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1
+
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+
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+
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+
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+
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+
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+ # --- Contents
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+
9
+ | | |
10
+ |-------------------------------------------------------------------------------------------|----|
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+ | Foreword ..... | 4 |
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+ | 1 Scope..... | 5 |
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+ | 2 References..... | 5 |
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+ | 3 Definitions and abbreviations ..... | 5 |
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+ | 3.1 Definitions..... | 5 |
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+ | 3.2 Abbreviations ..... | 5 |
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+ | 4 General..... | 6 |
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+ | 5 Test sequence format ..... | 6 |
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+ | 5.1 File format..... | 6 |
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+ | 5.2 Codec homing ..... | 6 |
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+ | 6 Speech codec test sequences..... | 7 |
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+ | 6.1 Codec configuration..... | 7 |
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+ | 6.2 Speech codec test sequences ..... | 7 |
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+ | 6.2.1 Speech encoder test sequences ..... | 7 |
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+ | 6.2.2 Speech decoder test sequences ..... | 8 |
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+ | 6.2.3 Codec homing sequence..... | 8 |
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+ | 7 Test sequences for source controlled rate operation ..... | 9 |
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+ | 7.1 Codec configuration..... | 9 |
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+ | 7.2 Test Sequences..... | 9 |
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+ | 7.2.1 Test sequences for background noise estimation ..... | 9 |
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+ | 7.2.2 Test sequences for pitch, tone and complex signal detection..... | 10 |
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+ | 7.2.3 Real speech and tones..... | 10 |
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+ | 7.2.4 Test sequence for signal-to-noise ratio estimation ..... | 10 |
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+ | 8 Sequences for finding the 20 ms framing of the adaptive multi-rate speech encoder ..... | 10 |
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+ | 8.1 Bit synchronisation..... | 10 |
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+ | 8.2 Frame synchronisation ..... | 11 |
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+ | 8.3 Formats and sizes of the synchronisation sequences ..... | 11 |
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+ | 9 Trau Testing with 8 Bit A- and $\mu$ -law PCM Test Sequences ..... | 12 |
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+ | Annex A (informative): Change history ..... | 13 |
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+
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+ # --- Foreword
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+
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+ This Technical Specification (TS) has been produced by the 3<sup>rd</sup> Generation Partnership Project (3GPP).
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+
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+ 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:
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+
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+ Version x.y.z
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+
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+ where:
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+
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+ - x the first digit:
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+ - 1 presented to TSG for information;
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+ - 2 presented to TSG for approval;
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+ - 3 or greater indicates TSG approved document under change control.
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+ - y the second digit is incremented for all changes of substance, i.e. technical enhancements, corrections, updates, etc.
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+ - z the third digit is incremented when editorial only changes have been incorporated in the document.
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+
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+ # --- 1 Scope
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+
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+ The present document specifies the digital test sequences for the adaptive multi-rate (AMR) speech codec. These sequences test for a bit exact implementation of the adaptive multi-rate speech transcoder (TS 26.090 [2]), voice activity detection (TS 26.094 [5]), comfort noise (TS 26.092 [3]), and source controlled rate operation (TS 26.093 [4]).
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+
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+ # --- 2 References
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+
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+ The following documents contain provisions which, through reference in this text, constitute provisions of the present document.
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+
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+ - References are either specific (identified by date of publication, edition number, version number, etc.) or non-specific.
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+ - For a specific reference, subsequent revisions do not apply.
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+ - 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*.
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+
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+ - [1] 3GPP TS 26.071: "AMR Speech Codec; General Description".
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+ - [2] 3GPP TS 26.090: "AMR Speech Codec; Speech Transcoding Functions".
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+ - [3] 3GPP TS 26.092: "AMR Speech Codec; Comfort Noise Aspects".
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+ - [4] 3GPP TS 26.093: "AMR Speech Codec; Source Controlled Rate Operation".
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+ - [5] 3GPP TS 26.094: "AMR Speech Codec; Voice Activity Detector".
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+ - [6] 3GPP TS 26.091: "AMR Speech Codec; Error Concealment of Lost Frames".
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+ - [7] 3GPP TS 26.073: "AMR Speech Codec; ANSI C-code".
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+ - [8] 3GPP TS 46.054: "Test sequences for the GSM Enhanced Full Rate (EFR) speech codec".
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+
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+ # --- 3 Definitions and abbreviations
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+
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+ ## 3.1 Definitions
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+
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+ For the purposes of the present document, the terms and definitions given in TS 26.090 [2], TS 26.091 [6], TS 26.092 [3], TS 26.093 [4] and TS 26.094 [5] apply.
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+
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+ ## 3.2 Abbreviations
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+
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+ For the purposes of the present document, the following abbreviations apply:
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+
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+ [T.B.A]
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+
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+ # --- 4 General
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+
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+ Digital test sequences are necessary to test for a bit exact implementation of the adaptive multi-rate speech transcoder (TS 26.090 [2]), voice activity detection (TS 26.094 [5]), comfort noise generation (TS 26.092 [3]), and source controlled rate operation (TS 26.093 [4]).
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+
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+ The test sequences may also be used to verify installations of the ANSI C code in TS 26.073 [7].
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+
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+ Clause 5 describes the format of the files which contain the digital test sequences. Clause 6 describes the test sequences for the speech transcoder. Clause 7 describes the test sequences for the VAD, comfort noise and source controlled rate operation.
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+
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+ Clause 8 describes the method by which synchronisation is obtained between the test sequences and the speech codec under test.
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+
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+ *[Clause 9 describes the alternative acceptance testing of the speech encoder and decoder in the TRAU by means of 8 bit A- or $\mu$ -law compressed test sequences on the A-Interface.]*
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+
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+ # --- 5 Test sequence format
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+
105
+ This clause provides information on the format of the digital test sequences for the GSM adaptive multi-rate speech transcoder (TS 26.090 [2]), voice activity detection (TS 26.094 [5]), comfort noise generation (TS 26.092 [3]), and source controlled rate operation (TS 26.093 [4]).
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+
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+ ## 5.1 File format
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+
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+ The test sequence files in PC (little-endian) byte order are provided in archive files (ZIP format) which accompany the present document.
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+
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+ Following decompression, three types of file are provided:
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+
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+ - | | |
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+ |-------------------------------------------------------------------------------------|---------|
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+ | - Files for input to the speech encoder: | *.INP |
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+ | - Files for comparison with the encoder output and for input to the speech decoder: | *.COD |
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+ | - Files for comparison with the decoder output: | *.OUT |
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+ | - One mode control file for the mode switching test | T21.MOD |
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+
120
+ All file formats are described in TS 26.073 [7].
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+
122
+ ## 5.2 Codec homing
123
+
124
+ Each \*.INP file includes two homing frames (see TS 26.073 [7]) at the start of the test sequence. The function of these frames is to reset the speech encoder state variables to their initial value. In the case of a correct installation of the ANSI-C simulation (TS 26.073 [7]), all speech encoder output frames shall be identical to the corresponding frame in the \*.COD file. In the case of a correct hardware implementation undergoing testing, the first speech encoder output frame is undefined and need not be identical to the first frame in the \*.COD file, but all remaining speech encoder output frames shall be identical to the corresponding frames in the \*.COD file.
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+
126
+ The function of the two homing frames in the \*.COD files is to reset the speech decoder state variables to their initial value. In the case of a correct installation of the ANSI-C simulation (TS 26.073 [7]), all speech decoder output frames shall be identical to the corresponding frame in the \*.OUT file. In the case of a correct hardware implementation undergoing testing, the first speech decoder output frame is undefined and need not be identical to first frame in the \*.OUT file, but all remaining speech decoder output frames shall be identical to the corresponding frames in the \*.OUT file.
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+
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+ # --- 6 Speech codec test sequences
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+
130
+ This clause describes the test sequences designed to exercise the adaptive multi-rate speech transcoder (TS 26.090 [2]).
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+
132
+ ## 6.1 Codec configuration
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+
134
+ The speech encoder shall be configured not to operate in the source controlled rate mode.
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+
136
+ ## 6.2 Speech codec test sequences
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+
138
+ ### 6.2.1 Speech encoder test sequences
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+
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+ Twenty-two encoder input sequences are provided. Note that for the input sequences T00.INP to T03.INP, the amplitude figures are given in 13-bit precision. The active speech levels are given in dBov.
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+
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+ - T00.INP - Synthetic harmonic signal. The pitch delay varies slowly from 18 to 143.5 samples. The minimum and maximum amplitudes are -997 and +971.
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+ - T01.INP - Synthetic harmonic signal. The pitch delay varies slowly from 144 down to 18.5 samples. Amplitudes at saturation point -4096 and +4095.
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+ - T02.INP - Sinusoidal sweep varying from 150 Hz to 3400 Hz. Amplitudes $\pm$ 1250.
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+ - T03.INP - Sinusoidal sweep varying from 150 Hz to 3400 Hz. Amplitudes $\pm$ 4000.
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+ - T04.INP - Female speech, active speech level: -19.4 dBov, flat frequency response.
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+ - T05.INP - Male speech, active speech level: -18.7 dBov, flat frequency response.
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+ - T06.INP - Female speech, ambient noise, active speech level: -35.0 dBov, flat frequency response.
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+ - T07.INP - Female speech, ambient noise, active speech level: -25.0 dBov, flat frequency response.
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+ - T08.INP - Female speech, ambient noise, active speech level: -15.6 dBov, flat frequency response.
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+ - T09.INP - Female speech, car noise, active speech level: -35.5 dBov, flat frequency response.
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+ - T10.INP - Female speech, car noise, active speech level: -26.1 dBov, flat frequency response.
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+ - T11.INP - Female speech, car noise, active speech level: -15.8 dBov, flat frequency response.
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+ - T12.INP - Male speech, ambient noise, active speech level: -34.9 dBov, flat frequency response.
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+ - T13.INP - Male speech, ambient noise, active speech level: -24.8 dBov, flat frequency response.
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+ - T14.INP - Male speech, ambient noise, active speech level: -15.0 dBov, flat frequency response.
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+ - T15.INP - Male speech, babble noise, active speech level: -34.1 dBov, flat frequency response.
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+ - T16.INP - Male speech, babble noise, active speech level: -24.3 dBov, flat frequency response.
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+ - T17.INP - Male speech, babble noise, active speech level: -14.4 dBov, flat frequency response.
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+ - T18.INP - Female speech, ambient noise, active speech level: -26.0 dBov, modified IRS frequency response, with many zero frames.
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+ - T19.INP - Male speech, ambient noise, active speech level: -36.0 dBov, modified IRS frequency response, with many zero frames.
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+ - T20.INP - Sequence for exercising the LPC vector quantisation codebooks and ROM tables of the codec.
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+ - T21.INP - Speech sequence for mode switching test.
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+
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+ The output using these input sequences will be different depending on the tested adaptive multi-rate mode. In the notation used below <mode> should be changed to the number of the tested mode, i.e. one of 122, 102, 795, 74, 67, 59, 515, or 475.
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+
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+ The T00.INP and T01.INP sequences were designed to test the pitch lag of the GSM adaptive multi-rate speech encoder. In a correct implementation, the resulting speech encoder output parameters shall be identical to those specified in the T00\_<mode>.COD and T01\_<mode>.COD sequences, respectively.
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+
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+ The T02.INP and T03.INP sequences are particularly suited for testing the LPC analysis, as well as for finding saturation problems. In a correct implementation, the resulting speech encoder output parameters shall be identical to those specified in the T02\_<mode>.COD and T03\_<mode>.COD sequences, respectively.
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+
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+ The T04.INP and T05.INP sequences contain a lot of low-frequency components. In a correct implementation, the resulting speech encoder output parameters shall be identical to those specified in the T04\_<mode>.COD and T05\_<mode>.COD sequences, respectively.
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+
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+ The T18.INP and T19.INP sequences contain some "all zeros" frames (silence) in between segments of speech. In a correct implementation, the resulting speech encoder output parameters shall be identical to those specified in the T18\_<mode>.COD and T19\_<mode>.COD sequences, respectively.
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+
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+ The T20.INP sequence was designed to exercise the LPC code indices and the ROM table indices of the codec.
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+
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+ The sequences T06.INP to T17.INP were selected on the basis of bringing various input characteristics (background noise) and levels to the test sequence set. In a correct implementation, the resulting speech encoder output parameters shall be identical to those specified in the T06\_<mode>.COD to T17\_<mode>.COD sequences, respectively.
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+
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+ The T21.INP sequence was designed to test mode switching in the encoder. For testing mode switching this sequence is used together with the mode control file T21.MOD. See TS 26.073 [7] for the format of the mode control file. In a correct implementation, the resulting speech encoder output parameters shall be identical to those specified in the sequence T21.COD. Note that T21.COD contains parameter frames in different codec modes.
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+
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+ ### 6.2.2 Speech decoder test sequences
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+
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+ Twenty-one times eight speech decoder input sequences TXX\_<mode>.COD (XX = 00..20, <mode> = {122, 102, 795, 74, 67, 59, 515, or 475}) are provided for the static mode tests. These are the output of the corresponding TXX.INP sequences, one set per mode. In a correct implementation, the resulting speech decoder output shall be identical to the corresponding TXX\_<mode>.OUT sequences.
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+
185
+ The switching test decoder input T21.COD shall result in decoder output identical to the T21.OUT sequence. For the decoder switching test no special mode control file is needed since the mode information is included in the .COD file according to the file format (see TS 26.073 [7]).
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+
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+ ### 6.2.3 Codec homing sequence
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+
189
+ In addition to the test sequences described above, the homing sequences are provided to assist in codec testing. T22.INP contains one encoder-homing-frame. The sequences T22\_<mode>.COD (<mode> = {122, 102, 795, 74, 67, 59, 515, or 475}) contain one decoder-homing-frame each for the corresponding mode. The use of these sequences is described in TS 26.071 [1].
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+
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+ All files are contained in the archive T.TGZ which accompanies the present document.
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+
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+ # --- 7 Test sequences for source controlled rate operation
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+
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+ This clause describes the test sequences designed to exercise the VAD algorithm options 1 and 2 (TS 26.094 [5]), comfort noise (TS 26.092 [3]), and source controlled rate operation (TS 26.093 [4]).
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+
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+ Test sequences DTX\*.\* are to be used with VAD option 1. DTX1.\* , DTX2.\* , and DTX4.\* shall be run only with speech codec mode MR122. Test sequence DTX3.\* shall be run for all the speech codec modes (MR122, MR102, MR795, MR67, MR59, MR515 and MR475).
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+
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+ Test sequences DT2.\* are to be used with VAD option 2. DT21.\* , DT23.\* , and DT24.\* shall be run only with speech codec mode MR122. Test sequence DT22.\* shall be run for all the speech codec modes (MR122, MR102, MR795, MR67, MR59, MR515 and MR475).
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+
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+ ## 7.1 Codec configuration
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+
203
+ The VAD, comfort noise and source controlled rate operation shall be tested in conjunction with the speech coder (TS 26.090 [2]). The speech encoder shall be configured to operate in the source controlled rate mode, with either VAD option 1 or VAD option 2.
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+
205
+ ## 7.2 Test Sequences
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+
207
+ Each DTX test sequence consists of three files:
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+
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+ - Files for input to the speech encoder: \*.INP
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+ - Files for comparison with the encoder output and input to the speech decoder: \*.COD
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+ - Files for comparison with the decoder output: \*.OUT
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+
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+ The \*.COD and \*.OUT file names has the format DTxA\_<mode>.\*, where "x" is the VAD option ( X for option 1 and 2 for option 2), "A" is the test case number (1, 2, 3 or 4) and <mode> is the speech codec mode.
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+
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+ In a correct implementation, the speech encoder parameters generated by the \*.INP file shall be identical to those specified in the \*.COD file; and the speech decoder output generated by the \*.COD file shall be identical to that specified in the \*.OUT file.
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+
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+ | Sequence name | No. of frames | Size (bytes) | | |
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+ |---------------|---------------|--------------|---------|---------|
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+ | | | *.INP | *.COD | *.OUT |
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+ | DTX1 | 710 | 227 200 | 355 000 | 227 200 |
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+ | DTX2 | 898 | 287 360 | 449 000 | 287 360 |
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+ | DTX3 | 1620 | 518 400 | 810 000 | 518 400 |
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+ | DTX4 | 1188 | 380 160 | 594 000 | 380 160 |
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+ | DT21 | 938 | 300 160 | 469 000 | 300 160 |
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+ | DT22 | 616 | 197 280 | 308 000 | 197 120 |
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+ | DT23 | 938 | 300 320 | 469 000 | 300 160 |
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+ | DT24 | 1188 | 380 160 | 594 000 | 380 160 |
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+
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+ ### 7.2.1 Test sequences for background noise estimation
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+
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+ Background noise estimation algorithm is tested by the following test sequences:
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+
233
+ DTX1.\*
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+
235
+ DTX2.\*
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+
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+ DT21.\*
238
+
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+ DT22.\*
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+
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+ (The sequence DTX1.INP in the same as in [8] 3GPP TS 46.054 sequence DTX01.INP)
242
+
243
+ ### 7.2.2 Test sequences for pitch, tone and complex signal detection
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+
245
+ Pitch, tone and complex signal detection algorithms are tested by the following test sequence:
246
+
247
+ DTX3.\*
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+
249
+ ### 7.2.3 Real speech and tones
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+
251
+ This test sequence consists of very clean speech, barely detectable speech and a swept frequency tone (The sequences DTX4.INP and DT24.INP are the same as in [8] 3GPP TS 46.054 sequence DTX07.INP):
252
+
253
+ DTX4.\*
254
+
255
+ DT24.\*
256
+
257
+ ### 7.2.4 Test sequence for signal-to-noise ratio estimation
258
+
259
+ The full range of SNR estimates are tested by the following test sequence:
260
+
261
+ DT23.\*
262
+
263
+ # 8 Sequences for finding the 20 ms framing of the adaptive multi-rate speech encoder
264
+
265
+ *[This clause needs further adaptation of the text and terminology from GSM to 3G context.]*
266
+
267
+ When testing the decoder, alignment of the test sequences used to the decoder framing is achieved by the air interface (testing of MS) or can be reached easily on the Abis-interface (testing on network side).
268
+
269
+ When testing the encoder, usually there is no information available about where the encoder starts its 20 ms segments of speech input to the encoder.
270
+
271
+ In the following, a procedure is described to find the 20 ms framing of the encoder using special synchronisation sequences. This procedure can be used for MS as well as for network side.
272
+
273
+ Synchronisation can be achieved in two steps. First, bit synchronisation has to be found. In a second step, frame synchronisation can be determined. This procedure takes advantage of the codec homing feature of the adaptive multi-rate codec, which puts the codec in a defined home state after the reception of the first homing frame. On the reception of further homing frames, the output of the codec is predefined and can be triggered to.
274
+
275
+ ## 8.1 Bit synchronisation
276
+
277
+ The input to the speech encoder is a series of 13 bit long words (104 kbits/s, 13 bit linear PCM). When starting to test the speech encoder, no knowledge is available on bit synchronisation, i.e., where the encoder expects its least significant bits, and where it expects the most significant bits.
278
+
279
+ The encoder homing frame consists of 160 samples, all set to zero with the exception of the least significant bit, which is set to one (0 0000 0000 0001 binary, or 0x0008 hex if written into 16 bit words left justified). If two such encoder homing frames are input to the encoder consecutively, the corresponding decoder homing frame of the used codec mode is expected at the output as a reaction of the second encoder homing frame.
280
+
281
+ Since there are only 13 possibilities for bit synchronisation, after a maximum of 13 trials bit synchronisation can be reached for each codec mode. In each trial three consecutive encoder homing frames are input to the encoder. If the corresponding decoder homing frame is not detected at the output, the relative bit position of the three input frames is shifted by one and another trial is performed. As soon as the decoder homing frame of the used codec mode is detected at the output, bit synchronisation is found, and the first step can be terminated.
282
+
283
+ The reason why three consecutive encoder homing frames are needed is that frame synchronisation is not known at this stage. To be sure that the encoder reads two complete homing frames, three frames have to be input. Wherever the encoder has its 20 ms segmentation, it will always read at least two complete encoder homing frames.
284
+
285
+ An example of the 13 different frame triplets is given in sequence BITSYNC.INP.
286
+
287
+ ## 8.2 Frame synchronisation
288
+
289
+ Once bit synchronisation is found, frame synchronisation can be found by inputting two identical frames consecutively to the encoder. There exist 160 different output sequences depending on the 160 different positions that the beginning of this sequence of frames can possibly have with respect to the encoder framing.
290
+
291
+ Before inputting this special synchronisation sequence to the encoder, again the encoder has to be reset by one encoder homing frame. A second encoder homing frame is needed to provoke a decoder homing frame at the output that can be triggered to. And since the framing of the encoder is not known at that stage, three encoder homing frames have to precede the special synchronisation sequence to ensure that the encoder reads at least two homing frames, and at least one decoder homing frame is produced at the output, serving as a trigger for recording.
292
+
293
+ After the last decoder homing frame of the used codec mode it is required to detect two consecutive output frames that are different from the preceding decoder homing frame. To achieve this in the 12.2 kbit/s mode (no lookahead in the linear prediction analysis [4]), the last 40 samples of the third encoder homing frame shall be different from 0x0008 hex. Only the first 120 samples of this frame were set to 0x0008 hex in this mode.
294
+
295
+ The special synchronisation sequence preceded by three encoder homing frames are given in SEQSYNC.INP. For the 12.2 kbit/s mode this sequence is different in the third frame and is given in SEQSYNC\_122.INP.
296
+
297
+ Generally, the output sequences will be different depending on the tested adaptive multi-rate mode. In the notation below <mode> should be changed to the number of the tested mode, i.e. one of 122, 102, 795, 74, 67, 59, 515 or 475.
298
+
299
+ In all 160 output sequences only the second frame after the last decoder homing frame is given in SYNC000\_<mode>.COD through SYNC159\_<mode>.COD. These output frames were calculated by shifting the sequence SEQSYNC.INP respectively SEQSYNC\_122.INP through the positions 0 to 159, where the samples at the beginning were set to zero. For each codec mode it was finally verified that the last frame in each of the 160 output sequences is different to all other last frames.
300
+
301
+ The three digit number in the filenames above indicates the number of samples by which the input was retarded with respect to the encoder framing. By a corresponding shift in the opposite direction, alignment with the encoder framing for the used codec mode can be reached.
302
+
303
+ ## 8.3 Formats and sizes of the synchronisation sequences
304
+
305
+ ### BITSYNC.INP:
306
+
307
+ This sequence consists of 13 frame triplets. It has the format of the speech encoder input test sequences (13 bit left justified with the three least significant bits set to zero).
308
+
309
+ The size of it is therefore:
310
+
311
+ $$\text{SIZE (BITSYNC.INP)} = 13 * 3 * 160 * 2 \text{ bytes} = 12480 \text{ bytes}$$
312
+
313
+ ### SEQSYNC.INP/SEQSYNC\_122.INP:
314
+
315
+ This sequence consists of a 3 frame header (see clause 8.2 for details) and the special synchronisation sequence, consisting of two identical frames. It has the format of the speech encoder input test sequences (13 bit left justified with the three least significant bits set to zero).
316
+
317
+ The size of it is therefore:
318
+
319
+ $$\text{SIZE (SEQSYNC.INP/SEQSYNC_122.INP)} = 5 * 160 * 2 \text{ bytes} = 1600 \text{ bytes}$$
320
+
321
+ ### SYNCXXX\_<mode>.COD:
322
+
323
+ These sequences consist of 1 encoder output frame each. They have the format of the speech encoder output test sequences (16 bit words right justified). In these frames the values of the FRAME\_TYPE and MODE\_INFO fields are set to the transmit frame type and to the corresponding encoding mode information [3].
324
+
325
+ The size of them is therefore:
326
+
327
+ $$\text{SIZE (SYNCXXX_<mode>.COD)} = (244 + 6) * 2 \text{ bytes} = 500 \text{ bytes}$$
328
+
329
+ All files are contained in the archive S.TGZ which accompanies the present document.
330
+
331
+ # 9 Trau Testing with 8 Bit A- and $\mu$ -law PCM Test Sequences
332
+
333
+ *[This clause needs further adaptation of the text and terminology from GSM to 3G context.]*
334
+
335
+ In the previous clauses, tests for the transcoder in the TRAU are described, using 13 bit linear test sequences. However, these 13 bit test sequences require a special interface in the TRAU and do not allow testing in the field. In most cases the TRAU has to be set in special mode before testing.
336
+
337
+ The 'Y' in the file names below stands for A (A-law) and U ( $\mu$ -law), respectively.
338
+
339
+ As an alternative, the speech codec tests in the TRAU can be performed using A- or $\mu$ -law compressed 8 bit PCM test sequences on the A interface. For this purpose modified input test sequences (TXX\_ Y.INP) are generated from the original sequences (see clause 6) by A or $\mu$ law compression. As an input to the encoder they result in modified encoder output sequences (TXX.COD). These modified (TXX.COD) sequences are used as decoder input sequences. The decoder will then produce the output sequences TXX\_ Y.OUT, which are A- or $\mu$ compressed.
340
+
341
+ The A- and $\mu$ -law compression and decompression does not change the homing frames at the encoder input. The format of all A- and $\mu$ -law PCM files TXX\_Y.INP and TXX\_Y.OUT is one sample (8 bit) written into 16 bit words. The format of the modified TXX.COD files is as described in clause 5.
342
+
343
+ All files are contained in the archives T\_A.TGZ (for the A-law sequences) and T\_U.TGZ (for the $\mu$ -law sequences) which accompany the present document.
344
+
345
+ In addition to the test sequences above, special input (SEQSYNC\_Y.INP/SEQSYNC\_122\_Y.INP) and output (SEQSYNC000\_<mode>.COD through SEQSYNC159\_<mode>.COD) sequences for frame synchronisation are provided. The Y again stands for A and $\mu$ law compressed PCM and <mode> is described in clause 6.
346
+
347
+ All files are contained in the archives S\_A.TGZ (for the A-law sequences) and S\_U.TGZ (for the $\mu$ -law sequences) which accompany the present document. The synchronization procedure is described in clause 8.
348
+
349
+ # Annex A (informative): Change history
350
+
351
+ | Change history | | | | | | | |
352
+ |----------------|-------|-----------|------|-----|-----------------------------------------------------------------------------------|--------|--------|
353
+ | Date | TSG # | TSG Doc. | CR | Rev | Subject/Comment | Old | New |
354
+ | 12-1999 | 6 | | | | V.3.0.0 Approved | | 3.0.0 |
355
+ | 02-2000 | | | | | Inclusion of test sequences | 3.0.0 | 3.0.1 |
356
+ | 09-2000 | | | | | Inclusion of corrected VAD2 test sequences | 3.0.1 | 3.0.2 |
357
+ | 02-2001 | 11 | SP-010101 | 001 | | Update of AMR codec test sequences after CRs to TS 26.073 | 3.0.2 | 3.1.0 |
358
+ | 03-2001 | 11 | SP-010101 | 002 | | Update of AMR codec test sequences after CRs to TS 26.073 | 3.0.2 | 4.0.0 |
359
+ | 06-2001 | | | | | Update of AMR codec test sequences (including also the synchronisation sequences) | 4.0.0 | 4.0.1 |
360
+ | 06-2002 | 16 | | | | Version for Release 5 | 4.0.1 | 5.0.0 |
361
+ | 12-2004 | 26 | | | | Version for Release 6 | 5.0.0 | 6.0.0 |
362
+ | 06-2007 | 36 | | | | Version for Release 7 | 6.0.0 | 7.0.0 |
363
+ | 12-2008 | 42 | | | | Version for Release 8 | 7.0.0 | 8.0.0 |
364
+ | 12-2009 | 46 | | | | Version for Release 9 | 8.0.0 | 9.0.0 |
365
+ | 11-2011 | 51 | | | | Version for Release 10 | 9.0.0 | 10.0.0 |
366
+ | 09-2012 | 57 | | | | Version for Release 11 | 10.0.0 | 11.0.0 |
367
+ | 12-2013 | 62 | SP-130568 | 0005 | | Correction to references | 11.0.0 | 11.1.0 |
marked/Rel-11/26_series/26092/raw.md ADDED
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1
+
2
+
3
+
4
+
5
+
6
+
7
+ # --- Contents
8
+
9
+ | | |
10
+ |------------------------------------------------------------------------------------|----|
11
+ | Foreword ..... | 4 |
12
+ | 1 Scope..... | 5 |
13
+ | 2 References..... | 5 |
14
+ | 3 Definitions, symbols and abbreviations ..... | 6 |
15
+ | 3.1 Definitions..... | 6 |
16
+ | 3.2 Symbols..... | 6 |
17
+ | 3.3 Abbreviations ..... | 6 |
18
+ | 4 General..... | 7 |
19
+ | 5 Functions on the transmit (TX) side ..... | 7 |
20
+ | 5.1 LSF evaluation ..... | 7 |
21
+ | 5.2 Frame energy calculation ..... | 8 |
22
+ | 5.3 Modification of the speech encoding algorithm during SID frame generation..... | 8 |
23
+ | 5.4 SID-frame encoding ..... | 9 |
24
+ | 6 Functions on the receive (RX) side..... | 9 |
25
+ | 6.1 Averaging and decoding of the LP and energy parameters ..... | 9 |
26
+ | 6.2 Comfort noise generation and updating ..... | 10 |
27
+ | 7 Computational details and bit allocation..... | 11 |
28
+ | Annex A (informative): Change history..... | 12 |
29
+
30
+ # --- Foreword
31
+
32
+ This Technical Specification has been produced by the 3<sup>rd</sup> Generation Partnership Project (3GPP).
33
+
34
+ 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:
35
+
36
+ Version x.y.z
37
+
38
+ where:
39
+
40
+ - x the first digit:
41
+ - 1 presented to TSG for information;
42
+ - 2 presented to TSG for approval;
43
+ - 3 or greater indicates TSG approved document under change control.
44
+ - y the second digit is incremented for all changes of substance, i.e. technical enhancements, corrections, updates, etc.
45
+ - z the third digit is incremented when editorial only changes have been incorporated in the document.
46
+
47
+ # --- 1 Scope
48
+
49
+ The present document gives the detailed requirements for the correct operation of the background acoustic noise evaluation, noise parameter encoding/decoding and comfort noise generation for the AMR speech codec during Source Controlled Rate (SCR) operation.
50
+
51
+ The requirements described in the present document are mandatory for implementation in all UEs capable of supporting the AMR speech codec.
52
+
53
+ The receiver requirements are mandatory for implementation in all networks capable of supporting the AMR speech codec, the transmitter requirements only for those where downlink SCR will be used.
54
+
55
+ In case of discrepancy between the requirements described in the present document and the fixed point computational description of these requirements contained in [1], the description in [1] will prevail.
56
+
57
+ # --- 2 References
58
+
59
+ The following documents contain provisions which, through reference in this text, constitute provisions of the present document.
60
+
61
+ - References are either specific (identified by date of publication, edition number, version number, etc.) or non-specific.
62
+ - For a specific reference, subsequent revisions do not apply.
63
+ - 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*.
64
+
65
+ - [1] 3GPP TS 26.073: "Adaptive Multi-Rate (AMR); ANSI C source code".
66
+ - [2] 3GPP TS 26.090 : "Transcoding functions".
67
+ - [3] 3GPP TS 26.091: "Mandatory Speech Codec speech processing functions; AMR Speech Codec; Error concealment of lost frames".
68
+ - [4] 3GPP TS 26.093: "Source Controlled Rate operation ".
69
+ - [5] 3GPP TS 26.101: "Frame Structure".
70
+
71
+ # 3 Definitions, symbols and abbreviations
72
+
73
+ ## 3.1 Definitions
74
+
75
+ For the purpose of the present document, the following terms and definitions apply.
76
+
77
+ **Frame:** time interval of 20 ms corresponding to the time segmentation of the adaptive multi-rate speech transcoder, also used as a short term for traffic frame.
78
+
79
+ **SID frames:** special Comfort Noise frames. It may convey information on the acoustic background noise or inform the decoder that it should start generating background noise.
80
+
81
+ **Speech frame:** traffic frame that cannot be classified as a SID frame.
82
+
83
+ **VAD flag:** voice Activity Detection flag.
84
+
85
+ **TX\_TYPE:** one of SPEECH, SID\_FIRST, SID\_UPD, NO\_DATA (defined in TS 26.093 [4]).
86
+
87
+ **RX\_TYPE:** classification of the received traffic frame (defined in TS 26.093 [4]).
88
+
89
+ Other definitions of terms used in the present document can be found in TS 26.090 [2] and TS 26.093 [4]. The overall operation of SCR is described in TS 26.093 [4].
90
+
91
+ ## 3.2 Symbols
92
+
93
+ For the purpose of the present document , the following symbols apply. Boldface symbols are used for vector variables.
94
+
95
+ $$\sum_{n=a}^b x(n) = x(a) + x(a+1) + \dots + x(b-1) + x(b)$$
96
+
97
+ $en_{\log}^{mean}$ Averaged logarithmic frame energy
98
+
99
+ $\mathbf{f}^{mean}$ Averaged LSF parameter vector
100
+
101
+ $\mathbf{e}$ Computed LSF parameter prediction residual
102
+
103
+ $en_{\log}$ Logarithmic frame energy
104
+
105
+ $\hat{\mathbf{e}}$ Quantized LSF parameter prediction residual
106
+
107
+ $\hat{\mathbf{f}}^T = [\hat{f}_1 \hat{f}_2 \dots \hat{f}_{10}]$ Quantized LSF vector
108
+
109
+ $\hat{\mathbf{f}}^{(m)}$ Quantized LSF vector of frame $m$
110
+
111
+ $\hat{\mathbf{f}}^{ref}$ Reference vector for LSF quantization
112
+
113
+ $\mathbf{f}^T = [f_1 f_2 \dots f_{10}]$ Unquantized LSF vector
114
+
115
+ $\mathbf{f}^{(m)}$ Unquantized LSF vector of frame $m$
116
+
117
+ ## 3.3 Abbreviations
118
+
119
+ For the purpose of the present document , the following abbreviations apply.
120
+
121
+ | | |
122
+ |-----|----------------------------------------------------------------------------|
123
+ | AMR | Adaptive Multi-Rate |
124
+ | SCR | Source Controlled Rate operation ( aka source discontinuous transmission ) |
125
+ | UE | User Equipment |
126
+ | SID | SIlence Descriptor |
127
+ | LP | Linear Prediction |
128
+ | LSP | Line Spectral Pair |
129
+ | LSF | Line Spectral Frequency |
130
+ | RX | Receive |
131
+ | TX | Transmit |
132
+
133
+ # --- 4 General
134
+
135
+ A basic problem when using SCR is that the background acoustic noise, which is transmitted together with the speech, would disappear when the transmission is cut, resulting in discontinuities of the background noise. Since the SCR switching can take place rapidly, it has been found that this effect can be very annoying for the listener - especially in a car environment with high background noise levels. In bad cases, the speech may be hardly intelligible.
136
+
137
+ This document specifies the way to overcome this problem by generating on the receive (RX) side synthetic noise similar to the transmit (TX) side background noise. The comfort noise parameters are estimated on the TX side and transmitted to the RX side at a regular rate when speech is not present. This allows the comfort noise to adapt to the changes of the noise on the TX side.
138
+
139
+ # --- 5 Functions on the transmit (TX) side
140
+
141
+ The comfort noise evaluation algorithm uses the following parameters of the AMR speech encoder, defined in [2]:
142
+
143
+ - the unquantized Linear Prediction (LP) parameters, using the Line Spectral Pair (LSP) representation, where the unquantized Line Spectral Frequency (LSF) vector is given by $\mathbf{f}^T = [f_1 \ f_2 \ \dots \ f_{10}]$ ;
144
+ - the unquantized LSF vector for the 12.2 kbit/s mode is given by the second set of LSF parameters in the frame.
145
+
146
+ The algorithm computes the following parameters to assist in comfort noise generation:
147
+
148
+ - the averaged LSF parameter vector $\mathbf{f}^{mean}$ (average of the LSF parameters of the eight most recent frames);
149
+ - the averaged logarithmic frame energy $en_{log}^{mean}$ (average of the logarithmic energy of the eight most recent frames).
150
+
151
+ These parameters give information on the level ( $en_{log}^{mean}$ ) and the spectrum ( $\mathbf{f}^{mean}$ ) of the background noise.
152
+
153
+ The evaluated comfort noise parameters ( $\mathbf{f}^{mean}$ and $en_{log}^{mean}$ ) are encoded into a special frame, called a Silence Descriptor (SID) frame for transmission to the RX side.
154
+
155
+ A hangover logic is used to enhance the quality of the silence descriptor frames. A hangover of seven frames is added to the VAD flag so that the coder waits with the switch from active to inactive mode for a period of seven frames, during that time the decoder can compute a silence descriptor frame from the quantized LSFs and the logarithmic frame energy of the decoded speech signal. Therefore, no comfort noise description is transmitted in the first SID frame after active speech. If the background noise contains transients which will cause the coder to switch to active mode and then back to inactive mode in a very short time period, no hangover is used. Instead the previously used comfort noise frames are used for comfort noise generation.
156
+
157
+ The first SID frame also serves to initiate the comfort noise generation on the receive side, as a first SID frame is always sent at the end of a speech burst, i.e., before the transmission is terminated.
158
+
159
+ The scheduling of SID or speech frames on the network path is described in [4].
160
+
161
+ ## 5.1 LSF evaluation
162
+
163
+ The comfort noise parameters to be encoded into a SID frame are calculated over $N = 8$ consecutive frames marked with VAD=0, as follows:
164
+
165
+ The averaged LSF parameter vector $\mathbf{f}^{mean}(i)$ of the frame $i$ shall be computed according to the equation:
166
+
167
+ $$\mathbf{f}^{mean}(i) = \frac{1}{8} \sum_{n=0}^7 \mathbf{f}(i-n) \quad (1)$$
168
+
169
+ where $\mathbf{f}(i-n)$ is the (unquantized) LSF parameter vector of the current frame $i$ ( $n=0$ ) and past frames ( $n=1, \dots, 7$ ).
170
+
171
+ The averaged LSF parameter vector $\mathbf{f}^{mean}(i)$ of the frame $i$ is encoded using the same encoding tables that are also used by the 7.4 kbit/s mode for the encoding of the non-averaged LSF parameter vectors in ordinary speech encoding mode, but the quantization algorithm is modified in order to support the quantization of comfort noise.
172
+
173
+ The LSF parameter prediction residual to be quantized for frame $i$ is obtained according to the following equation:
174
+
175
+ $$\mathbf{e}(i) = \mathbf{f}^{mean}(i) - \hat{\mathbf{f}}^{ref} \quad (2)$$
176
+
177
+ where $\hat{\mathbf{f}}^{ref}$ is a reference vector picked from a codebook.
178
+
179
+ The vector $\hat{\mathbf{f}}^{ref}$ used in eq (2) is encoded for each SID frame. A lookup table containing 8 vectors typical for background noise are searched. The vector which yields the lowest prediction residual energy is selected. After the above step the LSF parameter encoding procedure is performed. The 3-bit index for the reference vector and the 26 bits for LSF parameter are transmitted in the SID frame (see bit allocation in table 1).
180
+
181
+ ## 5.2 Frame energy calculation
182
+
183
+ The frame energy is computed for each frame marked with VAD=0 according to the equation :
184
+
185
+ $$en_{\log}(i) = \frac{1}{2} \log_2 \left( \frac{1}{N} \sum_{n=0}^{N-1} s^2(n) \right) \quad (3)$$
186
+
187
+ where $s(n)$ is the HP-filtered input speech signal of the current frame $i$ .
188
+
189
+ The averaged logarithmic energy is computed by:
190
+
191
+ $$en_{\log}^{mean}(i) = \frac{1}{8} \sum_{n=0}^7 en_{\log}(i-n) \quad (4)$$
192
+
193
+ The averaged logarithmic energy is quantized means of a 6 bit algorithmic quantizer. The 6 bits for the energy index are transmitted in the SID frame (see bit allocation in table 1).
194
+
195
+ ## 5.3 Modification of the speech encoding algorithm during SID frame generation
196
+
197
+ When the TX\_TYPE is not equal to SPEECH the speech encoding algorithm is modified in the following way:
198
+
199
+ - The non-averaged LP parameters which are used to derive the filter coefficients of the filters $H(z)$ and $W(z)$ of the speech encoder are not quantized;
200
+ - The open loop pitch lag search is performed, but the closed loop pitch lag search is inactivated. The adaptive codebook gain and memory is set to zero.
201
+ - No fixed codebook search is made.
202
+ - The memory of weighting filter $W(z)$ is set to zero, i.e., the memory of $W(z)$ is not updated.
203
+
204
+ - The ordinary LP parameter quantization algorithm is inactive. The averaged LSF parameter vector $\mathbf{f}^{mean}$ is calculated each time a new SID frame is to be sent to the AN. This parameter vector is encoded into the SID frame as defined in subclause 5.1.
205
+ - The ordinary gain quantization algorithm is inactive.
206
+ - The predictor memories of the ordinary LP parameter quantization and fixed codebook gain quantization algorithms are initialized when TX\_TYPE is not SPEECH, so that the quantizers start from known initial states when the speech activity begins again.
207
+
208
+ ## 5.4 SID-frame encoding
209
+
210
+ The encoding of the comfort noise bits in a SID frame is described in [5] where the indication of the first SID frame is also described. The bit allocation and sequence of the bits from comfort noise encoding is shown in Table 1.
211
+
212
+ # 6 Functions on the receive (RX) side
213
+
214
+ The situations in which comfort noise shall be generated on the receive side are defined in [4]. In general, the comfort noise generation is started or updated whenever a valid SID frame is received.
215
+
216
+ ## 6.1 Averaging and decoding of the LP and energy parameters
217
+
218
+ When speech frames are received by the decoder the LP and the energy parameters of the last seven speech frames shall be kept in memory. The decoder counts the number of frames elapsed since the last SID frame was updated and passed to the RSS by the encoder. Based on this count, the decoder determines whether or not there is a hangover period at the end of the speech burst (defined in [4]). The interpolation factor is also adapted to the SID update rate.
219
+
220
+ As soon as a SID frame is received comfort noise is generated at the decoder end. The first SID frame parameters are not received but computed from the parameters stored during the hangover period. If no hangover period is detected, the parameters from the previous SID update are used.
221
+
222
+ The averaging procedure for obtaining the comfort noise parameters for the first SID frame is as follows:
223
+
224
+ - when a speech frame is received, the LSF vector is decoded and stored in memory, moreover the logarithmic frame energy of the decoded signal is also stored in memory.
225
+ - the averaged values of the quantized LSF vectors and the averaged logarithmic frame energy of the decoded frames are computed and used for comfort noise generation.
226
+
227
+ The averaged value of the LSF vector for the first SID frame is given by:
228
+
229
+ $$\hat{\mathbf{f}}^{mean}(i) = \frac{1}{8} \sum_{n=0}^7 \hat{\mathbf{f}}(i-n) \quad (5)$$
230
+
231
+ where $\hat{\mathbf{f}}(i-n)$ , $n > 0$ is the quantized LSF vector of one of the frames of the hangover period and where $\hat{\mathbf{f}}(i-0) = \hat{\mathbf{f}}(i-1)$ . The averaged logarithmic frame energy for the first SID frame is given by:
232
+
233
+ $$\hat{en}_{log}^{mean}(i) = \frac{1}{8} \sum_{n=0}^7 \hat{en}_{log}(i-n) \quad (6)$$
234
+
235
+ where $\hat{en}_{log}(i-n)$ , $n > 0$ is the logarithmic vector of one of the frames of the hangover period computed for the decoded frames and where $\hat{en}_{log}(i-0) = \hat{en}_{log}(i-1)$ .
236
+
237
+ For ordinary SID frames, the LSF vector and logarithmic frame energy are computed by table lookup. The energy is also adjusted according to the signalled speech modes capabilities, as to provide high quality transitions from Comfort Noise to Speech. The LSF vector is given by the sum of the decoded reference vector and the decoded LSF prediction residual.
238
+
239
+ During comfort noise generation the spectrum and energy of the comfort noise is determined by interpolation between old and new SID frames.
240
+
241
+ In order to achieve a comfort noise that is less static in appearance the LSF vector is slightly perturbed for each frame by adding a small component based on parameters variations computed in the hangover period. The computation of the perturbation is made by computing the mean LSF vector from the matrix $\hat{\mathbf{f}}$ , this mean vector is then subtracted from each of the elements of $\hat{\mathbf{f}}$ forming a new matrix $\tilde{\mathbf{f}}$ . For every frame a mean removed LSF vector is randomly chosen from $\tilde{\mathbf{f}}$ and added to the interpolated LSF vector.
242
+
243
+ ## 6.2 Comfort noise generation and updating
244
+
245
+ The comfort noise generation procedure uses the adaptive multi-rate speech decoder algorithm defined in [2].
246
+
247
+ When comfort noise is to be generated, the various encoded parameters are set as follows:
248
+
249
+ In each subframe, the pulse positions and signs of the fixed codebook excitation are locally generated using uniformly distributed pseudo random numbers. The excitation pulses take values of +1 and -1 when comfort noise is generated. The fixed codebook comfort noise excitation generation algorithm works as follows:
250
+
251
+ ```
252
+ for (i = 0; i < 40; i++) code[i] = 0;
253
+ for (i = 0; i < 10; i++) {
254
+ j = random(4);
255
+ idx = j * 10 + i;
256
+ if (random(2) == 1) code[idx] = 1;
257
+ else code[idx] = -1;
258
+ }
259
+ ```
260
+
261
+ where:
262
+
263
+ **code[0..39]** fixed codebook excitation buffer;
264
+ **random(4)** generates a random integer value, uniformly distributed between 0 and 3;
265
+ **random(2)** generates a random integer value, uniformly distributed between 0 and 1.
266
+
267
+ The fixed codebook gain is computed from the logarithmic frame energy parameter by converting it to the linear domain and normalizing with the gain of LP synthesis filter.
268
+
269
+ The adaptive codebook gain values in each subframe are set to 0, also the memory of the adaptive codebook is set to zero.
270
+
271
+ The pitch delay values in each subframe are set to 40.
272
+
273
+ The LP filter parameters used are those received in the SID frame.
274
+
275
+ The predictor memories of the ordinary LP parameter and fixed codebook gain quantization algorithms are initialized when RX\_TYPE is not SPEECH, so that the quantizers start from given initial states when the speech activity begins again. With these parameters, the speech decoder now performs the standard operations described in [2] and synthesizes comfort noise.
276
+
277
+ Updating of the comfort noise parameters (energy and LP filter parameters) occurs each time a valid SID frame is received, as described in [4].
278
+
279
+ When updating the comfort noise, the parameters above should be interpolated over the SID update period to obtain smooth transitions.
280
+
281
+ # --- 7 Computational details and bit allocation
282
+
283
+ A bit exact computational description of comfort noise encoding and generation in form of an ANSI-C source code is found in [1].
284
+
285
+ The detailed bit allocation and the sequence of bits in the comfort noise encoding is shown in Table 1.
286
+
287
+ **Table 1: Source encoder output parameters in order of occurrence and bit allocation for comfort noise encoding.**
288
+
289
+ | Bits (MSB-LSB) | Description |
290
+ |----------------|-----------------------------------|
291
+ | s1 – s3 | index of reference vector |
292
+ | s4 - s11 | index of 1st LSF subvector |
293
+ | s12 – s20 | index of 2nd LSF subvector |
294
+ | s21 – s29 | index of 3rd LSF subvector |
295
+ | s30 – s35 | index of logarithmic frame energy |
296
+
297
+ # Annex A (informative): Change history
298
+
299
+ | <b>Document history</b> | | | | | | | | |
300
+ |-------------------------|-----------------|---------------------------------------|-----------|------------|------------------------|--|------------|------------|
301
+ | V. 2.0.0 | June 21, 1999 | Presented at S#4 Plenary for approval | | | | | | |
302
+ | V. 3.0.0 | June 22, 1999 | Approved at S#4 Plenary | | | | | | |
303
+ | V. 3.0.1 | August 22, 1999 | Reformatted in 3GPP style | | | | | | |
304
+ | <b>Change history</b> | | | | | | | | |
305
+ | <b>Date</b> | <b>TSG #</b> | <b>TSG Doc.</b> | <b>CR</b> | <b>Rev</b> | <b>Subject/Comment</b> | | <b>Old</b> | <b>New</b> |
306
+ | 03-2001 | 11 | | | | Version for Release 4 | | | 4.0.0 |
307
+ | 06-2002 | 16 | | | | Version for Release 5 | | 4.0.0 | 5.0.0 |
308
+ | 12-2004 | 26 | | | | Version for Release 6 | | 5.0.0 | 6.0.0 |
309
+ | 06-2007 | 36 | | | | Version for Release 7 | | 6.0.0 | 7.0.0 |
310
+ | 12-2008 | 42 | | | | Version for Release 8 | | 7.0.0 | 8.0.0 |
311
+ | 12-2009 | 46 | | | | Version for Release 9 | | 8.0.0 | 9.0.0 |
312
+ | 03-2011 | 51 | | | | Version for Release 10 | | 9.0.0 | 10.0.0 |
313
+ | 09-2012 | 57 | | | | Version for Release 11 | | 10.0.0 | 11.0.0 |
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1
+
2
+
3
+
4
+
5
+
6
+
7
+ # Contents
8
+
9
+ | | |
10
+ |--------------------------------------------------------------------------------------------------|-----------|
11
+ | Foreword ..... | 5 |
12
+ | 1 Scope..... | 6 |
13
+ | 2 References..... | 6 |
14
+ | 3 Definitions, symbols and abbreviations..... | 6 |
15
+ | 3.1 Definitions..... | 6 |
16
+ | 3.2 Symbols..... | 7 |
17
+ | 3.3 Abbreviations ..... | 7 |
18
+ | 4 General..... | 7 |
19
+ | 4.1 General organisation ..... | 7 |
20
+ | 5 AMR SCR operation..... | 8 |
21
+ | 5.1 Transmit (TX) side..... | 8 |
22
+ | 5.1.1 General operation ..... | 8 |
23
+ | 5.1.2 Functions of the TX SCR handler ..... | 9 |
24
+ | 5.1.2.1 AMR SCR Timing procedures..... | 9 |
25
+ | 5.1.3 The TX part of the AN ..... | 10 |
26
+ | 5.2 Receive (RX) side ..... | 11 |
27
+ | 5.2.1 General operation ..... | 11 |
28
+ | 5.2.3 Demands on the RX SCR handler ..... | 11 |
29
+ | 5.3 AMR SID Information format..... | 12 |
30
+ | <b>Annex A (normative): AMR DTX handler for the GSM system (corresponding to GSM 06.93).....</b> | <b>13</b> |
31
+ | A.1 Scope..... | 13 |
32
+ | A.2 References..... | 13 |
33
+ | A.3 Definitions, symbols and abbreviations..... | 14 |
34
+ | A.3.1 Definitions..... | 14 |
35
+ | A.3.2 Symbols..... | 14 |
36
+ | A.3.3 Abbreviations ..... | 15 |
37
+ | A.4 General..... | 15 |
38
+ | A.4.1 General organisation ..... | 15 |
39
+ | A.5 Transmit (TX) side..... | 16 |
40
+ | A.5.1 General operation ..... | 16 |
41
+ | A.5.1.1 Functions of the TX DTX handler..... | 17 |
42
+ | A.5.1.2 Functions of the TX Radio Subsystem..... | 18 |
43
+ | A.5.1.2.1 Functions of the TX Radio Subsystem for TCH/AFS ..... | 18 |
44
+ | A.5.1.2.2 Functions of the TX Radio Subsystem for TCH/AHS ..... | 18 |
45
+ | A.5.1.2.3 Functions of the Downlink TX Radio Subsystem for TFO ..... | 19 |
46
+ | A.5.1.2.4 Functions of the TX Radio Subsystem for RATSCCH ..... | 20 |
47
+ | A.6 Receive (RX) side ..... | 20 |
48
+ | A.6.1 General operation ..... | 20 |
49
+ | A.6.1.1 Functions of the RX radio subsystem..... | 21 |
50
+ | A.6.1.2 Functions of the RX DTX handler ..... | 21 |
51
+ | <b>Annex B (normative): ETSI GSM-EFR SCR handler .....</b> | <b>22</b> |
52
+ | Transmit (TX) side..... | 22 |
53
+ | General operation ..... | 22 |
54
+ | Functions of the TX SCR handler..... | 22 |
55
+ | GSM-EFR SCR Timing procedures..... | 22 |
56
+ | The TX part of the AN ..... | 22 |
57
+ | Demands on the TX part of the Access Network..... | 22 |
58
+
59
+ Receive (RX) side .....23
60
+
61
+ SID Information format.....23
62
+
63
+ **Annex C (normative): TIA IS-641 SCR Handler .....24**
64
+
65
+ TX-side.....24
66
+
67
+ RX-side.....24
68
+
69
+ SID Information format.....24
70
+
71
+ **Annex D (normative): TIA TDMA-US1 SCR Handler .....25**
72
+
73
+ TX-side.....25
74
+
75
+ RX-side.....25
76
+
77
+ SID Information format.....25
78
+
79
+ **Annex E (normative): ARIB PDC-EFR SCR Handler .....26**
80
+
81
+ Transmit (TX) side .....26
82
+
83
+ General operation .....26
84
+
85
+ Functions of the TX SCR handler.....26
86
+
87
+ PDC\_EFR SCR Timing procedures.....26
88
+
89
+ The TX part of the AN .....27
90
+
91
+ Demands on the TX part of the Access Network.....27
92
+
93
+ Receive (RX) side .....27
94
+
95
+ SID information format.....27
96
+
97
+ Annex F (informative): Change history.....28
98
+
99
+ # --- Foreword
100
+
101
+ This Technical Specification has been produced by the 3GPP.
