/(0.03*sampleRate)); +p = filter([1-g], [1, -g], abs(x)); + +``` + +``` + +q = filter((1-g), [1, -g], abs(p)); + +% Add 200ms hangover +hTimeInSamples = SPEECH_LEVEL_HANGOVER_TIME_IN_MS*sampleRate/1000; +qht = q; +for loop = 1:hTimeInSamples + qht = max(qht, [zeros(loop, 1); q(1:end-loop)]); +end + +% Compute cumulative histogram of signal power with hangover +nData = length(x); +cBins = 2.0.^(0:14)'; +histogramCsum = zeros(size(cBins)); + +for loop = 1:length(cBins) + histogramCsum(loop) = length(find(qht>cBins(loop))); +end + +% Get the levels +sumSquare = sum(x.^2); +refdB = 20*log10(32768); + +longTermLevel = 10*log10(sumSquare/nData) - refdB; +A = 10*log10(sumSquare./histogramCsum) - refdB; +C = 20*log10(cBins) - refdB; + +Diff = A-C; +if ((A(1) == 0) || ((A(1) - C(1)) <= 15.9)) + activeSpeechLevel = -100; +else + index = find(Diff <= 15.9, 1, 'first'); + + if (Diff(index) == 15.9) + activeSpeechLevel = A(index); + else + C_level = C(index) + ... + (15.9 - Diff(index))* ... + (C(index)-C(index-1))/(Diff(index)-Diff(index-1)); + activeSpeechLevel = A(index) + ... + (C_level - C(index))* ... + (A(index)-A(index-1))/(C(index)-C(index-1)); + end +end + +activityFactor = 10.0^((longTermLevel-activeSpeechLevel)/10); + +%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% +%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% +% +% Speech level meter according to IEC61672 +% +%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% +%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% + +function Rx = IEC61672(x, sampleRate, tc) +% +% +% This functions computes the power of a sampled signal +% using a discrete filter with time constant equivalent to a first order +% continuous time exponential averaging circuit, +% +% +% +$$Rx = \frac{1/tc}{s + 1/tc} x^2$$ + +% +% according to IEC 61672 (1993, section 7.2). +% + +T = 1/sampleRate; +tc = tc/1000; + +% +% Design H by sampling of Hc +% +la = exp(-T/tc); +B = 1-la; + +``` + +``` + +A = [1, -1a]; + +Rx = filter(B, A, x.^2); + +% +% Transform Rx to dBov (square wave), +% 0 dBov <=> power of maximum square wave signal, 32768 +% +% 10^0 = 32768^2/X => X = 32768^2 +% +% Avoid log(0) by using log(max(eps, Rx)) +% +Rx = 10*log10(max(eps, Rx)/32768/32768); + +``` + +### B.3.4 Level vs time computation + +``` + +%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% +%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% +% +% Computation of level and time statistics +% +%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% +%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% +function [... + levelVsDurationHistogram, ... + levelDifferenceAxis, ... + durationAxis] = ... + levelTimeStatistics(processedLevel, nearEndLevel) + +global MAX_DURATION_FRAMES MAX_LEVEL_DIFFERENCE MIN_LEVEL_DIFFERENCE + +FIRST_OCCURENCE = 1; + +% +% Compute level difference +% +levelDifference = processedLevel - nearEndLevel; + +% +% Only evaluate in integers (rounded towards 0) of dB and limit to max/min difference +% +levelDifference = fix(levelDifference); +levelDifference = min(levelDifference, MAX_LEVEL_DIFFERENCE); +levelDifference = max(levelDifference, MIN_LEVEL_DIFFERENCE); + +% +% Produce axis +% +levelDifferenceAxis = MIN_LEVEL_DIFFERENCE:MAX_LEVEL_DIFFERENCE; +durationAxis = 1:(MAX_DURATION_FRAMES+1); + +% +% Set initial values for computations and loop through all frames +% +numberOfEvaluatedFrames = length(levelDifference); + +levelIncludedInEvaluation = (MAX_LEVEL_DIFFERENCE+1)*... + ones(numberOfEvaluatedFrames, 1); +levelAndRunLength = zeros(numberOfEvaluatedFrames, 4); +levelVsDurationHistogram = zeros(MAX_LEVEL_DIFFERENCE+ ... + (-MIN_LEVEL_DIFFERENCE)+1, ... + MAX_DURATION_FRAMES+1); + +previousLevelDifference = 0; + +for frame = 1:numberOfEvaluatedFrames-1; + currentLevelDifference = levelDifference(frame); + + % + % Evaluate all levels from the previous level up to the current level + % + if currentLevelDifference <= 0 + +``` + +``` + + firstEvaluatedLevelDifference = max(min(0, previousLevelDifference), ... + currentLevelDifference); + step = -1; + else + firstEvaluatedLevelDifference = min(max(0, previousLevelDifference), ... + currentLevelDifference); + step = 1; + end + + % + % Loop the levels to be evaluated + % + for evaluatedLevelDifference = ... + firstEvaluatedLevelDifference:step:currentLevelDifference + % + % Check that the current frame is not already included + % in evaluation for earlier frames + % + if (evaluatedLevelDifference ~= levelIncludedInEvaluation(frame)) + if (evaluatedLevelDifference > 0) + duration = find(levelDifference(frame+1:end) < ... + evaluatedLevelDifference, FIRST_OCCURENCE); + else + duration = find(levelDifference(frame+1:end) > ... + evaluatedLevelDifference, FIRST_OCCURENCE); + end + + if (isempty(duration)) + duration = numberOfEvaluatedFrames-frame+1; + end + + % + % Set the frames during duration of the level difference + % as being evaluated + % + if (duration > 1) + levelIncludedInEvaluation(frame:(frame+duration-1)) = ... + evaluatedLevelDifference*ones(duration, 1); + end; + + % + % Add the number of frames in the duration that have + % absolute level diff greater or equal to evalutedLevel + % + durationIndex = min(duration, MAX_DURATION_FRAMES); + levelIndex = evaluatedLevelDifference+(-MIN_LEVEL_DIFFERENCE)+1; + levelVsDurationHistogram(levelIndex, durationIndex) = ... + levelVsDurationHistogram(levelIndex, durationIndex) + duration; + end + end + + previousLevelDifference = currentLevelDifference; +end + +``` + +### B.3.5 Categorization + +``` + +%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% +%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% +% +% Evaluate the histogram data +% +%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% +%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% +function [categories, averageLevelsInCategories] = ... + evaluateHistogram(... + histogramData, ... + levelDiff_ax, ... + duration_ax, ... + numberOfFrames) + +global D1 D2 D3 D4 L1 L2 L3 HISTOGRAM_RESOLUTION_MS; + +D1_scaled = D1/HISTOGRAM_RESOLUTION_MS; + +``` + +``` + +D2_scaled = D2/HISTOGRAM_RESOLUTION_MS; +D3_scaled = D3/HISTOGRAM_RESOLUTION_MS; +D4_scaled = D4/HISTOGRAM_RESOLUTION_MS; + +levelIndex_L1 = find(levelDiff_ax == L1); +levelIndex_L2 = levelDiff_ax == L2; +levelIndex_L3 = find(levelDiff_ax == L3); + +duration_A2 = duration_ax; +duration_B = duration_ax<=D1_scaled; +duration_C = (D1_scaledD2_scaled; +duration_E = duration_ax<=D3_scaled; +duration_F = (D3_scaledD4_scaled; + +framesInCategoryB = sum(histogramData(levelIndex_L3, duration_B)); +framesInCategoryC = sum(histogramData(levelIndex_L3, duration_C)); +framesInCategoryD = sum(histogramData(levelIndex_L3, duration_D)); +framesInCategoryE = sum(histogramData(levelIndex_L1, duration_E)); +framesInCategoryF = sum(histogramData(levelIndex_L1, duration_F)); +framesInCategoryG = sum(histogramData(levelIndex_L1, duration_G)); + +framesInCategoryA2 = sum(histogramData(levelIndex_L2, duration_A2)); +framesInCategoryA2 = framesInCategoryA2 - ... + framesInCategoryB - ... + framesInCategoryC - ... + framesInCategoryD; + +framesInCategoryA1 = numberOfFrames - ... + framesInCategoryA2 - ... + framesInCategoryB - ... + framesInCategoryC - ... + framesInCategoryD - ... + framesInCategoryE - ... + framesInCategoryF - ... + framesInCategoryG; + +categories = [framesInCategoryA1; + framesInCategoryA2; + framesInCategoryB; + framesInCategoryC; + framesInCategoryD; + framesInCategoryE; + framesInCategoryF; + framesInCategoryG]/numberOfFrames; + +averageLevelsInCategories = zeros(8, 1); + +% Category A1 +index = levelDiff_ax < L1; +index = levelDiff_ax(index) > L2; +weight = levelDiff_ax(index); + +duration = duration_ax; +levelTimesDuration = (weight*histogramData(index, duration)).*duration; +nData = sum(histogramData(index, duration)*duration'); +if (framesInCategoryA1 > 0) + averageLevelsInCategories(1) = sum(levelTimesDuration)/nData; +end + +% Category A2 +index = levelDiff_ax <= L2; +index = levelDiff_ax(index) > L3; +weight = levelDiff_ax(index); + +duration = duration_ax; +levelTimesDuration = (weight*histogramData(index, duration)).*duration; +nData = sum(histogramData(index, duration)*duration'); +if (framesInCategoryA2 > 0) + averageLevelsInCategories(2) = sum(levelTimesDuration)/nData; +end + +% Category B, C, D +index = find(levelDiff_ax <= L3); +weight = levelDiff_ax(index); + +``` + +``` + +duration = duration_ax(duration_B); +levelTimesDuration = (weight*histogramData(index, duration_B)).*duration; +nData = sum(histogramData(index, duration_B)*duration'); +if (framesInCategoryB > 0) + averageLevelsInCategories(3) = sum(levelTimesDuration)/nData; +end + +duration = duration_ax(duration_C); +levelTimesDuration = (weight*histogramData(index, duration_C)).*duration; +nData = sum(histogramData(index, duration_C)*duration'); +if (framesInCategoryC > 0) + averageLevelsInCategories(4) = sum(levelTimesDuration)/nData; +end + +duration = duration_ax(duration_D); +levelTimesDuration = (weight*histogramData(index, duration_D)).*duration; +nData = sum(histogramData(index, duration_D)*duration'); +if (framesInCategoryD > 0) + averageLevelsInCategories(5) = sum(levelTimesDuration)/nData; +end + +% Category E, F, G +index = find(levelDiff_ax >= L1); +weight = levelDiff_ax(index); + +duration = duration_ax(duration_E); +levelTimesDuration = (weight*histogramData(index, duration_E)).*duration; +nData = sum(histogramData(index, duration_E)*duration'); +if (framesInCategoryE > 0) + averageLevelsInCategories(6) = sum(levelTimesDuration)/nData; +end + +duration = duration_ax(duration_F); +levelTimesDuration = (weight*histogramData(index, duration_F)).*duration; +nData = sum(histogramData(index, duration_F)*duration'); +if (framesInCategoryF > 0) + averageLevelsInCategories(7) = sum(levelTimesDuration)/nData; +end + +duration = duration_ax(duration_G); +levelTimesDuration = (weight*histogramData(index, duration_G)).*duration; +nData = sum(histogramData(index, duration_G)*duration'); +if (framesInCategoryG > 0) + averageLevelsInCategories(8) = sum(levelTimesDuration)/nData; +end + +``` + +### B.3.6 Auxiliary functions for reporting data + +``` + +%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% +%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% +% +% Write the classification to file +% +%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% +%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% +function writeResultsToFile(fid, ... + PROC_FILE, ... + segment, ... + delay, ... + downlinkSystemDelay, ... + activityFactorDoubleTalk, ... + activityFactorSingleTalk, ... + C_dt, ... + C_st, ... + L_dt, ... + L_st) + +str = sprintf('%s; segm. %d; delay %d; DL delay %d; DT activity %1.3f; %1.3f; %1.3f; %1.3f; %1.3f; %1.3f; %1.3f; %1.3f; %1.3f;', ... + PROC_FILE, segment, delay, downlinkSystemDelay, activityFactorDoubleTalk, ... + C_dt(1), C_dt(2), C_dt(3), C_dt(4), ... + C_dt(5), C_dt(6), C_dt(7), C_dt(8)); + +``` + +``` + +disp(str); +if (fid > -1) + fprintf(fid, [str, '\n']); +end; + +str = sprintf('%s; segm. %d; delay %d; DL delay %d; DT level diff; %1.1f; %1.1f; %1.1f; %1.1f; %1.1f; %1.1f; %1.1f; %1.1f;', ... + PROC_FILE, segment, delay, downlinkSystemDelay, ... + L_dt(1), L_dt(2), L_dt(3), L_dt(4), L_dt(5), L_dt(6), L_dt(7), L_dt(8)); +disp(str); +if (fid > -1) + fprintf(fid, [str, '\n']); +end; + +str = sprintf('%s; segm. %d; delay %d; DL delay %d; ST activity %1.3f; %1.3f; %1.3f; %1.3f; %1.3f; %1.3f; %1.3f; %1.3f;', ... + PROC_FILE, segment, delay, downlinkSystemDelay, activityFactorSingleTalk, ... + C_st(1), C_st(2), C_st(3), C_st(4), ... + C_st(5), C_st(6), C_st(7), C_st(8)); +disp(str); +if (fid > -1) + fprintf(fid, [str, '\n']); +end; + +str = sprintf('%s; segm. %d; delay %d; DL delay %d; ST level diff; %1.1f; %1.1f; %1.1f; %1.1f; %1.1f; %1.1f; %1.1f; %1.1f;', ... + PROC_FILE, segment, delay, downlinkSystemDelay, ... + L_st(1), L_st(2), L_st(3), L_st(4), L_st(5), L_st(6), L_st(7), L_st(8)); +disp(str); +if (fid > -1) + fprintf(fid, [str, '\n']); +end; + +%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% +%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% +% +% Write the signal levels to file +% +%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% +%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% +function writeSpeechLevelsToFile(PROC_FILE, segment, fid, ... + activeSpeechLevelProcessed, ... + activeSpeechLevelNearEnd, ... + activeSpeechLevelDownlink, ... + longTermLevelProcessed, ... + longTermLevelNearEnd, ... + longTermLevelDownlink, ... + activityFactorProcessed, ... + activityFactorNearEnd, ... + activityFactorDownlink) + +str = sprintf('%s; segm. %d; Processed signal; active speech level [dBovl]; %3.1f; RMS level [dBovl]; %3.1f; speech activity; %1.3f', ... + PROC_FILE, segment, activeSpeechLevelProcessed, ... + longTermLevelProcessed, activityFactorProcessed); +disp(str); +if (fid > -1) + fprintf(fid, [str, '\n']); +end; + +str = sprintf('%s; segm. %d; Near end signal; active speech level [dBovl]; %3.1f; RMS level [dBovl]; %3.1f; speech activity; %1.3f', ... + PROC_FILE, segment, activeSpeechLevelNearEnd, ... + longTermLevelNearEnd, activityFactorNearEnd); +disp(str); +if (fid > -1) + fprintf(fid, [str, '\n']); +end; + +str = sprintf('%s; segm. %d; Downlink signal; active speech level [dBovl]; %3.1f; RMS level [dBovl]; %3.1f; speech activity; %1.3f', ... + PROC_FILE, segment, activeSpeechLevelDownlink, ... + longTermLevelDownlink, activityFactorDownlink); +disp(str); +if (fid > -1) + fprintf(fid, [str, '\n']); +end; + +``` + +``` + fprintf(fid, [str, '\n']); +end; +``` + +### B.3.7 Other helper functions + +``` +%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% +%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% +% +% Find & separate blocks with consecutive indices +% +%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% +%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% + +function [ConsecutiveBlocks] = findConsecutiveBlocks(FrameIndices) + +D = diff(FrameIndices); +Changes = find(D > 1); +ConsecutiveBlocks = zeros(length(Changes)+1,2); +ConsecutiveBlocks(1,1) = FrameIndices(1); + +for i = 1:length(Changes) + ConsecutiveBlocks(i,2) = FrameIndices(Changes(i)); + if i <= length(Changes) + ConsecutiveBlocks(i+1,1) = FrameIndices(Changes(i)+1); + end +end + +if ConsecutiveBlocks(end,2) == 0 + ConsecutiveBlocks(end,2) = FrameIndices(end); +end +``` + +# --- Annex C (informative): Change history + +| TSG SA# | TSG doc | Spec | CR | Rev | Cat | Vers | New Vers | Subject | +|---------|-----------|--------|------|-----|-----|--------|----------|---------------------------------------------------------------------------------------------------------------------------------------------| +| 08 | | | | | | | 3.0.0 | Approved | +| 09 | SP-000397 | 26.132 | 001 | | F | 3.0.0 | 3.1.0 | Handheld hands-free Test Setup | +| 11 | SP-010107 | 26.132 | 002 | 1 | F | 3.1.0 | 3.2.0 | Harmonisation of test methods for acoustics between 3GPP and GSM | +| 11 | | 26.132 | | | | 3.2.0 | 4.0.0 | Release 4 | +| 11 | SP-010107 | 26.132 | 003 | 1 | B | 4.0.0 | 5.0.0 | Compatibility with testing wideband telephony transmission performance | +| 13 | SP-010454 | 26.132 | 004 | | B | 5.0.0 | 5.1.0 | Extended scope of test signals for Ambient Noise Rejection | +| 13 | SP-010454 | 26.132 | 006 | | F | 5.0.0 | 5.1.0 | Restricted scope of ITU-T P.501 test signals for 3G acoustic tests | +| 13 | SP-010454 | 26.132 | 008 | | A | 5.0.0 | 5.1.0 | Bandwidth of test signals for acoustic testing | +| 15 | SP-020080 | 26.132 | 011 | 1 | A | 5.1.0 | 5.2.0 | Correction of references and editorial changes (wrong decimal separators) | +| 16 | SP-020435 | 26.132 | 016 | | F | 5.2.0 | 5.3.0 | Correction on ANR test for hands-free Ues | +| 21 | SP-030445 | 26.132 | 026 | | F | 5.3.0 | 5.4.0 | Loudness rating measurements at lower bit rates | +| 25 | SP-040649 | 26.132 | 028 | | C | 5.4.0 | 6.0.0 | Change of sending distortion test case | +| 35 | SP-070026 | 26.132 | 0030 | | F | 6.0.0 | 6.1.0 | Reference Update for ITU-T Recommendation P.57 "Artificial Ears" | +| 35 | SP-070026 | 26.132 | 0031 | 1 | F | 6.0.0 | 6.1.0 | Update of reference [16] to P.79-2001 Annex G | +| 35 | SP-070026 | 26.132 | 0032 | 1 | F | 6.0.0 | 6.1.0 | Distinction between narrow-band and wideband telephony in the frequency ranges and loudness rating and STMR weights, and in ANR calculation | +| 36 | | 26.132 | | | | 6.1.0 | 7.0.0 | Version for Release 7 | +| 38 | SP-070759 | 26.132 | 0034 | 2 | F | 7.0.0 | 7.1.0 | Changing the sidetone test to allow type 3.3 or 3.4 artificial ears | +| 42 | SP-080674 | 26.132 | 0035 | 1 | F | 7.1.0 | 7.2.0 | Correction to allow wideband testing for GSM terminals | +| 42 | SP-080685 | 26.132 | 0037 | 3 | C | 7.2.0 | 8.0.0 | Updated test methods for wideband terminal acoustics | +| 43 | SP-090015 | 26.132 | 0038 | | F | 8.0.0 | 8.1.0 | Clarification on Distortion with psophometric filter | +| 43 | SP-090018 | 26.132 | 0036 | 2 | C | 8.1.0 | 9.0.0 | Speech and video telephony terminal acoustic test | +| 45 | SP-090568 | 26.132 | 0040 | 2 | A | 9.0.0 | 9.1.0 | Correction of STMR calculation | +| 45 | SP-090573 | 26.132 | 0041 | 1 | F | 9.0.0 | 9.1.0 | Handling Acoustic Testing with Noise Suppression Algorithms Employed | +| 47 | SP-100021 | 26.132 | 0042 | 1 | F | 9.1.0 | 9.2.0 | Correction of distortion measurements | +| 51 | SP-110042 | 26.132 | 0043 | 5 | B | 9.2.0 | 10.0.0 | Alignment of 3GPP Audio Test Case Specification | +| 52 | SP-110304 | 26.132 | 0045 | 3 | C | 10.0.0 | 10.1.0 | Remaining modifications to EAAT WI | +| 53 | SP-110549 | 26.132 | 0047 | 1 | F | 10.1.0 | 10.2.0 | Correction to Acoustic Echo Control | +| 54 | SP-110793 | 26.132 | 0048 | 1 | F | 10.2.0 | 10.3.0 | Correction to Ambient Noise Rejection Test Procedure | +| 54 | SP-110793 | 26.132 | 0049 | | F | 10.2.0 | 10.3.0 | Clarification of Stability loss test signal | +| 55 | SP-120022 | 26.132 | 0050 | 1 | F | 10.3.0 | 10.4.0 | Correction of receiving distortion | +| 57 | SP-120501 | 26.132 | 0053 | 2 | F | 10.4.0 | 10.5.0 | Addition of 1/12 octave diffuse field table values with frequencies according to IEC 1260 (for information) | +| 57 | SP-120503 | 26.132 | 0052 | 3 | B | 10.5.0 | 11.0.0 | Addition of UE delay test method | +| 57 | SP-120503 | 26.132 | 0054 | 1 | B | 10.5.0 | 11.0.0 | Extension of Acoustic Tests | +| 58 | SP-120760 | 26.132 | 0055 | 1 | F | 11.0.0 | 11.1.0 | Correction of references and levels | +| 59 | SP-130017 | 26.132 | 0056 | | F | 11.1.0 | 11.2.0 | Alignment of free-field definition with ISO 3745 and ITU-T Rec. P.340 | +| 59 | SP-130017 | 26.132 | 0057 | 1 | F | 11.1.0 | 11.2.0 | Voiding of ambient noise rejection test cases | +| 60 | SP-130185 | 26.132 | 0058 | 1 | F | 11.2.0 | 11.3.0 | Update of the reference algorithm for echo control characteristics evaluation in Annex B | +| 60 | SP-130185 | 26.132 | 0061 | 2 | F | 11.2.0 | 11.3.0 | UE delay test method | +| 62 | SP-130568 | 26.132 | 0062 | | F | 11.3.0 | 11.4.0 | Correction to references | +| 62 | SP-130563 | 26.132 | 0066 | 2 | A | 11.3.0 | 11.4.0 | STMR - adaptation to modern form factors | +| 66 | SP-140721 | 26.132 | 0071 | - | F | 11.4.0 | 11.5.0 | Correction to UE receiving loudness rating performance test for wideband telephony | +| 66 | SP-140719 | 26.132 | 0078 | 1 | A | 11.4.0 | 11.5.0 | Correction of broadband signal level at the hands free reference point | + diff --git a/marked/Rel-11/26_series/26141/raw.md b/marked/Rel-11/26_series/26141/raw.md new file mode 100644 index 0000000000000000000000000000000000000000..bf9672fe1392a572c370010d77672854a16b5622 --- /dev/null +++ b/marked/Rel-11/26_series/26141/raw.md @@ -0,0 +1,361 @@ + + + + + + +# --- Contents + +| | | +|-------------------------------------------------------------------------|-----------| +| Foreword ..... | 4 | +| Introduction ..... | 4 | +| 1 Scope..... | 5 | +| 2 References..... | 5 | +| 3 Definitions, symbols and abbreviations..... | 7 | +| 3.1 Definitions..... | 7 | +| 3.2 Abbreviations ..... | 8 | +| 4 Formats for Static Media..... | 8 | +| 4.1 Text..... | 8 | +| 4.2 Still Image ..... | 9 | +| 4.3 Bitmap Graphics..... | 9 | +| 5 Formats for Continuous Media ..... | 9 | +| 5.1 Speech ..... | 9 | +| 5.2 Audio..... | 9 | +| 5.3 Video ..... | 10 | +| 5.4 File Format for video and associated speech/audio media types ..... | 10 | +| 5.5 Synthetic audio..... | 11 | +| 5.6 Vector graphics ..... | 11 | +| 6 Media synchronisation and presentation format ..... | 11 | +| Annex A (informative): CSI Handling ..... | 12 | +| A.1 Introduction..... | 12 | +| A.2 Sharing personal content during CS voice call ..... | 12 | +| A.3 Sharing personal content during CS multimedia call..... | 12 | +| Annex B (informative): Change history..... | 13 | + +# --- Foreword + +This Technical Specification has been produced by the 3rd Generation Partnership Project (3GPP). + +The contents of the present document are subject to continuing work within the TSG and may change following formal TSG approval. Should the TSG modify the contents of the present document, it will be re-released by the TSG with an identifying change of release date and an increase in version number as follows: + +Version x.y.z + +where: + +- x the first digit: + - 1 presented to TSG for information; + - 2 presented to TSG for approval; + - 3 or greater indicates TSG approved document under change control. +- y the second digit is incremented for all changes of substance, i.e. technical enhancements, corrections, updates, etc. +- z the third digit is incremented when editorial only changes have been incorporated in the document. + +# --- Introduction + +The 3GPP Technical Specifications TS 22.340 [55] and TS 22.141 [56] define the requirements for the 3GPP IP Multimedia Subsystem (IMS) based messaging and presence services. This Technical Specification takes the requirements into account when defining the minimal baseline and optional media codecs and message container format to be used by IMS Messaging and associated Presence service, when supported. + +IMS Messaging services incorporate one or more of the following messaging types Immediate messaging, Deferred delivery messaging, and Session based messaging. With Immediate messaging the sender expects immediate message delivery in what is perceived as real time compared with Deferred messaging where the sender expects the network to deliver the message as soon as the recipient becomes available. With Session based messaging a communications association is established between two or more users before communication can take place. In the simplest form Session based messaging may be a direct communication between two users. This specification defines the media types and container formats for both the Immediate message type and the Session based message type. + +The specification provides the ability to have an interoperable baseline set of media types for messaging and presence services, that will simultaneously maximise the technology re-use of the already existing 3GPP services with media types, defined in TS 26.140 [13] and TS 26.234 [14]. Simultaneously, the specification will provide the ability to indicate the IMS system about the complete set of UE media and storage capabilities relevant for the IMS messaging and presence service. + +For IMS terminals capable of Combined CS and IMS (CSI) operation [59][60], the specification provides an Annex with guidelines on how to combine IMS media with CS calls. + +# --- 1 Scope + +The present document specifies the basic media formats and codecs to be used in the IMS Messaging and Presence services, including CSI. It defines the mandatory 'baseline' set of media types for the services. Additionally, it also targets to allow possible message content type enhancements, either 3GPP-standardized or other generally used media types, in a flexible way. + +# --- 2 References + +The following documents contain provisions which, through reference in this text, constitute provisions of the present document. + +- References are either specific (identified by date of publication, edition number, version number, etc.) or non-specific. +- For a specific reference, subsequent revisions do not apply. +- For a non-specific reference, the latest version applies. In the case of a reference to a 3GPP document (including a GSM document), a non-specific reference implicitly refers to the latest version of that document *in the same Release as the present document*. + +- [1] 3GPP TR 21.905: "Vocabulary for 3GPP Specifications". +- [2] The Unicode Consortium: "The Unicode Standard", Version 2.0, Addison-Wesley Developers Press, 1996. URL: . +- [3] ANSI X3.4, 1986: "Information Systems; Coded Character Set 7 Bit; American National Standard Code for Information Interchange". +- [4] ISO/IEC 8859-1:1998: "Information technology; 8-bit single-byte coded graphic character sets; Part 1: Latin alphabet No. 1". +- [5] IETF; RFC 2279: "UTF-8, A Transformation format of ISO 10646", URL: . +- [6] 3GPP TS 24.011: "Point-to-Point (PP) Short Message Service (SMS) support on mobile radio interface". +- [7] 3GPP TS 26.090: "AMR speech Codec Transcoding functions". +- [8] ITU-T Recommendation T.81: "Information technology; Digital compression and coding of continuous-tone still images: Requirements and guidelines". +- [9] "JPEG File Interchange Format", Version 1.02, September 1, 1992. +- [10] ITU-T Recommendation H.263 (02/98): "Video coding for low bit rate communication". +- [11] ITU-T Recommendation H.263 – Annex X (03/04): "Annex X: Profiles and levels definition". +- [12] ISO/IEC 14496-2 (2004): "Information technology - Coding of audio-visual objects - Part 2: Visual". +- [13] 3GPP TS 26.140: "Multimedia Messaging Service (MMS); Media formats and codecs" +- [14] 