102
+
103
+ The present document describes the operation of the Adaptive Multi Rate speech codec during Source Controlled Rate (SCR) operation within the 3GPP system.
104
+
105
+ 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:
106
+
107
+ Version x.y.z
108
+
109
+ where:
110
+
111
+ - x the first digit:
112
+ - 1 presented to TSG for information;
113
+ - 2 presented to TSG for approval;
114
+ - 3 or greater indicates TSG approved document under change control.
115
+ - y the second digit is incremented for all changes of substance, i.e. technical enhancements, corrections, updates, etc.
116
+ - z the third digit is incremented when editorial only changes have been incorporated in the specification;
117
+
118
+ # --- 1 Scope
119
+
120
+ This document describes the Source Controlled Rate (SCR) operation of the Adaptive Multi-Rate speech Codec in Codec Types UMTS\_AMR and UMTS\_AMR2 for the UMTS system. The implementation of this SCR operation is mandatory in all UMTS equipment.
121
+
122
+ The description is structured according to the block diagram in figure 1. This structure of distributing the various functions between system entities is not mandatory for implementation, as long as the operation on the speech decoder output remains the same.
123
+
124
+ Annex A describes the Discontinuous Transmission (DTX) operation of the Adaptive Multi-Rate speech Codec in Codec Types FR\_AMR, HR\_AMR and OHR\_AMR for GERAN. This annex is the former GSM 06.93 (Release 98).
125
+
126
+ Annexes B to E describe the SCR operation of the Adaptive Multi-Rate speech Codec in Codec Types GSM\_EFR, TDMA\_EFR, TDMA\_US1 and PDC\_EFR for the UMTS system.
127
+
128
+ # --- 2 References
129
+
130
+ The following documents contain provisions which, through reference in this text, constitute provisions of the present document.
131
+
132
+ - References are either specific (identified by date of publication, edition number, version number, etc.) or non-specific.
133
+ - For a specific reference, subsequent revisions do not apply.
134
+ - 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*.
135
+ - [1] 3GPP TS 26.071 : "AMR Speech Codec; General description".
136
+ - [2] 3GPP TS 26.073 : "AMR Speech Codec; ANSI-C code".
137
+ - [3] 3GPP TS 26.074 : "AMR Speech Codec; Test sequences".
138
+ - [4] 3GPP TS 26.090 : "AMR Speech Codec; Transcoding functions".
139
+ - [5] 3GPP TS 26.091 : "AMR Speech Codec; Error concealment of lost frames".
140
+ - [6] 3GPP TS 26.092 : "AMR Speech Codec; Comfort noise aspects".
141
+ - [7] 3GPP TS 26.094 : "AMR Speech Codec; Voice Activity Detector (VAD)".
142
+ - [8] 3GPP TS 26.101 : "AMR Speech Codec; Frame structure".
143
+
144
+ # --- 3 Definitions, symbols and abbreviations
145
+
146
+ ## 3.1 Definitions
147
+
148
+ For the purpose of this document, the following definitions apply.
149
+
150
+ **frame:** Time interval of 20 ms, corresponding to the time segmentation of the Adaptive Multi Rate speech Codec, also used as a short term for a traffic frame.
151
+
152
+ **traffic frame:** Block of 95..244 information bits transmitted on the speech traffic channels.
153
+
154
+ **SID frame:** Frame that conveys information about the acoustic background noise.
155
+
156
+ **speech frame:** Traffic frame that has been classified as SPEECH\_GOOD or SPEECH\_BAD frame.
157
+
158
+ **VAD flag:** Boolean flag, generated by the VAD algorithm indicating the presence ("1") or the absence ("0") of a speech frame.
159
+
160
+ **RX\_TYPE:** classifies the received frame.
161
+
162
+ **TX\_TYPE:** classifies the frame to be transmitted.
163
+
164
+ **hangover period:** A period of frames added at the end of a speech burst in which VAD flag ="0" and TX\_TYPE is ="SPEECH\_GOOD", this period provides the encoder with an extra window to derive the Comfort Noise parameters .
165
+
166
+ ## 3.2 Symbols
167
+
168
+ For the purpose of this document, the following symbols apply.
169
+
170
+ | | |
171
+ |----------------------|------------------------------------------------------------|
172
+ | $N_{\text{elapsed}}$ | Number of elapsed frames since the last updated SID frame. |
173
+ |----------------------|------------------------------------------------------------|
174
+
175
+ ## 3.3 Abbreviations
176
+
177
+ For the purpose of this document , the following abbreviations apply.
178
+
179
+ | | |
180
+ |----------|------------------------------------------------------|
181
+ | AN | Access Network |
182
+ | SCR | Source Controlled Rate operation |
183
+ | TS | Telecommunication Standard, Technical Specification |
184
+ | GSM | Global System for Mobile Telecommunication |
185
+ | GSM-EFR | GSM Enhanced Full Rate speech Codec |
186
+ | UE | User Equipment |
187
+ | PDC-EFR | ARIB PDC-EFR 6.7 kBit/s speech Codec |
188
+ | RAN | Radio Access Network |
189
+ | RX | Receive |
190
+ | SID | Silence Descriptor |
191
+ | TDMA-EFR | TIA IS-641 Enhanced speech Codec |
192
+ | TDMA-US1 | TIA TDMA-US1 (12.2 kBit/s Codec, similar to GSM-EFR) |
193
+ | TX | Transmit |
194
+ | VAD | Voice Activity Detector |
195
+
196
+ # --- 4 General
197
+
198
+ Source Controlled Rate operation (SCR) is a mechanism for the AMR Speech Codec, which allows to encode the input signal at a lower average rate by taking speech inactivity into account. The SCR scheme may be used for the following purposes:
199
+
200
+ - to save power in the User Equipment;
201
+ - to reduce the overall interference and load in the networks.
202
+
203
+ SCR in the transmitting path (uplink) shall be in operation in UEs, if commanded so by the network. Note that for UMTS AMR and UMTS AMR2 codec types, the uplink SCR operation shall always be activated. The UE shall handle SCR in the receiving path (downlink) at any time, regardless, whether SCR in the transmitting path is enabled or not.
204
+
205
+ ## 4.1 General organisation
206
+
207
+ The default SCR mechanism described in this document requires the following functions:
208
+
209
+ - a Voice Activity Detector (VAD) on the transmit (TX) side;
210
+ - evaluation of the background acoustic noise on the transmit (TX) side, in order to transmit characteristic parameters to the receive (RX) side;
211
+ - generation on the receive (RX) side of a similar noise, called comfort noise, during periods where the transmission is switched off.
212
+
213
+ The Voice Activity Detector (VAD) is defined in [7] and the AMR-mode comfort noise functions in [6]. Both are based partly on the speech Codec and its internal variables, defined in [4].
214
+
215
+ In addition to these functions, if the parameters arriving at the RX side are detected to be seriously corrupted by errors, the speech or comfort noise must be generated from substituted data in order to avoid seriously annoying effects for the listener. These functions are defined in [5].
216
+
217
+ An overall description of the speech processing parts can be found in [1]. An overview of one link SCR operation is shown in Figure 1.
218
+
219
+ ![Figure 1: Block diagram of one link SCR operation. The diagram shows three main components: TX SCR handler, 'Network', and RX SCR handler. The TX SCR handler contains Speech Encoder, Voice Activity Detector, and Comfort Noise Parameter Computation. The 'Network' contains Information packeting, transport and classification. The RX SCR handler contains Speech Decoder, Error Concealment, and Comfort Noise Generation. Arrows show the flow of Information bits, Mode Indication, TX_TYPE, and RX_TYPE between these components.](5860ad6bd2a2dd8d1ab12864b8f90f37_img.jpg)
220
+
221
+ ```
222
+
223
+ graph LR
224
+ subgraph TX_SCR_handler [TX SCR handler]
225
+ SE[Speech Encoder]
226
+ VAD[Voice Activity Detector]
227
+ CNPC[Comfort Noise Parameter Computation]
228
+ end
229
+ subgraph Network ["Network"]
230
+ IPC[Information packeting, transport and classification]
231
+ end
232
+ subgraph RX_SCR_handler [RX SCR handler]
233
+ SD[Speech Decoder]
234
+ EC[Error Concealment]
235
+ CN[Comfort Noise Generation]
236
+ end
237
+ SE -- "Information bits" --> IPC
238
+ VAD -- "Mode Indication" --> IPC
239
+ CNPC -- "TX_TYPE" --> IPC
240
+ IPC -- "Information bits" --> SD
241
+ IPC -- "Mode Indication" --> EC
242
+ IPC -- "RX_TYPE" --> CN
243
+
244
+ ```
245
+
246
+ Figure 1: Block diagram of one link SCR operation. The diagram shows three main components: TX SCR handler, 'Network', and RX SCR handler. The TX SCR handler contains Speech Encoder, Voice Activity Detector, and Comfort Noise Parameter Computation. The 'Network' contains Information packeting, transport and classification. The RX SCR handler contains Speech Decoder, Error Concealment, and Comfort Noise Generation. Arrows show the flow of Information bits, Mode Indication, TX\_TYPE, and RX\_TYPE between these components.
247
+
248
+ Figure 1: Block diagram of one link SCR operation
249
+
250
+ # 5 AMR SCR operation
251
+
252
+ ## 5.1 Transmit (TX) side
253
+
254
+ A block diagram of the transmit side SCR functions is shown in Figure 2.
255
+
256
+ ![Figure 2: Block diagram of SCR functions at the TX side. The diagram shows three main components: TX SCR handler, Framing, and TX Access Network. The TX SCR handler contains Speech Encoder, Voice Activity Detector, and Comfort Noise Parameter Computation. The Framing unit contains bit fields T, S, 2, 6, ., 1, 0, 1. The TX Access Network contains TX of information. Arrows show the flow of Information bits, Mode Indication, TX_TYPE, and the resulting frame fields (Information bits, Frame Type, FQI) between these components.](1c94fd3cebf58af136144f14160d128e_img.jpg)
257
+
258
+ ```
259
+
260
+ graph LR
261
+ subgraph TX_SCR_handler [TX SCR handler]
262
+ SE[Speech Encoder]
263
+ VAD[Voice Activity Detector]
264
+ CNPC[Comfort Noise Parameter Computation]
265
+ end
266
+ subgraph Framing
267
+ T[T]
268
+ S[S]
269
+ 2[2]
270
+ 6[6]
271
+ dot[.]
272
+ 1[1]
273
+ 0[0]
274
+ 1[1]
275
+ end
276
+ subgraph TX_Access_Network [TX Access Network]
277
+ TXI[TX of information]
278
+ end
279
+ SE -- "Information bits" --> Framing
280
+ VAD -- "Mode Indication" --> Framing
281
+ CNPC -- "TX_TYPE" --> Framing
282
+ Framing -- "Information bits" --> TXI
283
+ Framing -- "Frame Type" --> TXI
284
+ Framing -- "FQI" --> TXI
285
+
286
+ ```
287
+
288
+ Figure 2: Block diagram of SCR functions at the TX side. The diagram shows three main components: TX SCR handler, Framing, and TX Access Network. The TX SCR handler contains Speech Encoder, Voice Activity Detector, and Comfort Noise Parameter Computation. The Framing unit contains bit fields T, S, 2, 6, ., 1, 0, 1. The TX Access Network contains TX of information. Arrows show the flow of Information bits, Mode Indication, TX\_TYPE, and the resulting frame fields (Information bits, Frame Type, FQI) between these components.
289
+
290
+ Figure 2: Block diagram of SCR functions at the TX side
291
+
292
+ ### 5.1.1 General operation
293
+
294
+ The TX SCR handler passes traffic frames, individually marked by TX\_TYPE, to the Framing unit. Each frame consists of bit fields containing the information bits, the codec mode indication, and the TX\_TYPE. TX\_TYPE shall be used to specify the contents of the frame. The table below provides an overview of the different TX\_TYPES used and explains the required contents in the information bit and the mode indication bit fields.
295
+
296
+ **Table 1: SCR TX\_TYPE identifiers for UMTS\_AMR and UMTS\_AMR2**
297
+
298
+ | <b>TX_TYPE</b> | <b>Information Bits</b> | <b>Mode Indication</b> |
299
+ |----------------|-----------------------------------------------------------------------------------------------|----------------------------------------------------------------------------|
300
+ | SPEECH_GOOD | Speech frame, size 95..244 bits, depending on codec mode | Current codec mode |
301
+ | SPEECH_BAD | Corrupt speech frame (bad CRC), size 95..244 bits, depending on codec mode | Current codec mode |
302
+ | SID_FIRST | Marker for the end of talkspurt, no further information, all 35 comfort noise bits set to "0" | The codec mode that would have been used if TX_TYPE had been "SPEECH_GOOD" |
303
+ | SID_UPDATE | 35 comfort noise bits | The codec mode that would have been used if TX_TYPE had been "SPEECH_GOOD" |
304
+ | SID_BAD | Corrupt SID update frame (bad CRC) | The codec mode that would have been used if TX_TYPE had been "SPEECH_GOOD" |
305
+ | NO_DATA | No useful information, nothing to be transmitted | No useful information |
306
+
307
+ TX\_TYPE = "NO\_DATA" indicates that the Information Bit and Codec Mode fields do not contain any useful data (and should not be transmitted over AN). The purpose of this TX\_TYPE is to provide the option to save network transmission between the transcoder and AN. Note, the TX\_TYPEs "SPEECH\_BAD" and "SID\_BAD" may occur in TFO and TrFO situations.
308
+
309
+ The scheduling of the frames for transmission on the Access Network is controlled by the TX SCR handler by the use of the TX\_TYPE field.
310
+
311
+ ### 5.1.2 Functions of the TX SCR handler
312
+
313
+ If TX SCR operation is disabled, the TX SCR handler continuously generates speech frames, i.e. frames marked with TX\_TYPE="SPEECH\_GOOD".
314
+
315
+ If the TX SCR operation is enabled, the VAD flag controls the TX SCR handler operation as described in the following paragraphs.
316
+
317
+ Note that the TX SCR operation is always enabled on the UE side for UMTS AMR and UMTS AMR2 codec types.
318
+
319
+ #### 5.1.2.1 AMR SCR Timing procedures
320
+
321
+ To allow an exact verification of the TX SCR handler functions, all frames before the reset of the system are treated as if there were speech frames of an infinitely long time. Therefore, and in order to ensure the correct estimation of comfort noise parameters at RX SCR side, the first 7 frames after the reset or after enabling the SCR operation shall always be marked with TX\_TYPE= "SPEECH\_GOOD", even if VAD flag ="0" (hangover period, see figure 3).
322
+
323
+ The Voice Activity Detector (VAD) shall operate all the time in order to assess whether the input signal contains speech or not. The output is a binary flag (VAD flag ="1" or VAD flag ="0", respectively) on a frame by frame basis (see [7]).
324
+
325
+ The VAD flag controls indirectly, via the TX SCR handler operations described below, the overall SCR operation on the transmit side.
326
+
327
+ Whenever VAD flag ="1", the speech encoder output frame along with mode information shall be passed directly to the AN, marked with TX\_TYPE = " SPEECH\_GOOD "
328
+
329
+ At the end of a speech burst (transition VAD flag ="1" to VAD flag ="0"), it takes eight consecutive frames to make a new updated SID analysis available (see [6]). Normally, the first seven speech encoder output frames after the end of
330
+
331
+ the speech burst shall therefore be passed directly to the AN, marked with TX\_TYPE = " SPEECH\_GOOD " ("hangover period").
332
+
333
+ The end of the speech is then indicated by passing frame eight after the end of the speech burst to the AN, marked with TX\_TYPE = "SID\_FIRST" (see figure 3). SID\_FIRST frames do not contain data.
334
+
335
+ ![Figure 3: Normal hangover procedure for AMR (N_elapsed > 23). The diagram shows a timeline of frames with VAD flags and TX types. A 'Hangover' period is indicated by a dotted line starting from the 'end of speech burst' (frame 42) and ending at the 'first pause frame' (frame 43). The 'last speech frame' is frame 42. The 'end of speech burst' is frame 42. The 'first pause frame' is frame 43. The 'Hangover' period is from frame 42 to frame 43. The 'Frame (20 ms)' is indicated by a double-headed arrow between frames 43 and 44. The 'N_elapsed' row shows values: e.g., 35, 36, 37, 38, 39, 40, 41, 42, 43, 44, 45, 0, 1. The 'TX Type' row shows: S, S, S, S, S, S, S, S, S, F, N, N, U, N. The 'Frames to AN' row shows: S, S, S, S, S, S, S, S, S, F, N, N, U, N. An arrow points from the 'TX Type' label to the 'S' in frame 35.](b8661c6c54f72ecc7ff6cb05e47b2891_img.jpg)
336
+
337
+ | | | | | | | | | | | | | | |
338
+ |---------------------------|----|----|----|----|----|----|----|----|----|----|----|---|---|
339
+ | N <sub>elapsed</sub> e.g. | 35 | 36 | 37 | 38 | 39 | 40 | 41 | 42 | 43 | 44 | 45 | 0 | 1 |
340
+ | TX Type | S | S | S | S | S | S | S | S | F | N | N | U | N |
341
+ | Frames to AN | S | S | S | S | S | S | S | S | F | N | N | U | N |
342
+
343
+ Figure 3: Normal hangover procedure for AMR (N\_elapsed > 23). The diagram shows a timeline of frames with VAD flags and TX types. A 'Hangover' period is indicated by a dotted line starting from the 'end of speech burst' (frame 42) and ending at the 'first pause frame' (frame 43). The 'last speech frame' is frame 42. The 'end of speech burst' is frame 42. The 'first pause frame' is frame 43. The 'Hangover' period is from frame 42 to frame 43. The 'Frame (20 ms)' is indicated by a double-headed arrow between frames 43 and 44. The 'N\_elapsed' row shows values: e.g., 35, 36, 37, 38, 39, 40, 41, 42, 43, 44, 45, 0, 1. The 'TX Type' row shows: S, S, S, S, S, S, S, S, S, F, N, N, U, N. The 'Frames to AN' row shows: S, S, S, S, S, S, S, S, S, F, N, N, U, N. An arrow points from the 'TX Type' label to the 'S' in frame 35.
344
+
345
+ TX Types: "S" = SPEECH; "F" = SID\_FIRST; "U" = "SID\_UPDATE; "N" = NO DATA
346
+ N<sub>elapsed</sub>: No. of elapsed frames since last SID\_UPDATE
347
+
348
+ **Figure 3: Normal hangover procedure for AMR (N<sub>elapsed</sub> > 23)**
349
+
350
+ If, however, at the end of the speech burst, less than 24 frames have elapsed since the last SID\_UPDATE frame was computed, then this last analysed SID\_UPDATE frame should be passed to the AN whenever a SID\_UPDATE frame is to be produced, until a new updated SID analysis is available (8 consecutive frames marked with VAD flag = "0"). This reduces the load on the network in cases where short background noise spikes are taken for speech, by avoiding the "hangover" waiting for the SID frame computation.
351
+
352
+ Once the SID\_FIRST frame has been passed to the AN, the TX SCR handler shall at regular intervals compute and pass updated SID\_UPDATE (Comfort Noise) frames to the AN as long as VAD flag = "0". SID\_UPDATE frames shall be generated every 8<sup>th</sup> frame. The first SID\_UPDATE shall be sent as the third frame after the SID\_FIRST frame.
353
+
354
+ The speech encoder is operated in full speech modality if TX\_TYPE = " SPEECH\_GOOD " and otherwise in a simplified mode, because not all encoder functions are required for the evaluation of comfort noise parameters and because comfort noise parameters are only to be generated at certain times.
355
+
356
+ ### 5.1.3 The TX part of the AN
357
+
358
+ The TX part of the AN has the following overall functionality. The transmission is cut after the transmission of a SID\_FIRST frame when the speaker stops talking. During speech pauses the transmission is resumed at regular intervals for transmission of one SID\_UPDATE frame, in order to update the generated comfort noise on the RX side. The operation of
359
+
360
+ the TX part of the AN is controlled by the TX SCR handler via the TX\_TYPE.
361
+
362
+ All frames, marked with SPEECH\_GOOD, SID\_FIRST or SID\_UPDATE shall be transmitted by the TX part of the AN.
363
+
364
+ ## 5.2 Receive (RX) side
365
+
366
+ A block diagram of the receive side SCR functions is shown in Figure 4 below.
367
+
368
+ ![Block diagram of the receive side SCR functions. The diagram shows three main components: RX SCR handler, De-framing, and RX Access Network. The RX Access Network contains 'AMR Frame Type Detection' and 'Error Correction & Detection'. It sends 'Information bits', 'Frame Type', and 'FQI' to the 'De-framing' block. The 'De-framing' block contains a vertical list of bits: T, S, 2, 6, ., 1, 0, 1. It sends 'Information bits', 'Mode Indication', and 'RX_TYPE' to the 'RX SCR handler'. The 'RX SCR handler' contains 'Speech Decoder', 'Error Concealment', and 'Comfort Noise Generation'.](7e670a2b556b53ea9002dfff3a420e08_img.jpg)
369
+
370
+ Block diagram of the receive side SCR functions. The diagram shows three main components: RX SCR handler, De-framing, and RX Access Network. The RX Access Network contains 'AMR Frame Type Detection' and 'Error Correction & Detection'. It sends 'Information bits', 'Frame Type', and 'FQI' to the 'De-framing' block. The 'De-framing' block contains a vertical list of bits: T, S, 2, 6, ., 1, 0, 1. It sends 'Information bits', 'Mode Indication', and 'RX\_TYPE' to the 'RX SCR handler'. The 'RX SCR handler' contains 'Speech Decoder', 'Error Concealment', and 'Comfort Noise Generation'.
371
+
372
+ Figure 4: Block diagram of the receive side SCR functions
373
+
374
+ ### 5.2.1 General operation
375
+
376
+ The AN passes all the received traffic frames to the RX SCR handler, classified with RX\_TYPE, as described in Table 2 (see TS 26.102). The RX SCR handles the frame accordingly.
377
+
378
+ Table 2: RX\_TYPE identifiers for UMTS\_AMR and UMTS\_AMR2
379
+
380
+ | RX_TYPE | Information Bits |
381
+ |--------------------|--------------------------------------------------------------------------------------------|
382
+ | <b>SPEECH_GOOD</b> | Speech frame without detected errors. |
383
+ | <b>SPEECH_BAD</b> | (likely) speech frame with bad CRC (or estimated to be very bad by the RX part of the AN ) |
384
+ | <b>SID_FIRST</b> | This SID-frame marks the beginning of a comfort noise period. |
385
+ | <b>SID_UPDATE</b> | Correct SID update frame |
386
+ | <b>SID_BAD</b> | Corrupt SID update frame (bad CRC; applicable only for SID_UPDATE frames) |
387
+ | <b>NO_DATA</b> | Nothing useable was received. The synthesis mode of the previous frame type is used. |
388
+
389
+ ### 5.2.3 Demands on the RX SCR handler
390
+
391
+ The RX SCR handler is responsible for the overall SCR operation on the RX side. It consists of two main modes: SPEECH and COMFORT\_NOISE. The initial mode shall be SPEECH.
392
+
393
+ The SCR operation on the RX side shall be as follows:
394
+
395
+ - The RX SCR handler shall enter mode SPEECH, when a frame classified as SPEECH\_GOOD is received.
396
+
397
+ whenever a frame classified as SPEECH\_GOOD is received the RX SCR handler shall pass it directly on to the speech decoder;
398
+
399
+ - if the RX SCR handler is in mode SPEECH, then frames classified as SPEECH\_BAD or NO\_DATA shall be substituted and muted as defined in [5]. Frames classified as NO\_DATA shall be handled like SPEECH\_BAD frames without valid speech information;
400
+
401
+ - frames classified as SID\_FIRST, SID\_UPDATE or SID\_BAD shall bring the RX SCR handler into mode COMFORT\_NOISE and shall result in comfort noise generation, as defined in [6]. SID\_BAD frames shall be substituted and muted as defined in [5];;
402
+ - in mode COMFORT\_NOISE the RX SCR handler shall ignore all unusable frames (NO\_DATA, SPEECH\_BAD); comfort noise generation shall continue, until timeout may apply ([5]).
403
+
404
+ ## 5.3 AMR SID Information format
405
+
406
+ The SID\_UPDATE frame format is according to [5]. This is the default and only mandatory operating mode of the SCR handler.
407
+
408
+ # --- Annex A (normative): AMR DTX handler for the GSM system (corresponding to GSM 06.93)
409
+
410
+ ## A.1 Scope
411
+
412
+ The present document gives a description of the general baseband operation of Adaptive Multi-Rate speech traffic channels in the transmitter and in the receiver of GSM Mobile Stations (MS)s and Base Station Systems (BSS)s during Discontinuous Transmission (DTX).
413
+
414
+ For clarity, the description is structured according to the block diagrams in figures 1 and 3. Except in the case described next, this structure of distributing the various functions between system entities is not mandatory for implementation, as long as the operation on the air interface and on the speech decoder output remains the same.
415
+
416
+ In the case of BSSs where the speech transcoder is located remote from the Base Transceiver Station (BTS), the implementation of the interfaces between the DTX handlers and the Radio Sub System (RSS) as described in the present document together with all their flags is mandatory, being part of the A-bis interface as described in GSM 08.60 and GSM 08.61.
417
+
418
+ The DTX functions described in this technical specification are mandatory for implementation in the GSM MSs. The receiver requirements are mandatory for implementation in all GSM BSSs, the transmitter requirements only for those where downlink DTX or Tandem Free Operation will be used.
419
+
420
+ ## --- A.2 References
421
+
422
+ The following documents contain provisions which, through reference in this text, constitute provisions of the present document.
423
+
424
+ - References are either specific (identified by date of publication, edition number, version number, etc.) or non-specific.
425
+ - For a specific reference, subsequent revisions do not apply.
426
+ - 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*.
427
+
428
+ - [1] 3GPP TR 21.905:"Vocabulary for 3GPP Specifications".
429
+ - [2] 3GPP TS 24.008: "Digital cellular telecommunication system (Phase 2+); Mobile radio interface layer 3 specification".
430
+ - [3] 3GPP TS 45.003: "Digital cellular telecommunication system (Phase 2+); Channel coding".
431
+ - [4] 3GPP TS 45.005: "Digital cellular telecommunication system (Phase 2+); Radio transmission and reception".
432
+ - [5] 3GPP TS 45.008: "Digital cellular telecommunication system (Phase 2+); Radio subsystem link control".
433
+ - [6] 3GPP TS 45.009: "Digital cellular telecommunication system (Phase 2+); Link adaptation".
434
+ - [7] 3GPP TS 26.071: "Digital cellular telecommunications system (Phase 2+); Adaptive Multi-Rate (AMR) speech processing functions; General description".
435
+ - [8] 3GPP TS 26.073: "Digital cellular telecommunications system (Phase 2+); ANSI-C code for the GSM Adaptive Multi-Rate speech codec".
436
+ - [9] 3GPP TS 26.074: "Digital cellular telecommunications system (Phase 2); Test vectors for the GSM Adaptive Multi-Rate speech codec".
437
+
438
+ - [10] 3GPP TS 26.090: "Digital cellular telecommunications system (Phase 2+); Adaptive Multi-Rate speech transcoding".
439
+ - [11] 3GPP TS 26.091: "Digital cellular telecommunications system (Phase 2+); Substitution and muting of lost frame for Adaptive Multi-Rate speech traffic channels".
440
+ - [12] 3GPP TS 26.092: "Digital cellular telecommunications system (Phase 2+); Comfort noise aspects for Adaptive Multi-Rate speech traffic channels".
441
+ - [13] 3GPP TS 26.094: "Digital cellular telecommunications system (Phase 2+); Voice Activity Detector (VAD) for Adaptive Multi-Rate speech traffic channels".
442
+ - [14] 3GPP TS 28.060: "Digital cellular telecommunication system (Phase 2+); Inband control of remote transcoders and rate adaptors for Full Rate traffic channels".
443
+ - [15] 3GPP TS 28.061: "Digital cellular telecommunication system (Phase 2+); Inband Control of Remote Transcoders and Rate Adaptors for Half Rate traffic channels".
444
+ - [16] 3GPP TS 28.062: " Digital cellular telecommunications system; Inband Tandem Free Operation (TFO) of Speech Codecs".
445
+
446
+ ## --- A.3 Definitions, symbols and abbreviations
447
+
448
+ ### A.3.1 Definitions
449
+
450
+ For the purpose of the present document, the following definitions apply.
451
+
452
+ **frame:** Time interval of 20 ms, corresponding to the time segmentation of the Adaptive Multi Rate speech transcoder (3GPP TS 26.090 [9]), also used as a short term for a traffic frame.
453
+
454
+ **traffic frame:** Block of 95..244 information bits transmitted on the TCH/AFS or TCH/AHS speech traffic channels.
455
+
456
+ **SID frame:** Frame characterised by the SID (Silence Descriptor) gross bit patterns. It may convey information on the acoustic background noise.
457
+
458
+ **speech frame:** Traffic frame that has been classified as a SPEECH frame.
459
+
460
+ **VAD flag:** Boolean flag, generated by the VAD algorithm defined in 3GPP TS 26.094 indicating the presence ("1") or the absence ("0") of a speech frame.
461
+
462
+ **RX\_TYPE:** flag with eight values, generated by the RX radio subsystem, indicating to the RX DTX handler the type of data in the current frame. Refer to Table 2.
463
+
464
+ **TX\_TYPE:** flag with eight values, generated by the TX DTX handler, indicating to the TX radio subsystem the type of data in the current frame. Refer to Table 1.
465
+
466
+ **hangover period:** A period of 7 frames added at the end of a speech burst in which VAD flag ="0" and TX\_TYPE is "SPEECH".
467
+
468
+ ### A.3.2 Symbols
469
+
470
+ For the purpose of the present document, the following symbols apply.
471
+
472
+ | | |
473
+ |----------------------|------------------------------------------------------------|
474
+ | $N_{\text{elapsed}}$ | Number of elapsed frames since the last updated SID frame. |
475
+ |----------------------|------------------------------------------------------------|
476
+
477
+ ### A.3.3 Abbreviations
478
+
479
+ For the purpose of the present document, the following abbreviations apply.
480
+
481
+ | | |
482
+ |-----|--------------------------|
483
+ | BSC | Base Station Controller |
484
+ | BSS | Base Station System |
485
+ | BTS | Base Transceiver Station |
486
+ | CHD | Channel Decoder |
487
+
488
+ | | |
489
+ |---------|-------------------------------------------------|
490
+ | CHE | Channel Encoder |
491
+ | DTX | Discontinuous Transmission |
492
+ | ETS | European Telecommunication Standard |
493
+ | FACCH | Fast Associated Control CHannel |
494
+ | GSM | Global System for Mobile Telecommunications |
495
+ | MS | Mobile Station |
496
+ | RATSCCH | Robust Amr Traffic Synchronised Control CHannel |
497
+ | RSS | Radio Sub System |
498
+ | RX | Receive |
499
+ | SACCH | Slow Associated Control CHannel |
500
+ | SID | SIlence Descriptor |
501
+ | TX | Transmit |
502
+ | VAD | Voice Activity Detector |
503
+
504
+ For abbreviations not given in this subclause, see 3GPP TS 21.004.
505
+
506
+ ## --- A.4 General
507
+
508
+ Discontinuous Transmission (DTX) is a mechanism, which allows the radio transmitter to be switched off most of the time during speech pauses for the following two purposes:
509
+
510
+ to save power in the Mobile Station (MS);
511
+
512
+ to reduce the overall interference level over the air interface.
513
+
514
+ DTX in uplink shall be in operation within the GSM MS, if commanded so by the network, see 3GPP TS 204.08. The MS shall handle DTX in downlink at any time, regardless, whether DTX in uplink is commanded or not.
515
+
516
+ ### A.4.1 General organisation
517
+
518
+ The overall DTX mechanism described in the present document requires the following functions:
519
+
520
+ a Voice Activity Detector (VAD) on the transmit (TX) side;
521
+
522
+ evaluation of the background acoustic noise on the transmit (TX) side, in order to transmit characteristic parameters to the receive (RX) side;
523
+
524
+ generation on the receive (RX) side of a similar noise, called comfort noise, during periods where the radio transmission is switched off.
525
+
526
+ The Voice Activity Detector (VAD) is defined in 3GPP TS 26.094 and the comfort noise functions in 3GPP TS 26.092. Both are based partly on the speech transcoder and its internal variables, defined in 3GPP TS 26.090.
527
+
528
+ In addition to these functions, if the parameters arriving at the RX side are detected to be seriously corrupted by errors, the speech or comfort noise must be generated from substituted data in order to avoid seriously annoying effects for the listener. This function is defined in 3GPP TS 26.091.
529
+
530
+ An overall description of the speech processing parts can be found in 3GPP TS 26.071.
531
+
532
+ The description for Tandem Free Operation is given in 3GPP TS 28.062.
533
+
534
+ ## --- A.5 Transmit (TX) side
535
+
536
+ A block diagram of the transmit side DTX functions is shown in figure 5.
537
+
538
+ ![Figure 5: Block diagram of the transmit side DTX functions. The diagram shows two main components: TX DTX handler and TX Radio subsystem. The TX DTX handler contains three sub-blocks: Speech encoder, Voice Activity Detector, and Comfort Noise Computation. The TX Radio subsystem contains two sub-blocks: Channel Encoder and TX_TYPE Monitoring. Arrows indicate the flow of data: Information bits from Speech encoder to Channel Encoder; Mode Indication from Voice Activity Detector to Channel Encoder; TX_TYPE from Comfort Noise Computation to TX_TYPE Monitoring; and NSYNC from TX_TYPE Monitoring back to Comfort Noise Computation.](1a827b10290f33d4fec04d0e8ef7a897_img.jpg)
539
+
540
+ ```
541
+
542
+ graph LR
543
+ subgraph TX_DTX_handler [TX DTX handler]
544
+ SE[Speech encoder]
545
+ VAD[Voice Activity Detector]
546
+ CNC[Comfort Noise Computation]
547
+ end
548
+ subgraph TX_Radio_subsystem [TX Radio subsystem]
549
+ CE[Channel Encoder]
550
+ TM[TX_TYPE Monitoring]
551
+ end
552
+ SE -- "Information bits" --> CE
553
+ VAD -- "Mode Indication" --> CE
554
+ CNC -- "TX_TYPE" --> TM
555
+ TM -- "NSYNC" --> CNC
556
+
557
+ ```
558
+
559
+ Figure 5: Block diagram of the transmit side DTX functions. The diagram shows two main components: TX DTX handler and TX Radio subsystem. The TX DTX handler contains three sub-blocks: Speech encoder, Voice Activity Detector, and Comfort Noise Computation. The TX Radio subsystem contains two sub-blocks: Channel Encoder and TX\_TYPE Monitoring. Arrows indicate the flow of data: Information bits from Speech encoder to Channel Encoder; Mode Indication from Voice Activity Detector to Channel Encoder; TX\_TYPE from Comfort Noise Computation to TX\_TYPE Monitoring; and NSYNC from TX\_TYPE Monitoring back to Comfort Noise Computation.
560
+
561
+ Figure 5: Block diagram of the transmit side DTX functions
562
+
563
+ ### A.5.1 General operation
564
+
565
+ The TX DTX handler passes traffic frames, individually marked by TX\_TYPE, to the Radio Subsystem (RSS). Each frame passed to the RSS consists of bit fields containing the information bits, the codec mode indication, and the TX\_TYPE. TX\_TYPE is used to specify the contents of the frame. The table below provides an overview of the different TX\_TYPEs used and explains the required contents in the information bit and the mode indication bit fields. In case of ongoing Tandem Free Operation (see 3GPP TS 28.062) frames with errors may arrive in downlink in the BTS.
566
+
567
+ Table 3: TX TYPE identifiers
568
+
569
+ | TX_TYPE Legend | Information Bits | Mode Indication |
570
+ |-----------------------------------------------------|----------------------------------------------------------------------------------------------------------------|---------------------------------------------------------------------|
571
+ | <b>SPEECH_GOOD</b> | speech frame, size 95..244 bits depending on codec mode; no errors known. | current code mode |
572
+ | <b>SPEECH_DEGRADED</b><br>(only in downlink in TFO) | Speech frame, size 95..244 bits, depending on codec mode; there might be errors in class 2 bits. | current codec mode |
573
+ | <b>SPEECH_BAD</b><br>(only in downlink in TFO) | Speech frame, size 95..244 bits, depending on codec mode; there are errors in class 1 bits. | current codec mode |
574
+ | <b>SID_FIRST</b> | marks the end of a talkspurt, respectively the beginning of a speech pause; does not contain information bits. | the codec mode that would have been used if TX_TYPE had been SPEECH |
575
+ | <b>SID_UPDATE</b> | comfort noise, 35 bits; no errors known | the codec mode that would have been used if TX_TYPE had been SPEECH |
576
+ | <b>SID_BAD</b><br>(only in downlink in TFO) | comfort noise, 35 bits; errors detected, parameters unusable | the codec mode that would have been used if TX_TYPE had been SPEECH |
577
+ | <b>ONSET</b><br>(only in downlink in TFO) | announces the beginning of a speech burst; does not contain information bits | the codec mode of the following speech frame |
578
+ | <b>NO_DATA</b> | no useful information | no useful information |
579
+
580
+ TX\_TYPE = “NO\_DATA” indicates that the Information Bit and Codec Mode fields do not contain any useful data (and shall not be transmitted over the air interface). The purpose of this TX\_TYPE is to provide the option to save transmission between the transcoder and the radio base station if a packet oriented transmission is used.
581
+
582
+ The scheduling of the frames for transmission on the air interface is controlled by the TX DTX handler by the use of the TX\_TYPE field.
583
+
584
+ #### A.5.1.1 Functions of the TX DTX handler
585
+
586
+ To allow an exact verification of the TX DTX handler functions, all frames before the reset of the system are treated as if there were speech frames of an infinitely long time. Therefore, and in order to ensure the correct estimation of comfort noise parameters at RX DTX side, the first 7 frames after the reset or after enabling the DTX operation shall always be marked with TX\_TYPE= " SPEECH\_GOOD ", even if VAD flag ="0" (hangover period, see figure 2).
587
+
588
+ The Voice Activity Detector (VAD) shall operate all the time in order to assess whether the input signal contains speech or not. The output is a binary flag (VAD flag = "1" or VAD flag = "0", respectively) on a frame by frame basis (see 3GPP TS 26.094).
589
+
590
+ The VAD flag controls indirectly, via the TX DTX handler operations described below, the overall DTX operation on the transmit side.
591
+
592
+ Whenever VAD flag = "1", the speech encoder output frame along with mode information shall be passed directly to the radio subsystem (RSS), marked with TX\_TYPE = " SPEECH\_GOOD "
593
+
594
+ At the end of a speech burst (transition VAD flag = "1" to VAD flag = "0"), it takes 8 consecutive frames to make a new updated SID analysis available at receiver side (see 3GPP TS 26.092). Normally, the first 7 speech encoder output frames after the end of the speech burst shall therefore be passed directly to the RSS, marked with TX\_TYPE = " SPEECH\_GOOD " ("hangover period"). The end of the speech is then indicated by passing frame 8 after the end of the speech burst to the RSS, marked with TX\_TYPE = "SID\_FIRST" (see figure 2).
595
+
596
+ ![Figure 6: Normal hangover procedure (N_elapsed > 23). The diagram shows a timeline of VAD flag states and a corresponding table of transmitted frames. The VAD flag is high (speech) until the 'last speech frame', then drops to low (pause) at the 'end of speech burst'. A 'Hangover' period of 8 frames follows, starting with the 'first pause frame'. The 'end of speech burst' occurs at the start of the 8th frame of the hangover. Below the timeline, a table shows frames 35 to 45, 0, and 1. Frames 35-42 are 'S' (Speech), frame 43 is 'F' (SID_FIRST), frames 44-45 are 'N' (No Data), frame 0 is 'U' (SID_UPDATE), and frame 1 is 'N' (No Data). An arrow points from the 'last speech frame' label to the start of the hangover period. Another arrow points from the 'end of speech burst' label to the start of frame 43. A third arrow points from the 'first pause frame' label to the start of frame 36. A fourth arrow points from the 'Hangover' label to the period from frame 36 to 42. A fifth arrow points from the 'Frame (20 ms)' label to a single frame interval. A sixth arrow points from the 'TX Type' label to the 'S' in frame 35.](36117e9cb27c58484cd8d3e5f9dc7ac3_img.jpg)
597
+
598
+ | | | | | | | | | | | | | | |
599
+ |---------------------------|---------|----|----|----|----|----|----|----|----|----|----|---|---|
600
+ | N <sub>elapsed</sub> e.g. | 35 | 36 | 37 | 38 | 39 | 40 | 41 | 42 | 43 | 44 | 45 | 0 | 1 |
601
+ | | S | S | S | S | S | S | S | S | F | N | N | U | N |
602
+ | Frames to RSS | TX Type | | | | | | | | | | | | |
603
+
604
+ Figure 6: Normal hangover procedure (N\_elapsed > 23). The diagram shows a timeline of VAD flag states and a corresponding table of transmitted frames. The VAD flag is high (speech) until the 'last speech frame', then drops to low (pause) at the 'end of speech burst'. A 'Hangover' period of 8 frames follows, starting with the 'first pause frame'. The 'end of speech burst' occurs at the start of the 8th frame of the hangover. Below the timeline, a table shows frames 35 to 45, 0, and 1. Frames 35-42 are 'S' (Speech), frame 43 is 'F' (SID\_FIRST), frames 44-45 are 'N' (No Data), frame 0 is 'U' (SID\_UPDATE), and frame 1 is 'N' (No Data). An arrow points from the 'last speech frame' label to the start of the hangover period. Another arrow points from the 'end of speech burst' label to the start of frame 43. A third arrow points from the 'first pause frame' label to the start of frame 36. A fourth arrow points from the 'Hangover' label to the period from frame 36 to 42. A fifth arrow points from the 'Frame (20 ms)' label to a single frame interval. A sixth arrow points from the 'TX Type' label to the 'S' in frame 35.
605
+
606
+ TX Types: "S" = SPEECH; "F" = SID\_FIRST; "U" = "SID\_UPDATE"; "N" = NO DATA
607
+ N<sub>elapsed</sub>: No. of elapsed frames since last SID\_UPDATE
608
+
609
+ **Figure 6: Normal hangover procedure (N<sub>elapsed</sub> > 23)**
610
+
611
+ If, however, at the end of the speech burst, less than 24 frames have elapsed since the last SID\_UPDATE frame was computed and passed to the RSS, then this last analysed SID\_UPDATE frame shall repeatedly be passed to the RSS whenever a SID\_UPDATE frame is to be produced, until a new updated SID analysis is available (8 consecutive frames marked with VAD flag = "0"). This reduces the activity on the air in cases where short background noise spikes are taken for speech, by avoiding the "hangover" waiting for the SID frame computation.
612
+
613
+ Once the first SID analysis after the end of a speech burst has been computed and the SID\_FIRST frame has been passed to the Radio Subsystem, the TX DTX handler shall at regular intervals compute and pass updated SID\_UPDATE (Comfort Noise) frames to the Radio Subsystem (RSS) as long as VAD flag = "0". SID\_UPDATE frames shall be generated every 8<sup>th</sup> frame. The first SID\_UPDATE shall be sent as the third frame after the SID\_FIRST frame.
614
+
615
+ The speech encoder is operated in full speech modality if TX\_TYPE = " SPEECH\_GOOD " and otherwise in a simplified mode, because not all encoder functions are required for the evaluation of comfort noise parameters and because comfort noise parameters are only to be generated at certain times.
616
+
617
+ In order to ensure TX/RX DTX handler synchronisation at handover, the uplink TX DTX handler in the MS shall accept messages from TX RSS with control parameter NSYNC, resulting in the following operation during a period of the next NSYNC frames:
618
+
619
+ - The TX DTX handler shall send SID\_UPDATE instead of NO\_DATA frames to the TX RSS.
620
+ - If, however, during this period of NSYNC frames, VAD flag is equal to 1 at least for one speech frame, TX DTX handler shall send SPEECH frames for the rest of the period of the NSYNC frames. Note the TX DTX handler shall send SPEECH frames at least for the duration of the hangover period.
621
+
622
+ #### A.5.1.2 Functions of the TX Radio Subsystem
623
+
624
+ The TX Radio Subsystem has the following overall functionality. The radio transmission is cut after the transmission of a SID\_FIRST frame when the speaker stops talking. During speech pauses the transmission is resumed at regular intervals for transmission of one SID\_UPDATE frame, in order to update the generated comfort noise on the RX side (and to improve the measurement of the link quality by the RSS). Note that the transcoder knows what frames to send. In the case when nothing is to be transmitted it outputs frames marked with TX\_TYPE = "NO\_DATA".
625
+
626
+ Within the TX Radio Subsystem the TX\_TYPE Monitoring unit controls the operation of the Channel Encoder (as specified in 3GPP TS 25.003) and the Transmission of the frame. Control input to the TX\_TYPE Monitoring unit is the TX\_TYPE. Control output and input to the Channel Encoder are indicators specifying the frame format. These frame format indicators are defined in 3GPP TS 25.003, they are different for TCH/AFS and TCH/AHS.
627
+
628
+ ##### A.5.1.2.1 Functions of the TX Radio Subsystem for TCH/AFS
629
+
630
+ The TX Radio Subsystem operates in the following way regarding DTX (without TFO):
631
+
632
+ - all frames marked with TX\_TYPE = " SPEECH\_GOOD " are scheduled for normal channel coding and transmission. The frame format for CHE operation shall be SPEECH. If, however, the previous frame was not of TX\_TYPE = "SPEECH\_GOOD", an ONSET frame format followed by SPEECH\_GOOD shall be signalled to the CHE;
633
+ - for frames marked with TX\_TYPE = "SID\_FIRST" a SID\_FIRST frame format is signalled to the CHE;
634
+ - frames marked with TX\_TYPE = "SID\_UPDATE" are scheduled for SID\_UPDATE frame channel coding and transmission. The frame format signalled to CHE is SID\_UPDATE;
635
+ - for frames marked with TX\_TYPE = "NO\_DATA" no processing or transmission is carried out.
636
+
637
+ If a FACCH needs to be sent during a speech pause, and:
638
+
639
+ - if the frame preceding the FACCH is not of TX\_TYPE="SPEECH\_GOOD", then an ONSET frame shall be signalled to the CHE, followed by the FACCH frame(s);
640
+ - if the frame following the FACCH frame is not of TX\_TYPE="SPEECH\_GOOD", then a SID\_FIRST shall be signalled to the CHE.
641
+
642
+ A stolen SID\_UPDATE should be rescheduled on the frame subsequent to the SID\_FIRST marker that follows the FACCH frame(s) (when that marker exists), unless that rescheduling would steal a frame of TX\_TYPE="SPEECH\_GOOD", or if a frame of TX\_TYPE="SID\_UPDATE" has been received before the rescheduling could take place. This rescheduling shall not affect the timing of subsequent SID\_UPDATE frames.
643
+
644
+ Note: a SID\_UPDATE frame is considered as stolen when this frame must be replaced by a FACCH frame.
645
+
646
+ At handover, TX/RX DTX handler synchronisation shall be initiated. At the time instant before the MS starts sending to the new base station, a message shall be sent to the uplink TX DTX handler with the parameter NSYNC = 12.
647
+
648
+ ##### A.5.1.2.2 Functions of the TX Radio Subsystem for TCH/AHS
649
+
650
+ The TX Radio Subsystem operates in the following way regarding DTX:
651
+
652
+ all frames marked with TX\_TYPE = " SPEECH\_GOOD " are scheduled for normal channel coding and transmission. The frame format for CHE operation shall be SPEECH. However, if the previous frame was of TX\_TYPE = "SID\_FIRST", a SID\_FIRST\_INH frame format followed by SPEECH\_GOOD shall be signalled to the CHE. If the previous frame was of TX\_TYPE = "SID\_UPDATE", a SID\_UPDATE\_INH frame format followed by SPEECH\_GOOD shall be signalled to the CHE. If the previous frame was of TX\_TYPE = "NO\_DATA", an ONSET frame format followed by SPEECH\_GOOD shall be signalled to the CHE;
653
+
654
+ for frames marked with TX\_TYPE = "SID\_FIRST" a SID\_FIRST\_P1 frame format is signalled to the CHE. Note: All 4 TDMA frames carrying the bits of this frame shall be transmitted. The Mode Indication received with the frame is stored for potential use in the next frame;
655
+
656
+ for frames marked with TX\_TYPE = "SID\_UPDATE" a SID\_UPDATE frame format is signalled to the CHE. All 4 TDMA frames carrying the bits of this frame shall be transmitted;
657
+
658
+ for frames marked with TX\_TYPE = "NO\_DATA", no processing or transmission is carried out. However, if the preceding frame was marked with TX\_TYPE = "SID\_FIRST", a SID\_FIRST\_P2 frame format is signalled to CHE. Note: The 2 TDMA frames carrying bits of this frame shall be transmitted. If, depending on the current frame number, the Mode Indication is to be transmitted with these TDMA frames, the Mode Indication shall be used that was stored during the processing of the preceding SID\_FIRST frame.
659
+
660
+ If a FACCH needs to be sent during a speech pause, and:
661
+
662
+ if the frame preceding the FACCH is of TX\_TYPE="SID\_FIRST", then a SID\_FIRST\_INH frame format followed by the FACCH shall be signalled to the CHE;
663
+
664
+ if the frame preceding the FACCH is of TX\_TYPE="SID\_UPDATE", then a SID\_UPDATE\_INH frame format followed by the FACCH shall be signalled to the CHE;
665
+
666
+ if the frame preceding the FACCH is of TX\_TYPE="NO\_DATA", then an ONSET frame format followed by the FACCH shall be signalled to the CHE;
667
+
668
+ if the frame following the FACCH is not of TX\_TYPE="SPEECH\_GOOD", then a SID\_FIRST shall be signalled to the CHE. Both SID\_FIRST\_P1 and SID\_FIRST\_P2 frame formats shall be consecutively signalled to CHE.
669
+
670
+ A stolen SID\_UPDATE should be rescheduled on the frame subsequent to the SID\_FIRST\_P1 and SID\_FIRST\_P2 markers that follow the FACCH frame(s) (when these markers exist), unless that rescheduling would steal a frame of TX\_TYPE="SPEECH\_GOOD", or if a frame of TX\_TYPE="SID\_UPDATE" has been received before the rescheduling could take place. This rescheduling shall not affect the timing of subsequent SID\_UPDATE frames.
671
+
672
+ Note: a SID\_UPDATE frame is considered as stolen when this frame must be replaced by a FACCH frame.
673
+
674
+ At handover, TX/RX DTX handler synchronisation shall be initiated. At the time instant before the MS starts sending to the new base station, a message shall be sent to the uplink TX DTX handler with the parameter NSYNC = 12.
675
+
676
+ ##### A.5.1.2.3 Functions of the Downlink TX Radio Subsystem for TFO
677
+
678
+ The TX Radio Subsystem in the BTS shall in addition operate in the following way regarding DTX, if TFO is ongoing (see 3GPP TS 28.062):
679
+
680
+ - Frames with TX\_TYPE = SPEECH\_GOOD, SID\_FIRST and SID\_UPDATE shall be handled as usual in DTX, regardless whether DTX in downlink is requested or not. Also NO\_DATA shall be handled as usual, if DTX is requested.
681
+ - Frames with TX\_TYPE = NO\_DATA shall be replaced by SID\_FILLER frames, if DTX in downlink is not requested. By this the radio transmission continues in downlink, although no parameters are transmitted in speech pauses on the Abis interface. The MS generates Comfort Noise in these speech pauses.
682
+ - Frames with TX\_TYPE = SPEECH\_DEGRADED shall be handled exactly like SPEECH\_GOOD frames.
683
+ - For frame with TX\_TYPE = SPEECH\_BAD and SID\_BAD the CHE shall perform its regular processing, but then shall invert the six, respectively 14 CRC bits before convolutional encoding and transmitting the frames on the air interface. By this the error concealment mechanism in the MS is triggered to handle these corrupted frames.
684
+ - ONSET frames may be ignored by the TX Radio Subsystem and need not to be processed.
685
+
686
+ **Definition:** SID\_FILLER frames are like SID\_BAD frames, but with all information bits set to "1". The 14 CRC bits shall artificially be inverted by the CHE before convolutional encoding and transmission.
687
+
688
+ ##### A.5.1.2.4 Functions of the TX Radio Subsystem for RATSCCH
689
+
690
+ During regular speech transmission (in the middle of a speech burst) RATSCCH replaces (steals) one (TCH/AFS) respectively two (TCH/AHS) speech frames (see 3GPP TS 25.009). Also in all non speech cases the RATSCCH shall be handled like speech. The respective RATSCCH frame formats (RATSCCH in case of TCH/AFS, respectively RATSCCH\_MARKER and RATSCCH\_DATA in case of TCH/AHS) shall be signalled to the CHE.
691
+
692
+ RATSCCH should be handled in the same way as a FACCH.
693
+
694
+ ## A.6 Receive (RX) side
695
+
696
+ A block diagram of the receive side DTX functions is shown in figure 7.
697
+
698
+ ![Block diagram of the receive side DTX functions. The diagram shows two main components: RX DTX handler and RX Radio subsystem. The RX DTX handler contains three sub-blocks: Speech Decoder, Comfort Noise Generation, and Error Concealment. The RX Radio subsystem contains three sub-blocks: Error Correction & Detection, SID frame Detection, and Mode Detection. Arrows indicate data flow: Information bits from Error Correction & Detection to Speech Decoder; Mode Indication from SID frame Detection to Comfort Noise Generation; and RX_TYPE from Mode Detection to Error Concealment.](3da1a07cb87051bf616c9876db958cf0_img.jpg)
699
+
700
+ ```
701
+
702
+ graph LR
703
+ subgraph RX_DTX_handler [RX DTX handler]
704
+ SD[Speech Decoder]
705
+ CNG[Comfort Noise Generation]
706
+ EC[Error Concealment]
707
+ end
708
+ subgraph RX_Radio_subsystem [RX Radio subsystem]
709
+ ECD[Error Correction & Detection]
710
+ SFD[SID frame Detection]
711
+ MD[Mode Detection]
712
+ end
713
+ ECD -- "Information bits" --> SD
714
+ SFD -- "Mode Indication" --> CNG
715
+ MD -- "RX_TYPE" --> EC
716
+
717
+ ```
718
+
719
+ Block diagram of the receive side DTX functions. The diagram shows two main components: RX DTX handler and RX Radio subsystem. The RX DTX handler contains three sub-blocks: Speech Decoder, Comfort Noise Generation, and Error Concealment. The RX Radio subsystem contains three sub-blocks: Error Correction & Detection, SID frame Detection, and Mode Detection. Arrows indicate data flow: Information bits from Error Correction & Detection to Speech Decoder; Mode Indication from SID frame Detection to Comfort Noise Generation; and RX\_TYPE from Mode Detection to Error Concealment.
720
+
721
+ Figure 7: Block diagram of the receive side DTX functions
722
+
723
+ ### A.6.1 General operation
724
+
725
+ Whatever their context (speech, SID, FACCH or none), the RSS continuously passes the received traffic frames to the RX DTX handler, individually marked by various pre-processing functions with RX\_TYPE as described in subclause 6.1.1 and table 2, which serves to classify the traffic frame. This classification allows the RX DTX handler to determine in a simple way how the received frame is to be handled.
726
+
727
+ Table 4: RX\_TYPE identifiers
728
+
729
+ | RX_TYPE Legend | Description |
730
+ |------------------------|-----------------------------------------------------------------------------------------------------------------------------------------------------------------------------|
731
+ | <b>SPEECH_GOOD</b> | Speech frame with CRC OK, Channel Decoder soft values also OK |
732
+ | <b>SPEECH_DEGRADED</b> | Speech frame with CRC OK, but 1B bits and class2 bits may be corrupted |
733
+ | <b>SPEECH_BAD</b> | ((likely) speech frame, bad CRC (or very bad Channel Decoder measures) |
734
+ | <b>SID_FIRST</b> | first SID marks the beginning of a comfort noise period |
735
+ | <b>SID_UPDATE</b> | SID update frame (with correct CRC) |
736
+ | <b>SID_BAD</b> | Corrupt SID update frame (bad CRC; applicable only for SID_UPDATE frames) |
737
+ | <b>ONSET</b> | ONSET frames precede the first speech frame of a speech burst |
738
+ | <b>NO_DATA</b> | Nothing useable (for the speech decoder) was received. This applies for the cases of no received frames (DTX) or received FACCH or RATSCCH or SID_FILLER signalling frames. |
739
+
740
+ #### A.6.1.1 Functions of the RX radio subsystem
741
+
742
+ The RX radio subsystem uses a combination of gross-bit markers, receiver measurements, and CRC checks to classify each received frame. The basic operation for each frame is outlined below:
743
+
744
+ the receiver first searches for the RATSCCH, SID\_UPDATE, SID\_FIRST or ONSET gross bit markers.
745
+
746
+ If the RATSCCH signalling is detected, then the RATSCCH frame (TCH/AFS) respectively the RATSCCH\_MARKER and RATSCCH\_DATA frames (TCH/AHS) shall be decoded and handled as described in 3GPP TS 25.009. They shall be passed to the RX DTX handler as a NO\_DATA frame(s).
747
+
748
+ If the SID\_FIRST marker is detected the frame is passed to the RX DTX handler as a SID\_FIRST frame.
749
+
750
+ If the SID\_UPDATE marker is detected, then the frame shall be decoded and passed to the RX DTX handler as a SID\_UPDATE or a SID\_BAD or a NO\_DATA frame, depending on the CRC and the information bits, along
751
+
752
+ with the comfort noise parameters, if applicable. A NO\_DATA frame shall be passed on, if all information bits of a SID\_UPDATE frame are set to "1" and the CRC is bad (see SID\_FILLER in subclause 5.1.2.3).
753
+
754
+ - If the ONSET marker is detected, then an ONSET frame shall be passed to the RX DTX handler.
755
+
756
+ if neither SID\_UPDATE nor SID\_FIRST markers are detected, the frame shall be channel decoded assuming it to be a speech frame. Depending on the CRC for speech frame channel decoding along with other receiver measurements the frame shall then be passed to the RX DTX handler marked as either SPEECH\_GOOD, SPEECH\_DEGRADED, SPEECH\_BAD or NO\_DATA frame.
757
+
758
+ #### A.6.1.2 Functions of the RX DTX handler
759
+
760
+ The RX DTX handler is responsible for the overall DTX operation on the RX side. It consists of two main modes: SPEECH and COMFORT\_NOISE. The initial mode shall be SPEECH.
761
+
762
+ The DTX operation on the RX side shall be as follows:
763
+
764
+ The RX DTX handler shall enter mode SPEECH, when a frame classified as SPEECH\_GOOD or SPEECH\_DEGRADED is received. ONSET frames may be taken into account to identify the beginning of a speech burst;
765
+
766
+ whenever a frame classified as SPEECH\_GOOD is received the RX DTX handler shall pass it directly on to the speech decoder;
767
+
768
+ if the RX DTX handler is in mode SPEECH, then frames classified as SPEECH\_DEGRADED, SPEECH\_BAD or NO\_DATA shall be substituted and muted as defined in 3GPP TS 26.091. Frames classified as NO\_DATA shall be handled like SPEECH\_BAD frames without valid speech information;
769
+
770
+ frames classified as SID\_FIRST, SID\_UPDATE or SID\_BAD shall bring the RX DTX handler into mode COMFORT\_NOISE and shall result in comfort noise generation, as defined in 3GPP TS 26.092. SID\_BAD frames shall be substituted and muted as defined in 3GPP TS 26.091. In mode COMFORT\_NOISE the RX DTX handler shall ignore all unusable frames (NO\_DATA, SPEECH\_BAD) delivered by the RSS; comfort noise generation shall continue, until timeout may apply (see 3GPP TS 26.091).
771
+
772
+ # --- Annex B (normative): ETSI GSM-EFR SCR handler
773
+
774
+ The interworking operation of AMR (only 12.2 kbit/s mode) with GSM-EFR is F.F.S. This annex provides the basis for the AMR operation under this condition.
775
+
776
+ ---
777
+
778
+ ## Transmit (TX) side
779
+
780
+ Speech coding internetworking aspects with ETSI-EFR are described in [TBD].
781
+
782
+ ### General operation
783
+
784
+ The TX SCR handler passes traffic frames, individually marked by TX\_TYPE, to the TX Framing Unit part of the Access Network. Each frame passed to the AN consists of bit fields containing the information bits, the codec mode indication, and the TX\_TYPE. TX\_TYPE shall be used to specify the contents of the frame. The table below provides an overview of the different TX\_TYPEs used and explains the required contents in the information bit and the mode indication bit fields.
785
+
786
+ **Table 5: TX TYPE identifiers for GSM-EFR**
787
+
788
+ | TX TYPE | Legend | Information Bits | Mode Indication |
789
+ |---------|------------|----------------------------------------------|--------------------------------------------------------------------------|
790
+ | 00 | SPEECH | speech frame, size depending on codec mode | current codec mode |
791
+ | 10 | SID_UPDATE | comfort noise information , information bits | the codec mode that would have been used if TX_TYPE had been 00 (SPEECH) |
792
+ | 11 | NO_DATA | no useful information | no useful information |
793
+
794
+ TX\_TYPE = “NO\_DATA” indicates that the Information Bit and Codec Mode fields do not contain any useful data (and should not be transmitted over AN). The purpose of this TX\_TYPE is to provide the option to save network transmission between the transcoder and AN.
795
+
796
+ The scheduling of the frames for transmission on the Access Network is controlled by the TX SCR handler by the use of the TX\_TYPE field and the given SCR operation mode.
797
+
798
+ ### Functions of the TX SCR handler
799
+
800
+ ## GSM-EFR SCR Timing procedures
801
+
802
+ To allow an exact verification of the TX SCR handler functions, all frames before the reset of the system are treated as if there were speech frames of an infinitely long time.
803
+
804
+ The SID\_UPDATE timing is according to ETSI GSM 06.81.
805
+
806
+ ## The TX part of the AN
807
+
808
+ ## Demands on the TX part of the Access Network
809
+
810
+ The TX part of the AN operates in the following way regarding SCR:
811
+
812
+ - frames marked with TX\_TYPE = (SPEECH) are scheduled for transmission.
813
+ - frames marked with TX\_TYPE = (SID\_UPDATE) are scheduled for transmission
814
+ - for frames marked with TX\_TYPE = (NO\_DATA) no processing or transmission is carried out.
815
+
816
+ SPEECH frames shall override other frames in these exceptional cases.
817
+
818
+ ## Receive (RX) side
819
+
820
+ Whatever their context (speech, SID, or none), the deframing unit after AN continuously passes the received traffic frames to the RX SCR handler, individually marked by various pre-processing functions with a 3 bit type indicator RX\_TYPE described in Table 4, which serve to classify the traffic frame. This classification allows the RX SCR handler to determine in a simple way how the received frame is to be handled.
821
+
822
+ **Table 6: RX\_TYPE identifiers for GSM-EFR**
823
+
824
+ | <b>RX TYPE</b> | <b>Legend</b> | <b>Description</b> |
825
+ |----------------|--------------------|----------------------------------------------------------------------------------------|
826
+ | 000 | <b>SPEECH_GOOD</b> | Speech frame with CRC OK, soft values in the RX part of AN also OK |
827
+ | 001 | <b>SPARE</b> | Spare |
828
+ | 010 | <b>SPEECH_BAD</b> | (likely) speech frame, bad CRC (or estimated to be very bad by the RX part of the AN ) |
829
+ | 011 | <b>SPARE</b> | Spare |
830
+ | 100 | <b>SPARE</b> | Spare |
831
+ | 101 | <b>SID_UPDATE</b> | Correct SID update frame |
832
+ | 110 | <b>SID_BAD</b> | Corrupt SID update frame (bad CRC ; applicable only for SID_UPDATE frames) |
833
+ | 111 | <b>NO_DATA</b> | Nothing useable was received. The synthesis mode of the previous frame type is used. |
834
+
835
+ ## SID Information format
836
+
837
+ When the TX SCR handler is ordered by the network to operate in ETSI GSM-EFR mode with SCR operation turned on the SID\_UPDATE frame format is according to ETSI GSM 06.62. (38 +5 bits ).
838
+
839
+ # --- Annex C (normative): TIA IS-641 SCR Handler
840
+
841
+ The interworking operation of AMR (only 7.4 kbit/s mode) with IS-641 (TDMA-EFR) is F.F.S. This annex provides the basis for the AMR operation under this condition.
842
+
843
+ ## --- TX-side
844
+
845
+ F.F.S
846
+
847
+ ## --- RX-side
848
+
849
+ F.F.S
850
+
851
+ ## --- SID Information format
852
+
853
+ When the TX SCR handler is ordered by the network to operate in IS-641 mode with SCR operation turned on the SID\_UPDATE frame format is according to IS-641-A Chapter 6 The SID\_UPDATE timing is according to IS 641-A.
854
+
855
+ Speech coding internetworking aspects with IS-641 are described in [TBD].
856
+
857
+ # --- Annex D (normative): TIA TDMA-US1 SCR Handler
858
+
859
+ The interworking operation of AMR (only 12.2 kbit/s mode) with TDMA-US1 is F.F.S. This annex provides the basis for the AMR operation under this condition.
860
+
861
+ ## --- TX-side
862
+
863
+ F.F.S
864
+
865
+ ## --- RX-side
866
+
867
+ F.F.S
868
+
869
+ ## --- SID Information format
870
+
871
+ F.F.S.
872
+
873
+ # --- Annex E (normative): ARIB PDC-EFR SCR Handler
874
+
875
+ The interworking operation of AMR (only 6.7 kbit/s mode) with PDC-EFR-6.7 is F.F.S. This annex provides the basis for the AMR operation under this condition.
876
+
877
+ ---
878
+
879
+ ## Transmit (TX) side
880
+
881
+ ### General operation
882
+
883
+ The TX SCR handler passes traffic frames, individually marked by TX\_TYPE, to the TX Framing Unit part of the Access Network. Each frame passed to the AN consists of bit fields containing the information bits, the codec mode indication, and the TX\_TYPE. TX\_TYPE shall be used to specify the contents of the frame. The table below provides an overview of the different TX\_TYPEs used and explains the required contents in the information bit and the mode indication bit fields.
884
+
885
+ **Table 7: TX TYPE identifiers for PDC\_EFR**
886
+
887
+ | TX TYPE | Legend | Information Bits | Mode Indication |
888
+ |---------|-----------------------------------------------------------------|----------------------------------------------|--------------------------------------------------------------------------|
889
+ | 000 | SPEECH | speech frame, size depending on codec mode | current code mode |
890
+ | 010 | POST1 | comfort noise information , information bits | the codec mode that would have been used if TX_TYPE had been 00 (SPEECH) |
891
+ | 011 | NO_DATA | no useful information | no useful information |
892
+ | 001 | POST0<br><br>(end of speech marker,<br>start of CN generation ) | no useful information, | the codec mode that would have been used if TX_TYPE had been 00 (SPEECH) |
893
+ | 100 | PRE<br><br>(end of CN marker, start<br>of speech generation ) | no useful information, | the codec mode that would have been used if TX_TYPE had been 00 (SPEECH) |
894
+
895
+ TX\_TYPE = “NO\_DATA” indicates that the Information Bit and Codec Mode fields do not contain any useful data (and should not be transmitted over AN). The purpose of this TX\_TYPE is to provide the option to save network transmission between the transcoder and AN.
896
+
897
+ The scheduling of the frames for transmission on the Access Network is controlled by the TX SCR handler by the use of the TX\_TYPE field and the given SCR operation mode.
898
+
899
+ Speech coding internetworking aspects with PDC-EFR are described in XX.YY.D.[TBD]
900
+
901
+ ### Functions of the TX SCR handler
902
+
903
+ ## PDC\_EFR SCR Timing procedures
904
+
905
+ To allow an exact verification of the TX SCR handler functions, all frames before the reset of the system are treated as if there were speech frames of an infinitely long time.
906
+
907
+ SID-frame timing is according to ARIB XXX.
908
+
909
+ ## The TX part of the AN
910
+
911
+ ## Demands on the TX part of the Access Network
912
+
913
+ The TX part of the AN operates in the following way regarding SCR:
914
+
915
+ - frames marked with TX\_TYPE = (SPEECH) are scheduled for transmission.
916
+ - frames marked with TX\_TYPE = (POST0) are scheduled for transmission.
917
+ - frames marked with TX\_TYPE = (POST1) are scheduled for transmission
918
+ - frames marked with TX\_TYPE = (PRE) are scheduled for transmission
919
+ - for frames marked with TX\_TYPE = (NO\_DATA) no processing or transmission is carried out.
920
+
921
+ SPEECH frames shall override other frames in these exceptional cases.
922
+
923
+ ## Receive (RX) side
924
+
925
+ Whatever their context (speech, SID, or none), the deframing unit after AN continuously passes the received traffic frames to the RX SCR handler, individually marked by various pre-processing functions with a 3 bit type indicator RX\_TYPE described in Table 6, which serve to classify the traffic frame. This classification allows the RX SCR handler to determine in a simple way how the received frame is to be handled.
926
+
927
+ **Table 8: RX\_TYPE identifiers for PDC\_EFR**
928
+
929
+ | <b>RX TYPE</b> | <b>Legend</b> | <b>Description</b> |
930
+ |----------------|--------------------|----------------------------------------------------------------------------------------|
931
+ | 000 | <b>SPEECH_GOOD</b> | Speech frame with CRC OK, soft values in the RX part of AN also OK |
932
+ | 001 | <b>PRE</b> | Next frame should be a Speech frame |
933
+ | 010 | <b>SPEECH_BAD</b> | (likely) speech frame, bad CRC (or estimated to be very bad by the RX part of the AN ) |
934
+ | 011 | <b>SPARE</b> | Spare |
935
+ | 100 | <b>POST0</b> | This SID-frame marks the beginning of a comfort noise period. |
936
+ | 101 | <b>POST1</b> | Correct SID update frame |
937
+ | 110 | <b>POST1_BAD</b> | Corrupt SID update frame (bad CRC; applicable only for POST1 frames) |
938
+ | 111 | <b>NO_DATA</b> | Nothing useable was received. The synthesis mode of the previous frame type is used. |
939
+
940
+ ## SID information format
941
+
942
+ When the TX SCR Handler is ordered by the network to operate in PDC-EFR mode with SCR turned on the SID\_UPDATE frame format is according to ARIB TBD (134 bits).
943
+
944
+ # Annex F (informative): Change history
945
+
946
+ | History | | | | | | | | | |
947
+ |----------------|---------------------------|-----------|------|-----|-----|-----------------------------------------------------------------------------------------------------------------------|--------|--------|-------|
948
+ | 1999-06 | Approved at S#4 Plenary | | | | | | | | 3.0.0 |
949
+ | 1999-08 | Reformatted in 3GPP style | | | | | | | | 3.0.1 |
950
+ | Change history | | | | | | | | | |
951
+ | Date | TSG # | TSG Doc. | CR | Rev | Cat | Subject/Comment | Old | New | |
952
+ | - | - | SP-99570 | 0001 | 2 | - | Alignment to GSM 06.93 | 3.0.1 | 3.1.0 | |
953
+ | - | - | SP-00262 | 0002 | | - | Re-scheduling of stolen SID_UPDATE Frames for AMR | 3.1.0 | 3.2.0 | |
954
+ | - | - | SP-000574 | 0003 | 1 | - | Re-scheduling of stolen SID_UPDATE frames for AMR (Part 2) | 3.2.0 | 3.3.0 | |
955
+ | - | - | SP-000574 | 0005 | | - | Re-scheduling of stolen SID_UPDATE frames by SID_FIRST frames for AMR Clarification of Hangover period after Handover | 3.2.0 | 3.3.0 | |
956
+ | - | - | SP-000574 | 0004 | | - | Re-scheduling of stolen SID_UPDATE frames for AMR (Part 2) | 3.2.0 | 4.0.0 | |
957
+ | - | - | SP-000574 | 0006 | | - | Re-scheduling of stolen SID_UPDATE frames by SID_FIRST frames for AMR Clarification of Hangover period after Handover | 3.2.0 | 4.0.0 | |
958
+ | - | - | | | | - | Version for Release 5 | 4.0.0 | 5.0.0 | |
959
+ | - | - | SP-020436 | 0009 | | - | Correction of Codec Type Names | 5.0.0 | 5.1.0 | |
960
+ | - | - | SP-020688 | 0010 | 3 | - | Correction of uplink SCR operation activation for UMTS AMR | 5.1.0 | 5.2.0 | |
961
+ | - | - | SP-030086 | 0011 | | - | Handling of FACCH and RATSCCH during AMR DTX | 5.2.0 | 6.0.0 | |
962
+ | - | - | SP-060356 | 0012 | 2 | - | Correction of references | 6.0.0 | 6.1.0 | |
963
+ | - | - | - | - | - | - | Version for Release 7 | 6.1.0 | 7.0.0 | |
964
+ | 2008-12 | SP-42 | - | - | - | - | Version for Release 8 | 7.0.0 | 8.0.0 | |
965
+ | 2009-12 | SP-46 | | | | | Version for Release 9 | 8.0.0 | 9.0.0 | |
966
+ | 2011-03 | SP-51 | | | | | Version for Release 10 | 9.0.0 | 10.0.0 | |
967
+ | 2012-09 | SP-57 | | | | | Version for Release 11 | 10.0.0 | 11.0.0 | |
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1
+
2
+
3
+
4
+
5
+
6
+
7
+ # Contents
8
+
9
+ | | |
10
+ |----------------------------------------------------------------|----|
11
+ | Foreword ..... | 4 |
12
+ | 1 Scope..... | 5 |
13
+ | 2 References..... | 5 |
14
+ | 3 Technical Description of VAD Option 1 ..... | 5 |
15
+ | 3.1 Definitions, symbols and abbreviations ..... | 5 |
16
+ | 3.1.1 Definitions ..... | 5 |
17
+ | 3.1.2 Symbols ..... | 5 |
18
+ | 3.1.2.1 Variables ..... | 5 |
19
+ | 3.1.2.2 Constants..... | 6 |
20
+ | 3.1.2.3 Functions..... | 7 |
21
+ | 3.1.3 Abbreviations ..... | 7 |
22
+ | 3.2 General ..... | 7 |
23
+ | 3.3 Functional description..... | 7 |
24
+ | 3.3.1 Filter bank and computation of sub-band levels..... | 8 |
25
+ | 3.3.2 Pitch detection ..... | 10 |
26
+ | 3.3.3 Tone detection ..... | 10 |
27
+ | 3.3.4 Correlated Complex Signal Analysis (and detection) ..... | 11 |
28
+ | 3.3.5 VAD decision..... | 11 |
29
+ | 3.3.5.1 Hangover addition..... | 12 |
30
+ | 3.3.5.2 Background noise estimation..... | 14 |
31
+ | 4 Technical Description of VAD Option 2 ..... | 16 |
32
+ | 4.1 Definitions, symbols and abbreviations ..... | 16 |
33
+ | 4.1.1 Definitions ..... | 16 |
34
+ | 4.1.2 Symbols ..... | 16 |
35
+ | 4.1.2.1 Variables ..... | 16 |
36
+ | 4.1.2.2 Constants..... | 17 |
37
+ | 4.1.2.3 Functions..... | 17 |
38
+ | 4.1.3 Abbreviations ..... | 18 |
39
+ | 4.2 General ..... | 18 |
40
+ | 4.3 Functional description..... | 18 |
41
+ | 4.3.1 Frequency Domain Conversion ..... | 19 |
42
+ | 4.3.2 Channel Energy Estimator..... | 19 |
43
+ | 4.3.3 Channel SNR Estimator ..... | 20 |
44
+ | 4.3.4 Voice Metric Calculation ..... | 20 |
45
+ | 4.3.5 Frame SNR and Long-Term Peak SNR Calculation..... | 20 |
46
+ | 4.3.6 Negative SNR Sensitivity Bias..... | 21 |
47
+ | 4.3.7 VAD Decision ..... | 21 |
48
+ | 4.3.8 Spectral Deviation Estimator..... | 22 |
49
+ | 4.3.9 Sinewave Detection..... | 22 |
50
+ | 4.3.10 Background Noise Update Decision ..... | 23 |
51
+ | 4.3.10 Background Noise Estimate Update..... | 23 |
52
+ | 5 Computational details ..... | 24 |
53
+ | Annex A (informative) : Change history..... | 25 |
54
+
55
+ # --- Foreword
56
+
57
+ This Technical Specification has been produced by the 3<sup>rd</sup> Generation Partnership Project (3GPP).
58
+
59
+ 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:
60
+
61
+ Version x.y.z
62
+
63
+ where:
64
+
65
+ - x the first digit:
66
+ - 1 presented to TSG for information;
67
+ - 2 presented to TSG for approval;
68
+ - 3 or greater indicates TSG approved document under change control.
69
+ - y the second digit is incremented for all changes of substance, i.e. technical enhancements, corrections, updates, etc.
70
+ - z the third digit is incremented when editorial only changes have been incorporated in the document.
71
+
72
+ # --- 1 Scope
73
+
74
+ The present document specifies two alternatives for the Voice Activity Detector (VAD) to be used in the Discontinuous Transmission (DTX) as described in [3]. Implementors of mobile station and infrastructure equipment conforming to the AMR specifications can choose which of the two VAD options to implement. There are no interoperability factors associated with this choice.
75
+
76
+ The requirements are mandatory on any VAD to be used either in User Equipment (UE) or Base Station Systems (BSS)s that utilize the AMR speech codec.
77
+
78
+ # --- 2 References
79
+
80
+ The following documents contain provisions which, through reference in this text, constitute provisions of the present document.
81
+
82
+ - References are either specific (identified by date of publication, edition number, version number, etc.) or non-specific.
83
+ - For a specific reference, subsequent revisions do not apply.
84
+ - 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*.
85
+ - [1] 3GPP TS 26.073: "Adaptive Multi-Rate (AMR); ANSI C source code".
86
+ - [2] 3GPP TS 26.090: "Transcoding functions".
87
+ - [3] 3 GPP TS 26.093: "Source Controlled Rate operation".
88
+ - [4] ITU, The International Telecommunications Union, Blue Book, Vol. III, Telephone Transmission Quality, IXth Plenary Assembly, Melbourne, 14-25 November, 1988, Recommendation G.711, Pulse code modulation (PCM) of voice frequencies.
89
+
90
+ # --- 3 Technical Description of VAD Option 1
91
+
92
+ ## 3.1 Definitions, symbols and abbreviations
93
+
94
+ ### 3.1.1 Definitions
95
+
96
+ For the purposes of the present document, the following terms and definitions apply:
97
+
98
+ **frame:** time interval of 20 ms corresponding to the time segmentation of the speech transcoder
99
+
100
+ ### 3.1.2 Symbols
101
+
102
+ For the purposes of the present document, the following symbols apply.
103
+
104
+ #### 3.1.2.1 Variables
105
+
106
+ | | |
107
+ |---------------------------|-------------------------------------------------------------------------------------------------------|
108
+ | <b>bckr_est[n]</b> | background noise estimate |
109
+ | <b>burst_count</b> | counts length of a speech burst, used by VAD hangover addition |
110
+ | <b>hang_count</b> | hangover counter, used by VAD hangover addition |
111
+ | <b>complex_hang_count</b> | hangover counter, used by CAD hangover addition |
112
+ | <b>complex_hang_timer</b> | hangover initiator, used for Complex Activity Estimation |
113
+ | <b>lagcount</b> | pitch detection counter |
114
+ | <b>level[n]</b> | signal level |
115
+ | <b>new_speech</b> | pointer of the speech encoder, points a buffer containing last received samples of a speech frame [2] |
116
+
117
+ | | |
118
+ |------------------------|--------------------------------------------------------------------------|
119
+ | <b>noise_level</b> | average level of the background noise estimate |
120
+ | <b>oldlagcount</b> | lagcount of the previous frame |
121
+ | <b>pitch</b> | flag indicating presence of a periodic signal |
122
+ | <b>complex_warning</b> | flag indicating the presence of a complex signal. |
123
+ | <b>best_corr_hp</b> | normalized and limited value from maximum HP filtered correlation vector |
124
+ | <b>corr_hp</b> | filtered best_corr_hp values |
125
+ | <b>pow_sum</b> | power of the input frame |
126
+ | <b>s(i)</b> | samples of the input frame |
127
+ | <b>snr_sum</b> | measure between input frame and noise estimate |
128
+ | <b>stat_count</b> | stationarity counter |
129
+ | <b>stat_rat</b> | measure indicating stationary |
130
+ | <b>T_op[n]</b> | open-loop lags [2] |
131
+ | <b>t0</b> | autocorrelation maxima calculated by the open-loop pitch analysis [2] |
132
+ | <b>t1</b> | signal power related to the autocorrelation maxima t0 [2] |
133
+ | <b>tone</b> | flag indicating the presence of a tone |
134
+ | <b>vad_thr</b> | VAD threshold |
135
+ | <b>VAD_flag</b> | boolean VAD flag |
136
+ | <b>vadreg</b> | intermediate VAD decision |
137
+ | <b>complex_low</b> | intermediate complex signal decisions |
138
+ | <b>complex_high</b> | intermediate complex signal decisions |
139
+
140
+ #### 3.1.2.2 Constants
141
+
142
+ | | |
143
+ |-------------------------------|---------------------------------------------------------------------|
144
+ | <b>ALPHA_UP1</b> | constant for updating noise estimate (see clause 3.3.5.2) |
145
+ | <b>ALPHA_DOWN1</b> | constant for updating noise estimate (see clause 3.3.5.2) |
146
+ | <b>ALPHA_UP2</b> | constant for updating noise estimate (see clause 3.3.5.2) |
147
+ | <b>ALPHA_DOWN2</b> | constant for updating noise estimate (see clause 3.3.5.2) |
148
+ | <b>ALPHA3</b> | constant for updating noise estimate (see clause 3.3.5.2) |
149
+ | <b>ALPHA4</b> | constant for updating average signal level (see clause 3.3.5.2) |
150
+ | <b>ALPHA5</b> | constant for updating average signal level (see clause 3.3.5.2) |
151
+ | <b>BURST_LEN_HIGH_NOISE</b> | constant for controlling VAD hangover addition (see clause 3.3.5.1) |
152
+ | <b>BURST_LEN_LOW_NOISE</b> | constant for controlling VAD hangover addition (see clause 3.3.5.1) |
153
+ | <b>COEFF3</b> | coefficient for the filter bank (see clause 3.3.1) |
154
+ | <b>COEFF5_1</b> | coefficient for the filter bank (see clause 3.3.1) |
155
+ | <b>COEFF5_2</b> | coefficient for the filter bank (see clause 3.3.1) |
156
+ | <b>HANG_LEN_HIGH_NOISE</b> | constant for controlling VAD hangover addition (see clause 3.3.5.1) |
157
+ | <b>HANG_LEN_LOW_NOISE</b> | constant for controlling VAD hangover addition (see clause 3.3.5.2) |
158
+ | <b>HANG_NOISE_THR</b> | constant for controlling VAD hangover addition (see clause 3.3.5.2) |
159
+ | <b>L_FRAME</b> | size of a speech frame, 160 |
160
+ | <b>L_NEXT</b> | length for the lookahead of the speech encoder, 40 |
161
+ | <b>LTHRESH</b> | threshold for pitch detection (see clause 3.3.2) |
162
+ | <b>NOISE_MAX</b> | maximum value for noise estimate (see clause 3.3.5.2) |
163
+ | <b>NOISE_MIN</b> | minimum value for noise estimate (see clause 3.3.5.2) |
164
+ | <b>NTHRESH</b> | threshold for pitch detection (see clause 3.3.2) |
165
+ | <b>POW_PITCH_THR</b> | threshold for pitch detection (see clause 3.3.5) |
166
+ | <b>POW_COMPLEX_THR</b> | threshold for complex detection (see clause 3.3.5) |
167
+ | <b>STAT_COUNT</b> | threshold for stationary detection (see clause 3.3.5.2) |
168
+ | <b>CAD_MIN_STAT_COUNT</b> | minimum threshold after complex warning |
169
+ | <b>STAT_THR</b> | threshold for stationary detection (see clause 3.3.5.2) |
170
+ | <b>STAT_THR_LEVEL</b> | threshold for stationary detection (see clause 3.3.5.2) |
171
+ | <b>TONE_THR</b> | threshold for tone detection (see clause 3.3.3) |
172
+ | <b>VAD_P1</b> | constant of computation for VAD threshold (see clause 3.3.5.2) |
173
+ | <b>VAD_POW_LOW</b> | constant for controlling VAD hangover addition (see clause 3.3.5.1) |
174
+ | <b>VAD_SLOPE</b> | constant of computation for VAD threshold (see clause 3.3.5) |
175
+ | <b>VAD_THR_HIGH</b> | constant of computation for VAD threshold (see clause 3.3.5) |
176
+ | <b>CVAD_THRESH_ADAPT_HIGH</b> | constant for updating complex_high |
177
+ | <b>CVAD_THRESH_ADAPT_LOW</b> | constant for updating complex_low |
178
+ | <b>CVAD_THRESH_HANG</b> | constant for updating complex_hang_timer |
179
+ | <b>CVAD_HANG_LIMIT</b> | constant for initiating complex_hang_count |
180
+ | <b>CVAD_HANG_LENGTH</b> | constant for resetting complex_hang_count |
181
+
182
+ #### 3.1.2.3 Functions
183
+
184
+ | | |
185
+ |---------------------|-------------------------------------------------------|
186
+ | + | addition |
187
+ | - | subtraction |
188
+ | * | multiplication |
189
+ | / | division |
190
+ | $ x $ | absolute value of x |
191
+ | <b>AND</b> | Boolean AND |
192
+ | <b>OR</b> | Boolean OR |
193
+ | $\sum_{n=a}^b x(n)$ | $= x(a) + x(a+1) + \dots + x(b-1) + x(b)$ |
194
+ | <b>MIN(x,y)</b> | $= \begin{cases} x, x \leq y \\ y, y < x \end{cases}$ |
195
+ | <b>MAX(x,y)</b> | $= \begin{cases} x, x \geq y \\ y, y > x \end{cases}$ |
196
+
197
+ ### 3.1.3 Abbreviations
198
+
199
+ For the purposes of the present document, the following abbreviations apply:
200
+
201
+ | | |
202
+ |-------------|---------------------------------------|
203
+ | <b>ANSI</b> | American National Standards Institute |
204
+ | <b>DTX</b> | Discontinuous Transmission |
205
+ | <b>VAD</b> | Voice Activity Detector |
206
+ | <b>CAD</b> | Complex Activity Detection |
207
+ | <b>CNG</b> | Comfort Noise Generation |
208
+
209
+ ## 3.2 General
210
+
211
+ The function of the VAD algorithm is to indicate whether each 20 ms frame contains signals that should be transmitted, i.e. speech, music or information tones. The output of the VAD algorithm is a Boolean flag (VAD\_flag) indicating presence of such signals.
212
+
213
+ ## 3.3 Functional description
214
+
215
+ The block diagram of the VAD algorithm is depicted in figure 1. The VAD algorithm uses parameters of the speech encoder to compute the Boolean VAD flag (VAD\_flag). Samples of the Input frame ( $s(i)$ ) are divided into sub-bands and level of the signal in each band ( $level[n]$ ) is calculated. Input for the pitch detection function are open-loop lags ( $T\_op[n]$ ), which are calculated by open-loop pitch analysis of the speech encoder. The pitch detection function computes a flag (pitch) which indicates presence of pitch. Tone detection function calculates a flag (tone), which indicates presence of an information tone. Tones are detected based on pitch gain of the open-loop pitch analysis. The pitch gain is estimated using autocorrelation values ( $t0$ and $t1$ ) received from the pitch analysis. Complex Signal Detection function calculates a flag (complex\_warning), which indicates presence of a correlated complex signal such as music. Correlate complex signals are detected based on analysis of the correlation vector available in the open-loop pitch analysis. The VAD decision function estimates background noise levels. Intermediate VAD decision is calculated based on the comparison of the background noise estimate and levels of the input frame ( $level[n]$ ). Finally, the VAD flag is calculated by adding hangover to the intermediate VAD decision.
216
+
217
+ ![Simplified block diagram of the VAD algorithm: Option 1. The diagram shows a central 'VAD decision' block receiving inputs from four other blocks: 'Filter bank and computation of sub-band levels' (input s(i), output level[n]), 'Pitch detection' (input T_op[n], outputs pitch and tone), 'Tone detection' (input t0,t1, output complex_warning), and 'Complex signal analysis' (input OL-LTP correlation vector, output complex_timer). The 'VAD decision' block outputs the 'VAD_flag'.](d0abac95583b52a3b35f74a215567334_img.jpg)
218
+
219
+ ```
220
+
221
+ graph LR
222
+ s_i["s(i)"] --> FB["Filter bank and computation of sub-band levels"]
223
+ T_op_n["T_op[n]"] --> PD["Pitch detection"]
224
+ t0_t1["t0,t1"] --> TD["Tone detection"]
225
+ OL_LTP["OL-LTP correlation vector"] --> CSA["Complex signal analysis"]
226
+ FB -- level[n] --> VD["VAD decision"]
227
+ PD -- pitch --> VD
228
+ PD -- tone --> VD
229
+ TD -- complex_warning --> VD
230
+ CSA -- complex_timer --> VD
231
+ VD -- VAD_flag --> VF["VAD_flag"]
232
+
233
+ ```
234
+
235
+ Simplified block diagram of the VAD algorithm: Option 1. The diagram shows a central 'VAD decision' block receiving inputs from four other blocks: 'Filter bank and computation of sub-band levels' (input s(i), output level[n]), 'Pitch detection' (input T\_op[n], outputs pitch and tone), 'Tone detection' (input t0,t1, output complex\_warning), and 'Complex signal analysis' (input OL-LTP correlation vector, output complex\_timer). The 'VAD decision' block outputs the 'VAD\_flag'.
236
+
237
+ Figure 3.1: Simplified block diagram of the VAD algorithm: Option 1
238
+
239
+ ### 3.3.1 Filter bank and computation of sub-band levels
240
+
241
+ The input signal is divided into frequency bands using a 9-band filter bank (figure 3.2). Cut-off frequencies for the filter bank are shown in table 3.1.
242
+
243
+ Table 3.1. Cut-off frequencies for the filter bank
244
+
245
+ | Band number | Frequencies |
246
+ |-------------|----------------|
247
+ | 1 | 0 - 250 Hz |
248
+ | 2 | 250 - 500 Hz |
249
+ | 3 | 500 - 750 Hz |
250
+ | 4 | 750 - 1000 Hz |
251
+ | 5 | 1000 - 1500 Hz |
252
+ | 6 | 1500 - 2000 Hz |
253
+ | 7 | 2000 - 2500 Hz |
254
+ | 8 | 2500 - 3000 Hz |
255
+ | 9 | 3000 - 4000 Hz |
256
+
257
+ Input for the filter bank is the speech frame pointed by the new\_speech pointer of the speech encoder [1]. Input values for the filter bank are scaled down by one bit. This ensures safe scaling, i.e. saturation can not occur during calculation of the filter bank.
258
+
259
+ ![Figure 3.2: Filter bank diagram. The diagram shows a hierarchical decomposition of an input signal into multiple frequency bands using 5th and 3rd order filter blocks. The input signal enters a 5th order filter block. Its high-pass output is labeled '3k - 4 kHz'. Its low-pass output enters another 5th order filter block. The high-pass output of this second 5th order block is labeled '2.5 - 3 kHz'. Its low-pass output enters a 3rd order filter block. The high-pass output of this 3rd order block is labeled '2 - 2.5 kHz'. Its low-pass output enters another 3rd order filter block. The high-pass output of this 3rd order block is labeled '1.5 - 2 kHz'. Its low-pass output enters a 5th order filter block. The high-pass output of this 5th order block is labeled '1 - 1.5 kHz'. Its low-pass output enters a 3rd order filter block. The high-pass output of this 3rd order block is labeled '750 - 1000 Hz'. Its low-pass output enters another 3rd order filter block. The high-pass output of this 3rd order block is labeled '500 - 750 Hz'. Its low-pass output enters a final 3rd order filter block. The high-pass output of this 3rd order block is labeled '250 - 500 Hz'. The low-pass output of this final 3rd order block is labeled '0 - 250 Hz'.](b3baf3a29b67c7425d2562ddbc52f0cc_img.jpg)
260
+
261
+ Figure 3.2: Filter bank diagram. The diagram shows a hierarchical decomposition of an input signal into multiple frequency bands using 5th and 3rd order filter blocks. The input signal enters a 5th order filter block. Its high-pass output is labeled '3k - 4 kHz'. Its low-pass output enters another 5th order filter block. The high-pass output of this second 5th order block is labeled '2.5 - 3 kHz'. Its low-pass output enters a 3rd order filter block. The high-pass output of this 3rd order block is labeled '2 - 2.5 kHz'. Its low-pass output enters another 3rd order filter block. The high-pass output of this 3rd order block is labeled '1.5 - 2 kHz'. Its low-pass output enters a 5th order filter block. The high-pass output of this 5th order block is labeled '1 - 1.5 kHz'. Its low-pass output enters a 3rd order filter block. The high-pass output of this 3rd order block is labeled '750 - 1000 Hz'. Its low-pass output enters another 3rd order filter block. The high-pass output of this 3rd order block is labeled '500 - 750 Hz'. Its low-pass output enters a final 3rd order filter block. The high-pass output of this 3rd order block is labeled '250 - 500 Hz'. The low-pass output of this final 3rd order block is labeled '0 - 250 Hz'.
262
+
263
+ **Figure 3.2: Filter bank**
264
+
265
+ The filter bank consists of 5<sup>th</sup> and 3<sup>rd</sup> order filter blocks. Each filter block divides the input into high-pass and low-pass parts and decimates the sampling frequency by 2. The 5<sup>th</sup> order filter block is calculated as follows:
266
+
267
+ $$x_{lp}(i) = 0.5 * (A_1(x(i-1)) + A_2(x(i))) \quad (3.1a)$$
268
+
269
+ $$x_{hp}(i) = 0.5 * (A_1(x(i-1)) - A_2(x(i))) \quad (3.1b)$$
270
+
271
+ where
272
+
273
+ $x(i)$ input signal for a filter block
274
+
275
+ $x_{lp}(i)$ low-pass component
276
+
277
+ $x_{hp}(i)$ high-pass component
278
+
279
+ The 3<sup>rd</sup> order filter block is calculated as follows:
280
+
281
+ $$x_{lp}(i) = 0.5 * (x(i) + A_3(x(i-1))) \quad (3.2a)$$
282
+
283
+ $$x_{hp}(i) = 0.5 * (x(i) - A_3(x(i-1))) \quad (3.2b)$$
284
+
285
+ The filters $A_1()$ , $A_2()$ , and $A_3()$ are first order direct form all-pass filters, whose transfer function is given by:
286
+
287
+ $$A(z) = \frac{C + z^{-1}}{1 + C^* z^{-1}}, \quad (3.3)$$
288
+
289
+ where $C$ is the filter coefficient.
290
+
291
+ Coefficients for the all-pass filters $A_1()$ , $A_2()$ , and $A_3()$ are COEFF5\_1, COEFF5\_2, and COEFF3, respectively.
292
+
293
+ Signal level is calculated at the output of the filter bank at each frequency band as follows:
294
+
295
+ $$level(n) = \sum_{i=START_n}^{END_n} |x_n(i)|, \quad (3.4)$$
296
+
297
+ where:
298
+
299
+ $n$ index for the frequency band
300
+
301
+ $x_n(i)$ sample $i$ at the output of the filter bank at frequency band $n$
302
+
303
+ $$START_n = \begin{cases} -2, & n \leq 4 \\ -4, & 5 \leq n \leq 8 \\ -8, & n = 9 \end{cases}$$
304
+
305
+ $$END_n = \begin{cases} 9, & n \leq 4 \\ 19, & 5 \leq n \leq 8 \\ 39, & n = 9 \end{cases}$$
306
+
307
+ Negative indices of $x_n(i)$ refer to the previous frame.
308
+
309
+ ### 3.3.2 Pitch detection
310
+
311
+ The purpose of the pitch detection function is to detect vowel sounds and other periodic signals. The pitch detection is based on comparison of open-loop lags ( $T\_op[n]$ ), which are calculated by the speech encoder [2]. If the difference of consecutive open-loop lags ( $T\_op[n]$ ) is smaller than a threshold, lagcount is incremented. If the sum of the lagcounts of two consecutive frames is high enough, the pitch flag is set. For 5.15 and 4.75 kbit/s rates, only one open-loop lag is calculated, and therefore only the first lag-comparison is made every frame. The pitch flag is calculated as follows:
312
+
313
+ Lagcount = 0;
314
+
315
+ If ( $|T\_op[-1] - T\_op[0]| < LTHRESH$ )
316
+
317
+ Lagcount = Lagcount + 1
318
+
319
+ If ( $|T\_op[0] - T\_op[1]| < LTHRESH$ )
320
+
321
+ Lagcount = Lagcount + 1
322
+
323
+ If (Lagcount + oldlagcount $\geq NTHRESH$ )
324
+
325
+ pitch = 1
326
+
327
+ else
328
+
329
+ pitch = 0
330
+
331
+ oldlagcount = Lagcount
332
+
333
+ $T\_op[-1]$ refers to the open-loop lag of the previous frame.
334
+
335
+ ### 3.3.3 Tone detection
336
+
337
+ Tone detection is used to detect information tones, since the pitch detection function can not always detect these signals. Also, other signals which contain very strong periodic component are detected, because it may sound annoying if these signals are replaced by comfort noise. If the open-loop pitch gain is higher than the constant $TONE\_THR$ , tone is detected and tone flag is set. The pitch gain can be tested by comparing variables $t0$ and $t1$ as follows:
338
+
339
+ if ( $t0 > TONE\_THR * t1$ )
340
+
341
+ tone = 1
342
+
343
+ The speech encoder calculates the pitch in three delay ranges, except for mode 10.2 kbit/s, where only one range is used. The above comparison is made once for each delay range and the tone flag should be set if the condition is true at least in one range. Otherwise, the tone flag should be set to zero.
344
+
345
+ The variables $t0$ and $t1$ are calculated by the open-loop pitch analysis of the speech encoder [2]. The variable $t0$ is autocorrelation maxima given by:
346
+
347
+ $$t0 = \sum_n s_w(n) s_w(n-k) \quad (3.5)$$
348
+
349
+ The variable $t1$ is the signal power related to the autocorrelation maxima $t0$ at the delay value $k$ :
350
+
351
+ $$t1 = \sum_n s_w^2(n-k) \quad (3.6)$$
352
+
353
+ The open-loop pitch search and correspondingly the tone flag is computed twice in each frame, except for modes 5.15 kbit/s and 4.75 kbit/s, where it is computed only once.
354
+
355
+ ### 3.3.4 Correlated Complex Signal Analysis (and detection)
356
+
357
+ Correlated complex signal detection is used to detect correlated signals in the highpass filtered weighted speech domain, since the pitch and tone detection functions can not always detect these signals. Signals which contain very strong correlation values in the high pass filtered domain are taken care of, because it may sound really annoying if these signals are replaced by comfort noise. If the statistics of the maximum normalized correlation value of a high pass filtered input signal indicates the presence of a correlated complex signal a flag **complex\_warning** is set. To reduce complexity the high band correlation analysis is performed in a simplified manner by analysing the high pass filtered fullband correlation vector which is available from the OL-LTP analysis performed by the speech encoder at least once in each frame.
358
+
359
+ $best\_corr\_hp_m$ is the maximum normalized value of the high pass filtered correlation in the range 19-146 limited to be in the range [1.0, 0.0]. (Note that the $best\_corr\_hp$ value is delayed one frame). The high pass filter is a simple first order filter with coefficients [1, -1] The $best\_corr\_hp$ value is filtered according to :
360
+
361
+ $$corr\_hp_{m+1} = (alpha) * corr\_hp_m + (1 - alpha) * best\_corr\_hp_m ,$$
362
+
363
+ where $alpha$ is varied between 0.98 and 0.8 as a function of $corr\_hp_m$ and $best\_corr\_hp_m$
364
+
365
+ The $corr\_hp$ output value is thresholded into two registers $complex\_high$ , $complex\_low$ and one $counter\_complex\_hang\_timer$ .
366
+
367
+ $complex\_low$ is set to 1 if the $corr\_hp$ value is greater than CVAD\_THRESH\_ADAPT\_LOW.
368
+
369
+ $complex\_high$ is set to 1 if the $corr\_hp$ value is greater than CVAD\_THRESH\_ADAPT\_HIGH.
370
+
371
+ $complex\_hang\_timer$ is increased by 1 if the $corr\_hp$ value is greater than CVAD\_THRESH\_HANG. If the $corr\_hp$ value is lower than or equal to CVAD\_THRESH\_HANG the $complex\_hang\_timer$ value is set to 0.
372
+
373
+ The flag **complex\_warning** is set if $complex\_low$ have been set for 15 consecutive frames or $complex\_high$ has been set for 8 consecutive frames.
374
+
375
+ The open-loop pitch search and correspondingly the tone flag is computed twice in each frame, except for modes 5.15 kbit/s and 4.75 kbit/s, where it is computed only once. The computation of the $corr\_hp$ value is however always done only once per frame using the newest correlation vector available.
376
+
377
+ ### 3.3.5 VAD decision
378
+
379
+ Power of the input frame is calculated as follows:
380
+
381
+ $$pow\_sum = \sum_{i=-L\_NEXT}^{L\_FRAME-L\_NEXT-1} s(i) * s(i) , \quad (3.7)$$
382
+
383
+ where samples $s(i)$ of the input frame are pointed by the new\_speech pointer of the speech encoder. If the power of the input frame ( $pow\_sum$ ) is lower than the constant POW\_PITCH\_THR, last pitch flag is set to zero. If the power of the input frame ( $pow\_sum$ ) is lower than the constant POW\_COMPLEX\_THR, last $complex\_low$ flag is set to zero.
384
+
385
+ The difference between the signal levels of the input frame and background noise estimate is calculated as follows:
386
+
387
+ $$snr\_sum = \sum_{n=1}^9 MAX(1.0, \frac{level[n]}{bckr\_est[n]})^2, \quad (3.8)$$
388
+
389
+ where:
390
+
391
+ level[n] signal level at band n
392
+
393
+ bckr\_est[n] level of background noise estimate at band n
394
+
395
+ VAD decision is made by comparing the variable snr\_sum to a threshold. The threshold (vad\_thr) is tuned to get desired sensitivity at each background noise level. The higher the noise level the lower is the threshold. Specially, a low threshold at high-level background noise is needed to detect speech reliably enough, although probability of detecting noise as speech also increases.
396
+
397
+ Average level of background noise is calculated by adding noise estimates at each band:
398
+
399
+ $$noise\_level = \sum_{n=1}^9 bckr\_est[n] \quad (3.9)$$
400
+
401
+ Threshold is calculated using average noise level as follows:
402
+
403
+ $$vad\_thr = VAD\_SLOPE * (noise\_level - VAD\_PI) + VAD\_THR\_HIGH, \quad (3.10)$$
404
+
405
+ where VAD\_SLOPE, VAD\_PI, and VAD\_THR\_HIGH are constants.