3GPP TS 26.234: "End-to-end transparent streaming Service; Protocols and codecs". +- [15] CompuServe Incorporated: "GIF Graphics Interchange Format: A Standard defining a mechanism for the storage and transmission of raster-based graphics information", Columbus, OH, USA, 1987. +- [16] Compuserve Incorporated, Columbus, Ohio (1990): "Graphics Interchange Format (Version 89a)". +- [17] IETF RFC 2083: "PNG (Portable Networks Graphics) Specification version 1.0 ", T. Boutell, et. al., March 1997. + +- [18] ITU-T Recommendation H.263 (1998): "Video coding for low bit rate communication - Annex X, Profiles and Levels Definition". +- [19] ISO/IEC 14496-3:2001, "Information technology -- Coding of audio-visual objects -- Part 3: Audio". +- [20] W3C Last Call Working Draft: "Scalable Vector Graphics (SVG) 1.2", , October 2004. +- [21] W3C Last Call Working Draft: "Mobile SVG Profile: SVG Tiny, Version 1.2", , August 2004. +- [22] 3GPP 22.140: "Service Aspects; Stage 1; Multimedia Messaging Service". +- [23] 3GPP 23.140: "Multimedia Messaging Service (MMS); Functional Description; Stage 2". +- [24] W3C Recommendation: "Synchronized Multimedia Integration Language (SMIL 2.0)", , August 2001. +- [25] IETF RFC 2046: "Multipurpose Internet Mail Extensions (MIME) Part Two: Media Types". +- [26] 3GPP TS 26.071: "Mandatory Speech Codec speech processing functions; AMR Speech Codec; General description". +- [27] 3GPP TS 26.171: "AMR speech codec; General description". +- [28] Scalable Polyphony MIDI Specification Version 1.0, RP-34, MIDI Manufacturers Association, Los Angeles, CA, February 2002. +- [29] Scalable Polyphony MIDI Device 5-to-24 Note Profile for 3GPP, RP-35, MIDI Manufacturers Association, Los Angeles, CA, February 2002. +- [30] WAP Forum Specification: "XHTML Mobile Profile", , October 2001. +- [31] "Standard MIDI Files 1.0", RP-001, in "The Complete MIDI 1.0 Detailed Specification, Document Version 96.1 " The MIDI Manufacturers Association, Los Angeles, CA, USA, February 1996. +- [32] IETF RFC 3267: " RTP payload format and file storage format for the Adaptive Multi-Rate (AMR) Adaptive Multi-Rate Wideband (AMR-WB) audio codecs ", March 2002. +- [33] 3GPP TS 26.244: "Transparent end-to-end packet switched streaming service (PSS); 3GPP file format (3GP)" +- [34] 3GPP TS 26.246: "Transparent end-to-end packet switched streaming service (PSS); 3GPP SMIL Language Profile". +- [35] 3GPP TS 26.245: "Transparent end-to-end packet switched streaming service (PSS); Timed text format" +- [36] IETF RFC 1952 "GZIP file format specification version 4.3", Deutsch P, May 1996. +- [37] (void) +- [38] Mobile DLS, MMA specification v1.0. RP-41 Los Angeles, CA, USA. 2004. +- [39] Mobile XMF Content Format Specification, MMA specification v1.0., RP-42, Los Angeles, CA, USA. 2004. +- [40] 3GPP TS 26.090: "Mandatory Speech Codec speech processing functions; Adaptive Multi-Rate (AMR) speech codec; Transcoding functions". +- [41] 3GPP TS 26.073: "ANSI-C code for the Adaptive Multi Rate (AMR) speech codec". +- [42] 3GPP TS 26.104: "ANSI-C code for the floating-point Adaptive Multi Rate (AMR) speech codec". + +- [43] 3GPP TS 26.190: "Speech Codec speech processing functions; AMR Wideband speech codec; Transcoding functions". +- [44] 3GPP TS 26.173: "ANSI-C code for the Adaptive Multi Rate - Wideband (AMR-WB) speech codec". +- [45] 3GPP TS 26.204: "ANSI-C code for the Floating-point Adaptive Multi-Rate Wideband (AMR-WB) speech codec". +- [46] 3GPP TS 26.290: "Extended AMR Wideband codec; Transcoding functions". +- [47] 3GPP TS 26.304: "ANSI-C code for the Floating-point; Extended AMR Wideband codec". +- [48] 3GPP TS 26.273: "ANSI-C code for the Fixed-point; Extended AMR Wideband codec". +- [49] 3GPP TS 26.401: "General audio codec audio processing functions; Enhanced aacPlus general audio codec; General description". +- [50] 3GPP TS 26.410: "General audio codec audio processing functions; Enhanced aacPlus general audio codec; Floating-point ANSI-C code". +- [51] 3GPP TS 26.411: "General audio codec audio processing functions; Enhanced aacPlus general audio codec; Fixed-point ANSI-C code". +- [52] ITU-T Recommendation H.264 (2003): "Advanced video coding for generic audiovisual services" | ISO/IEC 14496-10:2003: "Information technology – Coding of audio-visual objects – Part 10: Advanced Video Coding". +- [53] ISO/IEC 14496-10/FDAM1: "AVC Fidelity Range Extensions". +- [54] 3GPP TS 23.228; 3rd Generation Partnership Project; Technical Specification Group Services and System Aspects; IP Multimedia Subsystem (IMS); Stage 2. +- [55] 3GPP TS 22.340: "Stage 1, IMS Messaging Service". +- [56] 3GPP TS 22.141: "Stage 1, Presence Service". +- [57] Standard ECMA-327: "ECMAScript 3rd Edition Compact Profile", June 2001. +- [58] "Exchangeable image file format for digital still cameras: EXIF 2.2", Specification by the Japan Electronics and Information Technology Industries Association (JEITA), April 2002, URL: . +- [59] 3GPP TS 22.279: "Combining CS and IMS services; Stage 1". +- [60] 3GPP TS 23.279: "Combining CS and IMS services; Stage 2". +- [61] 3GPP TS 26.235: "Packet switched conversational multimedia applications; Default codecs". +- [62] 3GPP TR 26.936: "Performance characterization of 3GPP audio codecs". + +# --- 3 Definitions, symbols and abbreviations + +## 3.1 Definitions + +**Deferred delivery messaging:** A type of IMS Messaging service by which the sender expects the network to deliver the message as soon as the recipient becomes available. + +**Immediate messaging:** A type of IMS Messaging service by which the sender expects immediate message delivery in (near) real time fashion. + +**IMS Messaging services:** A group of services, supported by capabilities of the 3GPP IP Multimedia Subsystem 3GPP TS 22.228 [54], that allows an IMS user to send and receive messages to other users. IMS messaging services comprise of one or more types: Immediate messaging, Session based messaging and Deferred delivery messaging. + +**Session based messaging:** A type of IMS Messaging service by which the sender expects immediate message delivery in (near) real time fashion. In addition the sender(s) and the receiver(s) have to join to a messaging session e.g. chat room, before message exchange can take place. + +**continuous media:** media with an inherent notion of time, in the present document speech, audio, synthetic audio and video. + +**static media:** media that itself does not contain an element of time, in the present document all media not defined as continuous media. + +**scene description:** description of the spatial layout and temporal behaviour of a presentation, it can also contain hyperlinks. + +## 3.2 Abbreviations + +| | | +|------------------|-------------------------------------------------------------| +| 3GP | 3GPP file format | +| AAC | Advanced Audio Coding | +| AMR | Adaptive Multi-rate Codec | +| AVC | Advanced Video Coding | +| CC/PP | Composite Capability/Preference Profiles | +| CSI | Combination of CS and IMS services | +| DLS | Downloadable Sounds | +| Enhanced aacPlus | MPEG-4 High Efficiency AAC plus MPEG-4 Parametric Stereo | +| EXIF | Exchangeable image file format | +| GIF | Graphics Interchange Format | +| H.263 | ITU-T video codec | +| IP | Internet Protocol | +| IMS | IP Multimedia Subsystem | +| ITU-T | International Telecommunications Union - Telecommunications | +| JFIF | JPEG File Interchange Format | +| JPEG | Joint Picture Expert Group | +| MIDI | Musical Instrument Digital Interface | +| MIME | Multipurpose Internet Mail Extensions | +| MM | Multimedia Message | +| MMS | Multimedia Messaging Service | +| MPEG | Motion Picture Expert Group | +| MP4 | MPEG-4 file format | +| PSS | Packet-switched Streaming Service | +| SBR | Spectral Band Replication | +| SP-MIDI | Scalable Polyphony MIDI | +| SVG | Scalable Vector Graphics | +| UTF-8 | Unicode Transformation Format (the 8-bit form) | +| XMF | Extensible Music Format | + +# --- 4 Formats for Static Media + +Multiple media elements shall be combined into a composite single IMS message using MIME multipart content type format as defined in RFC 2046 [25]. The media type of a single IMS message element shall be identified by its appropriate MIME type whereas the media format shall be indicated by its appropriate MIME subtype. + +In order to guarantee a minimum support and compatibility between IMS Messaging and Presence Service capable terminals and OMA IMPS 1.1 capable terminals, IMS Messaging User Agent and IMS Presence User Agent supporting specific media types shall comply with the following selection of media formats: + +## 4.1 Text + +Plain text. Any character encoding (charset) that contains a subset of the logical characters in Unicode [2] shall be used (e.g. US-ASCII [3], ISO-8859-1 [4], UTF-8 [5], Shift\_JIS, etc.). + +Unrecognized subtypes of "text" shall be treated as subtype "plain" as long as the MIME implementation knows how to handle the charset. Any other unrecognized subtype and unrecognized charset shall be treated as "application/octet-stream". + +## 4.2 Still Image + +For IMS terminals supporting still images, ISO/IEC JPEG [8] together with JFIF [9] shall be supported. The support for ISO/IEC JPEG only apply to the following two modes: + +- mandatory: baseline DCT, non-differential, Huffman coding, as defined in table B.1, symbol 'SOF0' in [8]; +- optional: progressive DCT, non-differential, Huffman coding, as defined in table B.1, symbol 'SOF2' [8]. + +For JPEG baseline DCT, EXIF compressed image file format should also be supported, as defined in [58]. In that case there is no requirement for the MMS Messaging and Presence client to interpret or present the EXIF parameters recorded in the file. + +## 4.3 Bitmap Graphics + +For IMS terminals, supporting bitmap graphics, the following bitmap graphics formats should be supported: + +- GIF87a [15]; +- GIF89a [16]; +- PNG [17]. + +# --- 5 Formats for Continuous Media + +In order to guarantee a minimum support and compatibility between IMS Messaging and Presence Service capable terminals and MMS capable terminals that offer support of continuous media formats (section 5) and media synchronisation and scene description (see section 6), IMS Messaging User Agent and IMS Presence User Agent supporting specific media types should in addition to formats listed in section 4 of this document comply with the following selection of media formats: + +## 5.1 Speech + +For IMS terminals supporting speech, the AMR codec shall be supported for narrow-band speech [26][40][41][42]. + +The AMR wideband speech codec [27] [43][44][45] shall be supported when wideband speech working at 16 kHz sampling frequency is supported. + +When using speech media type alone, AMR or AMR-WB data stored according to the file format specified in [32] should be supported. The mandatory format is defined in clause 5.4. + +Multi-channel sessions shall not be used. + +## 5.2 Audio + +For IMS terminals supporting audio, one or both of the following two audio codecs should be supported: + +- Enhanced aacPlus [49][50][51] +- Extended AMR-WB [46][47][45] + +There is no requirement that a terminal supporting decoding by one of the codecs shall also support encoding by that codec. + +Specifically, based on the audio codec selection test results Extended AMR-WB is strong for the scenarios marked with blue, Enhanced aacPlus is strong for the scenarios marked with orange, and both are strong for the scenarios marked with green colour in the table below: + +| Content type | Music | Speech over Music | Speech between Music | Speech | +|----------------|-------|-------------------|----------------------|--------| +| Bit rate | | | | | +| 14 kbps mono | | | | | +| 18 kbps stereo | | | | | +| 24 kbps stereo | | | | | +| 24 kbps mono | | | | | +| 32 kbps stereo | | | | | +| 48 kbps stereo | | | | | + +More recent information on the performance of the codecs based on more recent versions of the codecs can be found in TR 26.936 [62]. + +Enhanced aacPlus decoder is also able to decode MPEG-4 AAC LC content. + +Extended AMR-WB decoder is also able to decode AMR-WB content. + +In addition, MPEG-4 AAC Low Complexity and MPEG-4 AAC Long Term Prediction object types [19] may be supported. The maximum sampling rate to be supported by the decoder is 48 kHz. The channel configurations to be supported are mono (1/0) and stereo (2/0). + +## 5.3 Video + +For IMS terminals supporting video, ITU-T Recommendation H.263 [10][11] profile 0 level 45 shall be supported. In addition: + +- H.263 Profile 3 Level 45 [10][11]; +- MPEG-4 Visual Simple Profile Level 0b, [12]; +- H.264 (AVC) Baseline Profile Level 1b [52][53] with constraint\_set1\_flag=1; + +should be supported. There are no requirements on output timing conformance of H.264 (AVC) decoding (Annex C of [52]). + +An optional video buffer model is given in Annex G of document [14]. It shall