406
+
407
+ The variable vadreg indicates intermediate VAD decision and it is calculated as follows:
408
+
409
+ if (snr\_sum > vad\_thr)
410
+
411
+ vadreg = 1
412
+
413
+ else
414
+
415
+ vadreg = 0
416
+
417
+ #### 3.3.5.1 Hangover addition
418
+
419
+ Before the final VAD flag is given, a hangover is added. The hangover addition helps to detect low power endings of speech bursts, which are subjectively important but difficult to detect. Also a long hangover is added if the signal has been found to be of very complex nature for a long time (2 seconds) since the VAD is not likely to work reliably for such a complex signal.
420
+
421
+ VAD flag is set to "1" if less than hang\_len frames with "0" decision have been elapsed since burst\_len consecutive "1" decisions have been detected. The variables hang\_len and burst\_len are set depending on the average noise level (noise\_level). The *vad\_flag* is also controlled by the *complex\_hang\_count* which indicates that the signal is too complex for the VAD and should not be used with a Comfort noise generation algorithm. The filtered correlation value *corr\_hp* is also used as an activity indication after the VAD has indicated noise for a while (during 200 ms), this will aid in situations where the VAD noise estimate has adapted to a rather stationary but still all to complex signal to make it sound well with CNG.
422
+
423
+ The power of the input frame is compared to a threshold (VAD\_POW\_LOW). If the power is lower, the VAD flag is set to "0" and no hangover is added. The VAD\_flag is calculated as follows:
424
+
425
+ if (noise\_level > HANG\_NOISE\_THR)
426
+
427
+ burst\_len = BURST\_LEN\_HIGH\_NOISE
428
+
429
+ hang\_len = HANG\_LEN\_HIGH\_NOISE
430
+
431
+ else
432
+
433
+ burst\_len = BURST\_LEN\_LOW\_NOISE
434
+
435
+ hang\_len = HANG\_LEN\_LOW\_NOISE
436
+
437
+ ```
438
+
439
+ if(complex_hang_timer > CVAD_HANG_LIMIT) {
440
+ if(complex_hang_count < CVAD_HANG_LENGTH) {
441
+ complex_hang_count = CVAD_HANG_LENGTH;
442
+ }
443
+ }
444
+
445
+ if(powsum < VAD_POW_LOW){
446
+ burst_count = 0
447
+ hang_count = 0
448
+ complex_hang_count = 0;
449
+ complex_hang_timer = 0;
450
+ Vad_flag=0;
451
+ Goto Exit;
452
+ }
453
+
454
+ VAD_flag=0;
455
+ if(complex_hang_count != 0){
456
+ burst_count = BURST_LEN_HIGH_NOISE;
457
+ complex_hang_count = complex_hang_count - 1 ;
458
+ VAD_flag=1;
459
+ goto Exit
460
+ } else {
461
+ if( ( (the 10 last out of 11 vadreg values all are zero) AND
462
+ (corr_hp > CVAD_THRESH_IN_NOISE) ) ) {
463
+ VAD_flag = 1;
464
+ Goto Exit
465
+ }
466
+ }
467
+
468
+ if(vadreg = 1){
469
+ burst_count = burst_count + 1}
470
+ if(burst_count >= burst_len){
471
+ hang_count = hang_len
472
+ }
473
+ VAD_flag = 1
474
+ } else {
475
+
476
+ ```
477
+
478
+ ```
479
+
480
+ burst_count = 0
481
+ if (hang_count > 0){
482
+ hang_count = hang_count - 1
483
+ VAD_flag=1
484
+ }
485
+ }
486
+ Label Exit
487
+
488
+ ```
489
+
490
+ #### 3.3.5.2 Background noise estimation
491
+
492
+ Background noise estimate ( $bckr\_est[n]$ ) is updated using amplitude levels of the previous frame. Thus, the update is delayed by one frame to avoid undetected start of speech bursts to corrupt the noise estimate. If the internal VAD decision is "1" or if pitch has been detected, the noise estimate is not updated upwards. The update speed for the current frame is selected as follows:
493
+
494
+ ```
495
+
496
+ if ((vadreg for the last 4 frames has been zero) AND
497
+ (pitch for the last 4 frames has been zero) AND
498
+ (we are not in complex signal hangover))
499
+ alpha_up = ALPHA_UP1
500
+ alpha_down = ALPHA_DOWN1
501
+ else
502
+ if ((stat_count = 0) AND (not in complex_signal hangover))
503
+ alpha_up = ALPHA_UP2
504
+ alpha_down = ALPHA_DOWN2
505
+ else
506
+ alpha_up = 0
507
+ alpha_down = ALPHA3
508
+
509
+ ```
510
+
511
+ The variable $stat\_count$ indicates stationary and its propose is explained later in this clause. The variables $alpha\_up$ and $alpha\_down$ define the update speed to upwards and downwards. The update speed for each band $n$ is selected as follows:
512
+
513
+ if ( $bckr\_est_m[n] < level_{m-1}[n]$ )
514
+
515
+ $alpha = alpha\_up$
516
+
517
+ else
518
+
519
+ $alpha = alpha\_down$
520
+
521
+ Finally, noise estimate is updated as follows:
522
+
523
+ $$bckr\_est_{m+1}[n] = (1.0 - alpha) * bckr\_est_m[n] + alpha * level_{m-1}[n], \quad (3.11)$$
524
+
525
+ where:
526
+
527
+ $n$ index of the frequency band
528
+
529
+ $m$ index of the frame
530
+
531
+ Level of the background estimate (bckr\_est[n]) is limited between constants NOISE\_MIN and NOISE\_MAX.
532
+
533
+ If level of background noise increases suddenly, vadreg will be set to "1" and background noise is not updated upwards. To recover from this situation, update of the background noise estimate is enabled if the intermediate VAD decision (vadreg) is "1" for enough long time and spectrum is stationary. Stationary (stat\_rat) is estimated using following equation:
534
+
535
+ $$stat\_rat = \sum_{n=1}^9 \frac{MAX(STAT\_THR\_LEVEL, MAX(ave\_level\_m[n], level\_m[n]))}{MAX(STAT\_THR\_LEVEL, MIN(ave\_level\_m[n], level\_m[n]))} \quad (3.12)$$
536
+
537
+ If the stationary estimate (stat\_rat) is higher than a threshold, the stationary counter (stat\_count) is set to the initial value defined by constant STAT\_COUNT. The stationary counter (stat\_count) is also initialised if pitch or tone or a complex\_warning is detected. If the signal is not stationary but speech has been detected (VAD decision is "1"), stat\_count is decreased by one in each frame until it is zero.
538
+
539
+ if (*complex\_warning*) {
540
+
541
+     If(stat\_count < CAD\_MIN\_STAT\_COUNT)
542
+
543
+         stat\_count = CAD\_MIN\_STAT\_COUNT
544
+
545
+ }
546
+
547
+ if ( ( 8 last vadreg flags have been zero) OR (2 last pitch flags have been one) OR (5 last tone flags have been one) )
548
+
549
+     stat\_count = STAT\_COUNT
550
+
551
+ else
552
+
553
+ if (stat\_rat > STAT\_THR)
554
+
555
+     stat\_count = STAT\_COUNT
556
+
557
+ else
558
+
559
+ if ((vadreg) AND (stat\_count $\neq$ 0))
560
+
561
+     stat\_count = stat\_count - 1
562
+
563
+ The average signal levels (ave\_level[n]) are calculated as follows:
564
+
565
+ $$ave\_level_{m+1}[n] = (1.0 - \alpha) * ave\_level_m[n] + \alpha * level_m[n] \quad (3.13)$$
566
+
567
+ The update speed ( $\alpha$ ) for the previous equation is selected as follows:
568
+
569
+ if (stat\_count = STAT\_COUNT)
570
+
571
+ $\alpha = 1.0$
572
+
573
+ else if (vadreg = 1)
574
+
575
+ $\alpha = \text{ALPHA5}$
576
+
577
+ else
578
+
579
+ $\alpha = \text{ALPHA4}$
580
+
581
+ # 4 Technical Description of VAD Option 2
582
+
583
+ ## 4.1 Definitions, symbols and abbreviations
584
+
585
+ ### 4.1.1 Definitions
586
+
587
+ For the purposes of the present document, the following terms and definitions apply:
588
+
589
+ **codec:** combination of an encoder and decoder in series (encoder/decoder)
590
+
591
+ **compress:** process of compressing and expanding a signal. In this text, the process is described in terms of PCM [4]
592
+
593
+ **Decoder:** generally, a device for the translation of a signal from a digital representation into an analog format. For the present document, a device which converts speech encoded in the format specified in the present document to analog or an equivalent PCM representation
594
+
595
+ **DFT:** see Discrete Fourier Transform
596
+
597
+ **Discrete Fourier Transform (DFT):** method of transforming a time domain sequence into a corresponding frequency domain sequence
598
+
599
+ **Encoder:** generally, a device for the translation of a signal into a digital representation. For the present document, a device which converts speech from an analog or its equivalent PCM representation to the digital representation described in the present document
600
+
601
+ **Fast Fourier Transform (FFT):** efficient implementation of the Discrete Fourier Transform
602
+
603
+ **FFT:** see Fast Fourier Transform
604
+
605
+ **Vocoder:** voice coder
606
+
607
+ **frame:** time interval of 20 ms corresponding to the time segmentation of the speech transcoder
608
+
609
+ ### 4.1.2 Symbols
610
+
611
+ For the purposes of the present document, the following symbols apply.
612
+
613
+ #### 4.1.2.1 Variables
614
+
615
+ | | |
616
+ |-------------------------------|-----------------------------------------------------------------------------------------------------------|
617
+ | $\alpha_{ch}(m)$ | channel energy smoothing factor |
618
+ | $\alpha(m)$ | exponential windowing factor |
619
+ | $\Delta_E(m)$ | estimated spectral deviation between current power spectrum and average long term power spectral estimate |
620
+ | $\phi(m)$ | spectral peak-to-average ratio |
621
+ | $\sigma_q^{(i)}$ | quantized channel SNR indices |
622
+ | $b(m)$ | burst count |
623
+ | $b_{th}$ | burst count threshold |
624
+ | $\{d(m)\}$ | overlapped portion of the frame buffer of input samples |
625
+ | $E_{ch}(m, i)$ | channel energy estimate; channel i, subframe m |
626
+ | $\mathbf{E}_{ch}(m)$ | vector of channel energy estimates, $0 \leq i < N_c$ |
627
+ | $E_{dB}(m, i)$ | estimated log power spectrum |
628
+ | $\mathbf{E}_{dB}(m)$ | vector of log power spectrum estimates, $0 \leq i < N_c$ |
629
+ | $\bar{E}_{dB}^{(m, i)}$ | average long term power spectral estimate |
630
+ | $\bar{\mathbf{E}}_{dB}^{(m)}$ | vector of average long term power spectral estimates, $0 \leq i < N_c$ |
631
+ | $E_n(m, i)$ | channel noise estimate |
632
+ | $\mathbf{E}_n(m)$ | vector of channel noise estimates, $0 \leq i < N_c$ |
633
+ | $E_{tn}(m)$ | total estimated noise energy |
634
+ | $E_{tot}(m)$ | total channel energy |
635
+ | $E'_{tot}(m)$ | modified total channel energy |
636
+ | $h(m)$ | hysteresis counter |
637
+ | $h_{cnt}$ | hangover count |
638
+
639
+ | | |
640
+ |---------------------|----------------------------------------------------------------------------------------------------------------------------------|
641
+ | $h_o(n)$ | overlap-and-add buffer of samples |
642
+ | $hyster\_cnt$ | hysteresis counter to avoid long term creeping of $update\_cnt$ |
643
+ | $last\_update\_cnt$ | previous value of $update\_cnt$ |
644
+ | $s_{hp}(n)$ | sample at the output of the speech encoder high pass filter |
645
+ | $sinewave\_flag$ | boolean flag, set TRUE when spectral peak-to-average ratio is greater than 10dB and the spectral deviation is less than DEV_THLD |
646
+ | $SNR$ | Signal to Noise ratio |
647
+ | $SNR_p(m)$ | long-term peak SNR |
648
+ | $SNR_q(m)$ | quantized version of $SNR_p(m)$ |
649
+ | $update\_cnt$ | counter gating noise estimate update process |
650
+ | $update\_flag$ | flag controlling noise estimate updating |
651
+ | $VAD(m)$ | boolean VAD flag for subframe m |
652
+ | $VAD\_flag$ | boolean VAD Flag |
653
+ | $v(m)$ | sum of voice metrics |
654
+ | $v_{th}$ | voice metric threshold |
655
+
656
+ #### 4.1.2.2 Constants
657
+
658
+ | | |
659
+ |-----------------|---------------------------------------------------------------|
660
+ | $\alpha_H$ | upper limit for values of $\alpha(m)$ |
661
+ | $\alpha_L$ | lower limit for values of $\alpha(m)$ |
662
+ | $\alpha_n$ | channel noise smoothing factor |
663
+ | $\zeta_p$ | pre-emphasis factor |
664
+ | $b_{table}$ | table to generate $b_{th}$ |
665
+ | D | overlap (delay) in sample intervals |
666
+ | DEV_THLD | threshold for setting $sinewave\_flag$ |
667
+ | $E_{floor}$ | low threshold for $E_{tot}(m)$ |
668
+ | $E_H$ | high energy endpoint for linear interpolation of $E_{tot}(m)$ |
669
+ | $E_{init}$ | minimum allowable channel noise initialisation energy |
670
+ | $E_L$ | low energy endpoint for linear interpolation of $E_{tot}(m)$ |
671
+ | $E_{min}$ | minimum allowable channel energy |
672
+ | $f_H$ | high channel combining table |
673
+ | $f_L$ | low channel combining table |
674
+ | $g(n)$ | trapezoidal window, $n = 0$ to $M$ |
675
+ | $G(k)$ | frequency domain transformation of $g(n)$ |
676
+ | $h_{table}$ | table to generate $h_{cnt}$ |
677
+ | HYSTER_CNT_THLD | threshold for $hyster\_cnt$ |
678
+ | L | subframe length in samples |
679
+ | M | DFT sequence length |
680
+ | $N_c$ | number of combined channels |
681
+ | NOISE_FLOOR_D | low threshold for $E_{tot}(m)$ in dB |
682
+ | UPDATE_CNT_THLD | threshold for $update\_cnt$ |
683
+ | UPDATE_THLD | threshold for $v(m)$ |
684
+ | V | voice metric table |
685
+ | $v_{table}$ | table to generate $v_{th}$ |
686
+
687
+ #### 4.1.2.3 Functions
688
+
689
+ | | |
690
+ |---------------------|--------------------------|
691
+ | + | addition |
692
+ | - | subtraction |
693
+ | * | multiplication |
694
+ | / | division |
695
+ | $\lfloor x \rfloor$ | largest integer $\leq x$ |
696
+ | AND | Boolean AND |
697
+ | OR | Boolean OR |
698
+
699
+ $$\sum_{n=a}^b x(n) = x(a) + x(a+1) + \dots + x(b-1) + x(b)$$
700
+
701
+ ### 4.1.3 Abbreviations
702
+
703
+ For the purposes of the present document, the following abbreviations apply:
704
+
705
+ | | |
706
+ |------|---------------------------------------|
707
+ | ANSI | American National Standards Institute |
708
+ | DTX | Discontinuous Transmission |
709
+ | VAD | Voice Activity Detector |
710
+ | CAD | Complex Activity Detection |
711
+ | CNG | Comfort Noise Generation |
712
+
713
+ ## 4.2 General
714
+
715
+ The function of the VAD algorithm is to indicate whether each 20 ms frame contains signals that should be transmitted, i.e. speech, music or information tones. The output of the VAD algorithm is a Boolean flag (*VAD\_flag*) indicating presence of such signals.
716
+
717
+ ## 4.3 Functional description
718
+
719
+ The block diagram of the VAD algorithm is depicted in figure 4.1. The VAD algorithm uses parameters of the speech encoder to compute the Boolean VAD flag (*VAD\_flag*).
720
+
721
+ ![Block Diagram of the VAD algorithm: Option 2](124c6108c63173818afb8ed49521e22d_img.jpg)
722
+
723
+ The block diagram illustrates the functional flow of the VAD algorithm. The input signal $s_{hp}(n)$ is processed through a 'Frequency Domain Conversion' block to produce $G(k)$ . This is then processed by a 'Channel Energy Estimator' to produce $E_{ch}(m)$ . The signal $s_{hp}(n)$ is also processed by three parallel blocks: 'Spectral Deviation Estimator', 'Peak-to-Average Ratio', and 'Voice Metric Calculation'. The 'Spectral Deviation Estimator' and 'Peak-to-Average Ratio' blocks both output to the 'Channel SNR Estimator', which also receives $E_{ch}(m)$ and produces $E_n(m)$ and $\{\sigma_q\}$ . The 'Voice Metric Calculation' block outputs to the 'Noise Update Decision' block. The 'Noise Update Decision' block also receives $E_{tot}(m)$ and $\Delta \epsilon(m)$ and produces $fupdate\_flag$ . The 'Background Noise Estimator' receives $E_n(m)$ and produces $E_{tot}(m)$ and $update\_flag$ . The 'VAD' block receives inputs from the 'Noise Update Decision' block ( $fupdate\_flag$ ), the 'Background Noise Estimator' ( $E_{tot}(m)$ and $v(m)$ ), and the 'Voice Metric Calculation' block. The output of the VAD block is *VAD\_flag*.
724
+
725
+ Block Diagram of the VAD algorithm: Option 2
726
+
727
+ Figure 4.1: Block Diagram of the VAD algorithm: Option 2
728
+
729
+ ### Input:
730
+
731
+ The output of the High-Pass Filter, $\{s_{hp}(n)\}$
732
+
733
+ - *LTP\_flag* is generated by the comparison of the long-term prediction gain to a constant threshold *LTP\_THLD*, where the long-term prediction gain $\beta$ is derived from the speech encoder[2] open-loop pitch predictor.
734
+
735
+ ### Output:
736
+
737
+ - The output of the vad is designated as *VAD\_flag*
738
+
739
+ #### Initialization:
740
+
741
+ The following variables shall be set to zero at initialization (frame $m = 0$ ):
742
+
743
+ - The pre-emphasis memory
744
+
745
+ The following shall be initialized to a startup value other than zero:
746
+
747
+ The channel energy estimate, $E_{ch}(m)$ , (see clause 4.3.2)
748
+
749
+ The long-term power spectral estimate, $\bar{E}_{dB}^{(m)}$ , (see clause 4.3.5)
750
+
751
+ The channel noise estimate, $E_n(m)$ , (see clause 4.3.8)
752
+
753
+ Processing: The following procedures shall be executed two times per 20 ms speech frame and the current 10 ms subframe shall be denoted $m$ .
754
+
755
+ ### 4.3.1 Frequency Domain Conversion
756
+
757
+ The input signal is pre-emphasised and windowed prior to frequency domain conversion. This process is defined as:
758
+
759
+ $$d(n) = s_{hp}(n) + \zeta_p s_{hp}(n-1), \quad 0 \leq n < L, \quad (4.1)$$
760
+
761
+ where $d(n)$ is the pre-emphasised speech buffer, $\zeta_p$ is the pre-emphasis factor, and $L$ is the subframe length. A rectangular window is then used to frame the speech prior to frequency domain conversion, which is expressed as:
762
+
763
+ $$g(n) = \begin{cases} 0, & 0 \leq n < D, L + D \leq n < M \\ d(n - D), & D \leq n < L + D \end{cases}, \quad (4.2)$$
764
+
765
+ where $D$ is the zero-padding offset into the DFT buffer, and $M$ is the DFT length. The transformation of $g(n)$ to the frequency domain is performed using the Discrete Fourier Transform (DFT) defined<sup>1</sup> as:
766
+
767
+ $$G(k) = \frac{2}{M} \sum_{n=0}^{M-1} g(n) e^{-j2\pi nk/M}, \quad 0 \leq k < M \quad (4.3)$$
768
+
769
+ where $e^{j\omega}$ is a unit amplitude complex phasor with instantaneous radial position $\omega$ .
770
+
771
+ ### 4.3.2 Channel Energy Estimator
772
+
773
+ Calculate the channel energy estimate $E_{ch}(m)$ for the current subframe, $m$ , as:
774
+
775
+ $$E_{ch}(m, i) = \max \left\{ E_{\min}, \alpha_{ch}(m) E_{ch}(m-1, i) + (1 - \alpha_{ch}(m)) \frac{1}{f_H(i) - f_L(i) + 1} \sum_{k=f_L(i)}^{f_H(i)} |G(k)|^2 \right\}, \quad 0 \leq i < N_c \quad (4.4)$$
776
+
777
+ where $E_{\min}$ is the minimum allowable channel energy, $\alpha_{ch}(m)$ is the channel energy smoothing factor (defined below), $N_c$ is the number of combined channels, and $f_L(i)$ and $f_H(i)$ are the $i$ -th elements of the respective low and high channel combining tables.
778
+
779
+ The channel energy smoothing factor, $\alpha_{ch}(m)$ , is defined as:
780
+
781
+ $$\alpha_{ch}(m) = \begin{cases} 0, & m \leq 1 \\ 0.45, & m > 1 \end{cases} \quad (4.5)$$
782
+
783
+ <sup>1</sup> This atypical definition is used to exploit the efficiencies of the complex Fast Fourier Transform (FFT). The $2/M$ scale factor results from preconditioning the $M$ point real sequence to form an $M/2$ point complex sequence that is transformed using an $M/2$ point complex FFT. Details on this technique can be found in Proakis, J. G. and Manolakis, D. G., *Introduction to Digital Signal Processing*, New York, Macmillan, 1988, pp. 721-722.
784
+
785
+ So, this means that $\alpha_{ch}(m)$ assumes a value of zero for the first frame ( $m = 1$ ) and a value of 0.45 for all subsequent frames. This allows the channel energy estimate to be initialized to the unfiltered channel energy of the first frame.
786
+
787
+ ### 4.3.3 Channel SNR Estimator
788
+
789
+ Estimate the channel SNR vector $\{ \sigma \}$ as:
790
+
791
+ $$\sigma(i) = 10 \log_{10} \left( \frac{E_{ch}(m,i)}{E_n(m,i)} \right), \quad 0 \leq i < N_c \quad (4.6)$$
792
+
793
+ where $E_n(m)$ is the current channel noise energy estimate (see clause 4.3.8), and then quantify the channel SNR estimate in 3/8 dB steps to yield the channel SNR indices $\{ \sigma_q \}$ given as:
794
+
795
+ $$\sigma_q(i) = \max \{ 0, \min \{ 89, \text{round} \{ \sigma(i) / 0.375 \} \} \}, \quad 0 \leq i < N_c \quad (4.7)$$
796
+
797
+ where the values of $\{ \sigma_q \}$ are constrained to be between 0 and 89, inclusive.
798
+
799
+ ### 4.3.4 Voice Metric Calculation
800
+
801
+ Next, calculate the sum of voice metrics as:
802
+
803
+ $$v(m) = \sum_{i=0}^{N_c-1} V(\sigma_q(i)) \quad (4.8)$$
804
+
805
+ where $V(k)$ is the $k^{\text{th}}$ value of the 90 element voice metric table V.
806
+
807
+ ### 4.3.5 Frame SNR and Long-Term Peak SNR Calculation
808
+
809
+ The instantaneous frame SNR, $SNR$ , and long-term peak SNR, $SNR_p(m)$ , are used to calibrate the responsiveness of the VAD decision. When the frame count is less than or equal to four ( $m \leq 4$ ) or the forced update flag (sec 4.3.10) is set (fupdate\_flag == TRUE), then the SNR's are initialized as:
810
+
811
+ $$SNR_p(m) = SNR = 56 - 10 \log_{10} \left( \sum_{i=0}^{N_c-1} E_n(m,i) \right) \quad (4.9)$$
812
+
813
+ Otherwise, the instantaneous frame SNR is generated by:
814
+
815
+ $$SNR = 10 \log_{10} \left( \frac{1}{N_c} \sum_{i=0}^{N_c-1} 10^{\sigma(i)/10} \right) \quad (4.10)$$
816
+
817
+ and the long-term peak SNR is derived by the following expression:
818
+
819
+ $$SNR_p(m) = \begin{cases} 0.9 SNR_p(m-1) + 0.1 SNR, & SNR > SNR_p(m-1) \\ 0.998 SNR_p(m-1) + 0.002 SNR, & 0.625 SNR_p(m-1) < SNR \leq SNR_p(m-1) \\ SNR_p(m-1), & \text{otherwise} \end{cases} \quad (4.11)$$
820
+
821
+ The long-term peak SNR is then quantized in 3 dB steps and limited to be between 0 and 19, as follows:
822
+
823
+ $$SNR_q = \max \{ \min \{ \lfloor SNR_p(m) / 3 \rfloor, 19 \}, 0 \} \quad (4.12)$$
824
+
825
+ where $\lfloor x \rfloor$ is the largest integer $\leq x$ (floor function).
826
+
827
+ ### 4.3.6 Negative SNR Sensitivity Bias
828
+
829
+ In order for the VAD decision to overcome the problem of being over-sensitive to fluctuating, non-stationary background noise conditions, a bias factor is used to increase the threshold on which the VAD decision is based. This bias factor is derived from an estimate of the variability of the background noise estimate. The variability estimate is further based on negative values of the instantaneous SNR. It is presumed that a negative SNR can only occur as a result of fluctuating background noise, and not from the presence of voice. Therefore, the bias factor $\mu(m)$ is derived by first calculating the variability factor $\psi(m)$ as:
830
+
831
+ $$\psi(m) = \begin{cases} 0.99\psi(m-1) + 0.01SNR^2, & SNR < 0 \\ \psi(m-1) & otherwise \end{cases} \quad (4.13)$$
832
+
833
+ which is then clamped in magnitude to $0 \leq \psi(m) \leq 4.0$ . In addition, the variability factor is reset to zero when the frame count is less than or equal to four ( $m \leq 4$ ) or the forced update flag (sec 4.3.10) is set (*fupdate\_flag* == TRUE). The bias factor $\mu(m)$ is then calculated as:
834
+
835
+ $$\mu(m) = \max\{12.0(\psi(m) - 0.65), 0\} \quad (4.14)$$
836
+
837
+ ### 4.3.7 VAD Decision
838
+
839
+ The quantized SNR $SNR_q$ is used to determine the respective voice metric threshold $v_{th}$ , hangover count $h_{cnt}$ , and burst count threshold $b_{th}$ parameters:
840
+
841
+ $$v_{th} = v_{table}(SNR_q), \quad h_{cnt} = h_{table}(SNR_q), \quad b_{th} = b_{table}(SNR_q) \quad (4.15)$$
842
+
843
+ where $SNR_Q$ is the index of the respective table elements. The VAD decision can then be made according to the following pseudocode:
844
+
845
+ ```
846
+ if ( v(m) > vth + μ(m) ) { /* if the voice metric > voice metric threshold*/
847
+ VAD(m) = ON
848
+ b(m) = b(m-1) + 1 /* increment burst counter */
849
+ if ( b(m) > bth ) { /* compare counter with threshold */
850
+ h(m) = hcnt /* set hangover */
851
+ }
852
+ } else {
853
+ b(m) = 0 /* clear burst counter */
854
+ h(m) = h(m-1) - 1 /* decrement hangover /
855
+ if ( h(m) <= 0 ) { /* check for expired hangover /
856
+ VAD(m) = OFF
857
+ h(m) = 0
858
+ } else {
859
+ VAD(m) = ON /* hangover not yet expired */
860
+ }
861
+ }
862
+ }
863
+ ```
864
+
865
+ Note that two 10 ms subframes are required to determine one VAD decision. The final decision is determined by the maximum of two subframe decisions, i.e.
866
+
867
+ ```
868
+ if(VAD(m) == ON OR VAD(m-1) == ON) {
869
+ VAD_flag = TRUE
870
+ } else {
871
+ VAD_flag = FALSE
872
+ }
873
+ ```
874
+
875
+ ### 4.3.8 Spectral Deviation Estimator
876
+
877
+ The spectral deviation estimator is used as a safeguard against erroneous updates of the background noise estimate. If the spectral deviation of the input signal is too high, then the background noise estimate update may not be permitted. Calculate the estimated log power spectrum as:
878
+
879
+ $$E_{dB}(m,i) = 10\log_{10}(E_{ch}(m,i)), \quad 0 \leq i < N_c \quad (4.16)$$
880
+
881
+ Then, calculate the estimated spectral deviation between the current power spectrum and the average long-term power spectral estimate:
882
+
883
+ $$\Delta_E(m) = \sum_{i=0}^{N_c-1} |E_{dB}(m,i) - \bar{E}_{dB}(m,i)| \quad (4.17)$$
884
+
885
+ where $\bar{E}_{dB}(m)$ is the average long-term power spectral estimate calculated during the previous subframe, as defined in Equation 4.20. The initial value of $\bar{E}_{dB}(m)$ , however, is defined to be the estimated log power spectrum of subframe 1, or:
886
+
887
+ $$\bar{E}_{dB}(m) = E_{dB}(m), \quad m = 1 \quad (4.18)$$
888
+
889
+ The exponential windowing factor, $\alpha(m)$ , is then calculated as a function of the instantaneous frame SNR $SNR$ and the long-term peak SNR $SNR_p(m)$ , as:
890
+
891
+ $$\alpha(m) = \alpha_H - \left( (\alpha_H - \alpha_L) \frac{SNR_p(m) - SNR}{SNR_p(m)} \right), \quad (4.19)$$
892
+
893
+ which is then limited to $\alpha_L \leq \alpha(m) \leq \alpha_H$ .
894
+
895
+ The average long-term power spectral estimate is then updated for the next frame by:
896
+
897
+ $$\bar{E}_{dB}(m+1,i) = \alpha(m)\bar{E}_{dB}(m,i) + (1-\alpha(m))E_{dB}(m,i), \quad 0 \leq i < N_c \quad (4.20)$$
898
+
899
+ where all the variables are previously defined.
900
+
901
+ ### 4.3.9 Sinewave Detection
902
+
903
+ Next the *sinewave\_flag* is set TRUE when the spectral peak-to-average ratio $\phi(m)$ is greater than 10, i.e.
904
+
905
+ $$\text{sinewave\_flag} = \begin{cases} \text{TRUE}, & \phi(m) > 10 \\ \text{FALSE}, & \text{otherwise} \end{cases} \quad (4.21)$$
906
+
907
+ where:
908
+
909
+ $$\phi(m) = 10\log_{10} \left( \frac{\max\{E_{ch}(m,i)\}}{\sum_{j=0}^{N_c-1} E_{ch}(m,j)/N_c} \right), \quad 2 \leq i < N_c \quad (4.22)$$
910
+
911
+ ### 4.3.10 Background Noise Update Decision
912
+
913
+ The following logic, as shown in pseudo-code, demonstrates how the noise estimate update decision is ultimately made:
914
+
915
+ ```
916
+
917
+ /* Normal update logic */
918
+ update_flag = fupdate_flag = FALSE
919
+ if ( v(m) ≤ UPDATE_THLD and b(m) == 0 ) {
920
+ update_flag = TRUE
921
+ update_cnt = 0
922
+ }
923
+
924
+ /* Forced update logic (for over-riding the normal update logic)*/
925
+ else if ( ( E_tot > NOISE_FLOOR) and ( Δ_E(m) < DEV_THLD ) )
926
+
927
+ ```
928
+
929
+ ```
930
+
931
+ and ( sinewave_flag == FALSE ) and ( LTP_flag == FALSE ) ) {
932
+ update_cnt = update_cnt + 1
933
+ if ( update_cnt ≥ UPDATE_CNT_THLD )
934
+ update_flag = fupdate_flag = TRUE
935
+ }
936
+
937
+ /* "Hysteresis" logic to prevent long-term creeping of update_cnt */
938
+
939
+ if ( update_cnt == last_update_cnt )
940
+ hyster_cnt = hyster_cnt + 1
941
+ else
942
+ hyster_cnt = 0
943
+ last_update_cnt = update_cnt
944
+ if ( hyster_cnt > HYSTER_CNT_THLD )
945
+ update_cnt = 0
946
+
947
+ ```
948
+
949
+ where $E_{tot}$ is the total channel energy defined as:
950
+
951
+ $$E_{tot} = \sum_{i=0}^{N_c-1} E_{ch}(m,i) \quad (4.23)$$
952
+
953
+ and *LTP\_flag* is generated by the comparison of the long-term prediction gain to a constant threshold LTP\_THLD, i.e.:
954
+
955
+ $$LTP\_flag = \begin{cases} \text{TRUE}, & \beta > \text{LTP\_THLD} \\ \text{FALSE}, & \text{otherwise} \end{cases} \quad (4.24)$$
956
+
957
+ where the long-term prediction gain $\beta$ is derived from the speech encoder [2] open-loop pitch predictor, and can be expressed as:
958
+
959
+ $$\beta = \frac{\sum_{n=0}^{N_p-1} s_w(n)s_w(n-k)}{\sum_{n=0}^{N_p-1} s_w^2(n)} \quad (4.25)$$
960
+
961
+ where $s_w(n)$ is the weighted speech, $k$ is the optimal open-loop lag, and $N_p$ is the pitch analysis frame length. This expression is calculated in the speech encoder on the previous frame.
962
+
963
+ ### 4.3.10 Background Noise Estimate Update
964
+
965
+ If (and only if) the update flag is set (*update\_flag* == **TRUE**), then update the channel noise estimate for the next subframe by:
966
+
967
+ $$E_n(m+1,i) = \max\{E_{min}, \alpha_n E_n(m,i) + (1-\alpha_n) E_{ch}(m,i)\}, \quad 0 \leq i < N_c \quad (4.26)$$
968
+
969
+ where $E_{min}$ is the minimum allowable channel energy, and $\alpha_n$ is the channel noise smoothing factor. The channel noise estimate shall be initialized for each of the first four frames to the estimated channel energy, i.e.:
970
+
971
+ $$E_n(m,i) = \max\{E_{init}, E_{ch}(m,i)\}, \quad m \leq 4, \quad 0 \leq i < N_c, \quad (4.27)$$
972
+
973
+ where $E_{init}$ is the minimum allowable channel noise initialization energy.
974
+
975
+ # 5 Computational details
976
+
977
+ A low level description has been prepared in form of ANSI C source code [1].
978
+
979
+ # Annex A (informative) : Change history
980
+
981
+ | <b>Document history</b> | | | | | | | | |
982
+ |-------------------------|--------------|--------------------------------------------------------------|-----------|------------|------------------------------------------------|------------|------------|------------|
983
+ | V. 2.0.0 | October 1999 | 3G TS 26.094 presented for approval to TSG#5 Plenary meeting | | | | | | |
984
+ | V. 3.0.0 | October 1999 | 3G TS 26.094 approved at TSG#5 Plenary meeting | | | | | | |
985
+ | <b>Change history</b> | | | | | | | | |
986
+ | <b>Date</b> | <b>TSG #</b> | <b>TSG Doc.</b> | <b>CR</b> | <b>Rev</b> | <b>Subject/Comment</b> | <b>Old</b> | <b>Old</b> | <b>New</b> |
987
+ | 03-2001 | 11 | | | | Version for Release 4 | | | 4.0.0 |
988
+ | 06-2002 | 16 | | | | Version for Release 5 | 4.0.0 | | 5.0.0 |
989
+ | 12-2004 | 26 | | | | Version for Release 6 | 5.0.0 | | 6.0.0 |
990
+ | 06-2006 | 32 | SP-060356 | 0001 | 2 | Correct the description error and syntax error | 6.0.0 | | 6.1.0 |
991
+ | 06-2007 | 36 | | | | Version for Release 7 | 6.1.0 | | 7.0.0 |
992
+ | 12-2008 | 42 | | | | Version for Release 8 | 7.0.0 | | 8.0.0 |
993
+ | 12-2009 | 46 | | | | Version for Release 9 | 8.0.0 | | 9.0.0 |
994
+ | 03-2011 | 51 | | | | Version for Release 10 | 9.0.0 | | 10.0.0 |
995
+ | 09-2011 | 57 | | | | Version for Release 11 | 10.0.0 | | 11.0.0 |
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1
+
2
+
3
+
4
+
5
+
6
+
7
+ # --- Contents
8
+
9
+ | | |
10
+ |-----------------------------------------------------------------------------------|-----------|
11
+ | Foreword ..... | 4 |
12
+ | 1 Scope..... | 5 |
13
+ | 2 References..... | 5 |
14
+ | 3 Definitions and Abbreviations ..... | 5 |
15
+ | 3.1 Definitions..... | 5 |
16
+ | 3.2 Abbreviations ..... | 5 |
17
+ | 4 AMR codec frame format (AMR IF1)..... | 6 |
18
+ | 4.1 AMR Header and AMR Auxiliary Information ..... | 6 |
19
+ | 4.1.1 Frame Type, Mode Indication, and Mode Request ..... | 6 |
20
+ | 4.1.2 Frame Quality Indicator ..... | 7 |
21
+ | 4.1.3 Mapping to TX_TYPE and RX_TYPE..... | 7 |
22
+ | 4.1.4 Codec CRC ..... | 8 |
23
+ | 4.2 AMR Core Frame..... | 8 |
24
+ | 4.2.1 AMR Core Frame with speech bits: Bit ordering..... | 8 |
25
+ | 4.2.2 AMR Core Frame with speech bits: Class division..... | 9 |
26
+ | 4.2.3 AMR Core Frame with comfort noise bits ..... | 9 |
27
+ | 4.3 AMR frame composition..... | 10 |
28
+ | 4.4 GSM-EFR Frame Composition..... | 11 |
29
+ | 4.4.1 GSM-EFR Frame with speech bits..... | 12 |
30
+ | 4.4.2 GSM-EFR Frame with comfort noise bits..... | 12 |
31
+ | <b>Annex A (informative): AMR Interface Format 2 (with octet alignment) .....</b> | <b>13</b> |
32
+ | <b>Annex B (normative): Tables for AMR Core Frame bit ordering.....</b> | <b>17</b> |
33
+ | Annex C (informative): Change history..... | 20 |
34
+
35
+ # --- Foreword
36
+
37
+ This Technical Specification has been produced by the 3<sup>rd</sup> Generation Partnership Project (3GPP).
38
+
39
+ 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:
40
+
41
+ Version x.y.z
42
+
43
+ where:
44
+
45
+ - x the first digit:
46
+ - 1 presented to TSG for information;
47
+ - 2 presented to TSG for approval;
48
+ - 3 or greater indicates TSG approved document under change control.
49
+ - y the second digit is incremented for all changes of substance, i.e. technical enhancements, corrections, updates, etc.
50
+ - z the third digit is incremented when editorial only changes have been incorporated in the document.
51
+
52
+ # --- 1 Scope
53
+
54
+ The present document describes a generic frame format for the Adaptive Multi-Rate (AMR) speech codec and the Enhanced Full Rate (GSM-EFR) speech codec. This format shall be used as a common reference point when interfacing speech frames between different elements of the 3G system and between different systems. Appropriate mappings to and from this generic frame format will be used within and between each system element.
55
+
56
+ Annex A describes a second frame format which shall be used when octet alignment of AMR frames is required.
57
+
58
+ # --- 2 References
59
+
60
+ The following documents contain provisions which, through reference in this text, constitute provisions of the present document.
61
+
62
+ - References are either specific (identified by date of publication, edition number, version number, etc.) or non-specific.
63
+ - For a specific reference, subsequent revisions do not apply.
64
+ - 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*.
65
+
66
+ - [1] TS 26.090: "AMR Speech Codec; Speech Transcoding Functions".
67
+ - [2] TS 26.093: "AMR Speech Codec; Source Controlled Rate Operation".
68
+ - [3] TS 26.092: "AMR Speech Codec; Comfort Noise Aspects".
69
+ - [4] TS 46.060: "EFR Speech Codec; Speech Transcoding Functions".
70
+ - [5] TS 46.062: "EFR Speech Codec; Comfort Noise Aspects".
71
+
72
+ # --- 3 Definitions and Abbreviations
73
+
74
+ ## 3.1 Definitions
75
+
76
+ For the purposes of the present document, the following terms and definitions apply:
77
+
78
+ **AMR mode:** one of the eight AMR codec bit-rates denoted also with indices 0 to 7 where 0 maps to the 4.75 kbit/s mode and 7 maps to the 12.2 kbit/s mode.
79
+
80
+ **AMR codec mode:** same as AMR mode.
81
+
82
+ **RX\_TYPE:** classification of the received frame as defined in [2].
83
+
84
+ **TX\_TYPE:** classification of the transmitted frame as defined in [2].
85
+
86
+ ## 3.2 Abbreviations
87
+
88
+ For the purposes of the present document, the following abbreviations apply:
89
+
90
+ | | |
91
+ |-----|------------------------------------------------|
92
+ | CRC | Cyclic Redundancy Check |
93
+ | FQI | Frame Quality Indicator |
94
+ | GSM | Global System for Mobile communication |
95
+ | LSB | Least Significant Bit |
96
+ | MSB | Most Significant Bit |
97
+ | RX | Receive |
98
+ | PDC | Personal Digital Communication (ARIB standard) |
99
+
100
+ | | |
101
+ |------|------------------------------------------|
102
+ | SCR | Source Controlled Rate operation |
103
+ | SID | Silence Descriptor (Comfort Noise Frame) |
104
+ | TDMA | Time Division Multiple Access (IS-641) |
105
+ | TX | Transmit |
106
+
107
+ # 4 AMR codec frame format (AMR IF1)
108
+
109
+ This clause describes the generic frame format for both the speech and comfort noise frames of the AMR speech codec. This format is referred to as AMR Interface Format 1 (AMR IF1). Annex A describes AMR Interface Format 2 (AMR IF2).
110
+
111
+ Each AMR codec mode follows the generic frame structure depicted in figure 1. The frame is divided into three parts: AMR Header, AMR Auxiliary Information, and AMR Core Frame. The AMR Header part includes the Frame Type and the Frame Quality Indicator fields. The AMR auxiliary information part includes the Mode Indication, Mode Request, and Codec CRC fields. The AMR Core Frame part consists of the speech parameter bits or, in case of a comfort noise frame, the comfort noise parameter bits. In case of a comfort noise frame, the comfort noise parameters replace Class A bits of AMR Core Frame while Class B and C bits are omitted.
112
+
113
+ ![Figure 1: Generic AMR frame structure. The diagram shows a vertical stack of fields grouped into three main sections: AMR Header, AMR Auxiliary Information, and AMR Core Frame. The AMR Header contains 'Frame Type (4 bits)' and 'Frame Quality Indicator (1 bit)'. The AMR Auxiliary Information contains 'Mode Indication (3 bits)', 'Mode Request (3 bits)', and 'Codec CRC (8 bits)', with a note '(for Mode Adaptation, and Error Detection)'. The AMR Core Frame contains 'Class A bits', 'Class B bits', and 'Class C bits', with a note '(speech or comfort noise data)'.](1d27fed9c01eb99f6535283f35fe3bbf_img.jpg)
114
+
115
+ | | |
116
+ |---------------------------------|-------------------------------------------------------------------------|
117
+ | Frame Type (4 bits) | AMR Header |
118
+ | Frame Quality Indicator (1 bit) | |
119
+ | Mode Indication (3 bits) | AMR Auxiliary Information<br>(for Mode Adaptation, and Error Detection) |
120
+ | Mode Request (3 bits) | |
121
+ | Codec CRC (8 bits) | |
122
+ | Class A bits | AMR Core Frame (speech or comfort noise data) |
123
+ | Class B bits | |
124
+ | Class C bits | |
125
+
126
+ Figure 1: Generic AMR frame structure. The diagram shows a vertical stack of fields grouped into three main sections: AMR Header, AMR Auxiliary Information, and AMR Core Frame. The AMR Header contains 'Frame Type (4 bits)' and 'Frame Quality Indicator (1 bit)'. The AMR Auxiliary Information contains 'Mode Indication (3 bits)', 'Mode Request (3 bits)', and 'Codec CRC (8 bits)', with a note '(for Mode Adaptation, and Error Detection)'. The AMR Core Frame contains 'Class A bits', 'Class B bits', and 'Class C bits', with a note '(speech or comfort noise data)'.
127
+
128
+ Figure 1: Generic AMR frame structure
129
+
130
+ ## 4.1 AMR Header and AMR Auxiliary Information
131
+
132
+ This subclause describes the AMR Header of figure 1.
133
+
134
+ ### 4.1.1 Frame Type, Mode Indication, and Mode Request
135
+
136
+ Table 1a defines the 4-bit Frame Type field. Frame Type can indicate the use of one of the eight AMR codec modes, one of four different comfort noise frames, or an empty frame. In addition, three Frame Type Indices are reserved for future use. The same table is reused for the Mode Indication and Mode Request fields which are 3-bit fields each and are defined only in the range 0...7 to specify one of the eight AMR codec modes.
137
+
138
+ **Table 1a: Interpretation of Frame Type, Mode Indication and Mode Request fields**
139
+
140
+ | Frame Type | Mode Indication | Mode Request | Frame content (AMR mode, comfort noise, or other) |
141
+ |------------|-----------------|--------------|---------------------------------------------------|
142
+ | 0 | 0 | 0 | AMR 4,75 kbit/s |
143
+ | 1 | 1 | 1 | AMR 5,15 kbit/s |
144
+ | 2 | 2 | 2 | AMR 5,90 kbit/s |
145
+ | 3 | 3 | 3 | AMR 6,70 kbit/s (PDC-EFR) |
146
+ | 4 | 4 | 4 | AMR 7,40 kbit/s (TDMA-EFR) |
147
+ | 5 | 5 | 5 | AMR 7,95 kbit/s |
148
+ | 6 | 6 | 6 | AMR 10,2 kbit/s |
149
+ | 7 | 7 | 7 | AMR 12,2 kbit/s (GSM-EFR) |
150
+ | 8 | - | - | AMR SID |
151
+ | 9 | - | - | GSM-EFR SID |
152
+ | 10 | - | - | TDMA-EFR SID |
153
+ | 11 | - | - | PDC-EFR SID |
154
+ | 12-14 | - | - | For future use |
155
+ | 15 | - | - | No Data (No transmission/No reception) |
156
+
157
+ ### 4.1.2 Frame Quality Indicator
158
+
159
+ The content of the Frame Quality Indicator field is defined in Table 1b. The field length is one bit. The Frame Quality Indicator indicates whether the data in the frame contains errors.
160
+
161
+ **Table 1b: Definition of Frame Quality Indicator**
162
+
163
+ | Frame Quality Indicator (FQI) | Quality of data |
164
+ |-------------------------------|--------------------------------------------------------------------------------|
165
+ | 0 | Bad frame or Corrupted frame<br>(bits may be used to assist error concealment) |
166
+ | 1 | Good frame |
167
+
168
+ ### 4.1.3 Mapping to TX\_TYPE and RX\_TYPE
169
+
170
+ Table 1c shows how the AMR Header data (FQI and Frame Type) maps to the TX\_TYPE and RX\_TYPE frames defined in [2].
171
+
172
+ **Table 1c: Mapping of Frame Quality Indicator and Frame Type to TX\_TYPE and RX\_TYPE [2], respectively**
173
+
174
+ | Frame Quality Indicator | Frame Type Index | TX_TYPE or RX_TYPE | Comment |
175
+ |-------------------------|------------------|-------------------------|-------------------------------------------------------------------------------------------------------------------------------|
176
+ | 1 | 0-7 | SPEECH_GOOD | The specific Frame Type Index depends on the bit-rate being used. |
177
+ | 0 | 0-7 | SPEECH_BAD | The specific Frame Type Index depends on the bit-rate being used. The corrupted data may be used to assist error concealment. |
178
+ | 1 | 8 | SID_FIRST or SID_UPDATE | For AMR: SID_FIRST and SID_UPDATE are differentiated using one Class A bit: STI. |
179
+ | 0 | 8 | SID_BAD | For AMR |
180
+ | 1 | 9 | GSM-EFR SID | For GSM-EFR |
181
+ | 0 | 9 | GSM-EFR SID_BAD | For GSM-EFR |
182
+ | 1 | 10-11 | SID_UPDATE | For TDMA-EFR and PDC-EFR |
183
+ | 0 | 10-11 | SID_BAD | For TDMA-EFR and PDC-EFR |
184
+ | 1 | 15 | NO_DATA | Typically a non-transmitted frame or an erased or stolen frame with no data usable to assist error concealment. |
185
+
186
+ ### 4.1.4 Codec CRC
187
+
188
+ Generic AMR codec frames with Frame Type 0.11 are associated with an 8-bit CRC for error-detection purposes. The Codec CRC field of AMR Auxiliary Information in figure 1 contains the value of this CRC. These eight parity bits are generated by the cyclic generator polynomial:
189
+
190
+ $$- G(x)=D^8 + D^6 + D^5 + D^4 + 1$$
191
+
192
+ which is computed over all Class A bits of AMR Core Frame. Class A bits for Frame Types 0.7 are defined in subclause 4.2.2 (for speech bits) and for Frame Types 8.11 in subclause 4.2.3 (for comfort noise bits).
193
+
194
+ When Frame Type Index of table 1a is 15 the CRC field is not included in the Generic AMR frame.
195
+
196
+ ## 4.2 AMR Core Frame
197
+
198
+ This subclause contains the description of AMR Core Frame of figure 1. The descriptions for AMR Core Frame with speech bits and with comfort noise bit are given separately.
199
+
200
+ ### 4.2.1 AMR Core Frame with speech bits: Bit ordering
201
+
202
+ This subclause describes how AMR Core Frame carries the coded speech data. The bits produced by the speech encoder are denoted as $\{s(1), s(2), \dots, s(K)\}$ , where $K$ refers to the number of bits produced by the speech encoder as shown in table 2. The notation $s(i)$ follows that of [1]. The speech encoder output bits are ordered according to their subjective importance. This bit ordering can be utilized for error protection purposes when the speech data is, for example, carried over a radio interface. Tables B.1 to B.8 in Annex B define the AMR IF1 bit ordering for all the eight AMR codec modes. In these tables the speech bits are numbered in the order they are produced by the corresponding speech encoder as described in the relevant tables of TS 26.090 [1]. The reordered bits are denoted below, in the order of decreasing importance, as $\{d(0), d(1), \dots, d(K-1)\}$ .
203
+
204
+ The ordering algorithm is described in pseudo code as:
205
+
206
+ - for $j = 0$ to $K-1$
207
+ - $d(j) := s(\text{table}_m(j)+1)$ ;
208
+
209
+ where $\text{table}_m(j)$ refers to the relevant table in Annex B depending on the AMR mode $m=0..7$ . The Annex B tables should be read line by line from left to right. The first element of the table has the index 0.
210
+
211
+ ### 4.2.2 AMR Core Frame with speech bits: Class division
212
+
213
+ The reordered bits are further divided into three indicative classes according to their subjective importance. This class division is only informative and provides supporting information for mapping this generic format into specific formats. The three different importance classes can then be subject to different error protection in the network.
214
+
215
+ The importance classes are Class A, Class B, and Class C. Class A contains the bits most sensitive to errors and any error in these bits typically results in a corrupted speech frame which should not be decoded without applying appropriate error concealment. This class is protected by the Codec CRC in AMR Auxiliary Information. Classes B and C contain bits where increasing error rates gradually reduce the speech quality, but decoding of an erroneous speech frame is usually possible without annoying artefacts. Class B bits are more sensitive to errors than Class C bits. The importance ordering applies also within the three different classes and there are no significant step-wise changes in subjective importance between neighbouring bits at the class borders.
216
+
217
+ The number of speech bits in each class (Class A, Class B, and Class C) for each AMR mode is shown in table 2. The classification in table 2 and the importance ordering $d(j)$ , together, are sufficient to assign all speech bits to their correct classes. For example, when the AMR codec mode is 4.75, then the Class A bits are $d(0)..d(41)$ , Class B bits are $d(42)..d(94)$ , and there are no Class C bits.
218
+
219
+ **Table 2: Number of bits in Classes A, B, and C for each AMR codec mode**
220
+
221
+ | Frame Type | AMR codec mode | Total number of bits | Class A | Class B | Class C |
222
+ |------------|----------------|----------------------|---------|---------|---------|
223
+ | 0 | 4,75 | 95 | 42 | 53 | 0 |
224
+ | 1 | 5,15 | 103 | 49 | 54 | 0 |
225
+ | 2 | 5,90 | 118 | 55 | 63 | 0 |
226
+ | 3 | 6,70 | 134 | 58 | 76 | 0 |
227
+ | 4 | 7,40 | 148 | 61 | 87 | 0 |
228
+ | 5 | 7,95 | 159 | 75 | 84 | 0 |
229
+ | 6 | 10,2 | 204 | 65 | 99 | 40 |
230
+ | 7 | 12,2 | 244 | 81 | 103 | 60 |
231
+
232
+ ### 4.2.3 AMR Core Frame with comfort noise bits
233
+
234
+ The AMR Core Frame content for the additional frame type with Frame Type Index 8 in table 1a is described in this subclause. This consists of the frame related to Source Controlled Rate Operation specified in [2].
235
+
236
+ The data content (comfort noise bits) of the additional frame types is carried in AMR Core Frame. The comfort noise bits are all mapped to Class A of AMR Core Frame and Classes B and C are not used. This is a notation convention only and the class division has no meaning for comfort noise bits.
237
+
238
+ The number of bits in each class (Class A, Class B, and Class C) for the AMR comfort noise bits (Frame Type Index 8) is shown in table 3. The contents of SID\_UPDATE and SID\_FIRST are divided into three parts (SID Type Indicator (STI), Mode Indication ( $mi(i)$ ), and Comfort Noise Parameters ( $s(i)$ ) as defined in [2]. In case of SID\_FIRST the Comfort Noise Parameters bits ( $s(i)$ ) shall be set to “0”.
239
+
240
+ The comfort noise parameter bits produced by the AMR speech encoder are denoted as $s(i) = \{s(1), s(2), \dots, s(35)\}$ . The notation $s(i)$ follows that of [3]. These bits are numbered in the order they are produced by the AMR encoder without any reordering. These bits are followed by the SID Type Indicator **STI** and the Mode Indication $mi(i) = \{mi(0), mi(1), mi(2)\} = \{LSB \dots MSB\}$ . Thus, the AMR SID or comfort noise bits $\{d(0), d(1), \dots, d(38)\}$ are formed as defined by the pseudo code below.
241
+
242
+ - for $j = 0$ to 34;
243
+ - $d(j) := s(j+1)$ ;
244
+ - $d(35) := STI$ ;
245
+ - for $j = 36$ to 38;
246
+ - $d(j) := mi(j-36)$ . Note: This mapping is different to the usual mapping: LSB first.
247
+
248
+ Note: The alternative would be: $d(j) := mi(38-j)$ : MSB first.
249
+
250
+ **Table 3. Bit classification for Frame Type 8 (AMR SID)**
251
+
252
+ | Frame Type Index | FQI | AMR TX_TYPE or RX_TYPE | Total number of bits | Class A | | | Class B | Class C |
253
+ |------------------|-----|------------------------|----------------------|--------------------------|-------------------------|--------------------------------|---------|---------|
254
+ | | | | | SID Type Indicator (STI) | Mode Indication $mi(i)$ | Comfort Noise Parameter $s(i)$ | | |
255
+ | 8 | 1 | SID_UPDATE | 39 | 1 (= "1") | 3 | 35 | 0 | 0 |
256
+ | 8 | 1 | SID_FIRST | 39 | 1 (= "0") | 3 | 35 (= "0") | 0 | 0 |
257
+ | 8 | 0 | SID_BAD | 39 | 1 | 3 | 35 | 0 | 0 |
258
+
259
+ The number of bits in each class (Class A, Class B, and Class C) for the comfort noise bits of Frame Types 9-11 is shown in Table 7.
260
+
261
+ **Table 4: void**
262
+
263
+ ## 4.3 AMR frame composition
264
+
265
+ The generic AMR frame is formed as a concatenation of AMR Header, AMR Auxiliary Information and the AMR Core Frame, in this order. The MSB of the Frame Type is placed in bit 8 of the first octet (see example in table 5 below), the LSB of the Frame Type is placed in bit 5. Then the next parameter follows, which is the Frame Quality Indicator, and so on. Between Mode Request and Codec CRC five spare bits are inserted to align the Codec CRC and the AMR Core frame to the octet boundary. The first bit of the AMR Core frame d(0) is placed in bit 8 of octet 4. The last bit of the generic AMR frame is the last bit of AMR Core Frame, which is the last bit of speech bits or the last bit of comfort noise bits, as defined in subclauses 4.2.1 and 4.2.3. Table 5 shows the composition for the example of the Codec Mode 6.7kbit/s and table 6 shows the composition for the AMR SID frame.
266
+
267
+ **Table 5: Mapping of an AMR speech coding mode into the generic AMR frame, AMR IF1, example: AMR 6.7 kbit/s, "good frame", Mode Request = 1.**
268
+
269
+ | Octet | MSB | Mapping of bits<br>AMR 6.7 | | | | | | LSB |
270
+ |-------|---------------------------|----------------------------|--------|--------|--------|----------------------|-----------|--------|
271
+ | | bit 8 | bit 7 | bit 6 | bit 5 | bit 4 | bit 3 | bit 2 | bit 1 |
272
+ | 1 | Frame Type (=3) | | | | FQI | Mode Indication (=3) | | |
273
+ | | 0 | 0 | 1 | 1 | 1 | 0 | 1 | 1 |
274
+ | 2 | Mode Request (=1) | | | spare | | | | |
275
+ | | 0 | 0 | 1 | 0 | 0 | 0 | 0 | 0 |
276
+ | 3 | Codec CRC | | | | | | | |
277
+ | | CRC(7) | CRC(6) | CRC(5) | CRC(4) | CRC(3) | CRC(2) | CRC(1) | CRC(0) |
278
+ | 4 | AMR Core Frame (octet 1) | | | | | | | |
279
+ | | d(0) | d(1) | d(2) | d(3) | d(4) | d(5) | d(6) | d(7) |
280
+ | 5..19 | | | | | | | | |
281
+ | 20 | AMR Core Frame (octet 17) | | | | | | undefined | |
282
+ | | d(128) | d(129) | d(130) | d(131) | d(132) | d(133) | | |
283
+
284
+ **Table 6: Mapping of an AMR SID frame into the generic AMR frame, AMR IF1, example: AMR SID\_Update, "good frame", Mode Indication = 3, Mode Request = 2.**
285
+
286
+ | Octet | MSB | Mapping of bits<br>AMR SID | | | | | | LSB |
287
+ |-------|--------------------------|----------------------------|------------------|--------|----------------------|-----------------|--------|--------|
288
+ | | bit 8 | bit 7 | bit 6 | bit 5 | bit 4 | bit 3 | bit 2 | bit 1 |
289
+ | 1 | Frame Type (=8) | | | | FQI | Mode Indication | | |
290
+ | | 1 | 0 | 0 | 0 | 1 | undefined | | |
291
+ | 2 | Mode Request (=2) | | | spare | | | | |
292
+ | | MSB | ... | LSB | | | | | |
293
+ | | 0 | 1 | 0 | 0 | 0 | 0 | 0 | 0 |
294
+ | 3 | Codec CRC | | | | | | | |
295
+ | | CRC(7) | CRC(6) | CRC(5) | CRC(4) | CRC(3) | CRC(2) | CRC(1) | CRC(0) |
296
+ | 4 | AMR Core Frame (octet 1) | | | | | | | |
297
+ | | d(0)=s(1) | d(1)=s(2) | d(2) | d(3) | d(4) | d(5) | d(6) | d(7) |
298
+ | 5..7 | | | | | | | | |
299
+ | 8 | | | | STI | Mode Indication (=3) | | | undef. |
300
+ | | | | | | LSB | ... | MSB | |
301
+ | | d(32) | d(33) | d(34) =<br>s(35) | 1 | 1 | 1 | 0 | |
302
+
303
+ Table 7 summarizes all possible AMR frame format combinations in terms of number of bits in each field.
304
+
305
+ **Table 7. Number of bits for different fields in different AMR frame compositions**
306
+
307
+ | Frame Type Index | Frame Type | Frame Quality Indicator | Mode Indication | Mode Request | Codec CRC | Class A | Class B | Class C | Total |
308
+ |------------------|-----------------------|-------------------------|-----------------|--------------|-----------|---------|---------|---------|-------|
309
+ | | <b>AMR Core Frame</b> | | | | | | | | |
310
+ | 0 | 4 | 1 | 3 | 3 | 8 | 42 | 53 | 0 | 114 |
311
+ | 1 | 4 | 1 | 3 | 3 | 8 | 49 | 54 | 0 | 122 |
312
+ | 2 | 4 | 1 | 3 | 3 | 8 | 55 | 63 | 0 | 137 |
313
+ | 3 | 4 | 1 | 3 | 3 | 8 | 58 | 76 | 0 | 153 |
314
+ | 4 | 4 | 1 | 3 | 3 | 8 | 61 | 87 | 0 | 167 |
315
+ | 5 | 4 | 1 | 3 | 3 | 8 | 75 | 84 | 0 | 178 |
316
+ | 6 | 4 | 1 | 3 | 3 | 8 | 65 | 99 | 40 | 223 |
317
+ | 7 | 4 | 1 | 3 | 3 | 8 | 81 | 103 | 60 | 263 |
318
+ | 8 | 4 | 1 | 3 | 3 | 8 | 39 | 0 | 0 | 58 |
319
+ | 9 | 4 | 1 | 3 | 3 | 8 | 43 | 0 | 0 | 62 |
320
+ | 10 | 4 | 1 | 3 | 3 | 8 | 38 | 0 | 0 | 57 |
321
+ | 11 | 4 | 1 | 3 | 3 | 8 | 37 | 0 | 0 | 56 |
322
+ | 12 | Not used | | | | | | | | |
323
+ | 13 | Not used | | | | | | | | |
324
+ | 14 | Not used | | | | | | | | |
325
+ | 15 | 4 | 0 | 0 | 0 | 0 | 0 | 0 | 0 | 4 |
326
+
327
+ ## 4.4 GSM-EFR Frame Composition
328
+
329
+ This subclause contains the description of the generic GSM-EFR Frame of Figure 1. The descriptions for the generic GSM-EFR Frame with speech bits and with comfort noise bit are given separately.
330
+
331
+ ### 4.4.1 GSM-EFR Frame with speech bits
332
+
333
+ The generic GSM-EFR frame for speech data bits is formed like for the AMR mode 12.2 kbit/s.
334
+
335
+ The same Frame Type (Frame Type 7) is used also for GSM-EFR. The Mode Indication and Mode Request fields are set to “7”. The GSM-EFR Core Frame for speech data bits is identical to the AMR Core Frame for speech data bits in the AMR mode with 12.2 kbit/s.
336
+
337
+ ### 4.4.2 GSM-EFR Frame with comfort noise bits
338
+
339
+ The GSM-EFR Frame content for the additional frame type with Frame Type Index 9 in Table 1a are described in this subclause. This consists of the frame related to GSM-EFR Comfort Noise Aspects as specified in [4] and [5].
340
+
341
+ The comfort noise bits are all mapped to Class A. Classes B and C are not used (see Table 7).
342
+
343
+ The contents of GSM-EFR SID is the Comfort Noise Parameters ( $s(i)$ ) as defined in [4]. The Comfort noise parameters are computed as described in [5] 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 [4] (Table 6). The notation $d(j) = \{d(0) \dots d(42)\}$ is local to the present document and is formed as defined by the pseudo code below.
344
+
345
+ ```
346
+
347
+ for j = 0 to 37
348
+ $d(j) := s(j+1)$ ; /* LSP parameters in s(1) to s(38) */
349
+
350
+ for j = 38 to 42
351
+ $d(j) := s(j+49)$ ; /* fixed codebook gain parameter in s(87)-s(91) */
352
+
353
+ ```
354
+
355
+ Table 8 shows the composition for the generic GSM-EFR SID frame.
356
+
357
+ **Table 8: Mapping of the GSM-EFR SID frame into the generic AMR frame format, AMR IF1,
358
+ Example of a good GSM-EFR SID frame (FQI=1).**
359
+
360
+ | Octet | MSB | Mapping of bits for GSM-EFR SID | | | | | | LSB |
361
+ |-------|----------------------|---------------------------------|-------------|--------|-------------|----------------------|-------------|-------------|
362
+ | | bit 8 | bit 7 | bit 6 | bit 5 | bit 4 | bit 3 | bit 2 | bit 1 |
363
+ | 1 | Frame Type (=9) | | | | FQI | Mode Indication (=7) | | |
364
+ | | 1 | 0 | 0 | 1 | 1 | 1 | 1 | 1 |
365
+ | 2 | Mode Request (=7) | | | spare | | | | |
366
+ | | 1 | 1 | 1 | 0 | 0 | 0 | 0 | 0 |
367
+ | 3 | Codec CRC | | | | | | | |
368
+ | | CRC(7) | CRC(6) | CRC(5) | CRC(4) | CRC(3) | CRC(2) | CRC(1) | CRC(0) |
369
+ | 4 | Core Frame (octet 1) | | | | | | | |
370
+ | | d(0)=s(1) | d(1)=s(2) | d(2) | d(3) | d(4) | d(5) | d(6) | d(7) |
371
+ | 5..7 | | | | | | | | |
372
+ | 8 | | | | | | | | |
373
+ | | d(32) | d(33) | d(34) | d(35) | d(36)=s(37) | d(37)=s(38) | d(38)=s(87) | d(39)=s(88) |
374
+ | 9 | spare | | | | | | | |
375
+ | | d(40)=s(89) | d(41)=s(90) | d(42)=s(91) | 0 | 0 | 0 | 0 | 0 |
376
+
377
+ # Annex A (informative): AMR Interface Format 2 (with octet alignment)
378
+
379
+ This annex defines an octet-aligned frame format for the AMR codec. This format is useful, for example, when the AMR codec is used in connection with applicable ITU-T H-series of recommendations. The format is referred to as AMR Interface Format 2 (AMR IF2).
380
+
381
+ The AMR IF2 frame is formed by concatenation of the 4-bit Frame Type field (as defined for AMR IF1 in subclause 4.1.1) and the AMR Core Frame (as defined for AMR IF1 in subclause 4.2) as shown in figure A.1. The length of the AMR Core Frame field depends on the particular Frame Type. The total number of bits in the AMR IF2 speech frames in the different modes is typically not a multiple of eight and bit stuffing is needed to achieve an octet structure.
382
+
383
+ ![Figure A.1: Frame structure for AMR IF2. The diagram shows a vertical stack of five boxes representing the frame components. From top to bottom: 'Frame Type (4 bits)', 'Class A bits', 'Class B bits', 'Class C bits', and 'Bit Stuffing'. A line connects the 'Class B bits' box to a separate box labeled 'AMR Core Frame (speech or comfort noise data)'.](ddc7460821484f1ae2835c67955c554c_img.jpg)
384
+
385
+ Figure A.1: Frame structure for AMR IF2. The diagram shows a vertical stack of five boxes representing the frame components. From top to bottom: 'Frame Type (4 bits)', 'Class A bits', 'Class B bits', 'Class C bits', and 'Bit Stuffing'. A line connects the 'Class B bits' box to a separate box labeled 'AMR Core Frame (speech or comfort noise data)'.
386
+
387
+ **Figure A.1: Frame structure for AMR IF2**
388
+
389
+ Table A.1a shows an example how the AMR 6.7 kbit/s mode is mapped into AMR IF2. The four LSBs of the first octet (octet 1) consist of the Frame Type(=3) for the AMR 6.7 kbit/s mode (see table 1a in AMR IF1 specification). This data field is followed by the 134 AMR Core Frame speech bits ( $d(0)\dots d(133)$ ) which consist of 58 Class A bits and 76 Class B bits as described in table 2 for AMR IF1. This results in a total of 138 bits and 6 bits are needed for Bit Stuffing to arrive to the closest multiple of 8 which is 144 bits.
390
+
391
+ **Table A.1a: Example mapping of the AMR speech coding mode 6.7kbit/s into AMR IF2.
392
+ The bits used for Bit Stuffing are denoted as UB (for "unused bit").**
393
+
394
+ | Octet | MSB | Mapping of bits<br>AMR 6.7 | | | | | | LSB |
395
+ |-------|---------------|----------------------------|-------|-------|------------------|-------|--------|--------|
396
+ | | bit 8 | bit 7 | bit 6 | bit 5 | bit 4 | bit 3 | bit 2 | bit 1 |
397
+ | 1 | | | | | Frame Type (= 3) | | | |
398
+ | | | | | | MSB | ..... | | LSB |
399
+ | 2 | d(3) | d(2) | d(1) | d(0) | 0 | 0 | 1 | 1 |
400
+ | 3 | d(11) | d(10) | d(9) | d(8) | d(7) | d(6) | d(5) | d(4) |
401
+ | 18 | ... | ... | ... | ... | ... | ... | ... | d(12) |
402
+ | 19 | Stuffing bits | | | | | | d(133) | d(132) |
403
+ | | UB | UB | UB | UB | UB | UB | d(133) | d(132) |
404
+
405
+ Table A.1b shows the composition of AMR IF2 frames for all Frame Types in terms of how many bits are used for each field of figure A.1.
406
+
407
+ Tables A.2 to A.5 specify how the AMR Core Frame comfort noise bits of Frame Types 8-11 are mapped to AMR IF2. Table A.6 specifies the mapping for an empty frame ("no transmission").
408
+
409
+ **Table A.1b: Composition of AMR IF2 Frames for all Frame Types.**
410
+
411
+ | Frame Type Index | Frame content | Number of bits in Frame Type | Number of Bits in AMR Core Frame | Number of Bits in Bit Stuffing | Number of octets (N) |
412
+ |------------------|----------------|------------------------------|----------------------------------|--------------------------------|----------------------|
413
+ | 0 | AMR 4,75 | 4 | 95 | 5 | 13 |
414
+ | 1 | AMR 5,15 | 4 | 103 | 5 | 14 |
415
+ | 2 | AMR 5,90 | 4 | 118 | 6 | 16 |
416
+ | 3 | AMR 6,70 | 4 | 134 | 6 | 18 |
417
+ | 4 | AMR 7,40 | 4 | 148 | 0 | 19 |
418
+ | 5 | AMR 7,95 | 4 | 159 | 5 | 21 |
419
+ | 6 | AMR 10,2 | 4 | 204 | 0 | 26 |
420
+ | 7 | AMR 12,2 | 4 | 244 | 0 | 31 |
421
+ | 8 | AMR SID | 4 | 39 | 5 | 6 |
422
+ | 9 | GSM-EFR SID | 4 | 43 | 1 | 6 |
423
+ | 10 | TDMA-EFR SID | 4 | 38 | 6 | 6 |
424
+ | 11 | PDC-EFR SID | 4 | 37 | 7 | 6 |
425
+ | 12-14 | For future use | - | - | - | - |
426
+ | 15 | No Data | 4 | 0 | 4 | 1 |
427
+
428
+ **Table A.2: Mapping of bits for Frame Type 8 (AMR SID)
429
+ (Bits s1 to s35 refer to TS 26.092)**
430
+
431
+ | | MSB | Mapping of bits<br>AMR SID | | | | | | LSB |
432
+ |-------|----------------------------------------|----------------------------------------|-------|-------|------------------|--------------------------|-------|----------------------------------------|
433
+ | Octet | bit 8 | bit 7 | bit 6 | bit 5 | bit 4 | bit 3 | bit 2 | bit 1 |
434
+ | 1 | Index of 1 <sup>st</sup> LSF subvector | index of LSF reference vector | | | Frame Type (= 8) | | | |
435
+ | | s4 | s3 | s2 | s1 | MSB<br>1 | .....<br>0 | 0 | LSB<br>0 |
436
+ | 2 | Index of 2 <sup>nd</sup> LSF subvector | index of 1 <sup>st</sup> LSF subvector | | | | | | |
437
+ | | s12 | s11 | s10 | s9 | s8 | s7 | s6 | s5 |
438
+ | 3 | | index of 2 <sup>nd</sup> LSF subvector | | | | | | |
439
+ | | s20 | s19 | s18 | s17 | s16 | s15 | s14 | s13 |
440
+ | 4 | | index of 3 <sup>rd</sup> LSF subvector | | | | | | |
441
+ | | s28 | s27 | s26 | s25 | s24 | s23 | s22 | s21 |
442
+ | 5 | SID Type Indicator | frame energy | | | | | | index of 3 <sup>rd</sup> LSF subvector |
443
+ | | t1 | s35 | s34 | s33 | s32 | s31 | s30 | s29 |
444
+ | 6 | | Stuffing bits | | | | Mode Indication<br>mi(i) | | |
445
+ | | UB | UB | UB | UB | UB | MSB<br>mi(2) | mi(1) | LSB<br>mi(0) |
446
+
447
+ Definitions of additional descriptor bits needed for the silence descriptor in the table are as follows: SID-type Indicator STI is {0=SID\_FIRST, 1=SID\_UPDATE }, Mode Indication (mi(0)- mi(2)) is the AMR codec mode according to the first eight entries in table 1a.
448
+
449
+ **Table A.3: Mapping of bits for Frame Type 9 (GSM-EFR SID)**
450
+ (Bits s1 to s91 refer to GSM 46.060)
451
+
452
+ | | MSB | Mapping of bits<br>GSM-EFR SID | | | | | | LSB |
453
+ |-------|----------------------------------------|--------------------------------|-------|-------|----------------------------------------|----------------------------------------|----------------------------------------|-------|
454
+ | Octet | bit 8 | bit 7 | bit 6 | bit 5 | bit 4 | bit 3 | bit 2 | bit 1 |
455
+ | 1 | Index of 1 <sup>st</sup> LSF subMatrix | | | | Frame Type (= 9) | | | |
456
+ | | s4 | s3 | s2 | s1 | 1 | 0 | 0 | 1 |
457
+ | 2 | Index of 2 <sup>nd</sup> LSF submatrix | | | | index of 1 <sup>st</sup> LSF subMatrix | | | |
458
+ | | s12 | s11 | s10 | s9 | s8 | s7 | s6 | s5 |
459
+ | 3 | Index of 3 <sup>rd</sup> LSF submatrix | | | | Index of 2 <sup>nd</sup> LSF submatrix | | | |
460
+ | | s20 | s19 | s18 | s17 | s16 | s15 | s14 | s13 |
461
+ | 4 | index of 4 <sup>th</sup> LSF submatrix | | | | sign of 3 <sup>rd</sup> LSF submatrix | index of 3 <sup>rd</sup> LSF submatrix | | |
462
+ | | s28 | s27 | s26 | s25 | s24 | s23 | s22 | s21 |
463
+ | 5 | index of 5 <sup>th</sup> LSF submatrix | | | | index of 4 <sup>th</sup> LSF submatrix | | | |
464
+ | | s36 | s35 | s34 | s33 | s32 | s31 | s30 | s29 |
465
+ | 6 | Stuffing bit | fixed codebook gain | | | | | index of 5 <sup>th</sup> LSF submatrix | |
466
+ | | UB | s91 | s90 | s89 | s88 | s87 | s38 | s37 |
467
+
468
+ **Table A.4: Mapping of bits for Frame Type 10 (TDMA-EFR SID)**
469
+ (Bits cn0 to cn37 refer to IS-641-A)
470
+
471
+ | | MSB | Mapping of bits<br>TDMA-EFR SID | | | | | | LSB |
472
+ |-------|-----------------------------------------|---------------------------------|----------------------------------------|----------------------------------------|----------------------------------------|-------|-----------------------------------------|-------|
473
+ | Octet | bit 8 | bit 7 | bit 6 | bit 5 | bit 4 | bit 3 | bit 2 | bit 1 |
474
+ | 1 | Index of 1 <sup>st</sup> LSF subvector | | | | Frame Type (= 10) | | | |
475
+ | | cn3 | cn2 | cn1 | cn0 | 1 | 0 | 1 | 0 |
476
+ | 2 | Index of 2 <sup>nd</sup> LSF subvector | | | | index of 1 <sup>st</sup> LSF subvector | | | |
477
+ | | cn11 | Cn10 | cn9 | cn8 | cn7 | cn6 | cn5 | cn4 |
478
+ | 3 | Index of 3 <sup>rd</sup> LSF subvector | | | Index of 2 <sup>nd</sup> LSF subvector | | | | |
479
+ | | cn19 | cn18 | cn17 | cn16 | cn15 | cn14 | cn13 | cn12 |
480
+ | 4 | Random Excitation Gain | | index of 3 <sup>rd</sup> LSF subvector | | | | | |
481
+ | | cn27 | cn26 | cn25 | cn24 | cn23 | cn22 | cn21 | cn20 |
482
+ | 5 | Index of 1 <sup>st</sup> RESC parameter | | Random Excitation Gain | | | | | |
483
+ | | cn35 | cn34 | cn33 | cn32 | cn31 | cn30 | cn29 | cn28 |
484
+ | 6 | Stuffing bits | | | | | | Index of 2 <sup>nd</sup> RESC parameter | |
485
+ | | UB | UB | UB | UB | UB | UB | cn37 | cn36 |
486
+
487
+ **Table A.5: Mapping of bits for Frame Type 11 (PDC-EFR SID)**
488
+ (Bits s1 to s35 refer to ARIB xx)
489
+
490
+ | | MSB | Mapping of bits<br>PDC-EFR SID | | | | | | LSB |
491
+ |-------|----------------------------------------|----------------------------------------|-------|-------|-------------------|-------|-------|----------------------------------------|
492
+ | Octet | bit 8 | bit 7 | bit 6 | bit 5 | bit 4 | bit 3 | bit 2 | bit 1 |
493
+ | 1 | index of 1 <sup>st</sup> LSF subvector | index of LSF reference vector | | | Frame Type (= 11) | | | |
494
+ | | s4 | s3 | s2 | s1 | 1 | 0 | 1 | 1 |
495
+ | 2 | index of 2 <sup>nd</sup> LSF subvector | index of 1 <sup>st</sup> LSF subvector | | | | | | |
496
+ | | s12 | s11 | s10 | s9 | s8 | s7 | s6 | s5 |
497
+ | 3 | index of 2 <sup>nd</sup> LSF subvector | | | | | | | |
498
+ | | s20 | s19 | s18 | s17 | s16 | s15 | s14 | s13 |
499
+ | 4 | index of 3 <sup>rd</sup> LSF subvector | | | | | | | |
500
+ | | s28 | s27 | s26 | s25 | s24 | s23 | s22 | s21 |
501
+ | 5 | SID type | frame energy | | | | | | Index of 3 <sup>rd</sup> LSF subvector |
502
+ | | t1 | s35 | s34 | s33 | s32 | s31 | s30 | s29 |
503
+ | 6 | Stuffing bits | | | | | | | SID type |
504
+ | | UB | UB | UB | UB | UB | UB | UB | t2 |
505
+
506
+ Definition of additional descriptor bits needed for the table is as follows: SID-type is {0=POST0, 1=POST1(SID\_UPDATE), 2=PRE, 3=POST1\_BAD }, where LSB of SID\_type is t1 and MSB of SID-type is t2.
507
+
508
+ **Table A.6: Mapping of bit for Frame Type 15 (No Data)**
509
+
510
+ | | MSB | Mapping of Bits<br>No Data | | | | | | LSB |
511
+ |-------|---------------|----------------------------|-------|-------|-------------------|-------|-------|-------|
512
+ | Octet | bit 8 | bit 7 | bit 6 | bit 5 | bit 4 | bit 3 | bit 2 | bit 1 |
513
+ | 1 | Stuffing bits | | | | Frame Type (= 15) | | | |
514
+ | | UB | UB | UB | UB | 1 | 1 | 1 | 1 |
515
+
516
+ # Annex B (normative): Tables for AMR Core Frame bit ordering
517
+
518
+ This annex contains the tables required for ordering the AMR Core Frame speech bits corresponding to the different AMR modes. These tables represent $table_m(j)$ in subclause 4.2.1 where $m=0..7$ is the AMR mode. The tables are read from left to right so that the first element (top left corner) of the table has index 0 and the last element (the rightmost element of the last row) has the index $K-1$ where $K$ is the total number of speech bits in the specific mode. For example, $table_0(20)=27$ , as defined in table B.1.
519
+
520
+ **Table B.1: Ordering of the speech encoder bits for the 4.75 kbit/s mode: $table_0(j)$**
521
+
522
+ | $j=0$ | $j=1$ | $j=2$ | ... | ... | ... | ... | ... | ... | ... |
523
+ |-------|-------|-------|-----|-----|-----|-----|-----|-----|-----|
524
+ | 0 | 1 | 2 | 3 | 4 | 5 | 6 | 7 | 8 | 9 |
525
+ | 10 | 11 | 12 | 13 | 14 | 15 | 23 | 24 | 25 | 26 |
526
+ | 27 | 28 | 48 | 49 | 61 | 62 | 82 | 83 | 47 | 46 |
527
+ | 45 | 44 | 81 | 80 | 79 | 78 | 17 | 18 | 20 | 22 |
528
+ | 77 | 76 | 75 | 74 | 29 | 30 | 43 | 42 | 41 | 40 |
529
+ | 38 | 39 | 16 | 19 | 21 | 50 | 51 | 59 | 60 | 63 |
530
+ | 64 | 72 | 73 | 84 | 85 | 93 | 94 | 32 | 33 | 35 |
531
+ | 36 | 53 | 54 | 56 | 57 | 66 | 67 | 69 | 70 | 87 |
532
+ | 88 | 90 | 91 | 34 | 55 | 68 | 89 | 37 | 58 | 71 |
533
+ | 92 | 31 | 52 | 65 | 86 | | | | | |
534
+
535
+ **Table B.2: Ordering of the speech encoder bits for the 5.15 kbit/s mode: $table_1(j)$**
536
+
537
+ | | | | | | | | | | |
538
+ |----|----|----|-----|-----|-----|----|----|----|----|
539
+ | 7 | 6 | 5 | 4 | 3 | 2 | 1 | 0 | 15 | 14 |
540
+ | 13 | 12 | 11 | 10 | 9 | 8 | 23 | 24 | 25 | 26 |
541
+ | 27 | 46 | 65 | 84 | 45 | 44 | 43 | 64 | 63 | 62 |
542
+ | 83 | 82 | 81 | 102 | 101 | 100 | 42 | 61 | 80 | 99 |
543
+ | 28 | 47 | 66 | 85 | 18 | 41 | 60 | 79 | 98 | 29 |
544
+ | 48 | 67 | 17 | 20 | 22 | 40 | 59 | 78 | 97 | 21 |
545
+ | 30 | 49 | 68 | 86 | 19 | 16 | 87 | 39 | 38 | 58 |
546
+ | 57 | 77 | 35 | 54 | 73 | 92 | 76 | 96 | 95 | 36 |
547
+ | 55 | 74 | 93 | 32 | 51 | 33 | 52 | 70 | 71 | 89 |
548
+ | 90 | 31 | 50 | 69 | 88 | 37 | 56 | 75 | 94 | 34 |
549
+ | 53 | 72 | 91 | | | | | | | |
550
+
551
+ **Table B.3: Ordering of the speech encoder bits for the 5.9 kbit/s mode: $table_2(j)$**
552
+
553
+ | | | | | | | | | | |
554
+ |-----|-----|-----|-----|-----|-----|----|-----|-----|-----|
555
+ | 0 | 1 | 4 | 5 | 3 | 6 | 7 | 2 | 13 | 15 |
556
+ | 8 | 9 | 11 | 12 | 14 | 10 | 16 | 28 | 74 | 29 |
557
+ | 75 | 27 | 73 | 26 | 72 | 30 | 76 | 51 | 97 | 50 |
558
+ | 71 | 96 | 117 | 31 | 77 | 52 | 98 | 49 | 70 | 95 |
559
+ | 116 | 53 | 99 | 32 | 78 | 33 | 79 | 48 | 69 | 94 |
560
+ | 115 | 47 | 68 | 93 | 114 | 46 | 67 | 92 | 113 | 19 |
561
+ | 21 | 23 | 22 | 18 | 17 | 20 | 24 | 111 | 43 | 89 |
562
+ | 110 | 64 | 65 | 44 | 90 | 25 | 45 | 66 | 91 | 112 |
563
+ | 54 | 100 | 40 | 61 | 86 | 107 | 39 | 60 | 85 | 106 |
564
+ | 36 | 57 | 82 | 103 | 35 | 56 | 81 | 102 | 34 | 55 |
565
+ | 80 | 101 | 42 | 63 | 88 | 109 | 41 | 62 | 87 | 108 |
566
+ | 38 | 59 | 84 | 105 | 37 | 58 | 83 | 104 | | |
567
+
568
+ **Table B.4: Ordering of the speech encoder bits for the 6.7 kbit/s mode: $table_3(j)$**
569
+
570
+ | | | | | | | | | | |
571
+ |-----|-----|-----|-----|-----|-----|-----|-----|-----|-----|
572
+ | 0 | 1 | 4 | 3 | 5 | 6 | 13 | 7 | 2 | 8 |
573
+ | 9 | 11 | 15 | 12 | 14 | 10 | 28 | 82 | 29 | 83 |
574
+ | 27 | 81 | 26 | 80 | 30 | 84 | 16 | 55 | 109 | 56 |
575
+ | 110 | 31 | 85 | 57 | 111 | 48 | 73 | 102 | 127 | 32 |
576
+ | 86 | 51 | 76 | 105 | 130 | 52 | 77 | 106 | 131 | 58 |
577
+ | 112 | 33 | 87 | 19 | 23 | 53 | 78 | 107 | 132 | 21 |
578
+ | 22 | 18 | 17 | 20 | 24 | 25 | 50 | 75 | 104 | 129 |
579
+ | 47 | 72 | 101 | 126 | 54 | 79 | 108 | 133 | 46 | 71 |
580
+ | 100 | 125 | 128 | 103 | 74 | 49 | 45 | 70 | 99 | 124 |
581
+ | 42 | 67 | 96 | 121 | 39 | 64 | 93 | 118 | 38 | 63 |
582
+ | 92 | 117 | 35 | 60 | 89 | 114 | 34 | 59 | 88 | 113 |
583
+ | 44 | 69 | 98 | 123 | 43 | 68 | 97 | 122 | 41 | 66 |
584
+ | 95 | 120 | 40 | 65 | 94 | 119 | 37 | 62 | 91 | 116 |
585
+ | 36 | 61 | 90 | 115 | | | | | | |
586
+
587
+ **Table B.5: Ordering of the speech encoder bits for the 7.4 kbit/s mode: $table_4(j)$**
588
+
589
+ | | | | | | | | | | |
590
+ |-----|-----|-----|-----|-----|-----|-----|-----|-----|-----|
591
+ | 0 | 1 | 2 | 3 | 4 | 5 | 6 | 7 | 8 | 9 |
592
+ | 10 | 11 | 12 | 13 | 14 | 15 | 16 | 26 | 87 | 27 |
593
+ | 88 | 28 | 89 | 29 | 90 | 30 | 91 | 51 | 80 | 112 |
594
+ | 141 | 52 | 81 | 113 | 142 | 54 | 83 | 115 | 144 | 55 |
595
+ | 84 | 116 | 145 | 58 | 119 | 59 | 120 | 21 | 22 | 23 |
596
+ | 17 | 18 | 19 | 31 | 60 | 92 | 121 | 56 | 85 | 117 |
597
+ | 146 | 20 | 24 | 25 | 50 | 79 | 111 | 140 | 57 | 86 |
598
+ | 118 | 147 | 49 | 78 | 110 | 139 | 48 | 77 | 53 | 82 |
599
+ | 114 | 143 | 109 | 138 | 47 | 76 | 108 | 137 | 32 | 33 |
600
+ | 61 | 62 | 93 | 94 | 122 | 123 | 41 | 42 | 43 | 44 |
601
+ | 45 | 46 | 70 | 71 | 72 | 73 | 74 | 75 | 102 | 103 |
602
+ | 104 | 105 | 106 | 107 | 131 | 132 | 133 | 134 | 135 | 136 |
603
+ | 34 | 63 | 95 | 124 | 35 | 64 | 96 | 125 | 36 | 65 |
604
+ | 97 | 126 | 37 | 66 | 98 | 127 | 38 | 67 | 99 | 128 |
605
+ | 39 | 68 | 100 | 129 | 40 | 69 | 101 | 130 | | |
606
+
607
+ **Table B.6: Ordering of the speech encoder bits for the 7.95 kbit/s mode: $table_5(j)$**
608
+
609
+ | | | | | | | | | | |
610
+ |-----|-----|-----|-----|-----|-----|-----|-----|-----|-----|
611
+ | 8 | 7 | 6 | 5 | 4 | 3 | 2 | 14 | 16 | 9 |
612
+ | 10 | 12 | 13 | 15 | 11 | 17 | 20 | 22 | 24 | 23 |
613
+ | 19 | 18 | 21 | 56 | 88 | 122 | 154 | 57 | 89 | 123 |
614
+ | 155 | 58 | 90 | 124 | 156 | 52 | 84 | 118 | 150 | 53 |
615
+ | 85 | 119 | 151 | 27 | 93 | 28 | 94 | 29 | 95 | 30 |
616
+ | 96 | 31 | 97 | 61 | 127 | 62 | 128 | 63 | 129 | 59 |
617
+ | 91 | 125 | 157 | 32 | 98 | 64 | 130 | 1 | 0 | 25 |
618
+ | 26 | 33 | 99 | 34 | 100 | 65 | 131 | 66 | 132 | 54 |
619
+ | 86 | 120 | 152 | 60 | 92 | 126 | 158 | 55 | 87 | 121 |
620
+ | 153 | 117 | 116 | 115 | 46 | 78 | 112 | 144 | 43 | 75 |
621
+ | 109 | 141 | 40 | 72 | 106 | 138 | 36 | 68 | 102 | 134 |
622
+ | 114 | 149 | 148 | 147 | 146 | 83 | 82 | 81 | 80 | 51 |
623
+ | 50 | 49 | 48 | 47 | 45 | 44 | 42 | 39 | 35 | 79 |
624
+ | 77 | 76 | 74 | 71 | 67 | 113 | 111 | 110 | 108 | 105 |
625
+ | 101 | 145 | 143 | 142 | 140 | 137 | 133 | 41 | 73 | 107 |
626
+ | 139 | 37 | 69 | 103 | 135 | 38 | 70 | 104 | 136 | |
627
+
628
+ **Table B.7: Ordering of the speech encoder bits for the 10.2 kbit/s mode: $table_6(j)$**
629
+
630
+ | | | | | | | | | | |
631
+ |-----|-----|-----|-----|-----|-----|-----|-----|-----|-----|
632
+ | 7 | 6 | 5 | 4 | 3 | 2 | 1 | 0 | 16 | 15 |
633
+ | 14 | 13 | 12 | 11 | 10 | 9 | 8 | 26 | 27 | 28 |
634
+ | 29 | 30 | 31 | 115 | 116 | 117 | 118 | 119 | 120 | 72 |
635
+ | 73 | 161 | 162 | 65 | 68 | 69 | 108 | 111 | 112 | 154 |
636
+ | 157 | 158 | 197 | 200 | 201 | 32 | 33 | 121 | 122 | 74 |
637
+ | 75 | 163 | 164 | 66 | 109 | 155 | 198 | 19 | 23 | 21 |
638
+ | 22 | 18 | 17 | 20 | 24 | 25 | 37 | 36 | 35 | 34 |
639
+ | 80 | 79 | 78 | 77 | 126 | 125 | 124 | 123 | 169 | 168 |
640
+ | 167 | 166 | 70 | 67 | 71 | 113 | 110 | 114 | 159 | 156 |
641
+ | 160 | 202 | 199 | 203 | 76 | 165 | 81 | 82 | 92 | 91 |
642
+ | 93 | 83 | 95 | 85 | 84 | 94 | 101 | 102 | 96 | 104 |
643
+ | 86 | 103 | 87 | 97 | 127 | 128 | 138 | 137 | 139 | 129 |
644
+ | 141 | 131 | 130 | 140 | 147 | 148 | 142 | 150 | 132 | 149 |
645
+ | 133 | 143 | 170 | 171 | 181 | 180 | 182 | 172 | 184 | 174 |
646
+ | 173 | 183 | 190 | 191 | 185 | 193 | 175 | 192 | 176 | 186 |
647
+ | 38 | 39 | 49 | 48 | 50 | 40 | 52 | 42 | 41 | 51 |
648
+ | 58 | 59 | 53 | 61 | 43 | 60 | 44 | 54 | 194 | 179 |
649
+ | 189 | 196 | 177 | 195 | 178 | 187 | 188 | 151 | 136 | 146 |
650
+ | 153 | 134 | 152 | 135 | 144 | 145 | 105 | 90 | 100 | 107 |
651
+ | 88 | 106 | 89 | 98 | 99 | 62 | 47 | 57 | 64 | 45 |
652
+ | 63 | 46 | 55 | 56 | | | | | | |
653
+
654
+ **Table B.8: Ordering of the speech encoder bits for the 12.2 kbit/s mode: $table_7(j)$**
655
+
656
+ | | | | | | | | | | |
657
+ |-----|-----|-----|-----|-----|-----|-----|-----|-----|-----|
658
+ | 0 | 1 | 2 | 3 | 4 | 5 | 6 | 7 | 8 | 9 |
659
+ | 10 | 11 | 12 | 13 | 14 | 23 | 15 | 16 | 17 | 18 |
660
+ | 19 | 20 | 21 | 22 | 24 | 25 | 26 | 27 | 28 | 38 |
661
+ | 141 | 39 | 142 | 40 | 143 | 41 | 144 | 42 | 145 | 43 |
662
+ | 146 | 44 | 147 | 45 | 148 | 46 | 149 | 47 | 97 | 150 |
663
+ | 200 | 48 | 98 | 151 | 201 | 49 | 99 | 152 | 202 | 86 |
664
+ | 136 | 189 | 239 | 87 | 137 | 190 | 240 | 88 | 138 | 191 |
665
+ | 241 | 91 | 194 | 92 | 195 | 93 | 196 | 94 | 197 | 95 |
666
+ | 198 | 29 | 30 | 31 | 32 | 33 | 34 | 35 | 50 | 100 |
667
+ | 153 | 203 | 89 | 139 | 192 | 242 | 51 | 101 | 154 | 204 |
668
+ | 55 | 105 | 158 | 208 | 90 | 140 | 193 | 243 | 59 | 109 |
669
+ | 162 | 212 | 63 | 113 | 166 | 216 | 67 | 117 | 170 | 220 |
670
+ | 36 | 37 | 54 | 53 | 52 | 58 | 57 | 56 | 62 | 61 |
671
+ | 60 | 66 | 65 | 64 | 70 | 69 | 68 | 104 | 103 | 102 |
672
+ | 108 | 107 | 106 | 112 | 111 | 110 | 116 | 115 | 114 | 120 |
673
+ | 119 | 118 | 157 | 156 | 155 | 161 | 160 | 159 | 165 | 164 |
674
+ | 163 | 169 | 168 | 167 | 173 | 172 | 171 | 207 | 206 | 205 |
675
+ | 211 | 210 | 209 | 215 | 214 | 213 | 219 | 218 | 217 | 223 |
676
+ | 222 | 221 | 73 | 72 | 71 | 76 | 75 | 74 | 79 | 78 |
677
+ | 77 | 82 | 81 | 80 | 85 | 84 | 83 | 123 | 122 | 121 |
678
+ | 126 | 125 | 124 | 129 | 128 | 127 | 132 | 131 | 130 | 135 |
679
+ | 134 | 133 | 176 | 175 | 174 | 179 | 178 | 177 | 182 | 181 |
680
+ | 180 | 185 | 184 | 183 | 188 | 187 | 186 | 226 | 225 | 224 |
681
+ | 229 | 228 | 227 | 232 | 231 | 230 | 235 | 234 | 233 | 238 |
682
+ | 237 | 236 | 96 | 199 | | | | | | |
683
+
684
+ # Annex C (informative): Change history
685
+
686
+ | Change history | | | | | | | |
687
+ |----------------|---------|-----------|-----|-----|-----------------------------------------------------------------------------------------------------------------|--------|--------|
688
+ | Date | TSG SA# | TSG Doc. | CR | Rev | Subject/Comment | Old | New |
689
+ | 1999-12 | 6 | SP-990562 | | | Approved at TSG-SA#6 Plenary | | 3.0.0 |
690
+ | 2000-03 | 7 | SP-000025 | 001 | | Correction of indices in Annex B table | 3.0.0 | 3.1.0 |
691
+ | 2000-03 | 7 | SP-000025 | 002 | | Addition of comfort noise bit ordering | 3.0.0 | 3.1.0 |
692
+ | 2000-03 | 7 | SP-000025 | 003 | | Correction of table indexing for AMR Core Frame class division | 3.0.0 | 3.1.0 |
693
+ | 2000-03 | 7 | SP-000025 | 004 | | Clarification of bit transmission order for AMR frame structure parameters for AMR Interface Format 1 (AMR IF1) | 3.0.0 | 3.1.0 |
694
+ | 2001-03 | 11 | | | | Version for Release 4 | | 4.0.0 |
695
+ | 2001-06 | 12 | SP-010305 | 006 | | Correction to SID Frame Mapping | 4.0.0 | 4.1.0 |
696
+ | 2002-03 | 15 | SP-020077 | 008 | | Correction of AMR codec output bitstream | 4.1.0 | 4.2.0 |
697
+ | 2002-06 | 16 | | | | Version for Release 5 | 4.2.0 | 5.0.0 |
698
+ | | | | | | | | |
699
+ | 2004-09 | 25 | SP-040644 | 009 | 2 | Generic Frame Structure for GSM-EFR SID | 5.0.0 | 6.0.0 |
700
+ | 2004-09 | 25 | SP-040644 | 010 | 1 | Error Corrections | 5.0.0 | 6.0.0 |
701
+ | 2007-06 | 36 | | | | Version for Release 7 | 6.0.0 | 7.0.0 |
702
+ | 2008-12 | 42 | | | | Version for Release 8 | 7.0.0 | 8.0.0 |
703
+ | 2009-12 | 46 | | | | Version for Release 9 | 8.0.0 | 9.0.0 |
704
+ | 2011-03 | 51 | | | | Version for Release 10 | 9.0.0 | 10.0.0 |
705
+ | 2012-09 | 57 | | | | Version for Release 11 | 10.0.0 | 11.0.0 |
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1
+
2
+
3
+
4
+
5
+
6
+
7
+ # --- Contents
8
+
9
+ | | |
10
+ |-------------------------------------------------------------------------|----|
11
+ | Foreword ..... | 4 |
12
+ | 1 Scope..... | 5 |
13
+ | 2 References..... | 5 |
14
+ | 3 Definitions and abbreviations ..... | 8 |
15
+ | 3.1 Definitions..... | 8 |
16
+ | 3.2 Abbreviations ..... | 8 |
17
+ | 4 Media formats ..... | 8 |
18
+ | 4.1 Text..... | 9 |
19
+ | 4.2 Speech ..... | 9 |
20
+ | 4.3 Audio..... | 9 |
21
+ | 4.4 Synthetic audio..... | 10 |
22
+ | 4.5 Still Image ..... | 10 |
23
+ | 4.6 Bitmap graphics..... | 10 |
24
+ | 4.7 Video ..... | 10 |
25
+ | 4.8 Vector graphics ..... | 11 |
26
+ | 4.9 File Format for video and associated speech/audio media types ..... | 11 |
27
+ | 4.10 Media synchronization and presentation format ..... | 11 |
28
+ | 4.11 Timed text ..... | 11 |
29
+ | 4.12 Digital Rights Management ..... | 12 |
30
+ | 4.13 PIM..... | 12 |
31
+ | 4.14 Dynamic and Interactive Multimedia Scene ..... | 12 |
32
+ | Annex A (informative): Change history..... | 13 |
33
+
34
+ # --- Foreword
35
+
36
+ This Technical Specification has been produced by the 3<sup>rd</sup> Generation Partnership Project (3GPP).
37
+
38
+ 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:
39
+
40
+ Version x.y.z
41
+
42
+ where:
43
+
44
+ - x the first digit:
45
+ - 1 presented to TSG for information;
46
+ - 2 presented to TSG for approval;
47
+ - 3 or greater indicates TSG approved document under change control.
48
+ - y the second digit is incremented for all changes of substance, i.e. technical enhancements, corrections, updates, etc.
49
+ - z the third digit is incremented when editorial only changes have been incorporated in the specification;
50
+
51
+ The 3GPP Multimedia messaging service (MMS) specification consists of three 3GPP TSs; 3GPP TS 22.140, 3GPP TS 23.140 and the present document. The TS 3GPP TS 22.140 [22] provides a set of requirements which shall be supported for the provision of non real-time multimedia messaging service, seen primarily from the subscriber's and service providers' points of view. The TS 23.140 [23] identifies the functional capabilities and information flows needed to support the MMS. The present document provides the details of media types, formats and codecs used by the MMS service
52
+
53
+ The issue of codecs ad for MMS services has been addressed initially in TS 23.140, owned by the 3GPP T2 group. During the TSG-T WG2 group meeting in Edinburgh in September 2001, the TSG-T WG2 group sent a Liaison statement (S4-AHP040) to the 3GPP SA WG4 group, requesting that the responsibility for the specification of codecs and formats to be used in MMS services is transferred to SA WG4 group starting with Release 5.
54
+
55
+ After the SA WG4 group agreed to take over this responsibility, and the present document is the result of such commitment on Release 6.
56
+
57
+ For the sake of interoperability and alignment it is important there is no contradiction between the recommendations made in the present document and in the 26.234 specification [14].
58
+
59
+ # --- 1 Scope
60
+
61
+ The present document specifies the media types, formats and codecs for the MMS within the 3GPP system. The scope of the present document extends to codecs for speech, audio, video, still images, bitmap graphics, and other media in general, as well as scene description, multimedia integration and synchronization schemes.
62
+
63
+ # --- 2 References
64
+
65
+ The following documents contain provisions which, through reference in this text, constitute provisions of the present document.
66
+
67
+ - References are either specific (identified by date of publication, edition number, version number, etc.) or non-specific.
68
+ - For a specific reference, subsequent revisions do not apply.
69
+ - 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*.
70
+ - [1] 3GPP TR 21.905: "Vocabulary for 3GPP Specifications".
71
+ - [2] The Unicode Consortium: "The Unicode Standard", Version 2.0, Addison-Wesley Developers Press, 1996. URL: <http://www.unicode.org/>.
72
+ - [3] ANSI X3.4, 1986: "Information Systems; Coded Character Set 7 Bit; American National Standard Code for Information Interchange".
73
+ - [4] ISO/IEC 8859-1:1998: "Information technology; 8-bit single-byte coded graphic character sets; Part 1: Latin alphabet No. 1".
74
+ - [5] IETF; RFC 2279: "UTF-8, A Transformation format of ISO 10646", URL: <http://www.ietf.org/rfc/rfc2279.txt>.
75
+ - [6] 3GPP TS 24.011: "Point-to-Point (PP) Short Message Service (SMS) support on mobile radio interface".
76
+ - [7] 3GPP TS 26.090: "AMR speech Codec Transcoding functions".
77
+ - [8] ITU-T Recommendation T.81: "Information technology; Digital compression and coding of continuous-tone still images: Requirements and guidelines".
78
+ - [9] "JPEG File Interchange Format", Version 1.02, September 1, 1992.
79
+ - [10] ITU-T Recommendation H.263 (02/98): "Video coding for low bit rate communication".
80
+ - [11] ITU-T Recommendation H.263 – Annex X (03/04): "Annex X: Profiles and levels definition".
81
+ - [12] ISO/IEC 14496-2 (2004): "Information technology - Coding of audio-visual objects - Part 2: Visual".
82
+ - [13] (void).
83
+ - [14] 3GPP TS 26.234: "End-to-end transparent streaming Service; Protocols and codecs".
84
+ - [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.
85
+ - [16] Compuserve Incorporated, Columbus, Ohio (1990): "Graphics Interchange Format (Version 89a)".
86
+ - [17] IETF RFC 2083: "PNG (Portable Networks Graphics) Specification version 1.0 ", T. Boutell, et. al., March 1997.
87
+ - [18] ITU-T Recommendation H.263 (1998): "Video coding for low bit rate communication - Annex X, Profiles and Levels Definition".
88
+
89
+ - [19] ISO/IEC 14496-3:2001, "Information technology -- Coding of audio-visual objects -- Part 3: Audio".
90
+ - [20] W3C Last Call Working Draft: "Scalable Vector Graphics (SVG) 1.2", <http://www.w3.org/TR/2004/WD-SVG12-20041027/>, October 2004.