not be used with H.264 (AVC). + +NOTE: ITU-T Recommendation H.263 profile 0 has been mandated to ensure that video-enabled IMS Messaging & Presence user agent supports a minimum baseline video capability. Both H.263 and MPEG-4 Visual decoders can decode an H.263 profile 0 bit stream. It is strongly recommended, though, that an H.263 profile 0 bit stream is transported and stored as H.263 and not as MPEG-4 visual (short header), as MPEG-4 Visual is not mandated by IMS Messaging & Presence services. + +## 5.4 File Format for video and associated speech/audio media types + +To ensure interoperability for the transport of video and associated speech/audio in an IMS Messaging and Presence client, the 3GPP file format with Basic profile shall be supported. + +The usage of the 3GPP file format shall follow the technical specifications and the implementation guidelines specified in TS 26.244 [33]. + +## 5.5 Synthetic audio + +For IMS terminals supporting synthetic audio, the Scalable Polyphony MIDI (SP-MIDI) content format defined in Scalable Polyphony MIDI Specification [28] and the device requirements defined in Scalable Polyphony MIDI Device 5-to-24 Note Profile for 3GPP [29] should be supported. + +SP-MIDI content is delivered in the structure specified in Standard MIDI Files 1.0 [31], either in format 0 or format 1. + +In addition the Mobile DLS instrument format defined in [38] and the Mobile XMF content format defined in [39] should be supported. + +A MSS client supporting Mobile DLS shall meet the minimum device requirements defined in [38] in section 1.3 and the requirements for the common part of the synthesizer voice as defined in [38] in sections 1.2.1.2. If Mobile DLS is supported, wavetables encoded with the G.711 A-law codec (wFormatTag value 0x0006, as defined in [38]) shall also be supported. The optional group of processing blocks as defined in [38] may be supported. Mobile DLS resources are delivered either in the file format defined in [38], or within Mobile XMF as defined in [39]. For Mobile DLS files delivered outside of Mobile XMF, the loading application should unload Mobile DLS instruments so that the sound bank required by the SP-MIDI profile [29] is not persistently altered by temporary loadings of Mobile DLS files. + +Content that pairs Mobile DLS and SP-MIDI resources is delivered in the structure specified in Mobile XMF [39]. As defined in [39], a Mobile XMF file shall contain one SP-MIDI SMF file and no more than one Mobile DLS file. MMS clients supporting Mobile XMF must not support any other resource types in the Mobile XMF file. Media handling behaviours for the SP-MIDI SMF and Mobile DLS resources contained within Mobile XMF are defined in [39]. + +## 5.6 Vector graphics + +For IMS terminals supporting 2D vector graphics, the Scalable Vector Graphics (SVG) Tiny 1.2 format [20][21] and ECMAScript [54] shall be supported. + +NOTE 1: The compression format for SVG content is GZIP [35], in accordance with the SVG specification [20]. + +NOTE 2: Only media formats supported by IMS Messaging and Presence, as specified in clauses 4 and 5 of this specification, shall be used. MMS Messaging and Presence clients do not support the Ogg Vorbis format. + +NOTE 3: Content creators of SVG Tiny 1.2 for IMS Messaging and Presence clients are strongly recommended to follow the content creation guidelines provided for PSS clients in Annex L of [14]. + +NOTE 4: If SVG Tiny 1.2 will not be published within a reasonable timeframe, the decision to adopt SVG Tiny 1.2 in favour of SVG Tiny 1.1 may be reconsidered. + +# --- 6 Media synchronisation and presentation format + +The 3GPP IMS Messaging and Presence uses a subset of SMIL 2.0 [24] for media synchronisation and scene description. IMS clients and servers with support for media synchronization and scene descriptions shall support the 3GPP SMIL Language Profile defined in [34]. + +- This profile is a subset of the SMIL 2.0 Language Profile but a superset of the SMIL 2.0 Basic Language Profile. Document [34] also includes an informative annex A that provides guidelines for SMIL content authors. + +Additionally, XHTML Mobile Profile [30] for scene description should be supported. IMS clients and servers with support for scene descriptions based on XHTML shall support XHTML Mobile Profile [30], defined by the WAP Forum. + +- XHTML Mobile Profile is a subset of XHTML 1.1 but a superset of XHTML Basic. + +# --- Annex A (informative): CSI Handling + +## A.1 Introduction + +The Combination of CS and IMS services (CSI) is an operation mode combining circuit switch calls and IMS services, where the UE presents the CS and IMS services within one context to the user [59][60]. However, the capability to simultaneously render certain media types of a CS call and IMS session may be limited by a UE and capability exchange alone may not be enough to resolve such conflicts. For instance: + +- During a CS speech call, a UE may not be able to render additional speech accompanying a video clip in an IMS session. This limitation is not clear if the UE has indicated that it is capable of receiving video clips. +- During a CS multimedia call, a UE may not be able to both display video from the CS call and images from the IMS session. Although the UE is not capable to fully render images and video simultaneously, it may be possible to view images in front of video. + +The above conflicts are resolved by applying default rules specified in [59]. This Annex describes the UE behaviour for a number of scenarios drawn from the rules in [59]. This list may be extended in future versions of this specification. + +Note that the IMS media types and formats applicable to CSI are specified in: + +- clauses 6 and 9 of reference [61] for streamed media; +- clauses 4 and 5 of the present document for media delivered in messages. + +## --- A.2 Sharing personal content during CS voice call + +In a person-2-person communication, participants can combine a CS voice call with an IMS session and share content such as still images and video. In particular participants may share media content that is (or has been) created by the participants in the session. + +TS 22.279 [59] defines that if media, or parts thereof, accepted by a user cannot be rendered by the UE simultaneously with the CS call, conflicts shall be resolved such that the user is presented with CS speech with preference over IMS speech/audio. + +## --- A.3 Sharing personal content during CS multimedia call + +In a person-2-person communication, participants can combine a CS multimedia call (3G-324M) with an IMS session and share content such as still images. In particular participants may share media content that is (or has been) created by the participants in the session. + +TS 22.279 [59] defines that if media, or parts thereof, accepted by a user cannot be rendered by the UE simultaneously with the CS call, conflicts shall be resolved such that the user is presented with: + +- CS speech with preference over IMS speech/audio; +- IMS video and images with preference over CS video. + +# Annex B (informative): Change history + +| Change history | | | | | | | | | +|----------------|-----------|--------|------|-----|-----|-------------------------------------------------|--------|--------| +| TSG SA# | SA Doc. | Spec | CR | Rev | Cat | Subject/Comment | Old | New | +| 26 | SP-040835 | 26.141 | | | | Version 1.0.0 approved at TSG SA#26 | 1.0.0 | 6.0.0 | +| 27 | SP-050098 | 26.141 | 001 | 1 | F | Editorial correction on missing IMS Presence UA | 6.0.0 | 6.1.0 | +| 30 | SP-050789 | 26.141 | 0003 | 2 | F | Adding missing reference numbers | 6.1.0 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0000000000000000000000000000000000000000..b5dd490dd3908807ab68ae8400ee8ae20055c75d --- /dev/null +++ b/marked/Rel-11/26_series/26142/ead623e6a8ec701df64567695a57d01c_img.jpg @@ -0,0 +1,3 @@ +version https://git-lfs.github.com/spec/v1 +oid sha256:cba005bbf32331cd72d728e1e61543c49544ee019393fd7988fba8d4434bb512 +size 6871 diff --git a/marked/Rel-11/26_series/26142/raw.md b/marked/Rel-11/26_series/26142/raw.md new file mode 100644 index 0000000000000000000000000000000000000000..e37fbbc8dcdd012245453f549a092878906dccf9 --- /dev/null +++ b/marked/Rel-11/26_series/26142/raw.md @@ -0,0 +1,1511 @@ + + + + + + +# Contents + +| | | +|----------------------------------------------------------|----| +| Foreword ..... | 5 | +| Introduction ..... | 5 | +| 1 Scope..... | 6 | +| 2 References..... | 6 | +| 3 Definitions and abbreviations ..... | 7 | +| 3.1 Definitions..... | 7 | +| 3.2 Abbreviations ..... | 8 | +| 4 Overview and architecture ..... | 9 | +| 5 Media-type definition..... | 9 | +| 5.1 Introduction ..... | 9 | +| 5.2 Media type components ..... | 9 | +| 5.3 Namespace ..... | 10 | +| 5.4 Scene description..... | 10 | +| 5.4.1 Base Scene Description ..... | 10 | +| 5.4.2 Scene Description Extensions ..... | 10 | +| 5.4.2.1 Introduction..... | 10 | +| 5.4.2.2 Rectangular clipping of a graphical object ..... | 10 | +| 5.4.2.3 Full-screen video..... | 10 | +| 5.4.2.4 Full-screen SVG ..... | 10 | +| 5.4.2.5 Attributes clipBegin and clipEnd..... | 11 | +| 5.4.2.6 Update Streams ..... | 11 | +| 5.4.2.7 Synchronization of Media Streams..... | 11 | +| 5.4.2.8 Screen orientation ..... | 11 | +| 5.4.2.9 Current-Time Indication ..... | 12 | +| 5.4.2.10 Active attribute ..... | 13 | +| 5.5 Scene Commands ..... | 13 | +| 5.5.1 Scene Updates ..... | 13 | +| 5.5.2 State management commands ..... | 13 | +| 5.5.3 Activate and Deactivate..... | 14 | +| 5.5.4 Distributed Random Access Points ..... | 15 | +| 5.5.4.1 Introduction..... | 15 | +| 5.5.4.2 DRAP syntax and semantics..... | 15 | +| 5.5.5 Immediate Script Execution ..... | 16 | +| 5.5.6 Seeking in the DIMS Stream..... | 16 | +| 5.6 DIMS Unit Definition ..... | 17 | +| 5.6.1 Definition..... | 17 | +| 5.6.2 DIMS Unit Header ..... | 17 | +| 5.7 Timing model..... | 18 | +| 5.8 Processing Model..... | 18 | +| 5.9 Random Access, Tune-in and Error Recovery..... | 20 | +| 5.9.1 Introduction ..... | 20 | +| 5.9.2 Random Access Points in Primary Streams ..... | 20 | +| 5.9.3 Random Access Points in Secondary Streams ..... | 20 | +| 5.9.4 Error Recovery ..... | 20 | +| 6 Interaction and Scripting..... | 21 | +| 6.1 Local interaction..... | 21 | +| 6.1.1 DOM Level 3 events ..... | 21 | +| 6.1.2 Media Access Events..... | 21 | +| 6.1.3 Screen Orientation Events ..... | 21 | +| 6.1.4 Other Events ..... | 21 | +| 6.2 Remote interaction ..... | 22 | +| 6.3 Scripting ..... | 22 | +| 7 Transport..... | 23 | +| 7.1 Overview ..... | 23 | + +| | | | +|-----------------------------|------------------------------------------------------|-----------| +| 7.2 | Storage in ISO Base Media File Format Files..... | 23 | +| 7.2.1 | Introduction ..... | 23 | +| 7.2.2 | Stream Type..... | 24 | +| 7.2.3 | Track and Media Header fields ..... | 24 | +| 7.2.4 | Sample Dependency Table ..... | 24 | +| 7.2.5 | Sample Entry Name and Format ..... | 25 | +| 7.2.6 | Sample Format..... | 26 | +| 7.2.7 | Other Resources..... | 26 | +| 7.2.8 | Sync Samples..... | 26 | +| 7.2.9 | Separate Redundant Track..... | 26 | +| 7.3 | RTP Payload format for DIMS Streams ..... | 27 | +| 7.3.1 | Priority ..... | 27 | +| 7.3.2 | RTP Packet format ..... | 27 | +| 7.3.2.1 | Introduction..... | 27 | +| 7.3.2.2 | RTP Header Usage..... | 28 | +| 7.3.2.3 | Common Packet Header ..... | 28 | +| 7.3.2.4 | Aggregation Packet..... | 29 | +| 7.3.2.5 | Fragmentation Packets ..... | 29 | +| 7.3.3 | SDP Parameters ..... | 30 | +| 7.3.4 | Separate Redundant Stream..... | 30 | +| 8 | Profiles and Levels..... | 31 | +| 8.1 | Profiles ..... | 31 | +| 8.1.1 | Introduction ..... | 31 | +| 8.1.2 | Mobile profile ..... | 31 | +| 8.2 | Levels ..... | 32 | +| 8.2.1 | Introduction ..... | 32 | +| 8.2.2 | Level Axes..... | 32 | +| 8.2.3 | Mobile Profile Level 10 definition ..... | 33 | +| 8.2.4 | Void ..... | 33 | +| 9 | Content usage guidelines ..... | 33 | +| 10 | Security and Content Protection Considerations ..... | 34 | +| 11 | Registered Types..... | 34 | +| 11.1 | RTP Payload format MIME Type..... | 34 | +| 11.2 | 'Codecs' Parameter for 3GP files..... | 36 | +| Annex A (normative): | Conformance Criteria..... | 37 | +| Annex B (informative): | Change history..... | 38 | + +# --- Foreword + +This Technical Specification has been produced by the 3rd Generation Partnership Project (3GPP). + +The contents of the present document are subject to continuing work within the TSG and may change following formal TSG approval. Should the TSG modify the contents of the present document, it will be re-released by the TSG with an identifying change of release date and an increase in version number as follows: + +Version x.y.z + +where: + +- x the first digit: + - 1 presented to TSG for information; + - 2 presented to TSG for approval; + - 3 or greater indicates TSG approved document under change control. +- y the second digit is incremented for all changes of substance, i.e. technical enhancements, corrections, updates, etc. +- z the third digit is incremented when editorial only changes have been incorporated in the document. + +# --- Introduction + +Dynamic and Interactive Multimedia Scenes (DIMS) is a dynamic, interactive, scene-based media system which enables display and interactive control of multimedia data such as audio, video, graphics, images and text. It ranges from a movie enriched with vector graphic overlays and interactivity (possibly enhanced with closed captions), to complex multi-step services with fluid interaction/interactivity and different media types at each step. The demand for such Rich Media service is increasing at a high pace, spurred by the development of the next generation mobile infrastructure and the generalization of TV content to new mobile environments. + +In the case of a video portal application, subscribers can watch TV, video and audio enriched with additional data (graphics, text, images) in streaming, progressive download or offline mode. DIMS provides a convenient and natural way to browse rich-media services, a web-like access (content available in less than three clicks, easy discovery, no learning curve), a permanent refresh of content through dynamic updates available on the fly and decreasing latency by allowing the visualization of data as soon as possible. + +Content can be synchronized up to a frame-accurate basis (e.g. to ensure content providers and operators that voting will start and stop at a precise time during a vote within an interactive show or to allow karaoke text flows). + +# --- 1 Scope + +DIMS defines a dynamic rich-media system, including a media type, its packaging, delivery, and interaction with the local terminal, user, and other local and remote sub-systems. Enhanced end-user experiences are provided by the coordinated management and synchronization of media and events, combined with end-user interaction. + +The DIMS media type can be used as a generic media type, allowing creating dynamic interactive rich-media services and can also benefit, or be used in association with other media types (e.g.: audio codecs, video codecs, XHTML browser, etc.). + +# --- 2 References + +The following documents contain provisions which, through reference in this text, constitute provisions of the present document. + +- References are either specific (identified by date of publication, edition number, version number, etc.) or non-specific. +- For a specific reference, subsequent revisions do not apply. +- For a non-specific reference, the latest version applies. In the case of a reference to a 3GPP document (including a GSM document), a non-specific reference implicitly refers to the latest version of that document *in the same Release as the present document*. + +[1] W3C Candidate Recommendation: "Scalable Vector Graphics (SVG) Tiny 1.2 Specification". + +NOTE: Available at: . + +[2] Open Mobile Alliance (July 2004): "ECMAScript Mobile Profile 1.0". + +[3] ISO/IEC 14496-20:2006: "Information technology - Coding of audio-visual objects - Part 20: Lightweight Application Scene Representation (LASEr) and Simple Aggregation Format (SAF)", including ISO/IEC 14496-20:2006/COR1, ISO/IEC 14496-20:2006/AMD1. + +[4] ISO/IEC 14496-22: "Information technology - Coding of audio-visual objects - Part 22: Open Font Format". + +[5] W3C Recommendation (December 2005): "Synchronized Multimedia Integration Language (SMIL 2.1)". + +NOTE: Available at: . + +[6] 3GPP TS 26.140: "Multimedia Messaging Service (MMS); Media format and codecs". + +[7] 3GPP TS 26.234: "Transparent end-to-end Packet-switched Streaming Service (PSS); Protocols and codecs". + +[8] 3GPP TS 26.244: "Transparent end-to-end packet switched streaming service (PSS); 3GPP file format (3GP)". + +[9] The Unicode Consortium: "The Unicode Standard", Version 5.0, . + +[10] ISO/IEC 14496-12: "Information technology - Coding of audio-visual objects - Part 12: ISO base media file format". + +[11] IETF RFC 1952 (May 1996): "GZIP file format specification version 4.3", P. Deutsch. + +[12] IETF RFC 2616 (June 1999): "Hypertext Transfer Protocol -- HTTP/1.1", R. Fielding, J. Gettys, J. Mogul, H. Frystyk, L. Masinter, P. Leach, T. Berners-Lee. + +[13] IETF RFC 4329 (April 2006): "Scripting Media Types", B. Hoehrmann. + +[14] IETF RFC 4281 (November 2005): "The Codecs Parameter for "Bucket" Media Types", R. Gellens, D. Singer, P. Frodjh. + +- [15] IETF STD 0064/RFC 3550 (July 2003) "RTP: A Transport Protocol for Real-Time Applications", H. Schulzrinne, S. Casner, R. Frederick, V. Jacobson. +- [16] IETF RFC 2326 (April 1998): "Real Time Streaming Protocol (RTSP)", H. Schulzrinne, A. Rao, R. Lanphier. +- [17] W3C Document Object Model (DOM) Level 3 Events Specification, Version 1.0, W3C Working Draft 13 April 2006. +- NOTE: Available at: . +- [18] IETF RFC 2046 (November 1996): "Multipurpose Internet Mail Extensions (MIME) Part Two: Media Types", N. Freed, N. Borenstein. +- [19] W3C XML Events, an Events Syntax for XML, W3C Recommendation 14 October 2003. +- NOTE: Available at: . +- [20] W3C Media Access Events +- [21] IETF STD 65, RFC 3551 (July 2003): "RTP Profile for Audio and Video Conferences with Minimal Control", H. Schulzrinne, S. Casner. +- [22] IETF RFC 3388 (December 2002): "Grouping of Media Lines in the Session Description Protocol (SDP)", G. Camarillo, G. Eriksson, J. Holler, H. Schulzrinne. +- [23] IETF RFC 2965 (October 2000): "HTTP State Management Mechanism", D. Kristol, L. Montulli. +- [24] IETF RFC 3926 (October 2004): "FLUTE - File Delivery over Unidirectional Transport", T. Paila, M. Luby, R. Lehtonen, V. Roca, R. Walsh. +- [25] W3C Extensible Markup Language (XML) 1.0, Fourth Edition [Recommendation]. +- NOTE: Available at . +- [26] Open Mobile Alliance "Rich Media Environment (RME)". +- [27] ISO/IEC 14496-4:2004/AMD25 "Information technology - Coding of audio-visual objects - Part 4: Conformance Testing: Amendment 25: LASEr and SAF Conformance". +- [28] ISO/IEC 14496-4:2004/AMD27 "Information technology - Coding of audio-visual objects - Part 4: Conformance Testing: Amendment 27: LASEr and SAF Extensions Conformance". + +# --- 3 Definitions and abbreviations + +## 3.1 Definitions + +For the purposes of the present document, the following terms and definitions apply: + +**DIMS Scene:** an SVG scene, which may include extensions, and may be updated over time + +**DIMS Scene Commands:** a set of one or more commands to modify the state of a DIMS Scene + +**DIMS Unit:** the basic unit of transport, processing, and compression, of DIMS content + +**New Scene:** a complete scene (containing an "svg" element), suitable for starting a session or completely replacing the current scene in a session +(Functions very similarly to an I-frame in video) + +**Normal DIMS Unit:** DIMS Units processed when processing a stream (cf. Redundant DIMS Unit) + +**Primary Stream:** a stream which defines the complete scene tree, i.e. in which all random access points are, or build, a complete DIMS Scene + +**Redundant DIMS Unit:** DIMS Units which supply a redundant 'summary' of the stream, and which can be used for random access, tune-in, or error recovery (cf. Normal DIMS Unit) + +**Scene Update:** a set of differences that make changes to the scene in the current session (Similar to a P-frame in video) + +**Secondary Stream:** a stream which manages only a portion of the scene tree + +## 3.2 Abbreviations + +For the purposes of the present document, the following abbreviations apply: + +| | | +|-------|----------------------------------------------| +| API | Application Program Interface | +| AVP | Audio/Video Profile | +| CTR | CounTeR | +| DIMS | Dynamic and Interactive Multimedia Scenes | +| DOM | Document Object Model | +| FLUTE | File deLivery over Unidirectional Transport | +| HTTP | Hyper Text Transfer Protocol | +| IANA | Internet Assigned Numbers Authority | +| ID | IDentifier | +| LASER | Lightweight Application Scene Representation | +| MIME | Multipurpose Internet Mail Extensions | +| MMS | Multimedia Messaging Service | +| MTU | Maximum Transmission Unit | +| PSS | Packet switched Streaming Service | +| RAP | Random Access Point | +| RTP | Real-Time transport Protocol | +| RTSP | Real Time Streaming Protocol | +| SDP | Session Description Protocol | +| SMIL | Synchronized Multimedia Integration Language | +| SVG | Scalable Vector Graphics | +| TCP | Transmission Control Protocol | +| uDOM | microDOM | +| UDP | User Datagram Protocol | +| UE | User Equipment | +| URL | Uniform Resource Locator | +| URN | Uniform Resource Name | +| W3C | World Wide Web Consortium | +| XHTML | eXtensible HyperText Markup Language | +| XML | eXtensible Markup Language | + +# 4 Overview and architecture + +![Figure 4-1: General architecture of the rich media system. The diagram shows three main components: Rich Media Server, Transport Mechanisms, and Rich Media Client. The server contains 'Rich Media content' and 'Container Format / Transport Packets'. The transport mechanisms include 'Forward Transmission' and 'Remote Interaction Mechanisms'. The client contains a 'Rich Media Player', 'Local Interaction Mechanisms', and a decision diamond 'Is the player's request remote in nature?'. Arrows show data flow from server to transport to client, and feedback from client to transport to server. A 'send request' arrow goes from the player to the decision diamond, and a 'no' arrow goes from the decision diamond to the local interaction mechanisms.](7a0db9703b68b3d06cdaeefc084c0006_img.jpg) + +``` + +graph LR + subgraph RMS [Rich Media Server] + RMC[Rich Media content scenes, scene updates, discrete and continuous media] --> CF[Container Format / Transport Packets] + end + subgraph TM [Transport Mechanisms] + FT[Forward Transmission Unicast, Multicast, Broadcast Download and Streaming Protocols] + RI[Remote Interaction Mechanisms] + end + subgraph RMCli [Rich Media Client] + RMP[Rich Media Player] + LIM[Local Interaction Mechanisms] + D{Is the player's request remote in nature?} + end + RMC --> FT + CF --> RI + FT --> RMP + RI --> D + D -- yes --> RI + D -- no --> LIM + RMP -- send request --> D + +``` + +Figure 4-1: General architecture of the rich media system. The diagram shows three main components: Rich Media Server, Transport Mechanisms, and Rich Media Client. The server contains 'Rich Media content' and 'Container Format / Transport Packets'. The transport mechanisms include 'Forward Transmission' and 'Remote Interaction Mechanisms'. The client contains a 'Rich Media Player', 'Local Interaction Mechanisms', and a decision diamond 'Is the player's request remote in nature?'. Arrows show data flow from server to transport to client, and feedback from client to transport to server. A 'send request' arrow goes from the player to the decision diamond, and a 'no' arrow goes from the decision diamond to the local interaction mechanisms. + +**Figure 4-1: General architecture of the rich media system** + +The rich media system can be perceived as client-server architecture, comprising 3 main components: The rich media server, transport mechanisms and the rich media client. Figure 4-1 illustrates the general architecture. The server takes as input, rich media content comprised of scene description, discrete (e.g. images) and continuous (e.g. audio, video) media. Scene description can be dynamically updated through scene updates. The rich media content can be encapsulated into a container format, containing additional information such as media synchronization, metadata, and hint tracks for packetization. The system then utilizes various transport mechanisms for 1-to-1 and 1-to-many protocols for download, progressive download and streaming scenarios. The content is played on the client, allowing for local and remote interactivity of feedback and data requests. + +# 5 Media-type definition + +## 5.1 Introduction + +The DIMS media type allows spatial and temporal layout of the multimedia scene. This scene can consist of any combination of still pictures, video, audio, and animated graphics. It includes an update mechanism that allows for partial updates of the existing scene, as well as updating the presentation with a completely new scene and streaming tune-in functionality. + +## 5.2 Media type components + +The DIMS media type consists of: + +- Base scene description, which is SVG Tiny 1.2 [1]. +- Scene description extensions. +- Scene commands. +- Event generation and processing. + +## 5.3 Namespace + +The namespace called DIMS here is associated with the URN "". + +## 5.4 Scene description + +### 5.4.1 Base Scene Description + +SVG Tiny 1.2 provides the basic DIMS Scene functionality; layout, inclusion and referencing of objects, synchronization of object timelines and a rendering