91
+ - [21] W3C Last Call Working Draft: "Mobile SVG Profile: SVG Tiny, Version 1.2", <http://www.w3.org/TR/2004/WD-SVGMobile12-20040813/>, August 2004.
92
+ - [22] 3GPP 22.140: "Service Aspects; Stage 1; Multimedia Messaging Service".
93
+ - [23] 3GPP 23.140: "Multimedia Messaging Service (MMS); Functional Description; Stage 2".
94
+ - [24] W3C Recommendation: "Synchronized Multimedia Integration Language (SMIL 2.0)", <http://www.w3.org/TR/2001/REC-smil20-20010807/>, August 2001.
95
+ - [25] IETF RFC 2046: "Multipurpose Internet Mail Extensions (MIME) Part Two: Media Types".
96
+ - [26] 3GPP TS 26.071: "Mandatory Speech Codec speech processing functions; AMR Speech Codec; General description".
97
+ - [27] 3GPP TS 26.171: "Speech codec speech processing functions; Adaptive Multi-Rate - Wideband (AMR-WB) speech codec; General description".
98
+ - [28] Scalable Polyphony MIDI Specification Version 1.0, RP-34, MIDI Manufacturers Association, Los Angeles, CA, February 2002.
99
+ - [29] Scalable Polyphony MIDI Device 5-to-24 Note Profile for 3GPP, RP-35, MIDI Manufacturers Association, Los Angeles, CA, February 2002.
100
+ - [30] WAP Forum Specification: "XHTML Mobile Profile", <http://www1.wapforum.org/tech/terms.asp?doc=WAP-277-XHTMLMP-20011029-a.pdf>, October 2001.
101
+ - [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.
102
+ - [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.
103
+ - [33] 3GPP TS 26.244: "Transparent end-to-end packet switched streaming service (PSS); 3GPP file format (3GP)".
104
+ - [34] 3GPP TS 26.246: "Transparent end-to-end packet switched streaming service (PSS); 3GPP SMIL Language Profile".
105
+ - [35] 3GPP TS 26.245: "Transparent end-to-end packet switched streaming service (PSS); Timed text format".
106
+ - [36] IETF RFC 1952 "GZIP file format specification version 4.3", Deutsch P, May 1996.
107
+ - [37] (void)
108
+ - [38] Mobile DLS, MMA specification v1.0. RP-41 Los Angeles, CA, USA. 2004.
109
+ - [39] Mobile XMF Content Format Specification, MMA specification v1.0., RP-42, Los Angeles, CA, USA. 2004.
110
+ - [40] 3GPP TS 26.090: "Mandatory Speech Codec speech processing functions; Adaptive Multi-Rate (AMR) speech codec; Transcoding functions".
111
+ - [41] 3GPP TS 26.073: "ANSI-C code for the Adaptive Multi Rate (AMR) speech codec".
112
+ - [42] 3GPP TS 26.104: "ANSI-C code for the floating-point Adaptive Multi Rate (AMR) speech codec".
113
+ - [43] 3GPP TS 26.190: "Speech Codec speech processing functions; AMR Wideband speech codec; Transcoding functions".
114
+
115
+ - [44] 3GPP TS 26.173: "ANSI-C code for the Adaptive Multi Rate - Wideband (AMR-WB) speech codec".
116
+ - [45] 3GPP TS 26.204: "ANSI-C code for the Floating-point Adaptive Multi-Rate Wideband (AMR-WB) speech codec".
117
+ - [46] 3GPP TS 26.290: "Extended AMR Wideband codec; Transcoding functions".
118
+ - [47] 3GPP TS 26.304: "ANSI-C code for the Floating-point; Extended AMR Wideband codec".
119
+ - [48] 3GPP TS 26.273: "ANSI-C code for the Fixed-point; Extended AMR Wideband codec".
120
+ - [49] 3GPP TS 26.401: "General audio codec audio processing functions; Enhanced aacPlus general audio codec; General description".
121
+ - [50] 3GPP TS 26.410: "General audio codec audio processing functions; Enhanced aacPlus general audio codec; Floating-point ANSI-C code".
122
+ - [51] 3GPP TS 26.411: "General audio codec audio processing functions; Enhanced aacPlus general audio codec; Fixed-point ANSI-C code".
123
+ - [52] ITU-T Recommendation H.264 (01/2012): "Advanced video coding for generic audiovisual services" | ISO/IEC 14496-10:2010: "Information technology – Coding of audio-visual objects – Part 10: Advanced Video Coding".
124
+ - [53] ISO/IEC 14496-10/FDAM1: "AVC Fidelity Range Extensions".
125
+ - [54] "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: <http://www.exif.org/>
126
+ - [55] Standard ECMA-327: "ECMAScript 3<sup>rd</sup> Edition Compact Profile", June 2001.
127
+ - [56] "Digital Rights Management", Open Mobile Alliance™, OMA-Download-DRM-v1\_0, <http://www.openmobilealliance.org/>
128
+ - [57] "DRM Rights Expression Language", Open Mobile Alliance™, OMA-Download-DRMREL-v1\_0, <http://www.openmobilealliance.org/>
129
+ - [58] "DRM Content Format", Open Mobile Alliance™, OMA-Download-DRMCF-v1\_0, <http://www.openmobilealliance.org/>
130
+ - [59] "vObject Minimum Interoperability Profile", Open Mobile Alliance™, OMA-TS-vObjectOMAProfile-V1\_0, <http://www.openmobilealliance.org/>
131
+ - [60] 3GPP TR 26.936: "Performance characterization of 3GPP audio codecs".
132
+ - [61] 3GPP TS 26.142: "Dynamic and Interactive Multimedia Scene".
133
+
134
+ # --- 3 Definitions and abbreviations
135
+
136
+ ## 3.1 Definitions
137
+
138
+ For the purposes of the present document, the following terms and definitions apply:
139
+
140
+ **continuous media:** media with an inherent notion of time, in the present document speech, audio and video
141
+
142
+ **discrete media:** media that itself does not contain an element of time, in the present document all media not defined as continuous media
143
+
144
+ **scene description:** description of the spatial layout and temporal behaviour of a presentation, it can also contain hyperlinks
145
+
146
+ ## 3.2 Abbreviations
147
+
148
+ For the purposes of the present document, the abbreviations given in 3GPP TR 21.905 [1] and the following apply:
149
+
150
+ | | |
151
+ |------------------|-------------------------------------------------------------|
152
+ | 3GP | 3GPP file format |
153
+ | AAC | Advanced Audio Coding |
154
+ | AVC | Advanced Video Coding |
155
+ | CC/PP | Composite Capability/Preference Profiles |
156
+ | CPB | Coding Picture Buffer |
157
+ | DIMS | Dynamic and Interactive Multimedia Scene |
158
+ | DLS | Downloadable Sounds |
159
+ | DRM | Digital Rights Management |
160
+ | Enhanced aacPlus | MPEG-4 High Efficiency AAC plus MPEG-4 Parametric Stereo |
161
+ | EXIF | Exchangeable image file format |
162
+ | GIF | Graphics Interchange Format |
163
+ | H.263 | ITU-T video codec |
164
+ | HDTV | High-definition television |
165
+ | ITU-T | International Telecommunications Union - Telecommunications |
166
+ | JFIF | JPEG File Interchange Format |
167
+ | JPEG | Joint Picture Expert Group |
168
+ | MIDI | Musical Instrument Digital Interface |
169
+ | MIME | Multipurpose Internet Mail Extensions |
170
+ | MM | Multimedia Message |
171
+ | MMS | Multimedia Messaging Service |
172
+ | MPEG | Motion Picture Expert Group |
173
+ | MP4 | MPEG-4 file format |
174
+ | PIM | Personal Information Manager |
175
+ | PSS | Packet-switched Streaming Service |
176
+ | SBR | Spectral Band Replication |
177
+ | SP-MIDI | Scalable Polyphony MIDI |
178
+ | SVG | Scalable Vector Graphics |
179
+ | UTF-8 | Unicode Transformation Format (the 8-bit form) |
180
+ | VCL | Video Coding Layer |
181
+ | XMF | Extensible Music Format |
182
+
183
+ # --- 4 Media formats
184
+
185
+ Multiple media elements shall be combined into a composite single MM using MIME multipart format as defined in RFC 2046 [25]. The media type of a single MM element shall be identified by its appropriate MIME type whereas the media format shall be indicated by its appropriate MIME subtype.
186
+
187
+ In order to guarantee a minimum support and compatibility between multimedia messaging capable terminals, MMS User Agent supporting specific media types shall comply with the following selection of media formats:
188
+
189
+ ## 4.1 Text
190
+
191
+ 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.).
192
+
193
+ 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".
194
+
195
+ Interoperability with SMS text type is according to [23].
196
+
197
+ ## 4.2 Speech
198
+
199
+ If speech is supported, the AMR codec shall be supported for narrow-band speech [26][40][41][42].
200
+
201
+ The AMR wideband speech codec [27][43][44][45] shall be supported when wideband speech working at 16 kHz sampling frequency is supported.
202
+
203
+ When using speech media type alone, AMR or AMR-WB data is stored according to the file format specified in [32].
204
+
205
+ Multi-channel sessions shall not be used.
206
+
207
+ ## 4.3 Audio
208
+
209
+ If audio is supported, then one or both of the following two audio codecs should be supported:
210
+
211
+ - Enhanced aacPlus [49] [50] [51]
212
+ - Extended AMR-WB [46] [47] [48]
213
+
214
+ There is no requirement that a terminal supporting decoding by one of the codecs shall also support encoding by that codec.
215
+
216
+ 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:
217
+
218
+ | Content type | Music | Speech over Music | Speech between Music | Speech |
219
+ |----------------|-------|-------------------|----------------------|--------|
220
+ | Bit rate | | | | |
221
+ | 14 kbps mono | | | | |
222
+ | 18 kbps stereo | | | | |
223
+ | 24 kbps stereo | | | | |
224
+ | 24 kbps mono | | | | |
225
+ | 32 kbps stereo | | | | |
226
+ | 48 kbps stereo | | | | |
227
+
228
+ More recent information on the performance of the codecs based on more recent versions of the codecs can be found in TR 26.936 [60].
229
+
230
+ Enhanced aacPlus decoder is also able to decode MPEG-4 AAC LC content.
231
+
232
+ Extended AMR-WB decoder is also able to decode AMR-WB content.
233
+
234
+ 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).
235
+
236
+ ## 4.4 Synthetic audio
237
+
238
+ If synthetic audio is supported, 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.
239
+
240
+ SP-MIDI content is delivered in the structure specified in Standard MIDI Files 1.0 [31], either in format 0 or format 1.
241
+
242
+ In addition the Mobile DLS instrument format defined in [38] and the Mobile XMF content format defined in [39] should be supported.
243
+
244
+ A MMS 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 [29] 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 [39] 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
245
+
246
+ 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.
247
+
248
+ 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].
249
+
250
+ ## 4.5 Still Image
251
+
252
+ If still images are supported, ISO/IEC JPEG [8] together with JFIF [9] shall be supported. The support for ISO/IEC JPEG only apply to the following two modes:
253
+
254
+ - mandatory: baseline DCT, non-differential, Huffman coding, as defined in table B.1, symbol 'SOF0' in [8];
255
+ - optional: progressive DCT, non-differential, Huffman coding, as defined in table B.1, symbol 'SOF2' [8].
256
+
257
+ For JPEG baseline DCT, EXIF compressed image file format should also be supported, as defined in [54]. In that case there is no requirement for the MMS client to interpret or present the EXIF parameters recorded in the file.
258
+
259
+ ## 4.6 Bitmap graphics
260
+
261
+ If bitmap graphics is supported, the following bitmap graphics formats should be supported:
262
+
263
+ - GIF87a [15];
264
+ - GIF89a, [16];
265
+ - PNG, [17].
266
+
267
+ ## 4.7 Video
268
+
269
+ If video is supported, ITU-T Recommendation H.264 / MPEG-4 (Part 10) AVC [52] Constrained Baseline Profile (CBP) Level 1.3 shall be supported. In addition,
270
+
271
+ - ITU-T Recommendation H.264 / MPEG-4 (Part 10) AVC [52] High Profile Level 3.1 with `frame_mbs_only_flag=1` should be supported by MMS clients supporting HDTV video content at a resolution of 1280x720 (720p) with progressive scan at 30 frames per second. Maximum VCL Bit Rate shall be constrained to 14Mbps by `cpbBrVclFactor` & `cpbBrNalFactor` being fixed to 1000 and 1200 respectively, irrespective of the profile. Note that peak Bit Rate is determined by the CPB size.
272
+ - ITU-T Recommendation H.263 profile 0 level 45 [10][11] should be supported for compatibility with earlier content and UEs.
273
+
274
+ If stereoscopic 3D video is supported, ITU-T Recommendation H.264 / MPEG-4 (Part 10) AVC [52] Stereo High Profile (SHP) Level 3.1 with `frame_mbs_only_flag=1` should be supported. When an H.264 (AVC) SHP sub-bitstream containing the base view only complies with Level 1.3 or below, it should be constrained as follows: the value of the `profile_idc` should be equal to 66 and the value of the `constraint_set1_flag` should be equal to 1 in all active sequence parameter sets, i.e. the H.264 (AVC) Constrained Baseline Profile should be indicated to be used for the base view.
275
+
276
+ NOTE: When the base view sub-bitstream of the MM complies with H.264 (AVC) CPB Level 1.3 or below, the base view of an MM can be played back by any MMS (Release 11) client supporting video, or the MM can be modified without re-encoding to an MM including 2D video to be played back in H.264 (AVC) CPB compatible MMS clients.
277
+
278
+ There are no requirements on output timing conformance of H.264 (AVC) decoding (Annex C of [52]).
279
+
280
+ A video buffer model defined in Annex G of document [14] should be used with H.263 and MPEG-4. It shall not be used with H.264 (AVC).
281
+
282
+ ## 4.8 Vector graphics
283
+
284
+ If 2D vector graphics is supported, Scalable Vector Graphics (SVG) Tiny 1.2 [20][21] and ECMAScript [55] shall be supported.
285
+
286
+ NOTE 1: The compression format for SVG content is GZIP [35], in accordance with the SVG specification [20].
287
+
288
+ NOTE 2: Only media formats supported by MMS, as specified in clause 4 of this specification, shall be used. MMS clients do not support the Ogg Vorbis format.
289
+
290
+ NOTE 3: Content creators of SVG Tiny 1.2 for MMS clients are strongly recommended to follow the content creation guidelines provided for PSS clients in Annex L of [14].
291
+
292
+ 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.
293
+
294
+ ## 4.9 File Format for video and associated speech/audio media types
295
+
296
+ To ensure interoperability for the transport of video and associated speech/audio and timed text in an MM, the 3GPP file format with Basic profile shall be supported.
297
+
298
+ The usage of the 3GPP file format shall follow the technical specifications and the implementation guidelines specified in TS 26.233 [33]
299
+
300
+ NOTE: When using speech media type alone, AMR or AMR-WB data is stored according to the file format specified in [32].
301
+
302
+ ## 4.10 Media synchronization and presentation format
303
+
304
+ The 3GPP MMS uses a subset of SMIL 2.0 [24] for media synchronization and scene description. MMS clients and servers with support for media synchronization and scene descriptions shall support the 3GPP SMIL Language Profile defined in [34].
305
+
306
+ 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.
307
+
308
+ Additionally, XHTML Mobile Profile [30] for scene description should be supported. MMS clients and servers with support for scene descriptions based on XHTML shall support XHTML Mobile Profile [30], defined by the WAP Forum.
309
+
310
+ XHTML Mobile Profile is a subset of XHTML 1.1 but a superset of XHTML Basic.
311
+
312
+ ## 4.11 Timed text
313
+
314
+ If timed text is supported, MMS clients shall support [35] with 3GP files using Basic profile [33].
315
+
316
+ ## 4.12 Digital Rights Management
317
+
318
+ If Rights Management is supported, OMA Digital Rights Management (DRM) 1.0 [56][57][58] shall be supported.
319
+
320
+ ## 4.13 PIM
321
+
322
+ If Personal Data Interchange is supported this shall be done according to the OMA vObject Minimum Interoperability Profile [59].
323
+
324
+ ## 4.14 Dynamic and Interactive Multimedia Scene
325
+
326
+ If dynamic and interactive multimedia scene is supported, MMS clients and servers shall support 3GPP TS 26.142 [61].
327
+
328
+ # Annex A (informative): Change history
329
+
330
+ | Change history | | | | | | | |
331
+ |----------------|-------|-----------|------|-----|-----------------------------------------------------------------------|--------|--------|
332
+ | Date | TSG # | TSG Doc. | CR | Rev | Subject/Comment | Old | New |
333
+ | 2002-03 | 15 | SP-020075 | | | Version 2.0.0 presented for approval | 2.0.0 | 5.0.0 |
334
+ | 2002-06 | 16 | SP-020224 | 001 | | Correcting the reference to AMR and AMR-WB RTP payload | 5.0.0 | 5.1.0 |
335
+ | 2002-12 | 18 | SP-020691 | 002 | | Code points for H.263 | 5.1.0 | 5.2.0 |
336
+ | 2002-12 | 18 | SP-020691 | 003 | 1 | File Format name change from MP4 to 3GP | 5.1.0 | 5.2.0 |
337
+ | | | | | | | | |
338
+ | 2004-09 | 25 | SP-040641 | 006 | 2 | Introduction of Extended AMR-WB and Enhanced aacPlus into MMS service | 5.2.0 | 6.0.0 |
339
+ | 2004-09 | 25 | SP-040650 | 007 | 1 | Update of MMS codecs and formats with Release 6 functionality | 5.2.0 | 6.0.0 |
340
+ | 2004-09 | 25 | SP-040655 | 008 | 1 | Update of MMS codecs and formats with H.264 | 5.2.0 | 6.0.0 |
341
+ | 2004-12 | 26 | SP-040838 | 009 | 1 | Support for EXIF in MMS | 6.0.0 | 6.1.0 |
342
+ | 2004-12 | 26 | SP-040838 | 010 | | Adoption of SVG Tiny 1.2 for MMS | 6.0.0 | 6.1.0 |
343
+ | 2005-12 | 27 | SP-050175 | 011 | 2 | Introduction of PIM and DRM | 6.1.0 | 6.2.0 |
344
+ | 2006-03 | 31 | SP-060009 | 0012 | 1 | Addition of a reference to TR 26.936 | 6.2.0 | 6.3.0 |
345
+ | 2006-09 | 33 | SP-060600 | 0013 | | Editorial correction of references | 6.3.0 | 7.0.0 |
346
+ | 2007-06 | 36 | SP-070319 | 0014 | 2 | Inclusion of DIMS in MMS | 7.0.0 | 7.1.0 |
347
+ | 2008-12 | 42 | | | | Version for Release 8 | 7.1.0 | 8.0.0 |
348
+ | 2009-12 | 46 | | | | Version for Release 9 | 8.0.0 | 9.0.0 |
349
+ | 2011-03 | 51 | | | | Version for Release 10 | 9.0.0 | 10.0.0 |
350
+ | 2012-03 | 55 | SP-120026 | 0016 | 4 | On MMS video enhancements | 10.0.0 | 11.0.0 |
351
+ | 2012-09 | 57 | SP-120509 | 0017 | 2 | Inclusion of MVC support for MMS | 11.0.0 | 11.1.0 |
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1
+
2
+
3
+
4
+
5
+
6
+
7
+ # --- Contents
8
+
9
+ | | |
10
+ |------------------------------------------------------------------------------------|----|
11
+ | Foreword ..... | 4 |
12
+ | 1 Scope..... | 5 |
13
+ | 2 Normative references ..... | 5 |
14
+ | 3 Definitions and abbreviations ..... | 5 |
15
+ | 3.1 Abbreviations ..... | 5 |
16
+ | 4 General ..... | 6 |
17
+ | 5 Adaptive Multi-Rate Wideband speech codec transcoding functions ..... | 8 |
18
+ | 6 Adaptive Multi-Rate Wideband speech codec ANSI C-code..... | 8 |
19
+ | 7 Adaptive Multi-Rate Wideband speech codec test vectors..... | 8 |
20
+ | 8 Adaptive Multi-Rate Wideband speech codec source controlled rate operation ..... | 9 |
21
+ | 9 Adaptive Multi-Rate Wideband speech codec voice activity detection..... | 9 |
22
+ | 10 Adaptive Multi-Rate Wideband speech codec comfort noise insertion..... | 10 |
23
+ | 11 Adaptive Multi-Rate Wideband speech codec error concealment of lost frames..... | 10 |
24
+ | 12 Adaptive Multi-Rate Wideband speech codec frame structure ..... | 10 |
25
+ | 13 Adaptive Multi-Rate Wideband speech codec interface to RAN ..... | 10 |
26
+ | 14 Adaptive Multi-Rate Wideband speech codec performance characterisation ..... | 11 |
27
+ | Annex A (informative): Change history..... | 12 |
28
+
29
+ # --- Foreword
30
+
31
+ This Technical Specification has been produced by the 3GPP.
32
+
33
+ The present document is an introduction to the speech processing parts of the wideband telephony speech service employing the Adaptive Multi-Rate Wideband (AMR-WB) speech coder within the 3GPP system.
34
+
35
+ 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:
36
+
37
+ Version x.y.z
38
+
39
+ where:
40
+
41
+ - x the first digit:
42
+ - 1 presented to TSG for information;
43
+ - 2 presented to TSG for approval;
44
+ - 3 Indicates TSG approved document under change control.
45
+ - y the second digit is incremented for all changes of substance, i.e. technical enhancements, corrections, updates, etc.
46
+ - z the third digit is incremented when editorial only changes have been incorporated in the specification;
47
+
48
+ # --- 1 Scope
49
+
50
+ The present document is an introduction to the speech processing parts of the wideband telephony speech service employing the Adaptive Multi-Rate Wideband (AMR-WB) speech coder. A general overview of the speech processing functions is given, with reference to the documents where each function is specified in detail.
51
+
52
+ # --- 2 Normative references
53
+
54
+ This TS incorporates by dated and undated reference, provisions from other publications. These normative references are cited at the appropriate places in the text and the publications are listed hereafter. For dated references, subsequent amendments to or revisions of any of these publications apply to this TS only when incorporated in it by amendment or revision. For undated references, the latest edition of the publication referred to applies.
55
+
56
+ - [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".
57
+ - [2] 3GPP TS 26.190 : "AMR Wideband Speech Codec; Transcoding functions".
58
+ - [3] 3GPP TS 26.173 : "AMR Wideband Speech Codec; ANSI-C code".
59
+ - [4] 3GPP TS 26.174 : "AMR Wideband Speech Codec; Test sequences".
60
+ - [5] 3GPP TS 26.193 : "AMR Wideband Speech Codec; Source Controlled Rate operation".
61
+ - [6] 3GPP TS 26.194 : "AMR Wideband Speech Codec; Voice Activity Detection (VAD)".
62
+ - [7] 3GPP TS 26.192 : "AMR Wideband Speech Codec; Comfort Noise Aspects".
63
+ - [8] 3GPP TS 26.191 : "AMR Wideband Speech Codec; Error Concealment of Lost Frames".
64
+ - [9] 3GPP TS 26.201 : "AMR Wideband Speech Codec; Frame Structure".
65
+ - [10] 3GPP TS 26.202 : "AMR Wideband Speech Codec; Interface to RAN".
66
+ - [11] 3GPP TR 26.976 : "Performance characterization of the Adaptive Multi-Rate Wideband (AMR-WB) speech codec".
67
+
68
+ # --- 3 Definitions and abbreviations
69
+
70
+ ## 3.1 Abbreviations
71
+
72
+ For the purposes of this TS, the following abbreviations apply:
73
+
74
+ | | |
75
+ |--------|-------------------------------------------------------------------------------------------------|
76
+ | ACELP | Algebraic Code Excited Linear Prediction |
77
+ | AMR | Adaptive Multi-Rate |
78
+ | AMR-WB | Adaptive Multi-Rate Wideband |
79
+ | BFI | Bad Frame Indication |
80
+ | CHD | Channel Decoder |
81
+ | CHE | Channel Encoder |
82
+ | GSM | Global System for Mobile communications |
83
+ | ITU-T | International Telecommunication Union – Telecommunication standardisation sector (former CCITT) |
84
+ | PCM | Pulse Code Modulation |
85
+ | PLMN | Public Land Mobile Network |
86
+ | PSTN | Public Switched Telephone Network |
87
+ | RX | Receive |
88
+ | SCR | Source Controlled Rate |
89
+ | SPD | SPeech Decoder |
90
+ | SPE | SPeech Encoder |
91
+ | TC | Transcoder |
92
+ | TX | Transmit |
93
+
94
+ UE
95
+
96
+ User Equipment (terminal)
97
+
98
+ # --- 4 General
99
+
100
+ The AMR-WB 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.
101
+
102
+ The multi-rate speech coder is a single integrated speech codec with nine source rates from 6.60 kbit/s to 23.85 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.
103
+
104
+ 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.
105
+
106
+ 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 this document. These aspects are only considered to the extent that the performance of the audio parts affect the performance of the speech transcoder.
107
+
108
+ ![Figure 1: Overview of audio processing functions. A complex block diagram showing the Transmit and Receive sides of an audio processing system with various functional blocks labeled with TS and GSM standards and circled numbers 1-11.](997233d405f0d4b89ddeb7683e047f66_img.jpg)
109
+
110
+ ```
111
+
112
+ graph TD
113
+ subgraph TRANSMIT_SIDE [TRANSMIT SIDE]
114
+ direction TB
115
+ BSS_WB[BSS side only wideband speech] --> WB_14bit[14-bit uniform 2]
116
+ BSS_NB[BSS side only narrowband speech] --> NB_Conv[8bit / A-law to 14-bit uniform 1]
117
+ NB_Conv -- TS 26.190 --> UpSamp[Up sampling 1:2]
118
+ MS_Side[MS side only] --> LPF_TX[LPF]
119
+ LPF_TX -- GSM 03.50 --> AD[A/D]
120
+
121
+ WB_14bit --> Sum1((2))
122
+ UpSamp --> Sum1
123
+ AD --> Sum1
124
+
125
+ Sum1 --> SpeechEnc[Speech Encoder]
126
+
127
+ subgraph TS_26_194 [TS 26.194]
128
+ VAD_Det[Voice Activity Detector] --> VAD_Flag((3))
129
+ end
130
+
131
+ SpeechEnc -- TS 26.190 --> SpeechFrame((4))
132
+ SpeechEnc -- TS 26.190 --> SID_TX((6))
133
+
134
+ subgraph TS_26_192 [TS 26.192]
135
+ CN_TX[Comfort Noise TX Functions] --> SID_Frame((5))
136
+ end
137
+
138
+ VAD_Flag --> DTX_TX[DTX Control and Operation TS 26.193]
139
+ SpeechFrame --> DTX_TX
140
+ SID_Frame --> DTX_TX
141
+
142
+ DTX_TX --> SP_Flag[SP flag 6]
143
+ DTX_TX --> Info_Bits_Out[Info. bits 7]
144
+ end
145
+
146
+ subgraph RECEIVE_SIDE [RECEIVE SIDE]
147
+ direction TB
148
+ Info_Bits_In[Info. bits 8] --> DTX_RX[DTX Control and Operation TS 26.193]
149
+ BFI_In[BFI 9] --> DTX_RX
150
+ SID_In[SID 10] --> DTX_RX
151
+ TAF_In[TAF 11] --> DTX_RX
152
+
153
+ DTX_RX -- TS 26.191 --> FrameSub[Speech frame substitution]
154
+ DTX_RX -- TS 26.190 --> SpeechFrame_RX((4))
155
+ DTX_RX -- TS 26.192 --> CN_RX[Comfort Noise RX Functions]
156
+ CN_RX --> SID_Frame_RX((5))
157
+
158
+ FrameSub --> SpeechDec[Speech Decoder TS 26.190]
159
+ SpeechFrame_RX --> SpeechDec
160
+ SID_Frame_RX --> SpeechDec
161
+
162
+ SpeechDec --> Sum2((2))
163
+
164
+ Sum2 --> WB_Out[BSS side only wideband speech: 14-bit uniform 2]
165
+ Sum2 --> NB_Out[BSS side only narrowband speech: Down sampling 2:1 to 14-bit uniform to 8bit / A-law 1]
166
+ Sum2 --> MS_Out[MS side only: D/A to LPF GSM 03.50]
167
+ end
168
+
169
+ ```
170
+
171
+ Figure 1: Overview of audio processing functions. A complex block diagram showing the Transmit and Receive sides of an audio processing system with various functional blocks labeled with TS and GSM standards and circled numbers 1-11.
172
+
173
+ **Figure 1: Overview of audio processing functions**
174
+
175
+ - 1) 8-bit A-law or $\mu$ -law PCM (ITU-T recommendation G.711), 8000 samples/s
176
+ - 2) 14-bit uniform PCM, 16 000 samples/s
177
+ - 3) Voice Activity Detector (VAD) flag
178
+ - 4) Encoded speech frame, 50 frames/s, number of bits/frame depending on the AMR-WB codec mode
179
+ - 5) Silence Descriptor (SID) frame.
180
+ - 6) TX\_TYPE, 3 bits, indicates whether information bits are available and if they are speech or SID information
181
+ - 7) Information bits delivered to the 3G AN
182
+ - 8) Information bits received from the 3G AN
183
+ - 9) RX\_TYPE, the type of frame received quantized into three bits
184
+ - 10) Silence Descriptor (SID) flag
185
+
186
+ 11) Time Alignment Flag (TAF), marks the position of the SID frame within the SACCH multiframe
187
+
188
+ # 5 Adaptive Multi-Rate Wideband speech codec transcoding functions
189
+
190
+ The adaptive multi-rate wideband speech codec is described in [2].
191
+
192
+ As shown in Figure 1, the speech encoder takes its input as a 14-bit uniform Pulse Code Modulated (PCM) signal either from the audio part of the UE or from the network side [TBD] or from the Public Switched Telephone Network (PSTN) via a narrowband 13-bit A-law or -law to wideband 14-bit uniform PCM conversion. An upsampling by factor of 2 has to be performed between narrowband and wideband speech signals. 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.
193
+
194
+ The detailed mapping between input blocks of 320 speech samples in 14-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 320 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.
195
+
196
+ An AMR-WB speech codec capable UE shall support all source rates listed in Table 1.
197
+
198
+ **Table 1: Source codec bit-rates for the AMR-WB codec**
199
+
200
+ | Codec mode | Source codec bit-rate |
201
+ |--------------|-----------------------|
202
+ | AMR-WB_23.85 | 23.85 kbit/s |
203
+ | AMR-WB_23.05 | 23.05 kbit/s |
204
+ | AMR-WB_19.85 | 19.85 kbit/s |
205
+ | AMR-WB_18.25 | 18.25 kbit/s |
206
+ | AMR-WB_15.85 | 15.85 kbit/s |
207
+ | AMR-WB_14.25 | 14.25 kbit/s |
208
+ | AMR-WB_12.65 | 12.65 kbit/s |
209
+ | AMR-WB_8.85 | 8.85 kbit/s |
210
+ | AMR-WB_6.60 | 6.60 kbit/s |
211
+ | AMR-WB_SID | 1.75 kbit/s * |
212
+
213
+ (\*) Assuming SID frames are continuously transmitted
214
+
215
+ # 6 Adaptive Multi-Rate Wideband speech codec ANSI C-code
216
+
217
+ The ANSI –C-code of the speech codec, VAD and CNG system are described in [3]. The ANSI C-code is mandatory.
218
+
219
+ # 7 Adaptive Multi-Rate Wideband speech codec test vectors
220
+
221
+ 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.
222
+
223
+ The test sequences are defined separately for:
224
+
225
+ - The speech codec described in [2],
226
+ - The VAD described in [6],
227
+ - The CN generation described in [7].
228
+
229
+ The adaptive multi-rate wideband 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
230
+
231
+ 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.
232
+
233
+ The input test sequences provided shall force the corresponding output test sequences, provided that the tested modules are in their home-state when starting.
234
+
235
+ The modules may be set into their home states by provoking the appropriate homing-functions.
236
+
237
+ NOTE: This is normally done during reset (initialisation of the codec).
238
+
239
+ 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.
240
+
241
+ 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.
242
+
243
+ # --- 8 Adaptive Multi-Rate Wideband speech codec source controlled rate operation
244
+
245
+ The source controlled rate operation of the adaptive multi-rate wideband speech codec is defined in [5].
246
+
247
+ 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:
248
+
249
+ - 1) In the UE, battery life will be prolonged or a smaller battery could be used for a given operational duration.
250
+ - 2) The average required bit-rate is reduced, leading to a more efficient transmission with decreased load and hence increased capacity.
251
+
252
+ The following functions are required for the source controlled rate operation:
253
+
254
+ - a Voice Activity Detector (VAD) on the TX side;
255
+ - evaluation of the background acoustic noise on the TX side, in order to transmit characteristic parameters to the RX side;
256
+ - generation of comfort noise on the RX side during periods when no normal speech frames are received.
257
+
258
+ 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.
259
+
260
+ # --- 9 Adaptive Multi-Rate Wideband speech codec voice activity detection
261
+
262
+ The adaptive multi-rate wideband VAD function is described in [6].
263
+
264
+ The input to the VAD is the input speech itself together with a set of parameters computed by the adaptive multi-rate wideband speech encoder. The VAD uses this information to decide whether each 20 ms speech coder frame contains speech or not.
265
+
266
+ 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.
267
+
268
+ # --- 10 Adaptive Multi-Rate Wideband speech codec comfort noise insertion
269
+
270
+ The adaptive multi-rate wideband comfort noise insertion function is described in [7].
271
+
272
+ 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.
273
+
274
+ The comfort noise generation process is as follows:
275
+
276
+ - the evaluation of the acoustic background noise in the transmitter;
277
+ - the noise parameter encoding (SID frames) and decoding, and
278
+ - the generation of comfort noise in the receiver.
279
+
280
+ 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 wideband speech codec defined in [2].
281
+
282
+ # --- 11 Adaptive Multi-Rate Wideband speech codec error concealment of lost frames
283
+
284
+ The adaptive multi-rate wideband speech codec error concealment of erroneous or lost frames is described in [8].
285
+
286
+ Frames may be erroneous due to transmission errors or frames may be lost due to frame stealing in a wireless environment or packet loss in a transport network.. The methods described in [8] may be used as a basis for error concealment.
287
+
288
+ In order to mask the effect of isolated erroneous/lost frames, the speech decoder shall be informed about erroneous/lost frames 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 erroneous/lost frames, a muting technique shall be used to indicate to the listener that transmission has been interrupted.
289
+
290
+ # --- 12 Adaptive Multi-Rate Wideband speech codec frame structure
291
+
292
+ The adaptive multi-rate wideband 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.
293
+
294
+ The interface format described in [9] is termed AMR-WB interface format 1 (AMR-WB IF1).
295
+
296
+ Annex A of [9] describes an octet aligned frame format which shall be used in applications requiring octet alignment, such as for 3G H.324. This format is termed AMR-WB interface format 2 (AMR-WB IF2).
297
+
298
+ # --- 13 Adaptive Multi-Rate Wideband speech codec interface to RAN
299
+
300
+ The adaptive multi-rate wideband speech service interface to RAN is described in [10].
301
+
302
+ # --- 14 Adaptive Multi-Rate Wideband speech codec performance characterisation
303
+
304
+ The adaptive multi-rate wideband speech channel performance characterisation is described in [11].
305
+
306
+ # --- Annex A (informative): Change history
307
+
308
+ | Change history | | | | | | | |
309
+ |----------------|-------|-----------|------|-----|-------------------------------------|--------|--------|
310
+ | Date | TSG # | TSG Doc. | CR | Rev | Subject/Comment | Old | New |
311
+ | 03-2001 | 11 | SP-010082 | | | Version 2.0.0 provided for approval | | 5.0.0 |
312
+ | 12-2004 | 26 | | | | Version for Release 6 | 5.0.0 | 6.0.0 |
313
+ | 09-2006 | 33 | SP-060600 | 0001 | | Editorial correction of reference | 6.0.0 | 7.0.0 |
314
+ | 12-2008 | 42 | | | | Version for Release 8 | 7.0.0 | 8.0.0 |
315
+ | 12-2009 | 46 | | | | Version for Release 9 | 8.0.0 | 9.0.0 |
316
+ | 03-2011 | 51 | | | | Version for Release 10 | 9.0.0 | 10.0.0 |
317
+ | 09-2012 | 57 | | | | Version for Release 11 | 10.0.0 | 11.0.0 |
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1
+
2
+
3
+
4
+
5
+
6
+
7
+ # Contents
8
+
9
+ | | |
10
+ |------------------------------------------------------------------------------------------|----|
11
+ | Foreword ..... | 4 |
12
+ | 1 Scope..... | 5 |
13
+ | 2 Normative references ..... | 5 |
14
+ | 3 Definitions and abbreviations ..... | 5 |
15
+ | 3.1 Definitions..... | 5 |
16
+ | 3.2 Abbreviations ..... | 5 |
17
+ | 4 General ..... | 6 |
18
+ | 5 Requirements ..... | 6 |
19
+ | 5.1 Error detection ..... | 6 |
20
+ | 5.2 Erroneous or lost speech frames..... | 6 |
21
+ | 5.3 First lost SID frame ..... | 6 |
22
+ | 5.4 Subsequent lost SID frames ..... | 6 |
23
+ | 6 Example ECU/BFH Solution ..... | 6 |
24
+ | 6.1 State Machine..... | 6 |
25
+ | 6.2 Substitution and muting of erroneous/lost speech frames..... | 8 |
26
+ | 6.2.1 BFI = 0, prevBFI = 0, State = 0 or 1 ..... | 8 |
27
+ | 6.2.2 BFI = 0, prevBFI = 1, State = 0 to 3..... | 9 |
28
+ | 6.2.3 BFI = 1, prevBFI = 0 or 1, State = 1...6 ..... | 9 |
29
+ | 6.2.3.1 LTP gain & fixed codebook gain concealment when RX_FRAMETYPE = SPEECH_BAD ..... | 9 |
30
+ | 6.2.3.2 LTP gain & fixed codebook gain concealment when RX_FRAMETYPE = SPEECH_LOST ..... | 10 |
31
+ | 6.2.3.3 ISF concealment ..... | 10 |
32
+ | 6.2.3.4 LTP-lag concealment ..... | 11 |
33
+ | 6.2.3.4.1 LTP-lag concealment when RX_FRAMETYPE = SPEECH_BAD..... | 11 |
34
+ | 6.2.3.4.2 LTP-lag concealment when RX_FRAMETYPE = SPEECH_LOST ..... | 12 |
35
+ | 6.2.4 Innovation sequence ..... | 12 |
36
+ | 6.2.5 High-band gain (for 23.85 kbit/s mode) ..... | 12 |
37
+ | 6.3 Substitution and muting of lost SID frames ..... | 13 |
38
+ | Annex A (informative): Change history..... | 14 |
39
+
40
+ # --- Foreword
41
+
42
+ This Technical Specification has been produced by the 3GPP.
43
+
44
+ The present document defines an error concealment procedure, also termed frame substitution and muting procedure, of the wideband telephony speech service employing the Adaptive Multi-Rate – Wideband (AMR-WB) speech coder within the 3GPP system.
45
+
46
+ 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:
47
+
48
+ Version x.y.z
49
+
50
+ where:
51
+
52
+ - x the first digit:
53
+ - 1 presented to TSG for information;
54
+ - 2 presented to TSG for approval;
55
+ - 3 Indicates TSG approved document under change control.
56
+ - y the second digit is incremented for all changes of substance, i.e. technical enhancements, corrections, updates, etc.
57
+ - z the third digit is incremented when editorial only changes have been incorporated in the specification;
58
+
59
+ # --- 1 Scope
60
+
61
+ This specification defines an error concealment procedure, also termed frame substitution and muting procedure, which shall be used by the AMR-WB speech codec receiving end when one or more erroneous/lost speech or lost Silence Descriptor (SID) frames are received.
62
+
63
+ The requirements of this document are mandatory for implementation in all networks and User Equipment (UE)s capable of supporting the AMR-WB speech codec. It is not mandatory to follow the bit exact implementation outlined in this document and the corresponding C source code.
64
+
65
+ # --- 2 Normative references
66
+
67
+ The following documents contain provisions which, through reference in this text, constitute provisions of the present document.
68
+
69
+ - References are either specific (identified by date of publication, edition number, version number, etc.) or non-specific.
70
+ - For a specific reference, subsequent revisions do not apply.
71
+ - 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*.
72
+
73
+ - [1] 3GPP TS 26.202"AMR Wideband Speech Codec; Interface to RAN".
74
+ - [2] 3GPP TS 26.190"AMR Wideband Speech Codec; Transcoding functions".
75
+ - [3] 3GPP TS 26.193"AMR Wideband Speech Codec; Source Controlled Rate operation".
76
+ - [4] 3GPP TS 26.201"AMR Wideband Speech Codec; Frame structure".
77
+
78
+ # --- 3 Definitions and abbreviations
79
+
80
+ ## 3.1 Definitions
81
+
82
+ For the purposes of this document, the following definition applies:
83
+
84
+ **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.
85
+
86
+ Further definitions of terms used in this document can be found in the references.
87
+
88
+ ## 3.2 Abbreviations
89
+
90
+ For the purposes of this document, the following abbreviations apply:
91
+
92
+ | | |
93
+ |----------|----------------------------------------------------------------|
94
+ | AMR-WB | Adaptive Multi Rate - WideBand |
95
+ | AN | Access Network |
96
+ | BFI | Bad Frame Indication from AN |
97
+ | BSI_netw | Bad Sub-block Indication obtained from AN interface CRC checks |
98
+ | prevBFI | Bad Frame Indication of previous frame |
99
+ | RX | Receive |
100
+ | SCR | Source Controlled Rate (operation) |
101
+ | SID | Silence Descriptor frame (Background noise) |
102
+ | CRC | Cyclic Redundancy Check |
103
+ | ECU | Error Concealment Unit |
104
+ | BFH | Bad Frame Handling |
105
+ | medianN | N-point median operation |
106
+
107
+ # --- 4 General
108
+
109
+ The purpose of the error concealment procedure is to conceal the effect of erroneous/lost AMR-WB speech frames. The purpose of muting the output in the case of several erroneous/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.
110
+
111
+ The network shall indicate erroneous/lost speech or lost SID frames by setting the RX\_TYPE values [3] to SPEECH\_BAD, SID\_BAD or SPEECH\_LOST. If these flags are set, the speech decoder shall perform parameter substitution to conceal errors.
112
+
113
+ The example solution provided in paragraph 6 apply only to bad frame handling on a complete speech frame basis. Sub-frame based error concealment may be derived using similar methods.
114
+
115
+ # --- 5 Requirements
116
+
117
+ ## 5.1 Error detection
118
+
119
+ If the most sensitive bits of the AMR-WB 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. When the frame is not received, the network shall indicate RX\_TYPE = RX\_SPEECH\_LOST in which case the BFI flag is set as well. If a SID frame is received in error, the network shall indicate RX\_TYPE = SID\_BAD..
120
+
121
+ ## 5.2 Erroneous or lost speech frames
122
+
123
+ Normal decoding of erroneous/lost speech frames would result in very unpleasant noise effects. In order to improve the subjective quality, erroneous/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. Subclause 6 provides example solution.
124
+
125
+ ## 5.3 First lost SID frame
126
+
127
+ 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].
128
+
129
+ ## 5.4 Subsequent lost SID frames
130
+
131
+ 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. Subclause 6 provides example solutions.
132
+
133
+ # --- 6 Example ECU/BFH Solution
134
+
135
+ ## 6.1 State Machine
136
+
137
+ This example solution for substitution and muting is based on a state machine with seven states (Figure 1).
138
+
139
+ 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 right-shifted by one. 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):
140
+
141
+ ```
142
+ if(BFI != 0 )
143
+ State = State + 1;
144
+ if(State > 6)
145
+ State = 6;
146
+ else
147
+ State = State >> 1;
148
+ ```
149
+
150
+ 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 6, the processing depends on the **BFI** flag.
151
+
152
+ The procedure can be described as follows:
153
+
154
+ ![State machine diagram for controlling bad frame substitution. It consists of seven states (STATE 0 to STATE 6) arranged vertically. Transitions are labeled as 'Bad frame (BFI=1)' (solid line) or 'Good frame (BFI=0)' (double line).](d793cf7c174b89eb024d132f00679787_img.jpg)
155
+
156
+ ```
157
+ graph TD; S0[STATE = 0<br/>BFI = 0<br/>PrevBFI = 0 or 1] --> S1[STATE = 1<br/>(BFI, prevBFI) = (1,0) or (0,1) or (0,0)]; S1 --> S2[STATE = 2<br/>(BFI, prevBFI) = (1,1) or (1,0) or (0,1)]; S2 --> S3[STATE = 3<br/>(BFI, prevBFI) = (1,1) or (1,0) or (0,1)]; S3 --> S4[STATE = 4<br/>BFI = 1<br/>prevBFI = 0 or 1]; S4 --> S5[STATE = 5<br/>BFI = 1<br/>prevBFI = 1]; S5 --> S6[STATE = 6<br/>BFI = 1<br/>prevBFI = 1]; S6 --> S0; S1 --> S0; S2 --> S1; S3 --> S2; S4 --> S1; S5 --> S2; S6 --> S3;
158
+ ```
159
+
160
+ Legend:
161
+
162
+ - Bad frame (BFI=1)
163
+ - ⇒ Good frame (BFI=0)
164
+
165
+ State machine diagram for controlling bad frame substitution. It consists of seven states (STATE 0 to STATE 6) arranged vertically. Transitions are labeled as 'Bad frame (BFI=1)' (solid line) or 'Good frame (BFI=0)' (double line).
166
+
167
+ Figure 1: State machine for controlling the bad frame substitution
168
+
169
+ ## 6.2 Substitution and muting of erroneous/lost speech frames
170
+
171
+ ### 6.2.1 BFI = 0, prevBFI = 0, State = 0 or 1
172
+
173
+ 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.
174
+
175
+ ### 6.2.2 BFI = 0, prevBFI = 1, State = 0 to 3
176
+
177
+ No error is detected in the received speech frame but the previous received speech frame was bad. The LTP gain is used normally in the speech synthesis and fixed codebook gain are limited below the values used for the last received good subframe:
178
+
179
+ $$g^c(n) = \{g_{\text{received}}^c, g_{\text{received}}^c \leq 100 \text{ or } g_{\text{received}}^c \leq g^c(n-1) \times 1.25\} \quad (1)$$
180
+
181
+ where
182
+
183
+ $g_{\text{received}}^c$ = current decoded fixed codebook-gain
184
+ $g^c(n-1)$ = fixed codebook gain used for the last good subframe (BFI = 0)
185
+ $g^c(n)$ = fixed codebook gain to be used for the current frame.
186
+
187
+ The rest of the received speech parameters are used normally in the speech synthesis. The current frame of speech parameters is saved.
188
+
189
+ ### 6.2.3 BFI = 1, prevBFI = 0 or 1, State = 1...6
190
+
191
+ An error is detected in the received speech frame and the substitution and muting procedure is started.
192
+
193
+ #### 6.2.3.1 LTP gain & fixed codebook gain concealment when RX\_FRAMETYPE = SPEECH\_BAD
194
+
195
+ The LTP gain $g^p$ and fixed codebook gain $g^c$ are replaced by attenuated values from the previous subframes:
196
+
197
+ $$g^p = P^p(\text{state}) * \text{median5}(g^p(n-1), \dots, g^p(n-5)) \quad (2)$$
198
+
199
+ $$g^c = \{P^c(\text{state}) * \text{median5}(g^c(n-1), \dots, g^c(n-5)), \text{VAD\_HIST} \leq 2\} \quad (3)$$
200
+
201
+ where:
202
+
203
+ $g^p$ = current decoded LTP gain,
204
+ $g^c$ = current decoded fixed codebook gain,
205
+ $g^p(n-1), \dots, g^p(n-5)$ = LTP gains used for the last 5 subframes,
206
+ $g^c(n-1), \dots, g^c(n-5)$ = fixed codebook gains used for the last 5 subframes,
207
+ $\text{median5}()$ = 5-point median operation,
208
+ $P^p(\text{state})$ = attenuation factor ( $P^p(1) = 0.98, P^p(2) = 0.96, P^p(3) = 0.75, P^p(4) = 0.23, P^p(5) = 0.05, P^p(6) = 0.01$ ),
209
+ $P^c(\text{state})$ = attenuation factor ( $P^c(1) = 0.98, P^c(2) = 0.98, P^c(3) = 0.98, P^c(4) = 0.98, P^c(5) = 0.98, P^c(6) = 0.70$ ),
210
+ $\text{state}$ = state number $\{0..6\}$ ,
211
+ $\text{VAD\_HIST}$ is number of consecutive VAD=0 decisions.
212
+
213
+ 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:
214
+
215
+ $$\text{ener}(0) = \frac{1}{4} [\sum_{i=1}^4 \text{ener}(n-i)] - 3 \quad (4)$$
216
+
217
+ #### 6.2.3.2 LTP gain & fixed codebook gain concealment when RX\_FRAMETYPE = SPEECH\_LOST
218
+
219
+ The LTP gain $g^p$ and fixed codebook gain $g^c$ are replaced by attenuated values from the previous subframes:
220
+
221
+ $$g^p = P^p(\text{state}) * \text{median5}(g^p(n-1), \dots, g^p(n-5)) \quad (5)$$
222
+
223
+ $$g^c = \{P^c(\text{state}) * \text{median5}(g^c(n-1), \dots, g^c(n-5)), \text{VAD\_HIST} \leq 2\} \quad (6)$$
224
+
225
+ where:
226
+
227
+ $g^p$ = current decoded LTP gain,
228
+
229
+ $g^c$ = current decoded fixed codebook gain,
230
+
231
+ $g^p(n-1), \dots, g^p(n-5)$ = LTP gains used for the last 5 subframes,
232
+
233
+ $g^c(n-1), \dots, g^c(n-5)$ = fixed codebook gains used for the last 5 subframes,
234
+
235
+ $\text{median5}()$ = 5-point median operation,
236
+
237
+ $P^p(\text{state})$ = attenuation factor ( $P^p(1) = 0.95, P^p(2) = 0.90, P^p(3) = 0.75, P^p(4) = 0.23, P^p(5) = 0.05, P^p(6) = 0.01$ ),
238
+
239
+ $P^c(\text{state})$ = attenuation factor ( $P^c(1) = 0.50, P^c(2) = 0.25, P^c(3) = 0.25, P^c(4) = 0.25, P^c(5) = 0.15, P^c(6) = 0.01$ ),
240
+
241
+ $\text{state}$ = state number $\{0..6\}$ ,
242
+
243
+ $\text{VAD\_HIST}$ is number of consecutive VAD=0 decisions.
244
+
245
+ 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:
246
+
247
+ $$\text{ener}(0) = \frac{1}{4} [\sum_{i=1}^4 \text{ener}(n-i)] - 3 \quad (7)$$
248
+
249
+ #### 6.2.3.3 ISF concealment
250
+
251
+ The past ISFs are shifted towards their partly adaptive mean:
252
+
253
+ $$\text{ISF}_q(i) = \alpha * \text{past\_ISF}_q(i) + (1 - \alpha) * \text{ISF\_mean}(i) \quad i = 0..16 \quad (8)$$
254
+
255
+ where
256
+
257
+ $\alpha = 0.9$ ,
258
+
259
+ $\text{ISF}_q(i)$ is ISF-vector for a current frame,
260
+
261
+ $\text{past\_ISF}_q(i)$ is ISF-vector from the previous frame,
262
+
263
+ $\text{ISF\_mean}(i)$ vector is combination of adaptive mean and constant mean ISF-vectors in the following manner:
264
+
265
+ $$\text{ISF\_mean}(i) = \beta * \text{ISF\_const\_mean}(i) + (1 - \beta) * \text{ISF\_adaptive\_mean}(i), \quad i = 0..16 \quad (9)$$
266
+
267
+ where
268
+
269
+ $\beta = 0.75$ ,
270
+
271
+ $\text{ISF\_adaptive\_mean}(i) = \frac{1}{3} \sum_{i=0}^2 \text{past\_ISF}_q(i)$ and is updated whenever BFI = 0.
272
+
273
+ $\text{ISF\_const\_mean}(i)$ is a vector containing long time average of ISF-vectors.