model. + +The full syntax and semantics of SVG Tiny 1.2 shall be supported for DIMS Scene functionality. The version and baseProfile attributes of the SVG element document the version and profile of SVG on which this scene is based. + +### 5.4.2 Scene Description Extensions + +#### 5.4.2.1 Introduction + +Extensions defined here are designed so that: + +- a) when the same functionality is present in profiles of SVG other than SVG Tiny 1.2, then the extension is compatible with that or a restricted version of that. +- b) A terminal implementing both the present document and SVG (any version) can use a common implementation of the DOM tree, scene graph, rendering model etc. without having variant handling that depends on whether the scene was built using DIMS or SVG. +- c) No extensions are required to be present in all documents; content authored to the SVG Tiny 1.2 specification may be used as the initial scene of a stream designed to the present document. + +The following extensions are defined here. + +#### 5.4.2.2 Rectangular clipping of a graphical object + +The lsr:rectClip mechanism provides pixel aligned clipping defined as a transformable rectangle. + +The lsr:rectClip element shall be supported. The definition of lsr:rectClip is defined in subclause 6.8.28 of [3]. + +#### 5.4.2.3 Full-screen video + +The full-screen video feature consists of the attribute lsr:fullscreen on the SVG video element. + +The lsr:fullscreen attribute shall be supported. The lsr:fullscreen attribute is defined in subclause 6.8.40.2 of [3]. + +See clause 10 for security considerations of fullscreen. + +#### 5.4.2.4 Full-screen SVG + +The fullscreen SVG feature in the DIMS namespace consists of an attribute 'fullscreen' on the element to hint that the scene should be rendered on the full screen. The possible values are "true" and "false" (default). With the attribute set to true the DIMS UE should negotiate the rendering area with its parent UE and get as large part of the screen as possible for the DIMS canvas. + +The dims:fullscreen attribute shall be supported on the svg element. + +See clause 10 for security considerations of fullscreen. + +#### 5.4.2.5 Attributes clipBegin and clipEnd + +Attributes clipBegin and clipEnd defined in subclause 7.6.1 of [5] shall be supported on the following elements: video, audio, animation, and the "updates" element as described in subclause 5.4.2.6. + +#### 5.4.2.6 Update Streams + +The present document defines a new element 'updates' in the DIMS namespace to link secondary streams of updates to a scene. This element has an implicit "simple duration" of 'indefinite'. The synchronization attributes defined in [1] subclause 12.6 can be used with this element. + +The dims:updates element shall be supported. + +NOTE: `lsr:updates` defined in subclause 6.8.53 of [3] is a superset of this element. + +##### *Attribute definitions:* + +All timing attributes defined in [1] subclause 16.2.7 are defined for this element, except the "fill" attribute. + +The attributes `clipBegin` and `clipEnd` defined in subclause 5.4.2.5, and `syncReference` defined in subclause 5.4.2.7, are defined for this element. + +`xlink:href = ""` + +An IRI reference to a source of updates, such as a DIMS stream/file. This attribute specifies the location of the stream of updates. In the absence of this attribute, this element does not have any effect. This attribute is not animatable and not inheritable. In DIMS, support is required only for DIMS streams or files. + +#### 5.4.2.7 Synchronization of Media Streams + +`lsr:syncReference = ""` + +The attribute `lsr:syncReference` from subclause 6.8.8.2 of [3] shall be supported on the elements `video`, `audio`, and `animation` from SVG, and 'updates' from the present document, with the associated synchronization behaviour. This attribute holds a reference to the stream or media element whose clock acts as a clock reference for the stream referred to by this element. This attribute is not animatable and not inheritable. + +#### 5.4.2.8 Screen orientation + +Two events and two extension strings are defined that make it possible for scenes to adapt to the screen layout. The events are defined in subclause 6.1.3. + +Whenever the terminal detects a change of orientation, angle, or screen size, one of these two events is dispatched. A portrait event is dispatched if the screen is taller than it is wide, and a landscape event is dispatched if the screen is wider than it is tall. It is the responsibility of the system below the scene to orient the screen buffer to user; the DIMS Scene author does not do this. + +In addition, the orientation is reported in degrees in `screenAngle`, to the best of the terminal's capability. This is measured as the angle between the positive X-axis of an un-rotated frame of reference, and the orientation of the longer of the positive X or Y axis of the screen, as rotated, as shown in Figure 5-1. Note that the SVG Y axis is downward. + +Specifically, for a screen that is normally portrait and in its normal position, the `screenAngle` is 90 degrees, since the longest axis is vertical, and the Y-axis is downward in SVG. Similarly, for a terminal that is normally landscape and in its normal position, the `screenAngle` is 0 degrees, since the longest axis is horizontal, and is the X-axis. This angle therefore would normally be close to 90 or 270 in portrait events, and close to 0 or 180 in landscape events, and 0 or 90 in terminals that are in their normal orientation, and 180 or 270 in terminals that are inverted. + +![Figure 5-1: Screen Orientation diagram. A rectangle representing the screen is tilted clockwise. A horizontal arrow pointing right is labeled 'reference X axis'. A vertical arrow pointing down from the top-left corner of the screen is labeled 'normal screen top or left'. The angle between the reference X axis and the normal screen top or left arrow is labeled 'a'.](6e5c78a85b8088d4ab0ddcced8a67ca7_img.jpg) + +Figure 5-1: Screen Orientation diagram. A rectangle representing the screen is tilted clockwise. A horizontal arrow pointing right is labeled 'reference X axis'. A vertical arrow pointing down from the top-left corner of the screen is labeled 'normal screen top or left'. The angle between the reference X axis and the normal screen top or left arrow is labeled 'a'. + +Figure 5-1: Screen Orientation + +These events shall map to the ScreenOrientationEvent interface as defined in subclause 6.1.3. + +The screen orientation events shall be supported in DIMS. If the terminal has an orientation sensor, or other physical adaptation that causes the available screen drawing area to change (e.g. a partial cover), events shall be generated whenever the terminal detects a change in any of the parameters to these events. + +The following extension strings shall also be supported, in order to allow the use of the switch element: + +- orientLandscape for typical 'landscape' orientation; +- orientPortrait for typical 'portrait' orientation. + +The namespace of these feature strings is DIMS. + +If the most recent event generated was a portrait event, then `requiredExtensions="orientPortrait"` tests as true; if the most recent event was a landscape event, `requiredExtensions="orientLandscape"` tests as true. At any time, exactly one of these `requiredExtensions` expressions shall test as true. If no event has been generated, the appropriate `requiredExtensions` expression tests as true. + +Information on softkey location and key mapping may be found in section 9 of [26]. + +#### 5.4.2.9 Current-Time Indication + +In a Primary Stream, Redundant Random-Access Point there is a need to establish the current SceneTime of the scene, so that terminals tuning-in, performing random-access, or recovering from a lost high-priority DIMS Unit achieve the same SceneTime as terminals which had processed the entire stream from the most recent non-redundant Random Access Point. This media-time (scene time) is indicated by the `currentSceneTime` attribute on the SVG element, and takes a valid clock value in the document timeline, from the SVG specification [1]. + +The scene state is set exactly as if the SVG document had been loaded and displayed at the non-zero time T in the current-time indicator. + +EXAMPLE: This is the same as if this SVG scene had been used in a SMIL document as the target of an "animation" element with `clipBegin` of T, or if conceptually all absolute times S in the document were replaced with S-T and the document instantiated at time 0. + +*Attribute definition:* + +`currentSceneTime = ""` + +Specifies the current scene time (a valid clock value) in the document timeline, at which the scene is displayed. The scene state is set exactly as if the SVG document had been loaded and displayed at the non-zero time T in the current-time indicator. This attribute is defined in DIMS namespace, and may be present on the root SVG element in redundant random access points. The default value is zero. + +#### 5.4.2.10 Active attribute + +On all elements, the following attribute is defined in the DIMS namespace: + +`active`: this attribute defines whether the element is active. The possible values are "true" (default) and "false". + +Setting the value of this attribute to true or false is equivalent to executing the commands `activate` and `deactivate`. See subclause 5.5.3 for the behaviour of deactivated elements. This attribute is not animatable and not inheritable. + +The `dims:active` attribute shall be supported. + +## 5.5 Scene Commands + +### 5.5.1 Scene Updates + +The scene update mechanism allows reception of updates that change parts of the current scene, without having to replace the entire scene. + +To account for the different update scenarios two update mechanisms are defined: + +- **Primary-stream updates:** Updates are delivered to the client in the same stream as the original scene. +- **Secondary-stream updates:** Updates are delivered to the client in separate streams from the original scene, e.g. in an interactive scenario or initiated from the scene mark-up. + +In a primary-stream case, the updates and/or scene replacements are sent in the same stream as the initial scene. The temporal management of samples in a primary stream is based upon transport level timestamps. A secondary stream is a stream that does not contain the initial scene. A secondary stream is initiated directly from the DIMS mark-up using the 'updates' element. + +The following LASEr commands from subclause 6.7 of [3] in LASEr ML format shall be supported. + +- The LASEr Insert command from subclause 6.7.5 of [3] shall be supported on elements, attributes and values in list attributes with the following relaxed constraints: values *may* be inserted on attributes x and y of the text element. +- The LASEr Delete command from subclause 6.7.4 of [3] shall be supported on elements, attributes and values in list attributes. +- The LASEr Replace command from subclause 6.7.8 of [3] shall be supported on elements, attributes and values in list attributes with the following relaxed constraints: attributes attributeName, id, type, xml:space, preserveAspectRatio and the x and y attributes of the text element *can* be replaced. There are no restrictions on the value of attributeName. The text regarding executionTime does not apply. +- The LASEr Add command from subclause 6.7.2 of [3] shall be supported. + +### 5.5.2 State management commands + +The following state management commands shall be supported: + +- The LASEr Save command from subclause 6.7.10 of [3] shall be supported. +- The LASEr Restore command from subclause 6.7.9 of [3] shall be supported. +- The LASEr Clean command from subclause 6.7.3 of [3] shall be supported. + +These LASEr commands are defined as an interface to persistent storage. Selected scene information is cached on a best effort basis. The security principles behind this caching are those of the state caching mechanism in HTTP, commonly called cookies [23]. + +The saved data is defined by a groupID (known as "name" in [23]) and scoped by the "service" defined by a domain-name and path; for a command to operate on the data, all must match. + +The LASEr command "save" saves the values of a set of attributes, each identified by element ID and attribute name. Each save operation uses a groupID. Any other saved state with the same domain-name, path, and groupID is replaced. + +The LASEr command "restore" restores the attributes (if any) previously saved and scoped by the domain-name and path. The set of data restored is defined below. + +The LASEr command "clean" erases the attributes (if any) previously saved and scoped by the domain-name and path. The set of data erased is defined below. + +The following two attributes are defined in the stream signalling, and define the security restrictions for the above commands: + +- **useFullRequestHost:** this Boolean attribute indicates whether the full domain-name of the request-host is used (true, 1) or the first component of the domain-name is elided (false, 0). For example, if the source material came from "www.example.org", then this differentiates between associating the "service" with "www.example.org" and ".example.org". (Note the definition of local names in [23], and the possibility to associate the "service" with locally loaded files, and that the domain-name may be either ".local" or ".local" in that case.) +- **pathComponents:** this 4-bit unsigned integer attribute indicates how much of the source path is used. If this takes the value 0, then the "service" is not associated with a path, and if it takes the special value 15 (or any value equal to or greater than the number of components in the path) then the entire path is used up to but excluding the final file-name. For example, if the source was "/user/laser-expert/demo/art.mp4" then a value of 3 + +or greater selects "/user/laser-expert/demo" as the path, the value 2 selects "/user/laser-expert" and the value zero sets no path. + +Data is saved as a set of four values, using the URI, pathComponents and useFullRequestHost from the stream containing the save command: + +- the domain-name formed from the URI and useFullRequestHost; +- the path formed from the URI and pathComponents; +- the groupID; +- the set of {element-ID, attribute-name, value} triplets. + +When a restore command is executed all saved sets with the same (equal) groupID, and also where the URI of the stream containing the restore command matches the saved domain-name and path, are restored. This matching is defined in section 3.3.4 of [23]. + +A clean command behaves exactly the same as a save command that saves no state; as is normal for the save command, any other saved state with the same domain-name, path, and groupID is replaced, in this case, with an empty set of saved data. This is functionally equivalent to deleting that saved state, as nothing would be restored. + +NOTE: Be aware that though the data saved and restored is scoped by stream, once it is restored into the tree it is globally visible. + +### 5.5.3 Activate and Deactivate + +The commands activate and deactivate as defined in subclauses 6.7.12 and 6.7.13 of [3] shall be supported, in a manner that is functionally equivalent to that specification. These commands have one attribute: + +- *ref*: the id of the element which is to be activated or de-activated. + +When an element is deactivated, the system then treats the DOM tree as if that element and its descendents were not present in the DOM tree, and invisible to everything except commands and scripts. Commands and scripts can reference it as if it were still in the DOM tree. When activated, the element is then restored to visibility, in the same location in the tree as if it had not been previously deactivated. + +The active attribute is not inherited. However, activated and de-activated events are generated if the effective active state of an element changes as a result of a change to the active state of a parent. This means that a deactivated event may occur for an element, as a result of deactivating a parent, when a test of its dims:active attribute value returns true. + +### 5.5.4 Distributed Random Access Points + +#### 5.5.4.1 Introduction + +A Distributed Random Access Point (DRAP) is a redundant DIMS tune-in point (either primary or secondary) that can, instead of explicitly defining all elements itself, reference elements in coming DIMS units. The commands in these following DIMS units are not executed, elements are simply copied according to references in the DRAP. These references can be used to reduce redundancy (i.e. not defining an element both in a RAP and an update) or to simply spread the size of the RAP over a period of time. + +After this copying operation, the pending action(s) in the DRAP are complete, and are then executed, and normal processing resumes. + +![Figure 5-2: Illustration of the DRAP Concept. The diagram shows a sequence of DIMS units (DU) over time. A DRAP element is shown on the left, containing 'References' and a 'Body'. A bracket labeled 'Units required' spans from DU(x+1) to DU(x+n). A dashed box encloses a DRAP element, its References, and its Body, with arrows pointing to DU(x+1) and DU(x+n). Text explains that the references are used to update the body, and the body is processed at time t=x+n, replacing DU(x+n). Another text note states that DU(x+n+1) is applied regardless of whether the DRAP was used.](7efae06af3af43ffe5d4b956a679cf54_img.jpg) + +The diagram illustrates the DRAP concept across a sequence of DIMS units (DU) over time. On the left, a DRAP element is shown with two internal components: 'References' and 'Body'. A bracket above a series of units labeled 'Units required' spans from $x+1$ to $x+n$ . Below this, a dashed box contains a DRAP element, its References, and its Body. Arrows point from this dashed box to the units $DU(x+1)$ and $DU(x+n)$ . A note indicates that the references of the DRAP are used to update the body. Another note states that the body of the DRAP is then processed at time $t=x+n$ , i.e., replacing $DU(x+n)$ . To the right, a unit $DU(x+n+1)$ is shown, with a note that it is applied in the same way irrespective of if the DRAP was used or not. A time axis arrow points from left to right, labeled $x$ . + +Figure 5-2: Illustration of the DRAP Concept. The diagram shows a sequence of DIMS units (DU) over time. A DRAP element is shown on the left, containing 'References' and a 'Body'. A bracket labeled 'Units required' spans from DU(x+1) to DU(x+n). A dashed box encloses a DRAP element, its References, and its Body, with arrows pointing to DU(x+1) and DU(x+n). Text explains that the references are used to update the body, and the body is processed at time t=x+n, replacing DU(x+n). Another text note states that DU(x+n+1) is applied regardless of whether the DRAP was used. + +Figure 5-2: Illustration of the DRAP Concept + +#### 5.5.4.2 DRAP syntax and semantics + +The rootmost element in a DRAP document shall be a element in the DIMS namespace. A DIMS Unit containing a DRAP shall contain only the DRAP. + +*Attribute definitions:* + +`unitsrequired="units-required"` + +Indicates the number of coming DIMS units required. + +NOTE 1: These DIMS units are *not* executed in the normal way when tuning in using DRAP; instead, they are used as needed as a source of material for the DRAP. + +The DRAP element contains one or more getfromupdate elements, which form the processing instructions, and one or more other elements that form the pending action(s). The processing instructions are applied to the pending action. The indicated number of DIMS units are processed for the DRAP, and the pending action(s) are performed at the time of the DIMS unit at the indicated distance, and normal DIMS unit processing resumes. + +NOTE 2: All the getfromupdates should have been resolved by the indicated distance. + +The getfromupdate element shall reference an element in another DIMS unit and an element in the pending actions. The element referred to in the other DIMS unit shall replace the element in the DRAP pending actions. + +*Attribute definitions:* + +`source="elementid"` + +Specifies an xml id appearing in an upcoming DIMS unit. If the same xml id appears in different DIMS units, it shall not make a difference which one the client chooses. + +`target="elementid"` + +Specifies an xml id appearing in the DRAP pending action(s). + +### 5.5.5 Immediate Script Execution + +The `doScript` command in the DIMS namespace shall be supported. This command supplies a script for immediate execution, including the ability to update the DOM. It has a single attribute, the type of the script. The script is in the body of the element. Processing this command involves executing the script in the context of the DIMS stream in which it occurs. + +*Attributes:* + +`type` - is a string that identifies the scripting language used. It takes a suitable MIME type [18] from the IANA registry, such as "application/ecmascript" (see [13]). + +An example is: + +``` + + var root = document.documentElement; + var myGroup = document.createElementNS( + "http://www.w3.org/2000/svg", "g"); + myGroup.id = "myGroup"; + root.appendChild(myGroup); + var myRect = document.createElementNS( + "http://www.w3.org/2000/svg", "rect"); + myRect.id = "myRect"; + var color = root.createRGBColor( 255, 0, 0); + myRect.setRGBColorTrait("fill", color); + myRect.setFloatTrait("x", 10); + myGroup.appendChild(myRect); + +``` + +### 5.5.6 Seeking in the DIMS Stream + +``` + +``` + +*Attributes:* + +`seekOffset`: A clock value from subclause 16.2.7 of [1]. + +The command `seek` in the DIMS namespace results in a seek, by the amount `seekOffset`, in the DIMS stream timeline. The target stream time is obtained by adding `seekOffset` to the current stream time. As a DIMS stream may contain multiple scenes, this seeking can result in a change of scene. A seek to the local time corresponding to the `seekOffset` shall be applied to the (possibly new) scene. The `seek` command shall be supported. + +**NOTE:** Seeking can be conceptually seen as a function where the global timeline and document timelines are moving forward in a synchronized manner, just as in normal playback, but more quickly and without rendering. A seek backwards in time (negative `seekOffset`) could be done in a similar way, but by starting again from zero and moving forward. + +## 5.6 DIMS Unit Definition + +### 5.6.1 Definition + +A DIMS Unit is built from a header and a body. The DIMS Unit Body is either: + +- a complete SVG document as specified in subclause 5.4, possibly using extensions; or +- a textually concatenated sequence of scene commands as specified in subclause 5.5; + +A DIMS Unit Body may be compressed. + +DIMS Units are framed by the transport layer. Each DIMS Unit has certain characteristics, signalled by the DIMS Unit Header. + +There are DIMS Units used in redundant processing, and DIMS Units used in normal processing. Redundant DIMS Units, and DIMS Units marked as random-access points, are used in random access, tune-in, and error recovery; for a full description of their processing model, see subclause 5.8. + +### 5.6.2 DIMS Unit Header + +DIMS Unit Header is 1 byte long. The length of a DIMS Unit is the length of the DIMS Unit Body plus the length of the DIMS Unit Header. DIMS Unit lengths are carried by the transport layer. + +The DIMS Unit Header has the following layout. + +![](21ad58fee90f2be50708ff541d225507_img.jpg) + +``` + ++-----+ +| 0 | 1 | 2 | 3 | 4 | 5 | 6 | 7 | ++---+---+---+---+---+---+---+---+ +| X | C | P | D | I | M | S | | ++-----+ + +``` + +**Figure 5-3: DIMS Unit header** + +These fields have the following definitions: + +- S: is-Scene:** when 1, indicates that the DIMS Unit contains a Scene Description as documented in subclause 5.4; when 0, indicates that the DIMS Unit contains one or more Scene Commands as documented in subclause 5.5. +- M: is-RAP:** when 1, indicates a Random Access Point; when 0, indicates a non-Random-access point. +- I: is-redundant:** when 0, indicates a main (normal processing) DIMS Unit; when 1, signals a redundant DIMS Unit. +- D: redundant-exit:** shall be 0 on DIMS Units with is-redundant==0; on DIMS Units with is-redundant==1, when 1, indicates that redundant processing is completed by this DIMS Unit, and normal processing should begin, and when 0, indicates that redundant processing should continue. +- P: priority** set to 1 indicates a high-priority unit; when set to 0 indicates a low-priority unit. A unit marked as low-priority indicates that if the unit, or any sequence of such low-priority units, is not processed by the terminal: +1. all succeeding DIMS Units can be decoded and operated on without error (e.g. their DOM updates do not depend on the possibly lost command(s).) + 2. the visual and semantic nature of the scene is satisfactory to the content author. + 3. The loss of those units does not require the terminal to enter into a tune-in or repair state. +- Units fulfilling the above criteria should be marked as low-priority, and shall be marked as high-priority otherwise. +- DIMS Units with is-redundant set to 1 should normally be marked as low-priority, to avoid their loss causing an un-needed entry into tune-in state when redundant and normal data are carried in the same transport. +- C: compression:** indicates the compression applied; +0 indicates no compression (textual format); +1 indicates that the content is compressed using the encoding signalled in stream setup. +- X: reserved:** shall be set to 0 and shall be ignored by the receiver + +**NOTE:** The setting of the priority field is, due to point 2 above, partly at the discretion of the content creator. An example of a simple method