274
+
275
+ #### 6.2.3.4 LTP-lag concealment
276
+
277
+ The histories of five last good LTP-lags and LTP-gains are used for finding the best method to update.
278
+
279
+ ##### 6.2.3.4.1 LTP-lag concealment when RX\_FRAMETYPE = SPEECH\_BAD
280
+
281
+ The usability of the received LTP lag ( $Q_{\text{lag}}$ ) is defined as follows: (Predicts if the received lag is most probably very close to one that was sent and therefore its usage should not introduce any bad artifacts)
282
+
283
+ $Q_{\text{lag}} =$
284
+
285
+ $$\{1, T_{\text{dif}} < 10 \text{ and } T_{\text{min}} - 5 < T_{\text{received}} < T_{\text{min}} + 5\} \quad \{1, g^p(n-1) > 0.5 \text{ and } g^p(n-2) > 0.5 \text{ and } T(n-1) - 10 < T_{\text{received}} < T(n-1) + 10\} \quad (10)$$
286
+
287
+ where:
288
+
289
+ $T(n-1)$ is LTP lag from the previous good frame,
290
+
291
+ $T_{\text{dif}} = |T_{\text{received}} - T(n-1)|$ ,
292
+
293
+ $T_{\min} = \min(T_{\text{buffer}}),$
294
+ $T_{\max} = \max(T_{\text{buffer}}),$
295
+ $T_{\text{received}}$ is received lag,
296
+ $g_{\min}^p = \min(g_{\text{buffer}}^p),$
297
+ $g^p$ is LTP gain of the current frame,
298
+ $g^p(-1)$ is LTP gain of the previous good frame,
299
+ $g^p(-2)$ is LTP gain of the frame before previous good frame,
300
+ $T_{\text{mean}} = \text{average}(T_{\text{buffer}})$
301
+
302
+ LPT lag value for the current frame is defined as follows:
303
+
304
+ $$T = \{T_{\text{received}}, Q_{\text{lag}} = 1\} \quad (11)$$
305
+
306
+ where:
307
+
308
+ $T_{\max} = \max(T_{\text{buffer}}),$
309
+ $T_{\max-1}$ is second largest value in $T_{\text{buffer}},$
310
+ $T_{\max-2}$ is second largest value in $T_{\text{buffer}},$
311
+ $\text{RND}(x)$ is random value generated to range $[-\frac{x}{2}, +\frac{x}{2}]$
312
+
313
+ ##### 6.2.3.4.2 LTP-lag concealment when RX\_FRAMETYPE = SPEECH\_LOST
314
+
315
+ The usability of the LTP lag from last good frame ( $Q_{\text{lag},t-1}$ ) is defined as follows: (Predicts if the received lag is most probably very close to one that was sent and therefore its usage should not introduce any bad artifacts)
316
+
317
+ $$Q_{\text{lag},t-1} = \{1, g_{\min}^p > 0.5 \text{ and } T_{\text{dif}} < 10 \mid \{1, g^p(n-1) > 0.5 \text{ and } g^p(n-2) > 0.5\} \} \quad (12)$$
318
+
319
+ where:
320
+
321
+ $g_{\min}^p = \min(g_{\text{buffer}}^p),$
322
+ $g^p(n-1)$ is LTP gain of the previous good frame,
323
+ $g^p(n-2)$ is LTP gain of the frame before previous good frame
324
+
325
+ LPT lag value for the current frame is defined as follows:
326
+
327
+ $$T = \{T(n-1), Q_{\text{lag},t-1} = 1\} \quad (13)$$
328
+
329
+ where:
330
+
331
+ $T(n-1)$ is LTP lag from the previous good frame,
332
+ $T_{\max} = \max(T_{\text{buffer}}),$
333
+ $T_{\max-1}$ is second largest value in $T_{\text{buffer}},$
334
+ $T_{\max-2}$ is second largest value in $T_{\text{buffer}},$
335
+ $\text{RND}(x)$ is random value generated to range $[-\frac{x}{2}, +\frac{x}{2}]$
336
+
337
+ ### 6.2.4
338
+
339
+ #### Innovation sequence
340
+
341
+ When RX\_FRAMETYPE = SPEECH\_BAD, the received fixed codebook innovation pulses from the erroneous frame are used as they are received.
342
+
343
+ When RX\_FRAMETYPE = SPEECH\_LOST, the received fixed codebook innovation pulses from the erroneous frame are not used and the fixed codebook innovation vector is filled with random signal (values limited to range [-1, +1]).
344
+
345
+ ### 6.2.5 High-band gain (for 23.85 kbit/s mode)
346
+
347
+ When RX\_FRAMETYPE = SPEECH\_BAD or RX\_FRAMETYPE = SPEECH\_LOST the received high-band energy parameter of the frame is not used and the estimation for the high-band gain is used instead. This means that in case of bad/lost speech frames, the high-band reconstruction operates in the same way for all the modes.
348
+
349
+ ## 6.3
350
+
351
+ ## Substitution and muting of lost SID frames
352
+
353
+ 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.
354
+
355
+ # Annex A (informative): Change history
356
+
357
+ | Change history | | | | | | | | |
358
+ |----------------|---------|-----------|-----|-----|----------------------------------------------------------|--|--------|--------|
359
+ | Date | TSG SA# | TSG Doc. | CR | Rev | Subject/Comment | | Old | New |
360
+ | 03-2001 | 11 | SP-010086 | | | Version 2.0.0 produced for approval | | | 5.0.0 |
361
+ | 03-2002 | 15 | SP-020083 | 001 | | Error concealment of high band gain in 23.85 kbit/s mode | | 5.0.0 | 5.1.0 |
362
+ | 12-2004 | 26 | | | | Version for Release 6 | | 5.1.0 | 6.0.0 |
363
+ | 06-2007 | 36 | | | | Version for Release 7 | | 6.0.0 | 7.0.0 |
364
+ | 12-2008 | 42 | | | | Version for Release 8 | | 7.0.0 | 8.0.0 |
365
+ | 12-2009 | 46 | | | | Version for Release 9 | | 8.0.0 | 9.0.0 |
366
+ | 03-2011 | 51 | | | | Version for Release 10 | | 9.0.0 | 10.0.0 |
367
+ | 09-2012 | 57 | | | | Version for Release 11 | | 10.0.0 | 11.0.0 |
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1
+
2
+
3
+
4
+
5
+
6
+
7
+ # --- Contents
8
+
9
+ | | |
10
+ |------------------------------------------------------------------------------------|-----------|
11
+ | Foreword ..... | 4 |
12
+ | 1 Scope..... | 5 |
13
+ | 2 References..... | 5 |
14
+ | 3 Definitions and Abbreviations ..... | 5 |
15
+ | 3.1 Definitions..... | 5 |
16
+ | 3.2 Abbreviations ..... | 5 |
17
+ | 4 AMR-WB codec Interface format 1 (AMR-WB IF1) ..... | 6 |
18
+ | 4.1 AMR-WB Header and AMR-WB Auxiliary Information ..... | 6 |
19
+ | 4.1.1 Frame Type, Mode Indication, and Mode Request ..... | 6 |
20
+ | 4.1.2 Frame Quality Indicator ..... | 7 |
21
+ | 4.1.3 Mapping to TX_TYPE and RX_TYPE..... | 7 |
22
+ | 4.1.4 Codec CRC ..... | 8 |
23
+ | 4.2 AMR-WB Core Frame..... | 8 |
24
+ | 4.2.1 AMR-WB Core Frame with speech bits: Bit ordering ..... | 8 |
25
+ | 4.2.2 AMR-WB Core Frame with speech bits: Class division..... | 8 |
26
+ | 4.2.3 AMR-WB Core Frame with comfort noise bits ..... | 9 |
27
+ | 4.3 Generic AMR-WB Frame Composition ..... | 10 |
28
+ | <b>Annex A (normative): AMR-WB Interface Format 2 (with octet alignment) .....</b> | <b>12</b> |
29
+ | <b>Annex B (normative): Tables for AMR-WB Core Frame bit ordering.....</b> | <b>14</b> |
30
+ | Annex C (informative): Change history..... | 23 |
31
+
32
+ # --- Foreword
33
+
34
+ This Technical Specification (TS) has been produced by the 3<sup>rd</sup> Generation Partnership Project (3GPP).
35
+
36
+ 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:
37
+
38
+ Version x.y.z
39
+
40
+ where:
41
+
42
+ - x the first digit:
43
+ - 1 presented to TSG for information;
44
+ - 2 presented to TSG for approval;
45
+ - 3 or greater indicates TSG approved document under change control.
46
+ - y the second digit is incremented for all changes of substance, i.e. technical enhancements, corrections, updates, etc.
47
+ - z the third digit is incremented when editorial only changes have been incorporated in the document.
48
+
49
+ # --- 1 Scope
50
+
51
+ The present document describes a generic frame format for the Adaptive Multi-Rate Wideband (AMR-WB) speech codec. This format shall be used as a common reference point when interfacing speech frames between different elements of the 3G system and between different systems. Appropriate mappings to and from this generic frame format will be used within and between each system element.
52
+
53
+ Annex A describes a second frame format which shall be used when octet alignment of AMR-WB frames is required.
54
+
55
+ # --- 2 References
56
+
57
+ The following documents contain provisions which, through reference in this text, constitute provisions of the present document.
58
+
59
+ - References are either specific (identified by date of publication, edition number, version number, etc.) or non-specific.
60
+ - For a specific reference, subsequent revisions do not apply.
61
+ - For a non-specific reference, the latest version applies.
62
+
63
+ - [1] 3GPP TS 26.190: "AMR Wideband Speech Codec; Speech Transcoding Functions".
64
+ - [2] 3GPP TS 26.193: "AMR Wideband Speech Codec; Source Controlled Rate Operation".
65
+ - [3] 3GPP TS 26.192: "AMR Wideband Speech Codec; Comfort Noise Aspects".
66
+
67
+ # --- 3 Definitions and Abbreviations
68
+
69
+ ## 3.1 Definitions
70
+
71
+ For the purposes of the present document, the following terms and definitions apply:
72
+
73
+ **AMR-WB mode:** one of the nine AMR-WB codec bit-rates denoted also with indices 0 to 8 where 0 maps to the 6.60 kbit/s mode and 8 maps to the 23.85 kbit/s mode.
74
+
75
+ **AMR-WB codec mode:** same as AMR-WB mode.
76
+
77
+ **RX\_TYPE:** classification of the received frame as defined in [2].
78
+
79
+ **TX\_TYPE:** classification of the transmitted frame as defined in [2].
80
+
81
+ ## 3.2 Abbreviations
82
+
83
+ For the purposes of the present document, the following abbreviations apply:
84
+
85
+ | | |
86
+ |-----|------------------------------------------|
87
+ | CRC | Cyclic Redundancy Check |
88
+ | FQI | Frame Quality Indicator |
89
+ | GSM | Global System for Mobile communication |
90
+ | LSB | Least Significant Bit |
91
+ | MSB | Most Significant Bit |
92
+ | RX | Receive |
93
+ | SCR | Source Controlled Rate operation |
94
+ | SID | Silence Descriptor (Comfort Noise Frame) |
95
+ | TX | Transmit |
96
+
97
+ # 4 AMR-WB codec Interface format 1 (AMR-WB IF1)
98
+
99
+ This clause describes the generic frame format for both the speech and comfort noise frames of the AMR-WB speech codec. This format is referred to as AMR-WB Interface Format 1 (AMR-WB IF1). Annex A describes AMR-WB Interface Format 2 (AMR-WB IF2).
100
+
101
+ Each AMR-WB codec mode follows the generic frame structure depicted in figure 1. The frame is divided into three parts: AMR-WB Header, AMR-WB Auxiliary Information, and AMR-WB Core Frame. The AMR-WB Header part includes the Frame Type and the Frame Quality Indicator fields. The AMR-WB auxiliary information part includes the Mode Indication, Mode Request, and Codec CRC fields. The AMR-WB Core Frame part consists of the speech parameter bits or, in case of a comfort noise frame, the comfort noise parameter bits. In case of a comfort noise frame, the comfort noise parameters replace Class A bits of AMR-WB Core Frame while Class B and C bits are omitted.
102
+
103
+ It is to be noted that unlike AMR, AMR-WB uses only two bit protection classes A and B. In no case bits are assigned to protection class C and, correspondingly, no RAB subflow needs to be allocated for such a class. The mentioning of class C is only done for the purpose of keeping uniformity of this specification with the corresponding specification for AMR.
104
+
105
+ ![Figure 1: Generic AMR-WB frame structure diagram. The diagram shows a vertical stack of fields. The top section is the AMR-WB Header, containing 'Frame Type (4 bits)' and 'Frame Quality Indicator (1 bit)'. Below it is the AMR-WB Auxiliary Information section, containing 'Mode Indication (4 bits)', 'Mode Request (4 bits)', and 'Codec CRC (8 bits)'. A label 'AMR-WB Auxiliary Information (for Tandem Free Operation, Mode Adaptation, and Error Detection)' points to this section. The bottom section is the AMR-WB Core Frame, containing 'Class A bits', 'Class B bits', and 'Class C bits'. A label 'AMR-WB Core Frame' points to this section. To the right, a label 'AMR-WB Header' points to the top section, and a label 'AMR-WB comfort noise frame' points to the bottom section.](1d27fed9c01eb99f6535283f35fe3bbf_img.jpg)
106
+
107
+ | |
108
+ |---------------------------------|
109
+ | Frame Type (4 bits) |
110
+ | Frame Quality Indicator (1 bit) |
111
+ | |
112
+ | Mode Indication (4 bits) |
113
+ | Mode Request (4 bits) |
114
+ | Codec CRC (8 bits) |
115
+ | |
116
+ | Class A bits |
117
+ | Class B bits |
118
+ | Class C bits |
119
+
120
+ AMR-WB Header (indicated by arrow)
121
+
122
+ AMR-WB Auxiliary Information
123
+ (for Tandem Free Operation, Mode Adaptation, and Error Detection)
124
+
125
+ AMR-WB Core Frame (indicated by arrow)
126
+ AMR-WB comfort noise frame (indicated by arrow)
127
+
128
+ Figure 1: Generic AMR-WB frame structure diagram. The diagram shows a vertical stack of fields. The top section is the AMR-WB Header, containing 'Frame Type (4 bits)' and 'Frame Quality Indicator (1 bit)'. Below it is the AMR-WB Auxiliary Information section, containing 'Mode Indication (4 bits)', 'Mode Request (4 bits)', and 'Codec CRC (8 bits)'. A label 'AMR-WB Auxiliary Information (for Tandem Free Operation, Mode Adaptation, and Error Detection)' points to this section. The bottom section is the AMR-WB Core Frame, containing 'Class A bits', 'Class B bits', and 'Class C bits'. A label 'AMR-WB Core Frame' points to this section. To the right, a label 'AMR-WB Header' points to the top section, and a label 'AMR-WB comfort noise frame' points to the bottom section.
129
+
130
+ Figure 1. Generic AMR-WB frame structure
131
+
132
+ ## 4.1 AMR-WB Header and AMR-WB Auxiliary Information
133
+
134
+ This subclause describes the AMR-WB Header of figure 1.
135
+
136
+ ### 4.1.1 Frame Type, Mode Indication, and Mode Request
137
+
138
+ Table 1a defines the 4-bit Frame Type field. Frame Type can indicate the use of one of the nine AMR-WB codec modes, comfort noise frame, lost speech frame, or an empty frame. In addition, four Frame Type Indices are reserved for future use. The same table is reused for the Mode Indication and Mode Request fields which are 4-bit fields each and are defined only in the range 0...8 to specify one of the nine AMR-WB codec modes.
139
+
140
+ **Table 1a: Interpretation of Frame Type, Mode Indication and Mode Request fields.**
141
+
142
+ | Frame Type Index | Mode Indication | Mode Request | Frame content (AMR-WB mode, comfort noise, or other) |
143
+ |------------------|-----------------|--------------|------------------------------------------------------|
144
+ | 0 | 0 | 0 | AMR-WB 6.60 kbit/s |
145
+ | 1 | 1 | 1 | AMR-WB 8.85 kbit/s |
146
+ | 2 | 2 | 2 | AMR-WB 12.65 kbit/s |
147
+ | 3 | 3 | 3 | AMR-WB 14.25 kbit/s |
148
+ | 4 | 4 | 4 | AMR-WB 15.85 kbit/s |
149
+ | 5 | 5 | 5 | AMR-WB 18.25 kbit/s |
150
+ | 6 | 6 | 6 | AMR-WB 19.85 kbit/s |
151
+ | 7 | 7 | 7 | AMR-WB 23.05 kbit/s |
152
+ | 8 | 8 | 8 | AMR-WB 23.85 kbit/s |
153
+ | 9 | - | - | AMR-WB SID (Comfort Noise Frame) |
154
+ | 10-13 | - | - | For future use |
155
+ | 14 | - | - | speech lost |
156
+ | 15 | - | - | No Data (No transmission/No reception) |
157
+ | | - | - | |
158
+
159
+ ### 4.1.2 Frame Quality Indicator
160
+
161
+ The content of the Frame Quality Indicator field is defined in Table 1b. The field length is one bit. The Frame Quality Indicator indicates whether the data in the frame contains errors.
162
+
163
+ **Table 1b: Definition of Frame Quality Indicator**
164
+
165
+ | Frame Quality Indicator (FQI) | Quality of data |
166
+ |-------------------------------|--------------------------------------------------------------------------------|
167
+ | 0 | Bad frame or Corrupted frame<br>(bits may be used to assist error concealment) |
168
+ | 1 | Good frame |
169
+
170
+ ### 4.1.3 Mapping to TX\_TYPE and RX\_TYPE
171
+
172
+ Table 1c shows how the AMR-WB Header data (FQI and Frame Type) maps to the TX\_TYPE and RX\_TYPE frames defined in [2].
173
+
174
+ **Table 1c: Mapping of Frame Quality Indicator and Frame Type to TX\_TYPE and RX\_TYPE [2], respectively**
175
+
176
+ | Frame Quality Indicator | Frame Type Index | TX_TYPE or RX_TYPE | Comment |
177
+ |-------------------------|------------------|--------------------|-------------------------------------------------------------------------------------------------------------------------------|
178
+ | 1 | 0-8 | SPEECH_GOOD | The specific Frame Type Index depends on the bit-rate being used. |
179
+ | 0 | 0-8 | SPEECH_BAD | The specific Frame Type Index depends on the bit-rate being used. The corrupted data may be used to assist error concealment. |
180
+ | 0 | 14 | SPEECH_LOST | No useful information. An erased or stolen frame with no data usable to assist error concealment. |
181
+ | 1 | 9 | SID_FIRST or | SID_FIRST and SID_UPDATE |
182
+ | | 9 | SID_UPDATE | are differentiated using one Class A bit: STI. |
183
+ | 0 | 9 | SID_BAD | |
184
+ | 1 | 15 | NO_DATA | Typically a non-transmitted frame. |
185
+
186
+ ### 4.1.4 Codec CRC
187
+
188
+ Generic AMR-WB codec frames with Frame Type 0...9 are associated with an 8-bit CRC for error-detection purposes. The Codec CRC field of AMR-WB Auxiliary Information in figure 1 contains the value of this CRC. These eight parity bits are generated by the cyclic generator polynomial:
189
+
190
+ $$- G(x)=D^8 + D^6 + D^5 + D^4 + 1$$
191
+
192
+ which is computed over all Class A bits of AMR-WB Core Frame. Class A bits for Frame Types 0...8 are defined in subclause 4.2.2 (for speech bits) and for Frame Type 9 in subclause 4.2.3 (for comfort noise bits).
193
+
194
+ When Frame Type Index of table 1a is 14 or 15, the CRC field is not included in the Generic AMR-WB frame.
195
+
196
+ ## 4.2 AMR-WB Core Frame
197
+
198
+ This subclause contains the description of AMR-WB Core Frame of figure 1. The descriptions for AMR-WB Core Frame with speech bits and with comfort noise bit are given separately.
199
+
200
+ ### 4.2.1 AMR-WB Core Frame with speech bits: Bit ordering
201
+
202
+ This subclause describes how AMR-WB Core Frame carries the coded speech data. The bits produced by the speech encoder are denoted as $\{s(1),s(2),\dots,s(K)\}$ , where $K$ refers to the number of bits produced by the speech encoder as shown in table 2. The notation $s(i)$ follows that of [1]. The speech encoder output bits are ordered according to their subjective importance. This bit ordering can be utilized for error protection purposes when the speech data is, for example, carried over a radio interface. Tables B.1 to B.9 in Annex B define the AMR-WB IF1 bit ordering for all the nine AMR-WB codec modes. In these tables the speech bits are numbered in the order they are produced by the corresponding speech encoder as described in the relevant tables of 3GPP TS 26.190 [1]. The reordered bits are denoted below, in the order of decreasing importance, as $\{d(0),d(1),\dots,d(K-1)\}$ .
203
+
204
+ The ordering algorithm is described in pseudo code as:
205
+
206
+ - for $j = 0$ to $K-1$
207
+ - $d(j) := s(table_m(j)+1)$ ;
208
+
209
+ where $table_m(j)$ refers to the relevant table in Annex B depending on the AMR-WB mode $m=0..8$ . The Annex B tables should be read line by line from left to right. The first element of the table has the index 0.
210
+
211
+ ### 4.2.2 AMR-WB Core Frame with speech bits: Class division
212
+
213
+ The reordered bits are further divided into three indicative classes according to their subjective importance. The three different importance classes can then be subject to different error protection in the network.
214
+
215
+ The importance classes are Class A, Class B, and Class C. Class A contains the bits most sensitive to errors and any error in these bits typically results in a corrupted speech frame which should not be decoded without applying appropriate error concealment. This class is protected by the Codec CRC in AMR-WB Auxiliary Information. Classes B and C contain bits where increasing error rates gradually reduce the speech quality, but decoding of an erroneous speech frame is usually possible without annoying artifacts. Class B bits are more sensitive to errors than Class C bits. The importance ordering applies also within the three different classes and there are no significant step-wise changes in subjective importance between neighbouring bits at the class borders.
216
+
217
+ The number of speech bits in each class (Class A, Class B, and Class C) for each AMR-WB mode is shown in table 2. The classification in table 2 and the importance ordering $d(j)$ , together, are sufficient to assign all speech bits to their correct classes. For example, when the AMR-WB codec mode is 6.60, then the Class A bits are $d(0)..d(53)$ , Class B bits are $d(54)..d(131)$ , and there are no Class C bits.
218
+
219
+ **Table 2: Number of bits in Classes A, B, and C for each AMR-WB codec mode**
220
+
221
+ | Frame Type | AMR-WB codec mode | Total number of bits | Class A | Class B | Class C |
222
+ |------------|-------------------|----------------------|---------|---------|---------|
223
+ | 0 | 6.60 | 132 | 54 | 78 | 0 |
224
+ | 1 | 8.85 | 177 | 64 | 113 | 0 |
225
+ | 2 | 12.65 | 253 | 72 | 181 | 0 |
226
+ | 3 | 14.25 | 285 | 72 | 213 | 0 |
227
+ | 4 | 15.85 | 317 | 72 | 245 | 0 |
228
+ | 5 | 18.25 | 365 | 72 | 293 | 0 |
229
+ | 6 | 19.85 | 397 | 72 | 325 | 0 |
230
+ | 7 | 23.05 | 461 | 72 | 389 | 0 |
231
+ | 8 | 23.85 | 477 | 72 | 405 | 0 |
232
+
233
+ ### 4.2.3 AMR-WB Core Frame with comfort noise bits
234
+
235
+ The AMR-WB Core Frame content for the additional frame types with Frame Type Indices 9-15 in table 1a are described in this subclause. These mainly consist of the frames related to Source Controlled Rate Operation specified in [2].
236
+
237
+ The data content (comfort noise bits) of the additional frame types is carried in AMR-WB Core Frame. The comfort noise bits are all mapped to Class A of AMR-WB Core Frame and Classes B and C are not used. This is a notation convention only and the class division has no meaning for comfort noise bits.
238
+
239
+ The number of bits in each class (Class A, Class B, and Class C) for the AMR-WB comfort noise bits (Frame Type Index 9) is shown in table 3. The contents of SID\_UPDATE and SID\_FIRST are divided into three parts (SID Type Indicator (STI), Mode Indication ( $mi(i)$ ), and Comfort Noise Parameters ( $s(i)$ ) as defined in [2].
240
+
241
+ The comfort noise parameter bits produced by the AMR-WB speech encoder are denoted as $s(i) = \{s(1), s(2), \dots, s(35)\}$ . The notation $s(i)$ follows that of [3]. These bits are numbered in the order they are produced by the AMR-WB encoder without any reordering. These bits are followed by the SID Type Indicator STI and the Mode Indication bits $mi(i) = \{mi(0), mi(1), mi(2), mi(3)\} = \{\text{LSB} \dots \text{MSB}\}$ . Thus, the AMR-WB SID or comfort noise bits $\{d(0), d(1), \dots, d(39)\}$ are formed as defined by the pseudo code below.
242
+
243
+ - for $j = 0$ to 34;
244
+ - $d(j) := s(j+1)$ ;
245
+ - $d(35) := STI$ ;
246
+ - for $j = 36$ to 39;
247
+ - $d(j) := smi(39-j)$ .
248
+
249
+ **Table 3. Bit classification for Frame Type 9: AMR-WB SID (Comfort Noise Frame)**
250
+
251
+ | Frame Type Index | FQI | AMR-WB TX_TYPE or RX_TYPE | Total number of bits | Class A | | | Class B | Class C |
252
+ |------------------|-----|---------------------------|----------------------|------------------------|-------------------------|--------------------------------|---------|---------|
253
+ | | | | | SID Type Indicator STI | Mode Indication $mi(i)$ | Comfort Noise Parameter $s(i)$ | | |
254
+ | 9 | 1 | SID_UPDATE | 40 | 1 (= "1") | 4 | 35 | 0 | 0 |
255
+ | 9 | 1 | SID_FIRST | 40 | 1 (= "0") | 4 | 35 (= "0") | 0 | 0 |
256
+ | 9 | 0 | SID_BAD | 40 | 1 | 4 | 35 | 0 | 0 |
257
+
258
+ AMR-WB no transmission frame type (14 or 15) contains the AMR-WB Header information (as defined in Figure 1), while AMR-WB Auxiliary Information and AMR-WB Core frame are omitted. The AMR-WB Header includes the corresponding Frame Type and the Frame Quality Indicator (as defined in table 1c).
259
+
260
+ ## 4.3 Generic AMR-WB Frame Composition
261
+
262
+ The generic AMR-WB frame is formed as a concatenation of AMR-WB Header, AMR-WB Auxiliary Information and the AMR-WB Core Frame, in this order. The MSB of the Frame Type is placed in bit 8 of the first octet (see example in table 5 below), the LSB of the Frame Type is placed in bit 5. Then the next parameter follows, which is the Frame Quality Indicator, and so on. After FQI, three spare bits are inserted to align the Codec CRC and the AMR-WB Core frame to the octet boundary. The first bit of the AMR-WB Core frame d(0) is placed in bit 8 of octet 4. The last bit of the generic AMR-WB frame is the last bit of AMR-WB Core Frame, which is the last bit of speech bits or the last bit of comfort noise bits, as defined in subclauses 4.2.1 and 4.2.3. Table 5 shows the composition for the example of the Codec Mode 12.65 kbit/s and table 6 shows the composition for the AMR-WB SID frame.
263
+
264
+ **Table 5: Mapping of an AMR-WB speech coding mode into the generic AMR-WB frame, AMR-WB IF1, example: AMR-WB 12.65 kbit/s (Mode Indication = 3), "good frame", Mode Request = 1.**
265
+
266
+ | Octet | MSB | Mapping of bits<br>AMR-WB 12.65 | | | | | | LSB |
267
+ |-------|------------------------------------|---------------------------------|--------|--------|-------------------|--------|--------|--------|
268
+ | | bit 8 | bit 7 | bit 6 | bit 5 | bit 4 | bit 3 | bit 2 | bit 1 |
269
+ | 1 | Frame Type (=3) | | | | FQI | spare | | |
270
+ | | 0 | 0 | 1 | 1 | 1 | 0 | 0 | 0 |
271
+ | 2 | Mode Indication (=3) | | | | Mode Request (=1) | | | |
272
+ | | MSB | ... | LSB | | MSB | ... | LSB | |
273
+ | | 0 | 0 | 1 | 1 | 0 | 0 | 0 | 1 |
274
+ | 3 | Codec CRC | | | | | | | |
275
+ | | CRC(7) | CRC(6) | CRC(5) | CRC(4) | CRC(3) | CRC(2) | CRC(1) | CRC(0) |
276
+ | 4 | AMR-WB Core Frame (octet 1) | | | | | | | |
277
+ | | d(0) | d(1) | d(2) | d(3) | d(4) | d(5) | d(6) | d(7) |
278
+ | 5..34 | AMR-WB Core Frame (octets 2 to 31) | | | | | | | |
279
+ | | d(8) | ... | ... | ... | ... | ... | ... | ... |
280
+ | 35 | AMR-WB Core Frame (octet 32) | | | | undefined | | | |
281
+ | | d(248) | d(249) | d(250) | d(251) | d(252) | | | |
282
+
283
+ **Table 6: Mapping of an AMR-WB SID frame into the generic AMR-WB frame, AMR-WB IF1, example: AMR-WB SID\_Update, "good frame", Mode Indication = 3, Mode Request = 2.**
284
+
285
+ | Octet | MSB | Mapping of bits<br>AMR-WB SID | | | | | | LSB |
286
+ |-------|-----------------------------------|-------------------------------|------------------|--------|----------------------|--------|--------|--------|
287
+ | | bit 8 | bit 7 | bit 6 | bit 5 | bit 4 | bit 3 | bit 2 | bit 1 |
288
+ | 1 | Frame Type (=9) | | | | FQI | spare | | |
289
+ | | 1 | 0 | 0 | 0 | 1 | 0 | 0 | 0 |
290
+ | 2 | Mode Indication | | | | Mode Request (=2) | | | |
291
+ | | MSB | ... | LSB | | MSB | ... | LSB | |
292
+ | | undefined | | | | 0 | 0 | 1 | 0 |
293
+ | 3 | Codec CRC | | | | | | | |
294
+ | | CRC(7) | CRC(6) | CRC(5) | CRC(4) | CRC(3) | CRC(2) | CRC(1) | CRC(0) |
295
+ | 4 | AMR-WB Core Frame (octet 1) | | | | | | | |
296
+ | | d(0)=s(1) | d(1)=s(2) | d(2) | d(3) | d(4) | d(5) | d(6) | d(7) |
297
+ | 5..7 | AMR-WB Core Frame (octets 2 to 4) | | | | | | | |
298
+ | | d(8) | ... | ... | ... | ... | ... | ... | ... |
299
+ | 8 | AMR-WB Core Frame (octet 5) | | | STI | Mode Indication (=3) | | | |
300
+ | | d(32) | d(33) | d(34) =<br>s(35) | 1 | 0 | 0 | 1 | 1 |
301
+
302
+ Table 7 summarizes all possible AMR-WB frame format combinations in terms of number of bits in each field.
303
+
304
+ **Table 7. Number of bits for different fields in different AMR-WB frame compositions**
305
+
306
+ | Frame Type Index | Frame Type | Frame Quality Indicator | Mode Indication | Mode Request | Codec CRC | Class A | Class B | Class C | Total |
307
+ |------------------|--------------------------|-------------------------|-----------------|--------------|-----------|---------|---------|---------|-------|
308
+ | | <b>AMR-WB Core Frame</b> | | | | | | | | |
309
+ | 0 | 4 | 1 | 4 | 4 | 8 | 54 | 78 | 0 | 153 |
310
+ | 1 | 4 | 1 | 4 | 4 | 8 | 64 | 113 | 0 | 198 |
311
+ | 2 | 4 | 1 | 4 | 4 | 8 | 72 | 181 | 0 | 274 |
312
+ | 3 | 4 | 1 | 4 | 4 | 8 | 72 | 213 | 0 | 306 |
313
+ | 4 | 4 | 1 | 4 | 4 | 8 | 72 | 245 | 0 | 338 |
314
+ | 5 | 4 | 1 | 4 | 4 | 8 | 72 | 293 | 0 | 386 |
315
+ | 6 | 4 | 1 | 4 | 4 | 8 | 72 | 325 | 0 | 418 |
316
+ | 7 | 4 | 1 | 4 | 4 | 8 | 72 | 389 | 0 | 482 |
317
+ | 8 | 4 | 1 | 4 | 4 | 8 | 72 | 405 | 0 | 498 |
318
+ | 9 | 4 | 1 | 4 | 4 | 8 | 40 | 0 | 0 | 61 |
319
+ | 10-13 | Not used | | | | | | | | |
320
+ | 14 | 4 | 1 | 0 | 0 | 0 | 0 | 0 | 0 | 4 |
321
+ | 15 | 4 | 1 | 0 | 0 | 0 | 0 | 0 | 0 | 4 |
322
+
323
+ # Annex A (normative): AMR-WB Interface Format 2 (with octet alignment)
324
+
325
+ This annex defines an octet-aligned frame format for the AMR-WB codec. This format is useful, for example, when the AMR-WB codec is used in connection with applicable ITU-T H-series of recommendations. The format is referred to as AMR-WB Interface Format 2 (AMR-WB IF2).
326
+
327
+ The AMR-WB IF2 frame is formed by concatenation of the 4-bit Frame Type field (as defined for AMR-WB IF1 in subclause 4.1.1), the 1-bit Frame Quality Indicator field (as defined for AMR-WB IF1 in subclause 4.1.2) and the AMR-WB Core Frame (as defined for AMR-WB IF1 in subclause 4.2) as shown in figure A.1. The length of the AMR-WB Core Frame field depends on the particular Frame Type. The total number of bits in the AMR-WB IF2 speech frames in the different modes is typically not a multiple of eight and bit stuffing is needed to achieve an octet structure.
328
+
329
+ ![Figure A.1: Frame structure for AMR-WB IF2. The diagram shows a vertical stack of fields. At the top, 'Frame Type (4 bits)' and 'Frame Quality Indicator (1 bit)' are stacked. Below them are 'Class A bits', 'Class B bits', and 'Class C bits'. At the bottom is 'Bit Stuffing'. To the right, a box labeled 'AMR-WB Core Frame (speech or comfort noise data)' has an arrow pointing to the 'Class A bits', 'Class B bits', and 'Class C bits' section.](08441fa90c5fd11994626f662ac13f19_img.jpg)
330
+
331
+ Figure A.1: Frame structure for AMR-WB IF2. The diagram shows a vertical stack of fields. At the top, 'Frame Type (4 bits)' and 'Frame Quality Indicator (1 bit)' are stacked. Below them are 'Class A bits', 'Class B bits', and 'Class C bits'. At the bottom is 'Bit Stuffing'. To the right, a box labeled 'AMR-WB Core Frame (speech or comfort noise data)' has an arrow pointing to the 'Class A bits', 'Class B bits', and 'Class C bits' section.
332
+
333
+ Figure A.1: Frame structure for AMR-WB IF2
334
+
335
+ Table A.1a shows an example how the AMR-WB 8.85 kbit/s mode is mapped into AMR-WB IF2. The four MSBs of the first octet (octet 1) consist of the Frame Type (=1) for the AMR-WB 8.85 kbit/s mode (see table 1a in AMR-WB IF1 specification) and the Frame Quality Indicator bit. This field is followed by the 177 AMR-WB Core Frame speech bits ( $d(0)\dots d(176)$ ) which consist of 64 Class A bits and 113 Class B bits as described in table 2 for AMR-WB IF1. This results in a total of 182 bits and 2 bits are needed for Bit Stuffing to arrive to the closest multiple of 8 which is 184 bits.
336
+
337
+ Table A.1a: Example mapping of the AMR-WB speech coding mode 8.85kbit/s into AMR-WB IF2.
338
+ The bits used for Bit Stuffing are denoted as UB (for "unused bit").
339
+
340
+ | Octet | MSB | Mapping of bits<br>AMR-WB 8.85 kbit/s | | | | | | LSB |
341
+ |--------|----------------------------------------------------------------------|---------------------------------------|--------|--------|--------|--------|---------------|-------|
342
+ | | bit 8 | bit 7 | bit 6 | bit 5 | bit 4 | bit 3 | bit 2 | bit 1 |
343
+ | | Frame Type (= 1)<br>MSB <span style="margin-left: 150px;">LSB</span> | | | | | | | |
344
+ | 1 | 0 | 0 | 0 | 1 | FQI | d(0) | d(1) | d(2) |
345
+ | 2 | d(3) | d(4) | d(5) | d(6) | d(7) | d(8) | d(9) | d(10) |
346
+ | 3...22 | d(11) | ... | ... | ... | ... | ... | ... | ... |
347
+ | 23 | d(171) | d(172) | d(173) | d(174) | d(175) | d(176) | Stuffing bits | |
348
+ | | d(171) | d(172) | d(173) | d(174) | d(175) | d(176) | UB | UB |
349
+
350
+ Table A.1b shows the composition of AMR-WB IF2 frames for all Frame Types in terms of how many bits are used for each field of figure A.1.
351
+
352
+ Table A.2 specify how the AMR-WB Core Frame comfort noise bits of Frame Type 9 is mapped to AMR-WB IF2. Table A.3 specifies the mapping for an empty or lost frame ("no transmission" or "speech lost").
353
+
354
+ **Table A.1b: Composition of AMR-WB IF2 Frames for all Frame Types**
355
+
356
+ | Frame Type Index | Frame content | Number of bits in Frame Type | Number of bits in Frame Quality Indicator | Number of Bits in AMR-WB Core Frame | Number of Bits in Bit Stuffing | Number of octets (N) |
357
+ |------------------|----------------------------------------|------------------------------|-------------------------------------------|-------------------------------------|--------------------------------|----------------------|
358
+ | 0 | AMR-WB 6.60 kbit/s | 4 | 1 | 132 | 7 | 18 |
359
+ | 1 | AMR-WB 8.85 kbit/s | 4 | 1 | 177 | 2 | 23 |
360
+ | 2 | AMR-WB 12.65 kbit/s | 4 | 1 | 253 | 6 | 33 |
361
+ | 3 | AMR-WB 14.25 kbit/s | 4 | 1 | 285 | 6 | 37 |
362
+ | 4 | AMR-WB 15.85 kbit/s | 4 | 1 | 317 | 6 | 41 |
363
+ | 5 | AMR-WB 18.25 kbit/s | 4 | 1 | 365 | 6 | 47 |
364
+ | 6 | AMR-WB 19.85 kbit/s | 4 | 1 | 397 | 6 | 51 |
365
+ | 7 | AMR-WB 23.05 kbit/s | 4 | 1 | 461 | 6 | 59 |
366
+ | 8 | AMR-WB 23.85 kbit/s | 4 | 1 | 477 | 6 | 61 |
367
+ | 9 | AMR-WB SID (Comfort Noise Frame) | 4 | 1 | 40 | 3 | 6 |
368
+ | 10-13 | For future use | - | - | - | - | - |
369
+ | 14 | speech lost | 4 | 1 | 0 | 3 | 1 |
370
+ | 15 | No Data (No transmission/No reception) | 4 | 1 | 0 | 3 | 1 |
371
+
372
+ **Table A.2: Mapping of bits for Frame Type 9 (AMR-WB SID)
373
+ (Bits s1 to s35 refer to TS 26.192)**
374
+
375
+ | Octet | MSB | Mapping of bits<br>AMR-WB SID | | | | | | LSB |
376
+ |-------|-----------------------------------|-------------------------------------------|-------|-------|-------|---------------|-------|-------|
377
+ | | bit 8 | bit 7 | bit 6 | bit 5 | bit 4 | bit 3 | bit 2 | bit 1 |
378
+ | | Frame Type (= 9)<br>MSB ..... LSB | | | | | | | |
379
+ | 1 | 1 | 0 | 0 | 1 | FQI | s1 | s2 | s3 |
380
+ | 2 | s4 | s5 | s6 | s7 | s8 | s9 | s10 | s11 |
381
+ | 3 | s12 | s13 | s14 | s15 | s16 | s17 | s18 | s19 |
382
+ | 4 | s20 | s21 | s22 | s23 | s24 | s25 | s26 | s27 |
383
+ | 5 | s28 | s29 | s30 | s31 | s32 | s33 | s34 | s35 |
384
+ | | SID Type Indicator | Mode Indication<br>mi(i)<br>MSB ..... LSB | | | | Stuffing bits | | |
385
+ | 6 | t1 | mi(3) | mi(2) | mi(1) | mi(0) | UB | UB | UB |
386
+
387
+ Definitions of additional descriptor bits needed for the silence descriptor in the table are as follows: SID-type Indicator STI is {0=SID\_FIRST, 1=SID\_UPDATE}, Speech Mode Indication (mi(0)- mi(3)) is the AMR-WB codec mode according to the first nine entries in table 1a. Note that in parameter mi the index 3 refers to MSB.
388
+
389
+ **Table A.3: Mapping of bit for Frame Type 14 (Speech Lost) and for Frame Type 15 (No Data)**
390
+
391
+ | Transmitted Octets | MSB | Mapping of bits | | | | | | LSB |
392
+ |--------------------|----------------------------------------------------|-----------------|-------|-------|-------|-------|---------------|-------|
393
+ | | bit 8 | bit 7 | bit 6 | bit 5 | bit 4 | bit 3 | bit 2 | bit 1 |
394
+ | | Frame Type 14 = 1 1 1 0<br>Frame Type 15 = 1 1 1 1 | | | | | | Stuffing bits | |
395
+ | 1 | mi(3) | mi(2) | mi(1) | mi(0) | FQI | UB | UB | UB |
396
+
397
+ # Annex B (normative): Tables for AMR-WB Core Frame bit ordering
398
+
399
+ This annex contains the tables required for ordering the AMR-WB Core Frame speech bits corresponding to the different AMR-WB modes. These tables represent $table_m(j)$ in subclause 4.2.1 where $m=0..8$ is the AMR-WB mode. The tables are read from left to right so that the first element (top left corner) of the table has index 0 and the last element (the rightmost element of the last row) has the index $K-1$ where $K$ is the total number of speech bits in the specific mode. For example, $table_0(20)=60$ , as defined in table B.1.