of evaluating point 2 is to see if, when the next packet is received, the terminal state is identical to what it would have been if the DIMS Unit(s) had not been lost in the first place. + +## 5.7 Timing model + +DIMS inherits the timing model from [1] in its entirety. This section defines the timing of DIMS units. + +DIMS units are associated with a media timestamp which marks + +- a) the time when the SVG scene is instantiated and the scene time resets to 0, for New Scenes, or + +- b) the time when DIMS units are applied and affect the scene, for other DIMS units. + +The processing order for scene updates is the same as for script and event processing. The present document does not mandate any relative timing or processing order for DIMS units, scripts or events that shall be processed at the same scene time. DIMS Units from the same stream shall be processed in decoding order, i.e. sequence number order in RTP or order inside a sample in the 3GP file format. + +In the main stream the SVG time has periodic resets to 0 on each New Scene (what SVG calls "document time" and is called here "SVG Scene Time"). The SVG scene time, and the media timestamps of the DIMS presentation, are related by a piecewise linear relationship, with discontinuities only at the New Scenes. Secondary streams are synchronized to the main stream using the normal mechanisms for media streams. + +DIMS Units containing commands, with the same timestamp in the same stream, are applied "instantaneously" in scene time, that is, the scene time does not change while they are being applied. + +DIMS units containing commands are applied atomically and have exclusive access to the scene tree during processing; only events and scripts directly resulting from the processing of the DIMS unit, are run. + +## 5.8 Processing Model + +A Scene Description is processed as a complete replacement for the current scene tree. That is, the entire DOM is discarded and replaced with the result of parsing the SVG element. All other DIMS Units retain (and possibly modify) the current scene tree. + +All high-priority data units not marked as redundant shall be processed during normal decoding. All low-priority data units not marked as redundant should be processed during normal decoding. Data units marked as redundant should be ignored during normal processing. All data units marked as RAP are suitable tune-in points. When tuning-in to a stream, decoding shall begin by, at the latest, the first unit marked as RAP irrespective of the value of its is-redundant flag. + +If a normal (non-redundant) random access point is identified during redundant processing or DRAP processing, the normal random access point should take precedence. + +Commands that cannot be executed (e.g. they refer to a DOM node which does not exist) shall be ignored when in tune-in or redundant-processing. This condition should not arise in normal processing, and their handling in this state is not defined by the present document. + +The following state diagram, and processing pseudo-code for each state, illustrate the states and the use of the various flags in the DIMS unit header, and comply with the mandatory and recommended processing requirements. The state diagram and pseudo-code, or better, should be implemented in DIMS clients. + +In the state diagram and pseudo-code, the terminal may be processing a stream under one of three conditions: + +- a) normal processing, 'normal'; +- b) after tuning in, performing random access, or when loss is detected, 'tune-in'; +- c) while processing redundant DIMS units, 'process-redundant'. + +Tune-in state is entered under any of the following circumstances: + +- a) after opening a stream; +- b) after performing random access; +- c) after loss of a high-priority DIMS Unit in normal processing; +- d) after loss of any DIMS Unit in redundant processing. + +![Figure 5-4: DIMS Client State Diagram. The diagram shows three states: 'tune-in', 'redundant processing', and 'normal'. Transitions are as follows: 'tune-in' to 'redundant processing' on 'redundant RAP'; 'tune-in' to 'normal' on 'normal RAP'; 'redundant processing' to 'tune-in' on 'any loss'; 'redundant processing' to 'normal' on 'normal RAP or redundant-exit'; 'normal' to 'tune-in' on 'high-priority loss'. An external entry point 'tune-in or random access' leads to the 'tune-in' state.](8fa679f79a1bb1f527cba9f29e784e89_img.jpg) + +``` + +stateDiagram-v2 + [*] --> tune-in : tune-in or random access + tune-in --> redundant_processing : redundant RAP + tune-in --> normal : normal RAP + redundant_processing --> tune-in : any loss + redundant_processing --> normal : normal RAP or redundant-exit + normal --> tune-in : high-priority loss + +``` + +Figure 5-4: DIMS Client State Diagram. The diagram shows three states: 'tune-in', 'redundant processing', and 'normal'. Transitions are as follows: 'tune-in' to 'redundant processing' on 'redundant RAP'; 'tune-in' to 'normal' on 'normal RAP'; 'redundant processing' to 'tune-in' on 'any loss'; 'redundant processing' to 'normal' on 'normal RAP or redundant-exit'; 'normal' to 'tune-in' on 'high-priority loss'. An external entry point 'tune-in or random access' leads to the 'tune-in' state. + +**Figure 5-4: DIMS Client State Diagram** + +The following behaviour is performed for each DIMS Unit in each state, and then a state transition is performed as indicated by the state diagram. + +Normal state: + +``` + +if DIMS-Unit.is-redundant + then Discard(DIMS-Unit) + else Process(DIMS-Unit); + +``` + +Tune-in state: + +``` + +if DIMS-Unit.is-RAP + then Process(DIMS-Unit) + else Discard(DIMS-Unit); + +``` + +Redundant-processing: + +``` + +if ( DIMS-unit.is-redundant) || +((not DIMS-Unit.is-redundant) && DIMS-unit.is-RAP) + then Process(DIMS-unit) + else Discard(DIMS-Unit); + +``` + +Where this pseudo-code indicates that a DIMS Unit is processed, then if a Distributed Random Access Point (DRAP) is in process, elements required by the DRAP are extracted from this unit. If a DRAP is not in process, the DIMS Unit is processed as normal. If one of the DIMS units identified by units-required is a normal (non-redundant) random access point, DRAP processing should be abandoned, and that normal RAP processed in the usual way. + +## 5.9 Random Access, Tune-in and Error Recovery + +### 5.9.1 Introduction + +Random access points in streams are either normal random access points or redundant random access points. Normal random access points are processed by terminals in all states. Redundant random access points should only be processed by terminals when performing random access, tune-in, or error recovery. + +Random access points are indicated in the DIMS Unit header using the is-RAP flag. Redundant random access points have the is-Redundant flag set to 1; normal Random Access points have this flag set to 0. + +### 5.9.2 Random Access Points in Primary Streams + +A Random Access Point (normal or redundant RAP) in a primary stream shall either contain an entire scene (i.e. be a Scene Description) or the mechanism to build an entire scene (such as DRAP). When used, this scene becomes the current scene and replaces all previous data. There may be further DIMS Units with the same timestamp that modify the scene tree. + +A redundant Random Access Point in a primary stream shall have the currentSceneTime attribute on the SVG element. Any following commands in subsequent DIMS Units with the same timestamp are processed at this time. + +### 5.9.3 Random Access Points in Secondary Streams + +A Random Access Point (normal or redundant RAP) in a secondary stream shall either contain an entire update (i.e. a series of commands) or the mechanism to build an entire update (such as DRAP). The command(s) provided set the scene (specifically, the portion of the scene managed by the secondary stream) into an appropriate state, whether the random access point is used for initial tune-in, or for error recovery. + +**NOTE:** The secondary stream needs to be encoded in such a way that it does not matter which packets were lost or this is an initial tune-in or random access - the appropriate state is set. This would include removing any elements or attributes which should have been removed, etc. A simple way of encoding such a stream would be to only let updates in a secondary stream make modifications to a few nodes. Then this operation could be as simple as removing these few nodes and reinserting them, removing all potential errors. + +### 5.9.4 Error Recovery + +There are several error resilience mechanisms available in DIMS. Among these are: + +- **Priority:** By separating essential and non-essential units one can determine if a loss need repair or not. This is described in subclause 7.3.1. +- **Periodic Random Access Points (RAPs):** Random Access Points can be placed periodically in a stream. In the case of error one can tune-in to the channel again. +- **Separation of static and dynamic data.** This can even increase the efficiency of Distributed Random Access Points. + +A combination of these methods can be used. + +# --- 6 Interaction and Scripting + +## 6.1 Local interaction + +### 6.1.1 DOM Level 3 events + +The supported local events and their management in DIMS are built upon the events model described in [1]. + +They include DOM Events (focus, activate, etc.), SVG Events (connection, load, etc.) and general XML events [19] (user events, timing, key, and pointer events). + +### 6.1.2 Media Access Events + +The media access events defined in [20] shall be supported. + +### 6.1.3 Screen Orientation Events + +The following events shall be supported. + +| Event Type | Namespace | Description | Interface | Bubble | Canc | +|------------------------------|-----------|-----------------------------------------------------------------------|------------------------|--------|------| +| "screenOrientationPortrait" | DIMS | The screen orientation has changed to typical 'landscape' orientation | ScreenOrientationEvent | No | No | +| "screenOrientationLandscape" | DIMS | The screen orientation has changed to typical 'portrait' orientation | ScreenOrientationEvent | No | No | + +``` +interface ScreenOrientationEvent : Event +{ + readonly attribute unsigned long screenWidth; + readonly attribute unsigned long screenHeight; + readonly attribute unsigned long screenAngle; +} +``` + +**screenWidth** - contains the new screen display or viewport width. + +**screenHeight** -contains the new screen display or viewport height. + +**screenAngle** - documents the angle between the primary axis of the screen, and an unrotated horizontal axis (see 5.4.2.8) with a value between 0 and 359 inclusive. + +NOTE: A superset of the ScreenOrientationEvent interface is specified in [26]. + +### 6.1.4 Other Events + +The events pauseevent and resumeevent from subclause 6.5.2 of [3] shall be supported. + +The following events shall be supported. + +| Event Type | Namespace | Description | Interface | Bubble | Canc | +|--------------------|-----------|--------------------------------------------------------------------|---------------------------|--------|------| +| "activatedEvent" | DIMS | Occurs when an element changes state from deactivated to activated | Event
(Annex A of [1]) | No | No | +| "deactivatedEvent" | DIMS | Occurs when an element changes state from activated to deactivated | Event
(Annex A of [1]) | No | No | + +## 6.2 Remote interaction + +Client-server communication is possible in the DIMS system using three different mechanisms: + +- The client can open a suitable URL. The set of valid URL forms is not specified in DIMS, and may include, for example, protocols such as HTTP [12], RTSP [16] or MailTo. +- By establishing a socket connection between the client and the server using the Connection API in the uDOM [17]. +- By using the HTTP specific SVG uDOM methods getURL or postURL [17]. + +## 6.3 Scripting + +SVG Tiny 1.2 contains a uDOM interface that provides linkage to a script engine and adds the possibility to modify the DOM representation of the scene from scripts. + +ECMAScript mobile profile (MP) [2] can be used in conjunction with the script and handler elements and SVG μDOM API (Appendix A of [1]) in order to provide more powerful DOM manipulation, and interaction. + +UEs supporting the DIMS media type shall support ECMAScript mobile profile (MP) [2] with the following extensions to uDOM API. + +Table 1 adds to the table in subclause A.8.12 of [1]. It contains trait access rules for DIMS extensions. + +**Table 1: Trait access rules for DIMS extensions** + +| Attribute | Trait Getter | Trait Setter | Default Values | Description | +|-----------------|--------------------------|--------------------------|----------------|--------------------------------------------------------------------| +| lsr:fullscreen | getTraitNS[true false] | setTraitNS[true false] | false | Available on