400
+
401
+ **Table B.1: Ordering of the speech encoder bits for the 6.60 kbit/s mode: $table_0(j)$**
402
+
403
+ | | | | | | | | | | |
404
+ |-----|-----|-----|-----|-----|-----|-----|-----|-----|-----|
405
+ | 0 | 5 | 6 | 7 | 61 | 84 | 107 | 130 | 62 | 85 |
406
+ | 8 | 4 | 37 | 38 | 39 | 40 | 58 | 81 | 104 | 127 |
407
+ | 60 | 83 | 106 | 129 | 108 | 131 | 128 | 41 | 42 | 80 |
408
+ | 126 | 1 | 3 | 57 | 103 | 82 | 105 | 59 | 2 | 63 |
409
+ | 109 | 110 | 86 | 19 | 22 | 23 | 64 | 87 | 18 | 20 |
410
+ | 21 | 17 | 13 | 88 | 43 | 89 | 65 | 111 | 14 | 24 |
411
+ | 25 | 26 | 27 | 28 | 15 | 16 | 44 | 90 | 66 | 112 |
412
+ | 9 | 11 | 10 | 12 | 67 | 113 | 29 | 30 | 31 | 32 |
413
+ | 34 | 33 | 35 | 36 | 45 | 51 | 68 | 74 | 91 | 97 |
414
+ | 114 | 120 | 46 | 69 | 92 | 115 | 52 | 75 | 98 | 121 |
415
+ | 47 | 70 | 93 | 116 | 53 | 76 | 99 | 122 | 48 | 71 |
416
+ | 94 | 117 | 54 | 77 | 100 | 123 | 49 | 72 | 95 | 118 |
417
+ | 55 | 78 | 101 | 124 | 50 | 73 | 96 | 119 | 56 | 79 |
418
+ | 102 | 125 | | | | | | | | |
419
+
420
+ **Table B.2: Ordering of the speech encoder bits for the 8.85 kbit/s mode: $table_1(j)$**
421
+
422
+ | | | | | | | | | | |
423
+ |-----|-----|-----|-----|-----|-----|-----|-----|-----|-----|
424
+ | 0 | 4 | 6 | 7 | 5 | 3 | 47 | 48 | 49 | 112 |
425
+ | 113 | 114 | 75 | 106 | 140 | 171 | 80 | 111 | 145 | 176 |
426
+ | 77 | 108 | 142 | 173 | 78 | 109 | 143 | 174 | 79 | 110 |
427
+ | 144 | 175 | 76 | 107 | 141 | 172 | 50 | 115 | 51 | 2 |
428
+ | 1 | 81 | 116 | 146 | 19 | 21 | 12 | 17 | 18 | 20 |
429
+ | 16 | 25 | 13 | 10 | 14 | 24 | 23 | 22 | 26 | 8 |
430
+ | 15 | 52 | 117 | 31 | 82 | 147 | 9 | 33 | 11 | 83 |
431
+ | 148 | 53 | 118 | 28 | 27 | 84 | 149 | 34 | 35 | 29 |
432
+ | 46 | 32 | 30 | 54 | 119 | 37 | 36 | 39 | 38 | 40 |
433
+ | 85 | 150 | 41 | 42 | 43 | 44 | 45 | 55 | 60 | 65 |
434
+ | 70 | 86 | 91 | 96 | 101 | 120 | 125 | 130 | 135 | 151 |
435
+ | 156 | 161 | 166 | 56 | 87 | 121 | 152 | 61 | 92 | 126 |
436
+ | 157 | 66 | 97 | 131 | 162 | 71 | 102 | 136 | 167 | 57 |
437
+ | 88 | 122 | 153 | 62 | 93 | 127 | 158 | 67 | 98 | 132 |
438
+ | 163 | 72 | 103 | 137 | 168 | 58 | 89 | 123 | 154 | 63 |
439
+ | 94 | 128 | 159 | 68 | 99 | 133 | 164 | 73 | 104 | 138 |
440
+ | 169 | 59 | 90 | 124 | 155 | 64 | 95 | 129 | 160 | 69 |
441
+ | 100 | 134 | 165 | 74 | 105 | 139 | 170 | | | |
442
+
443
+ **Table B.3: Ordering of the speech encoder bits for the 12.65 kbit/s mode: $table_2(j)$**
444
+
445
+ | | | | | | | | | | |
446
+ |-----|-----|-----|-----|-----|-----|-----|-----|-----|-----|
447
+ | 0 | 4 | 6 | 93 | 143 | 196 | 246 | 7 | 5 | 3 |
448
+ | 47 | 48 | 49 | 50 | 51 | 150 | 151 | 152 | 153 | 154 |
449
+ | 94 | 144 | 197 | 247 | 99 | 149 | 202 | 252 | 96 | 146 |
450
+ | 199 | 249 | 97 | 147 | 200 | 250 | 100 | 203 | 98 | 148 |
451
+ | 201 | 251 | 95 | 145 | 198 | 248 | 52 | 2 | 1 | 101 |
452
+ | 204 | 155 | 19 | 21 | 12 | 17 | 18 | 20 | 16 | 25 |
453
+ | 13 | 10 | 14 | 24 | 23 | 22 | 26 | 8 | 15 | 53 |
454
+ | 156 | 31 | 102 | 205 | 9 | 33 | 11 | 103 | 206 | 54 |
455
+ | 157 | 28 | 27 | 104 | 207 | 34 | 35 | 29 | 46 | 32 |
456
+ | 30 | 55 | 158 | 37 | 36 | 39 | 38 | 40 | 105 | 208 |
457
+ | 41 | 42 | 43 | 44 | 45 | 56 | 106 | 159 | 209 | 57 |
458
+ | 66 | 75 | 84 | 107 | 116 | 125 | 134 | 160 | 169 | 178 |
459
+ | 187 | 210 | 219 | 228 | 237 | 58 | 108 | 161 | 211 | 62 |
460
+ | 112 | 165 | 215 | 67 | 117 | 170 | 220 | 71 | 121 | 174 |
461
+ | 224 | 76 | 126 | 179 | 229 | 80 | 130 | 183 | 233 | 85 |
462
+ | 135 | 188 | 238 | 89 | 139 | 192 | 242 | 59 | 109 | 162 |
463
+ | 212 | 63 | 113 | 166 | 216 | 68 | 118 | 171 | 221 | 72 |
464
+ | 122 | 175 | 225 | 77 | 127 | 180 | 230 | 81 | 131 | 184 |
465
+ | 234 | 86 | 136 | 189 | 239 | 90 | 140 | 193 | 243 | 60 |
466
+ | 110 | 163 | 213 | 64 | 114 | 167 | 217 | 69 | 119 | 172 |
467
+ | 222 | 73 | 123 | 176 | 226 | 78 | 128 | 181 | 231 | 82 |
468
+ | 132 | 185 | 235 | 87 | 137 | 190 | 240 | 91 | 141 | 194 |
469
+ | 244 | 61 | 111 | 164 | 214 | 65 | 115 | 168 | 218 | 70 |
470
+ | 120 | 173 | 223 | 74 | 124 | 177 | 227 | 79 | 129 | 182 |
471
+ | 232 | 83 | 133 | 186 | 236 | 88 | 138 | 191 | 241 | 92 |
472
+ | 142 | 195 | 245 | | | | | | | |
473
+
474
+ **Table B.4: Ordering of the speech encoder bits for the 14.25 kbit/s mode: $table_3(j)$**
475
+
476
+ | | | | | | | | | | |
477
+ |-----|-----|-----|-----|-----|-----|-----|-----|-----|-----|
478
+ | 0 | 4 | 6 | 101 | 159 | 220 | 278 | 7 | 5 | 3 |
479
+ | 47 | 48 | 49 | 50 | 51 | 166 | 167 | 168 | 169 | 170 |
480
+ | 102 | 160 | 221 | 279 | 107 | 165 | 226 | 284 | 104 | 162 |
481
+ | 223 | 281 | 105 | 163 | 224 | 282 | 108 | 227 | 106 | 164 |
482
+ | 225 | 283 | 103 | 161 | 222 | 280 | 52 | 2 | 1 | 109 |
483
+ | 228 | 171 | 19 | 21 | 12 | 17 | 18 | 20 | 16 | 25 |
484
+ | 13 | 10 | 14 | 24 | 23 | 22 | 26 | 8 | 15 | 53 |
485
+ | 172 | 31 | 110 | 229 | 9 | 33 | 11 | 111 | 230 | 54 |
486
+ | 173 | 28 | 27 | 112 | 231 | 34 | 35 | 29 | 46 | 32 |
487
+ | 30 | 55 | 174 | 37 | 36 | 39 | 38 | 40 | 113 | 232 |
488
+ | 41 | 42 | 43 | 44 | 45 | 56 | 114 | 175 | 233 | 62 |
489
+ | 120 | 181 | 239 | 75 | 133 | 194 | 252 | 57 | 115 | 176 |
490
+ | 234 | 63 | 121 | 182 | 240 | 70 | 128 | 189 | 247 | 76 |
491
+ | 134 | 195 | 253 | 83 | 141 | 202 | 260 | 92 | 150 | 211 |
492
+ | 269 | 84 | 142 | 203 | 261 | 93 | 151 | 212 | 270 | 85 |
493
+ | 143 | 204 | 262 | 94 | 152 | 213 | 271 | 86 | 144 | 205 |
494
+ | 263 | 95 | 153 | 214 | 272 | 64 | 122 | 183 | 241 | 77 |
495
+ | 135 | 196 | 254 | 65 | 123 | 184 | 242 | 78 | 136 | 197 |
496
+ | 255 | 87 | 145 | 206 | 264 | 96 | 154 | 215 | 273 | 58 |
497
+ | 116 | 177 | 235 | 66 | 124 | 185 | 243 | 71 | 129 | 190 |
498
+ | 248 | 79 | 137 | 198 | 256 | 88 | 146 | 207 | 265 | 97 |
499
+ | 155 | 216 | 274 | 59 | 117 | 178 | 236 | 67 | 125 | 186 |
500
+ | 244 | 72 | 130 | 191 | 249 | 80 | 138 | 199 | 257 | 89 |
501
+ | 147 | 208 | 266 | 98 | 156 | 217 | 275 | 60 | 118 | 179 |
502
+ | 237 | 68 | 126 | 187 | 245 | 73 | 131 | 192 | 250 | 81 |
503
+ | 139 | 200 | 258 | 90 | 148 | 209 | 267 | 99 | 157 | 218 |
504
+ | 276 | 61 | 119 | 180 | 238 | 69 | 127 | 188 | 246 | 74 |
505
+ | 132 | 193 | 251 | 82 | 140 | 201 | 259 | 91 | 149 | 210 |
506
+ | 268 | 100 | 158 | 219 | 277 | | | | | |
507
+
508
+ **Table B.5: Ordering of the speech encoder bits for the 15.85 kbit/s mode: $table_4(j)$**
509
+
510
+ | | | | | | | | | | |
511
+ |-----|-----|-----|-----|-----|-----|-----|-----|-----|-----|
512
+ | 0 | 4 | 6 | 109 | 175 | 244 | 310 | 7 | 5 | 3 |
513
+ | 47 | 48 | 49 | 50 | 51 | 182 | 183 | 184 | 185 | 186 |
514
+ | 110 | 176 | 245 | 311 | 115 | 181 | 250 | 316 | 112 | 178 |
515
+ | 247 | 313 | 113 | 179 | 248 | 314 | 116 | 251 | 114 | 180 |
516
+ | 249 | 315 | 111 | 177 | 246 | 312 | 52 | 2 | 1 | 117 |
517
+ | 252 | 187 | 19 | 21 | 12 | 17 | 18 | 20 | 16 | 25 |
518
+ | 13 | 10 | 14 | 24 | 23 | 22 | 26 | 8 | 15 | 53 |
519
+ | 188 | 31 | 118 | 253 | 9 | 33 | 11 | 119 | 254 | 54 |
520
+ | 189 | 28 | 27 | 120 | 255 | 34 | 35 | 29 | 46 | 32 |
521
+ | 30 | 55 | 190 | 37 | 36 | 39 | 38 | 40 | 121 | 256 |
522
+ | 41 | 42 | 43 | 44 | 45 | 56 | 122 | 191 | 257 | 63 |
523
+ | 129 | 198 | 264 | 76 | 142 | 211 | 277 | 89 | 155 | 224 |
524
+ | 290 | 102 | 168 | 237 | 303 | 57 | 123 | 192 | 258 | 70 |
525
+ | 136 | 205 | 271 | 83 | 149 | 218 | 284 | 96 | 162 | 231 |
526
+ | 297 | 62 | 128 | 197 | 263 | 75 | 141 | 210 | 276 | 88 |
527
+ | 154 | 223 | 289 | 101 | 167 | 236 | 302 | 58 | 124 | 193 |
528
+ | 259 | 71 | 137 | 206 | 272 | 84 | 150 | 219 | 285 | 97 |
529
+ | 163 | 232 | 298 | 59 | 125 | 194 | 260 | 64 | 130 | 199 |
530
+ | 265 | 67 | 133 | 202 | 268 | 72 | 138 | 207 | 273 | 77 |
531
+ | 143 | 212 | 278 | 80 | 146 | 215 | 281 | 85 | 151 | 220 |
532
+ | 286 | 90 | 156 | 225 | 291 | 93 | 159 | 228 | 294 | 98 |
533
+ | 164 | 233 | 299 | 103 | 169 | 238 | 304 | 106 | 172 | 241 |
534
+ | 307 | 60 | 126 | 195 | 261 | 65 | 131 | 200 | 266 | 68 |
535
+ | 134 | 203 | 269 | 73 | 139 | 208 | 274 | 78 | 144 | 213 |
536
+ | 279 | 81 | 147 | 216 | 282 | 86 | 152 | 221 | 287 | 91 |
537
+ | 157 | 226 | 292 | 94 | 160 | 229 | 295 | 99 | 165 | 234 |
538
+ | 300 | 104 | 170 | 239 | 305 | 107 | 173 | 242 | 308 | 61 |
539
+ | 127 | 196 | 262 | 66 | 132 | 201 | 267 | 69 | 135 | 204 |
540
+ | 270 | 74 | 140 | 209 | 275 | 79 | 145 | 214 | 280 | 82 |
541
+ | 148 | 217 | 283 | 87 | 153 | 222 | 288 | 92 | 158 | 227 |
542
+ | 293 | 95 | 161 | 230 | 296 | 100 | 166 | 235 | 301 | 105 |
543
+ | 171 | 240 | 306 | 108 | 174 | 243 | 309 | | | |
544
+
545
+ **Table B.6: Ordering of the speech encoder bits for the 18.25 kbit/s mode: $table_5(j)$**
546
+
547
+ | | | | | | | | | | |
548
+ |-----|-----|-----|-----|-----|-----|-----|-----|-----|-----|
549
+ | 0 | 4 | 6 | 121 | 199 | 280 | 358 | 7 | 5 | 3 |
550
+ | 47 | 48 | 49 | 50 | 51 | 206 | 207 | 208 | 209 | 210 |
551
+ | 122 | 200 | 281 | 359 | 127 | 205 | 286 | 364 | 124 | 202 |
552
+ | 283 | 361 | 125 | 203 | 284 | 362 | 128 | 287 | 126 | 204 |
553
+ | 285 | 363 | 123 | 201 | 282 | 360 | 52 | 2 | 1 | 129 |
554
+ | 288 | 211 | 19 | 21 | 12 | 17 | 18 | 20 | 16 | 25 |
555
+ | 13 | 10 | 14 | 24 | 23 | 22 | 26 | 8 | 15 | 53 |
556
+ | 212 | 31 | 130 | 289 | 9 | 33 | 11 | 131 | 290 | 54 |
557
+ | 213 | 28 | 27 | 132 | 291 | 34 | 35 | 29 | 46 | 32 |
558
+ | 30 | 55 | 214 | 37 | 36 | 39 | 38 | 40 | 133 | 292 |
559
+ | 41 | 42 | 43 | 44 | 45 | 56 | 134 | 215 | 293 | 198 |
560
+ | 299 | 136 | 120 | 138 | 60 | 279 | 58 | 62 | 357 | 139 |
561
+ | 140 | 295 | 156 | 57 | 219 | 297 | 63 | 217 | 137 | 170 |
562
+ | 300 | 222 | 64 | 106 | 61 | 78 | 294 | 92 | 142 | 141 |
563
+ | 135 | 221 | 296 | 301 | 343 | 59 | 298 | 184 | 329 | 315 |
564
+ | 220 | 216 | 265 | 251 | 218 | 237 | 352 | 223 | 157 | 86 |
565
+ | 171 | 87 | 164 | 351 | 111 | 302 | 65 | 178 | 115 | 323 |
566
+ | 72 | 192 | 101 | 179 | 93 | 73 | 193 | 151 | 337 | 309 |
567
+ | 143 | 274 | 69 | 324 | 165 | 150 | 97 | 338 | 110 | 310 |
568
+ | 330 | 273 | 68 | 107 | 175 | 245 | 114 | 79 | 113 | 189 |
569
+ | 246 | 259 | 174 | 71 | 185 | 96 | 344 | 100 | 322 | 83 |
570
+ | 334 | 316 | 333 | 252 | 161 | 348 | 147 | 82 | 269 | 232 |
571
+ | 260 | 308 | 353 | 347 | 163 | 231 | 306 | 320 | 188 | 270 |
572
+ | 146 | 177 | 266 | 350 | 256 | 85 | 149 | 116 | 191 | 160 |
573
+ | 238 | 258 | 336 | 305 | 255 | 88 | 224 | 99 | 339 | 230 |
574
+ | 228 | 227 | 272 | 242 | 241 | 319 | 233 | 311 | 102 | 74 |
575
+ | 180 | 275 | 66 | 194 | 152 | 325 | 172 | 247 | 244 | 261 |
576
+ | 117 | 158 | 166 | 354 | 75 | 144 | 108 | 312 | 94 | 186 |
577
+ | 303 | 80 | 234 | 89 | 195 | 112 | 340 | 181 | 345 | 317 |
578
+ | 326 | 276 | 239 | 167 | 118 | 313 | 70 | 355 | 327 | 253 |
579
+ | 190 | 176 | 271 | 104 | 98 | 153 | 103 | 90 | 76 | 267 |
580
+ | 277 | 248 | 225 | 262 | 182 | 84 | 154 | 235 | 335 | 168 |
581
+ | 331 | 196 | 341 | 249 | 162 | 307 | 148 | 349 | 263 | 321 |
582
+ | 257 | 243 | 229 | 356 | 159 | 119 | 67 | 187 | 173 | 145 |
583
+ | 240 | 77 | 304 | 332 | 314 | 342 | 109 | 254 | 81 | 278 |
584
+ | 105 | 91 | 346 | 318 | 183 | 250 | 197 | 328 | 95 | 155 |
585
+ | 169 | 268 | 226 | 236 | 264 | | | | | |
586
+
587
+ **Table B.7: Ordering of the speech encoder bits for the 19.85 kbit/s mode: $table_6(j)$**
588
+
589
+ | | | | | | | | | | |
590
+ |-----|-----|-----|-----|-----|-----|-----|-----|-----|-----|
591
+ | 0 | 4 | 6 | 129 | 215 | 304 | 390 | 7 | 5 | 3 |
592
+ | 47 | 48 | 49 | 50 | 51 | 222 | 223 | 224 | 225 | 226 |
593
+ | 130 | 216 | 305 | 391 | 135 | 221 | 310 | 396 | 132 | 218 |
594
+ | 307 | 393 | 133 | 219 | 308 | 394 | 136 | 311 | 134 | 220 |
595
+ | 309 | 395 | 131 | 217 | 306 | 392 | 52 | 2 | 1 | 137 |
596
+ | 312 | 227 | 19 | 21 | 12 | 17 | 18 | 20 | 16 | 25 |
597
+ | 13 | 10 | 14 | 24 | 23 | 22 | 26 | 8 | 15 | 53 |
598
+ | 228 | 31 | 138 | 313 | 9 | 33 | 11 | 139 | 314 | 54 |
599
+ | 229 | 28 | 27 | 140 | 315 | 34 | 35 | 29 | 46 | 32 |
600
+ | 30 | 55 | 230 | 37 | 36 | 39 | 38 | 40 | 141 | 316 |
601
+ | 41 | 42 | 43 | 44 | 45 | 56 | 142 | 231 | 317 | 63 |
602
+ | 73 | 92 | 340 | 82 | 324 | 149 | 353 | 159 | 334 | 165 |
603
+ | 338 | 178 | 163 | 254 | 77 | 168 | 257 | 153 | 343 | 57 |
604
+ | 248 | 238 | 79 | 252 | 166 | 67 | 80 | 201 | 101 | 267 |
605
+ | 143 | 164 | 341 | 255 | 339 | 187 | 376 | 318 | 78 | 328 |
606
+ | 362 | 115 | 232 | 242 | 253 | 290 | 276 | 62 | 58 | 158 |
607
+ | 68 | 93 | 179 | 319 | 148 | 169 | 154 | 72 | 385 | 329 |
608
+ | 333 | 344 | 102 | 83 | 144 | 233 | 323 | 124 | 243 | 192 |
609
+ | 354 | 237 | 64 | 247 | 202 | 209 | 150 | 116 | 335 | 268 |
610
+ | 239 | 299 | 188 | 196 | 298 | 94 | 195 | 258 | 123 | 363 |
611
+ | 384 | 109 | 325 | 371 | 170 | 370 | 84 | 110 | 295 | 180 |
612
+ | 74 | 210 | 191 | 106 | 291 | 205 | 367 | 381 | 377 | 206 |
613
+ | 355 | 122 | 119 | 120 | 383 | 160 | 105 | 108 | 277 | 380 |
614
+ | 294 | 284 | 285 | 345 | 208 | 269 | 249 | 366 | 386 | 300 |
615
+ | 297 | 259 | 125 | 369 | 197 | 97 | 194 | 286 | 211 | 281 |
616
+ | 280 | 183 | 372 | 87 | 155 | 283 | 59 | 348 | 327 | 184 |
617
+ | 76 | 111 | 330 | 203 | 349 | 69 | 98 | 152 | 145 | 189 |
618
+ | 66 | 320 | 337 | 173 | 358 | 251 | 198 | 174 | 263 | 262 |
619
+ | 126 | 241 | 193 | 88 | 388 | 117 | 95 | 387 | 112 | 359 |
620
+ | 287 | 244 | 103 | 272 | 301 | 171 | 162 | 234 | 273 | 127 |
621
+ | 373 | 181 | 292 | 85 | 378 | 302 | 121 | 107 | 364 | 346 |
622
+ | 356 | 212 | 278 | 213 | 65 | 382 | 288 | 207 | 113 | 175 |
623
+ | 99 | 296 | 374 | 368 | 199 | 260 | 185 | 336 | 331 | 161 |
624
+ | 270 | 264 | 250 | 240 | 75 | 350 | 151 | 60 | 89 | 321 |
625
+ | 156 | 274 | 360 | 326 | 70 | 282 | 167 | 146 | 352 | 81 |
626
+ | 91 | 389 | 266 | 245 | 177 | 235 | 190 | 256 | 204 | 342 |
627
+ | 128 | 118 | 303 | 104 | 379 | 182 | 114 | 375 | 200 | 96 |
628
+ | 293 | 172 | 214 | 365 | 279 | 86 | 289 | 351 | 347 | 357 |
629
+ | 261 | 186 | 176 | 271 | 90 | 100 | 147 | 322 | 275 | 361 |
630
+ | 71 | 332 | 61 | 265 | 157 | 246 | 236 | | | |
631
+
632
+ **Table B.8: Ordering of the speech encoder bits for the 23.05 kbit/s mode: $table_{7}(j)$**
633
+
634
+ | | | | | | | | | | |
635
+ |-----|-----|-----|-----|-----|-----|-----|-----|-----|-----|
636
+ | 0 | 4 | 6 | 145 | 247 | 352 | 454 | 7 | 5 | 3 |
637
+ | 47 | 48 | 49 | 50 | 51 | 254 | 255 | 256 | 257 | 258 |
638
+ | 146 | 248 | 353 | 455 | 151 | 253 | 358 | 460 | 148 | 250 |
639
+ | 355 | 457 | 149 | 251 | 356 | 458 | 152 | 359 | 150 | 252 |
640
+ | 357 | 459 | 147 | 249 | 354 | 456 | 52 | 2 | 1 | 153 |
641
+ | 360 | 259 | 19 | 21 | 12 | 17 | 18 | 20 | 16 | 25 |
642
+ | 13 | 10 | 14 | 24 | 23 | 22 | 26 | 8 | 15 | 53 |
643
+ | 260 | 31 | 154 | 361 | 9 | 33 | 11 | 155 | 362 | 54 |
644
+ | 261 | 28 | 27 | 156 | 363 | 34 | 35 | 29 | 46 | 32 |
645
+ | 30 | 55 | 262 | 37 | 36 | 39 | 38 | 40 | 157 | 364 |
646
+ | 41 | 42 | 43 | 44 | 45 | 56 | 158 | 263 | 365 | 181 |
647
+ | 192 | 170 | 79 | 57 | 399 | 90 | 159 | 297 | 377 | 366 |
648
+ | 275 | 68 | 183 | 388 | 286 | 194 | 299 | 92 | 70 | 182 |
649
+ | 401 | 172 | 59 | 91 | 58 | 400 | 368 | 161 | 81 | 160 |
650
+ | 264 | 171 | 80 | 389 | 390 | 378 | 379 | 193 | 298 | 69 |
651
+ | 266 | 265 | 367 | 277 | 288 | 276 | 287 | 184 | 60 | 195 |
652
+ | 82 | 93 | 71 | 369 | 402 | 173 | 162 | 444 | 300 | 391 |
653
+ | 98 | 76 | 278 | 61 | 267 | 374 | 135 | 411 | 167 | 102 |
654
+ | 380 | 200 | 87 | 178 | 65 | 94 | 204 | 124 | 72 | 342 |
655
+ | 189 | 305 | 381 | 396 | 433 | 301 | 226 | 407 | 289 | 237 |
656
+ | 113 | 215 | 185 | 128 | 309 | 403 | 116 | 320 | 196 | 331 |
657
+ | 370 | 422 | 174 | 64 | 392 | 83 | 425 | 219 | 134 | 188 |
658
+ | 432 | 112 | 427 | 139 | 279 | 163 | 436 | 208 | 447 | 218 |
659
+ | 236 | 229 | 97 | 294 | 385 | 230 | 166 | 268 | 177 | 443 |
660
+ | 225 | 426 | 101 | 272 | 138 | 127 | 290 | 117 | 347 | 199 |
661
+ | 414 | 95 | 140 | 240 | 410 | 395 | 209 | 129 | 283 | 346 |
662
+ | 105 | 241 | 437 | 86 | 308 | 448 | 203 | 345 | 186 | 107 |
663
+ | 220 | 415 | 334 | 319 | 106 | 313 | 118 | 123 | 73 | 207 |
664
+ | 421 | 214 | 384 | 373 | 438 | 62 | 371 | 341 | 75 | 449 |
665
+ | 168 | 323 | 164 | 242 | 416 | 324 | 304 | 197 | 335 | 404 |
666
+ | 271 | 63 | 191 | 325 | 96 | 169 | 231 | 280 | 312 | 187 |
667
+ | 406 | 84 | 201 | 100 | 67 | 382 | 175 | 336 | 202 | 330 |
668
+ | 269 | 393 | 376 | 383 | 293 | 307 | 409 | 179 | 285 | 314 |
669
+ | 302 | 372 | 398 | 190 | 180 | 89 | 99 | 103 | 232 | 78 |
670
+ | 88 | 77 | 136 | 387 | 165 | 198 | 394 | 125 | 176 | 428 |
671
+ | 74 | 375 | 238 | 227 | 66 | 273 | 282 | 141 | 306 | 412 |
672
+ | 114 | 85 | 130 | 348 | 119 | 291 | 296 | 386 | 233 | 397 |
673
+ | 303 | 405 | 284 | 445 | 423 | 221 | 210 | 205 | 450 | 108 |
674
+ | 274 | 434 | 216 | 343 | 337 | 142 | 243 | 321 | 408 | 451 |
675
+ | 310 | 292 | 120 | 109 | 281 | 439 | 270 | 429 | 332 | 295 |
676
+ | 418 | 211 | 315 | 222 | 326 | 131 | 430 | 244 | 327 | 349 |
677
+ | 417 | 316 | 143 | 338 | 440 | 234 | 110 | 212 | 452 | 245 |
678
+ | 121 | 419 | 350 | 223 | 132 | 441 | 328 | 413 | 317 | 339 |
679
+ | 126 | 104 | 137 | 446 | 344 | 239 | 435 | 115 | 333 | 206 |
680
+ | 322 | 217 | 228 | 424 | 453 | 311 | 351 | 111 | 442 | 224 |
681
+ | 213 | 122 | 431 | 340 | 235 | 246 | 133 | 144 | 420 | 329 |
682
+ | 318 | | | | | | | | | |
683
+
684
+ Table B.9: Ordering of the speech encoder bits for the 23.85 kbit/s mode: $table_8(j)$
685
+
686
+ | | | | | | | | | | |
687
+ |-----|-----|-----|-----|-----|-----|-----|-----|-----|-----|
688
+ | 0 | 4 | 6 | 145 | 251 | 360 | 466 | 7 | 5 | 3 |
689
+ | 47 | 48 | 49 | 50 | 51 | 262 | 263 | 264 | 265 | 266 |
690
+ | 146 | 252 | 361 | 467 | 151 | 257 | 366 | 472 | 148 | 254 |
691
+ | 363 | 469 | 149 | 255 | 364 | 470 | 156 | 371 | 150 | 256 |
692
+ | 365 | 471 | 147 | 253 | 362 | 468 | 52 | 2 | 1 | 157 |
693
+ | 372 | 267 | 19 | 21 | 12 | 17 | 18 | 20 | 16 | 25 |
694
+ | 13 | 10 | 14 | 24 | 23 | 22 | 26 | 8 | 15 | 53 |
695
+ | 268 | 31 | 152 | 153 | 154 | 155 | 258 | 259 | 260 | 261 |
696
+ | 367 | 368 | 369 | 370 | 473 | 474 | 475 | 476 | 158 | 373 |
697
+ | 9 | 33 | 11 | 159 | 374 | 54 | 269 | 28 | 27 | 160 |
698
+ | 375 | 34 | 35 | 29 | 46 | 32 | 30 | 55 | 270 | 37 |
699
+ | 36 | 39 | 38 | 40 | 161 | 376 | 41 | 42 | 43 | 44 |
700
+ | 45 | 56 | 162 | 271 | 377 | 185 | 196 | 174 | 79 | 57 |
701
+ | 411 | 90 | 163 | 305 | 389 | 378 | 283 | 68 | 187 | 400 |
702
+ | 294 | 198 | 307 | 92 | 70 | 186 | 413 | 176 | 59 | 91 |
703
+ | 58 | 412 | 380 | 165 | 81 | 164 | 272 | 175 | 80 | 401 |
704
+ | 402 | 390 | 391 | 197 | 306 | 69 | 274 | 273 | 379 | 285 |
705
+ | 296 | 284 | 295 | 188 | 60 | 199 | 82 | 93 | 71 | 381 |
706
+ | 414 | 177 | 166 | 456 | 308 | 403 | 98 | 76 | 286 | 61 |
707
+ | 275 | 386 | 135 | 423 | 171 | 102 | 392 | 204 | 87 | 182 |
708
+ | 65 | 94 | 208 | 124 | 72 | 350 | 193 | 313 | 393 | 408 |
709
+ | 445 | 309 | 230 | 419 | 297 | 241 | 113 | 219 | 189 | 128 |
710
+ | 317 | 415 | 116 | 328 | 200 | 339 | 382 | 434 | 178 | 64 |
711
+ | 404 | 83 | 437 | 223 | 134 | 192 | 444 | 112 | 439 | 139 |
712
+ | 287 | 167 | 448 | 212 | 459 | 222 | 240 | 233 | 97 | 302 |
713
+ | 397 | 234 | 170 | 276 | 181 | 455 | 229 | 438 | 101 | 280 |
714
+ | 138 | 127 | 298 | 117 | 355 | 203 | 426 | 95 | 140 | 244 |
715
+ | 422 | 407 | 213 | 129 | 291 | 354 | 105 | 245 | 449 | 86 |
716
+ | 316 | 460 | 207 | 353 | 190 | 107 | 224 | 427 | 342 | 327 |
717
+ | 106 | 321 | 118 | 123 | 73 | 211 | 433 | 218 | 396 | 385 |
718
+ | 450 | 62 | 383 | 349 | 75 | 461 | 172 | 331 | 168 | 246 |
719
+ | 428 | 332 | 312 | 201 | 343 | 416 | 279 | 63 | 195 | 333 |
720
+ | 96 | 173 | 235 | 288 | 320 | 191 | 418 | 84 | 205 | 100 |
721
+ | 67 | 394 | 179 | 344 | 206 | 338 | 277 | 405 | 388 | 395 |
722
+ | 301 | 315 | 421 | 183 | 293 | 322 | 310 | 384 | 410 | 194 |
723
+ | 184 | 89 | 99 | 103 | 236 | 78 | 88 | 77 | 136 | 399 |
724
+ | 169 | 202 | 406 | 125 | 180 | 440 | 74 | 387 | 242 | 231 |
725
+ | 66 | 281 | 290 | 141 | 314 | 424 | 114 | 85 | 130 | 356 |
726
+ | 119 | 299 | 304 | 398 | 237 | 409 | 311 | 417 | 292 | 457 |
727
+ | 435 | 225 | 214 | 209 | 462 | 108 | 282 | 446 | 220 | 351 |
728
+ | 345 | 142 | 247 | 329 | 420 | 463 | 318 | 300 | 120 | 109 |
729
+ | 289 | 451 | 278 | 441 | 340 | 303 | 430 | 215 | 323 | 226 |
730
+ | 334 | 131 | 442 | 248 | 335 | 357 | 429 | 324 | 143 | 346 |
731
+ | 452 | 238 | 110 | 216 | 464 | 249 | 121 | 431 | 358 | 227 |
732
+ | 132 | 453 | 336 | 425 | 325 | 347 | 126 | 104 | 137 | 458 |
733
+ | 352 | 243 | 447 | 115 | 341 | 210 | 330 | 221 | 232 | 436 |
734
+ | 465 | 319 | 359 | 111 | 454 | 228 | 217 | 122 | 443 | 348 |
735
+ | 239 | 250 | 133 | 144 | 432 | 337 | 326 | | | |
736
+
737
+ # --- Annex C (informative): Change history
738
+
739
+ | Change history | | | | | | | |
740
+ |----------------|-------|-----------|------|-----|-----------------------------------------|--------|--------|
741
+ | Date | TSG # | TSG Doc. | CR | Rev | Subject/Comment | Old | New |
742
+ | 03-2001 | 11 | SP-010090 | | | Presented as version 2.0.0 for approval | | 5.0.0 |
743
+ | 12-2004 | 26 | | | | Version for Release 6 | 5.0.0 | 6.0.0 |
744
+ | 06-2007 | 36 | | | | Version for Release 7 | 6.0.0 | 7.0.0 |
745
+ | 06-2008 | 38 | SP-080250 | 0001 | 2 | Obsolescence of class C definition | 7.0.0 | 7.1.0 |
746
+ | 12-2008 | 42 | | | | Version for Release 8 | 7.1.0 | 8.0.0 |
747
+ | 12-2009 | 46 | | | | Version for Release 9 | 8.0.0 | 9.0.0 |
748
+ | 03-2011 | 51 | | | | Version for Release 10 | 9.0.0 | 10.0.0 |
749
+ | 09-2012 | 57 | | | | Version for Release 11 | 10.0.0 | 11.0.0 |
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1
+
2
+
3
+
4
+
5
+
6
+
7
+ # --- Contents
8
+
9
+ | | | |
10
+ |------------------------|----------------------------------------------------------|----|
11
+ | 0 | Scope..... | 5 |
12
+ | 1 | Normative references ..... | 5 |
13
+ | 2 | Definitions and Abbreviations ..... | 5 |
14
+ | 3 | C code structure ..... | 5 |
15
+ | 3.1 | Contents of the C source code..... | 6 |
16
+ | 3.2 | Program execution..... | 6 |
17
+ | 3.3 | Code hierarchy ..... | 10 |
18
+ | 3.3.1 | Initialization routines..... | 10 |
19
+ | 3.3.2 | Signal Processing Functions..... | 11 |
20
+ | 3.4 | Description of global constants used in the C-code ..... | 12 |
21
+ | 3.5 | Type Definitions..... | 13 |
22
+ | 3.6 | Functions of the C Code..... | 13 |
23
+ | Annex A (informative): | Change history..... | 27 |
24
+
25
+ # --- Foreword
26
+
27
+ This Technical Specification has been produced by T1P1.
28
+
29
+ The contents of the present document are subject to continuing work within the 3GPP TSG and may change following formal 3GPP approval. Should the 3GPP TSG modify the contents of this TS, it will be re-released by the 3GPP TSG with an identifying change of release date and an increase in version number as follows:
30
+
31
+ Version x.y.z
32
+
33
+ where:
34
+
35
+ - x the first digit:
36
+ - 1 presented to 3GPP for information;
37
+ - 2 presented to 3GPP for approval;
38
+ - 3 Indicates 3GPP approved document under change control.
39
+ - y the second digit is incremented for all changes of substance, i.e. technical enhancements, corrections, updates, etc.
40
+ - z the third digit is incremented when editorial only changes have been incorporated in the specification;
41
+
42
+ # --- 0 Scope
43
+
44
+ This Technical Standard (TS) contains an electronic copy of the ANSI-C code for the Cellular Text Telephone Modem (CTM) for reliable transmission of text telephone text via the speech channel of cellular networks. While CTM is generally usable with text in UCS coding, the example application linked to CTM in this document is limited to use the signals and character set of the Baudot type.
45
+
46
+ # --- 1 Normative references
47
+
48
+ This TS incorporates by dated and undated reference, provisions from other publications. These normative references are cited at the appropriate places in the text and the publications are listed hereafter. For dated references, subsequent amendments to or revisions of any of these publications apply to this TS only when incorporated in it by amendment or revision. For undated references, the latest edition of the publication referred to applies.
49
+
50
+ - [1] 3GPP TS 26.226: "Cellular text telephone modem; General description".
51
+ - [2] ISO/IEC 10646-1: "Information technology – Universal Multiple-Octet Coded Character Set (UCS) – Part 1: Architecture and Basic Multilingual Plane".
52
+
53
+ # --- 2 Definitions and Abbreviations
54
+
55
+ For the purposes of this TS, the following abbreviations apply:
56
+
57
+ | | |
58
+ |-----|---------------------------------------------------------------------------------------------------------------------|
59
+ | CTM | Cellular Text Telephone Modem |
60
+ | FEC | Forward Error Correction |
61
+ | FSK | Frequency Shift Key |
62
+ | HCO | Hearing Carry Over, (individual may be able to hear, but cannot speak) Alternating transmission of speech and text. |
63
+ | PCM | Pulse Code Modulation |
64
+ | RX | Receive |
65
+ | TX | Transmit |
66
+ | TTY | Text Telephone |
67
+ | UCS | Universal Multiple-Octet Coded Character Set |
68
+ | UTF | UCS transformation format |
69
+ | VAD | Voice Activity Detection |
70
+ | VCO | Voice Carry Over, Alternating transmission of speech and text |
71
+
72
+ # --- 3 C code structure
73
+
74
+ This clause gives an overview of the structure of the bit-exact C code and provides an overview of the contents and organization of the C code attached to this document.
75
+
76
+ The C code has been verified on the following system.
77
+
78
+ - Sun Microsystems workstations with SUN Solaris™ operating system and the the Gnu C Compiler (gcc version 2.7.2.3) and GNU Make 3.77;
79
+
80
+ The C code has also been successfully compiled and used in the following environment, with the exception that it cannot be guaranteed that the upper part of the UCS code table in file `ucs_functions.c` will be compiled correctly since it depends on the codepage setting of the environment.
81
+
82
+ - IBM PC/AT compatible computers with Windows™ NT 4.0 operating system and Microsoft Visual C++ 6.0™ compiler.
83
+
84
+ ## 3.1 Contents of the C source code
85
+
86
+ The distributed files with suffix "c" contain the source code and the files with suffix "h" are the header files. All these files are in the root level of the ZIP-archive.
87
+
88
+ Makefiles are provided for the platforms in which the C code has been verified (listed above). They are called "Makefile" for GNU Make and "Makefile.vc" for Microsoft Visual C++™.
89
+
90
+ For the Sun Microsystems platform, an example shell script for a transmission via two signal adaptation modules is given in "test\_negotiation". For the Microsoft Windows™ platform, no shell script or batch program is provided.
91
+
92
+ The software can be compiled using the commands
93
+
94
+ ```
95
+ make all or gmake all in case of Gnu Make
96
+ ```
97
+
98
+ ```
99
+ nmake /f Makefile.vc in case of Microsoft Visual C++.
100
+ ```
101
+
102
+ The executables are compiled into the directory `./solaris` (in case of Gnu Make) or into the actual directory in case of Microsoft Visual C++™.
103
+
104
+ The directory `./patterns` provides the file `baudot.pcm` that serves as input signal for the test script `test_negotiation`. All output data of `test_negotiation` will be stored into the directory `./output`. If required, this directory will be created by `test_negotiation` automatically.
105
+
106
+ ## 3.2 Program execution
107
+
108
+ The CTM signal adaptation module is implemented in the executable `adaptation_switch` (in case of Sun Solaris™ platform) or `adaptation_switch.exe` (in case of the Micorsoft Windows™ platform).
109
+
110
+ The program should be called like:
111
+
112
+ ```
113
+ adaptation_switch -ctmin <file> -ctmout <file>
114
+ -baudotin <file> -baudotout <file>
115
+ ```
116
+
117
+ using the following parameters:
118
+
119
+ | | | |
120
+ |-----------------------------|----------------------------------|-----------------------------------------------|
121
+ | <code>-ctmin</code> | <code>&lt;input_file&gt;</code> | input file with CTM signal |
122
+ | <code>-ctmout</code> | <code>&lt;output_file&gt;</code> | output file for CTM signal |
123
+ | <code>-baudotin</code> | <code>&lt;input_file&gt;</code> | input file with Baudot Tones |
124
+ | <code>-baudotout</code> | <code>&lt;output_file&gt;</code> | output file for Baudot Tones |
125
+ | <code>-textout</code> | <code>&lt;text_file&gt;</code> | output text file from CTM receiver (optional) |
126
+ | <code>-numsamples</code> | <code>&lt;number&gt;</code> | number of samples to process (optional) |
127
+ | <code>-nonegotiation</code> | | disables the negotiation (optional) |
128
+
129
+ All files contain 16-bit linear encoded PCM audio samples, which are swapped according to the platform's endian type (Sun Microsystems platforms use big endian, Intel platforms use little endian). An example file `baudot.pcm` containing a Baudot Code modem signal (big endian) is provided in the subdirectory `./patterns`.
130
+
131
+ Due to the fact that the signal adaptation module expects a successful negotiation before Baudot Code signals can be converted to CTM signals, the signal adaptation module has to be executed several times in two instances in order to execute a successful negotiation. For the Sun Microsystems platform, a shell script `test_negotiation` is provided for executing the following structure:
132
+
133
+ ![](997233d405f0d4b89ddeb7683e047f66_img.jpg)
134
+
135
+ ```
136
+
137
+ ---------- ctm_forward ----------
138
+ baudot.pcm --->| | -------------------->| |---> baudot_out.pcm
139
+ | adapt#1 | | adapt#2 |
140
+ /dev/null <---| | <--------------------| |<--- /dev/zero
141
+ ---------- ctm_backward ----------
142
+
143
+ ```
144
+
145
+ First, the adaptation module #1 is executed. At this first run, the signal ctm\_backward is not known. Therefore, the negotiation does not get a positive acknowledge, so that the transmission falls back to Baudot Tones.
146
+
147
+ Then signal adaptation module #2 is executed for the first time.
148
+
149
+ After that, adaptation module #1 is executed for the second time. With this second run, the signal ctm\_backward is valid. Therefore, the negotiation receives a valid acknowledge, so that CTM signals are transmitted.
150
+
151
+ At last, adaptation module #2 is executed for the second time. With this run, adaptation module #2 receives a valid CTM signal so that the baudot\_out.pcm signal can be generated.
152
+
153
+ After executing each of the modules twice, the signal baudot\_out.pcm is analyzed. This analysis is also performed by the program adaptation\_switch. First, the Baudot detector of adaptation\_switch is used for this analysis in order to examine whether the regenerated Baudot signal can be decoded correctly. In a second step it is examined whether the regenerated signal still contains any CTM preambles. This investigation is performed by means of the CTM detector that is integrated in adaptation\_switch. This last test fails if the CTM detector is able to detect any CTM preamble in the regenerated signal.
154
+
155
+ During the execution of the script test\_negotiation the following text output shall be generated:
156
+
157
+ ```
158
+
159
+ ============================================================
160
+ Execute adaptation module #1 (first pass)
161
+ ============================================================
162
+
163
+ ***************************************************************************
164
+ Cellular Text Telephone Modem (CTM) - Example Implementation for
165
+ Conversion between CTM and Baudot Code (use option -h for help)
166
+ ***************************************************************************
167
+
168
+ number of samples to process: 100000
169
+
170
+ >>> Enquiry Burst generated! <<<
171
+ THE>>> Enquiry Burst generated! <<<
172
+ >>> Enquiry Burst generated! <<<
173
+ CELL
174
+
175
+ ============================================================
176
+ Execute adaptation module #2 (first pass)
177
+
178
+ ```
179
+
180
+ =====
181
+
182
+ \*\*\*\*\*
183
+
184
+ Cellular Text Telephone Modem (CTM) - Example Implementation for
185
+ Conversion between CTM and Baudot Code (use option -h for help)
186
+
187
+ \*\*\*\*\*
188
+
189
+ >>> CTM from far-end detected! <<<
190
+
191
+ >>> Enquiry From Far End Detected! <<<
192
+
193
+ THE>>> Enquiry From Far End Detected! <<<
194
+
195
+ >>> Enquiry From Far End Detected! <<<
196
+
197
+ CELL
198
+
199
+ =====
200
+
201
+ Execute adaptation module #1 (second pass)
202
+
203
+ =====
204
+
205
+ \*\*\*\*\*
206
+
207
+ Cellular Text Telephone Modem (CTM) - Example Implementation for
208
+ Conversion between CTM and Baudot Code (use option -h for help)
209
+
210
+ \*\*\*\*\*
211
+
212
+ >>> Enquiry Burst generated! <<<
213
+
214
+ THE>>> CTM from far-end detected! <<<
215
+
216
+ CELLULAR TEXT TELEPHONE MODEM (CTM) ALLOWS RELIABLE
217
+ TRANSMISSION OF A TEXT TELEPHONE CONVERSATION ALTERNATING
218
+ WITH A SPEECH CONVERSATION THROUGH THE EXISTING SPEECH
219
+ COMMUNICATION PATHS IN CELLULAR MOBILE PHONE SYSTEMS.
220
+ THIS RELIABILITY IS ACHIEVED BY AN IMPROVED MODULATION
221
+ TECHNIQUE, INCLUDING ERROR PROTECTION, INTERLEAVING AND
222
+ SYNCHRONIZATION.
223
+
224
+ =====
225
+
226
+ Execute adaptation module #2 (second pass)
227
+
228
+ =====
229
+
230
+ \*\*\*\*\*
231
+
232
+ Cellular Text Telephone Modem (CTM) - Example Implementation for
233
+ Conversion between CTM and Baudot Code (use option -h for help)
234
+
235
+ \*\*\*\*\*
236
+
237
+ >>> CTM from far-end detected! <<<
238
+
239
+ >>> Enquiry From Far End Detected! <<<
240
+
241
+ THE CELLULAR TEXT TELEPHONE MODEM (CTM) ALLOWS RELIABLE
242
+ TRANSMISSION OF A TEXT TELEPHONE CONVERSATION ALTERNATING
243
+ WITH A SPEECH CONVERSATION THROUGH THE EXISTING SPEECH
244
+ COMMUNICATION PATHS IN CELLULAR MOBILE PHONE SYSTEMS.
245
+ THIS RELIABILITY IS ACHIEVED BY AN IMPROVED MODULATION
246
+ TECHNIQUE, INCLUDING ERROR PROTECTION, INTERLEAVING AND
247
+ SYNCHRONIZATION.
248
+
249
+ =====
250
+
251
+ Now we try to decode the regenerated Baudot signal. The text message
252
+ shall be decoded completely now...
253
+
254
+ =====
255
+
256
+ \*\*\*\*\*
257
+
258
+ Cellular Text Telephone Modem (CTM) - Example Implementation for
259
+ Conversion between CTM and Baudot Code (use option -h for help)
260
+
261
+ \*\*\*\*\*
262
+
263
+ THE CELLULAR TEXT TELEPHONE MODEM (CTM) ALLOWS RELIABLE
264
+ TRANSMISSION OF A TEXT TELEPHONE CONVERSATION ALTERNATING
265
+ WITH A SPEECH CONVERSATION THROUGH THE EXISTING SPEECH
266
+ COMMUNICATION PATHS IN CELLULAR MOBILE PHONE SYSTEMS.
267
+ THIS RELIABILITY IS ACHIEVED BY AN IMPROVED MODULATION
268
+ TECHNIQUE, INCLUDING ERROR PROTECTION, INTERLEAVING AND
269
+ SYNCHRONIZATION.
270
+
271
+ =====
272
+
273
+ Testing whether the regenerated Baudot signal is free of CTM headers.
274
+
275
+ No CTM burst shall be detected now...
276
+
277
+ ```
278
+ =====
279
+
280
+ *****
281
+ Cellular Text Telephone Modem (CTM) - Example Implementation for
282
+ Conversion between CTM and Baudot Code (use option -h for help)
283
+ *****
284
+ ```
285
+
286
+ ## 3.3 Code hierarchy
287
+
288
+ This section gives an overview of the hierarchy how the functions are used in the signal adaptation module. All standard C functions: printf(), fwrite(), etc. have been omitted. Also, all functions related to the asynchronous transfer between the signal processing functions by means of FIFO buffers (Shortint\_fifo\_push, Shortint\_fifo\_pop, etc.) are not listed in the charts.
289
+
290
+ The following functions are not part of the actual CTM bit exact specification but are included to allow demonstration of CTM in a Baudot environment:
291
+
292
+ - init\_baudot\_tonedemod
293
+ - init\_baudot\_tonemod
294
+ - baudot\_tonedemod
295
+ - convertUCScode2char
296
+ - convertChar2TTYcode
297
+ - baudot\_tonemod
298
+ - convertTTYcode2char
299
+ - convertChar2UCScode
300
+
301
+ ### 3.3.1 Initialization routines
302
+
303
+ The following functions are called for the initialization of the signal adaptation module.
304
+
305
+ | | | |
306
+ |-----------------------|--------------------------|--------------------------------------------|
307
+ | init_baudot_tonedemod | | |
308
+ | init_baudot_tonemod | | |
309
+ | init_ctm_transmitter | init_interleaver | generate_scrambling_sequence<br>m_sequence |
310
+ | | init_tonemod | |
311
+ | | conv_encoder_init | |
312
+ | | generate_resync_sequence | m_sequence |
313
+ | | calc_mute_positions | |
314
+ | init_ctm_receiver | init_tonedemod | sin_fip |
315
+ | | viterbi_init | |
316
+ | | calc_mute_positions | |
317
+ | | init_deinterleaver | generate_scrambling_sequence |
318
+ | | init_wait_for_sync | m_sequence<br>generate_scrambling_sequence |
319
+
320
+ ### 3.3.2 Signal Processing Functions
321
+
322
+ The following functions are called during the main signal processing loop.
323
+
324
+ | | | |
325
+ |---------------------|------------------------|--------------|
326
+ | baudot_tonedemod | iir_filt | |
327
+ | ctm_receiver | tonedemod | rotate_right |
328
+ | | | rotate_left |
329
+ | | wait_for_sync | |
330
+ | | reinit_deinterleaver | |
331
+ | | viterbi_reinit | |
332
+ | | diag_deinterleaver | |
333
+ | | shift_deinterleaver | |
334
+ | | mutingRequired | |
335
+ | | viterbi_exec | |
336
+ | | reinit_wait_for_sync | |
337
+ | | reinit_deinterleaver | |
338
+ | | viterbi_reinit | |
339
+ | | transformUTF2UCS | |
340
+ | convertUCScode2char | | |
341
+ | convertChar2TTYcode | | |
342
+ | baudot_tonemod | | |
343
+ | convertTTYcode2char | | |
344
+ | convertChar2UCScode | | |
345
+ | ctm_transmitter | transformUCS2UTF | |
346
+ | | reinit_interleaver | |
347
+ | | conv_encoder_exec | |
348
+ | | mutingRequired | |
349
+ | | diag_interleaver | |
350
+ | | diag_interleaver_flush | |
351
+ | | tonemod | |
352
+ | | | |
353
+
354
+ ## 3.4 Description of global constants used in the C-code
355
+
356
+ The following constants are defined in the file `ctm_defines.h`
357
+
358
+ | Constant | Value | Description |
359
+ |----------------------------------------------|-------|---------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------|
360
+ | MAX_IDLE_SYMB | 5 | Number of Idle Symbols at End of Burst |
361
+ | CHC_RATE | 4 | Rate of the Error Protection |
362
+ | CHC_K | 5 | Constraint Length of the Error Protection |
363
+ | SYMB_LEN | 40 | Length of one CTM symbol |
364
+ | LENGTH_TONE_VEC | 1 | frame size |
365
+ | LENGTH_TX_BITS | 8 | number of bits per 20 ms frame |
366
+ | BITS_PER_SYMB | 8 | bits per symbol |
367
+ | NCYCLES_0 | 2 | Number of periods for symbol #0 |
368
+ | NCYCLES_1 | 3 | Number of periods for symbol #1 |
369
+ | NCYCLES_2 | 4 | Number of periods for symbol #2 |
370
+ | NCYCLES_3 | 5 | Number of periods for symbol #3 |
371
+ | THRESHOLD_RELIABILITY_FOR_SUPPRESSING_OUTPUT | 100 | Characters with lower reliability are suppressed |
372
+ | THRESHOLD_RELIABILITY_FOR_XCORR | 200 | Bits with lower reliability don't contribute to xcorr |
373
+ | THRESHOLD_RELIABILITY_FOR_GOING_OFFLINE | 100 | Threshold for regarding a bit as unreliable |
374
+ | MAX_NUM_UNRELIABLE_GROSS_BITS | 400 | Receiver goes offline after 400 unreliable bits |
375
+ | NUM_BITS_GUARD_INTERVAL | 6 | Number of muted bits between two bursts |
376
+ | WAIT_SYNC_REL_THRESHOLD_0 | 20316 | (=0.62) rel. threshold for preamble |
377
+ | WAIT_SYNC_REL_THRESHOLD_1 | 17039 | (=0.52) rel. threshold for preamble |
378
+ | WAIT_SYNC_REL_THRESHOLD_2 | 23065 | (=0.71) dto. in case that RX is already online |
379
+ | RESYNC_REL_THRESHOLD | 26542 | Threshold for Resynchronization (=0.81) |
380
+ | GUARD_BIT_SYMBOL | 10 | magic number indicating that a bit shall be muted |
381
+ | intlVB | 8 | Interleaver block length (number of rows) |
382
+ | intlVD | 2 | Interleaver block distance (interlace factor) |
383
+ | demodSyncLns | 1 | Number of demodulator sync lines |
384
+ | deintSyncLns | 0 | Number of deinterleaver sync lines |
385
+ | IDLE_SYMB | 0x16 | UCS code for Idle Symbol |
386
+ | ENQU_SYMB | 0x05 | UCS code for Enquiry Symbol |
387
+ | ENQUIRY_TIMEOUT | 3040 | number of 20-ms frames for negotiation |
388
+ | NUM_ENQUIRY_BURSTS | 3 | number of enquiry attempts |
389
+ | NUM_MUTE_ROWS | 4 | Number of Intl. rows that shall be muted |
390
+ | RESYNC_SEQ_LENGTH | 32 | length of the resynchronization sequence, must be a multiple of 8 |
391
+ | NUM_BITS_BETWEEN_RESYNC | 352 | Distance between two resync sequences, the value<br><code>NUM_BITS_BETWEEN_RESYNC+RESYNC_SEQ_LENGTH</code><br>must be a multiple of <code>CHC_RATE</code> , <code>intlVB</code> , and<br><code>BITS_PER_CHAR</code> , and must be greater than<br><code>intlVB*((intlVB-1)*intlVD+NUM_MUTE_ROWS)</code> |
392
+ | BAUDOT_NUM_INFO_BITS | 5 | number of information bits per Baudot character |
393
+ | BAUDOT_SHIFT_FIGURES | 27 | code of shift to figures symbol |
394
+
395
+ | | | |
396
+ |-----------------------|-----|---------------------------------------------------------------------------------------|
397
+ | BAUDOT_SHIFT_LETTERS | 31 | code of shift to letters symbol |
398
+ | BAUDOT_BIT_DURATION | 176 | must be 176 (for 45.45 baud) or 160 (50 baud) |
399
+ | BAUDOT_LP_FILTERORDER | 1 | Order of the low-pass filters in function<br>baudot_tonedemod() |
400
+ | BAUDOT_BP_FILTERORDER | 2 | Order of the according band-pass filters, must<br>be equal to 2*BAUDOT_BP_FILTERORDER |
401
+
402
+ ## 3.5 Type Definitions
403
+
404
+ In order to make the C code platform-independent, the following type definitions have been used, which are defined in `typedefs.h`:
405
+
406
+ | defined type | meaning | corresponding constants |
407
+ |--------------|-----------------|----------------------------|
408
+ | ----- | | |
409
+ | Char | character | (none) |
410
+ | Bool | boolean | true, false |
411
+ | Shortint | 16-bit signed | minShortint, maxShortint |
412
+ | UShortint | 16-bit unsigned | minUShortint, maxUShortint |
413
+ | Longint | 32-bit signed | minLongint, maxLongint |
414
+ | ULongint | 32-bit unsigned | minULongint, maxULongint |
415
+
416
+ ## 3.6 Functions of the C Code
417
+
418
+ ---
419
+
420
+ ```
421
+ void baudot_tonedemod(Shortint* toneVec, Shortint numSamples,
422
+ fifo_state_t* ptrOutFifoState,
423
+ baudot_tonedemod_state_t* state);
424
+ ```
425
+
426
+ Purpose: Demodulator for Baudot Tones
427
+ Defined in: `baudot_functions.c`
428
+
429
+ ### Input Variables:
430
+
431
+ `toneVec` Vector containing the input audio signal
432
+ `numSamples` Length of `toneVec`
433
+
434
+ ### Input/Output Variables:
435
+
436
+ `ptrOutFifoState` Pointer to the state of the output shift register
437
+ containing the demodulated TTY codes
438
+ `state` Pointer to the state variable of `baudot_tonedemod()`
439
+
440
+ ---
441
+
442
+ ```
443
+ void baudot_tonemod(Shortint inputTTYcode,
444
+ Shortint *outputToneVec,
445
+ Shortint lengthToneVec,
446
+ Shortint *ptrNumBitsStillToModulate,
447
+ baudot_tonemod_state_t* state);
448
+ ```
449
+
450
+ Purpose: Modulator for Baudot Tones
451
+ Defined in: `baudot_functions.c`
452
+
453
+ ### Input Variables:
454
+
455
+ inputTTYcode                TTY code of the character that has to be modulated.
456
+ inputTTYcode must be in the range 0...63, otherwise
457
+ it is assumed that there is no character to
458
+ modulate.
459
+ lengthToneVec              Indicates how many samples have to be generated.
460
+
461
+ Output Variables:
462
+ outputToneVec              Vector where the output samples are written to.
463
+ ptrNumBitsStillToModulate    Indicates how many bits are still in the fifo
464
+ buffer.
465
+
466
+ Input/Output Variables:
467
+ state                      Pointer to the state variable of baudot\_tonedemod()
468
+
469
+ ---
470
+
471
+ ```
472
+ void calc_mute_positions(Shortint *mute_positions,
473
+                           Shortint num_rows_to_mute,
474
+                           Shortint start_position,
475
+                           Shortint B,
476
+                           Shortint D);
477
+ ```
478
+
479
+ Purpose:                Calculation of the indices of the bits that have to be muted
480
+                           within one burst. The indices are returned in the vector
481
+                           mute\_positions.
482
+ Defined in:            init\_interleaver.c
483
+
484
+ ---
485
+
486
+ ```
487
+ Shortint convertChar2ttyCode(char inChar);
488
+ ```
489
+
490
+ Purpose:                Conversion from character into TTY code
491
+ Defined in:            baudot\_functions.c
492
+
493
+ Input Variables:
494
+
495
+ inChar                    character that shall be converted
496
+
497
+ Return Value:            baudot code of the input or -1 in case that inChar
498
+                           is not valid (e.g. inChar=='\0')
499
+
500
+ ---
501
+
502
+ ```
503
+ UShortint convertChar2UCScode(char inChar);
504
+ ```
505
+
506
+ Purpose:                Conversion from character into UCS code (Universal Multiple-
507
+                           Octet Coded Character Set, Row 00 of the Multilingual plane
508
+                           according to ISO/IEC 10646-1). This routine only handles
509
+                           characters in the range 0..255 since that is all that is
510
+                           required for demonstration of Baudot support.
511
+
512
+ Defined in:            ucs\_functions.c
513
+
514
+ Input Variables:
515
+
516
+ inChar                    character that shall be converted
517
+
518
+ Return Value:            UCS code of the input or 0x0016 <IDLE> in case that
519
+                           inChar is not valid (e.g. inChar=='\0')
520
+
521
+ ---
522
+
523
+ ```
524
+ char convertTTYcode2char(Shortint ttyCode);
525
+ ```
526
+
527
+ Purpose: Conversion from TTY code into Character
528
+ Defined in: baudot\_functions.c
529
+
530
+ ### Input Variables:
531
+
532
+ ttyCode Baudot code (must be within the range 0...63) or -1
533
+ if there is nothing to convert
534
+
535
+ ### Return Value:
536
+
537
+ character (or '\0' if ttyCode is not valid)
538
+
539
+ ---
540
+
541
+ ```
542
+ char convertUCScode2char(ushortint ucsCode);
543
+ ```
544
+
545
+ Purpose: Conversion from UCS code into character (Universal Multiple-Octet Coded Character Set, Row 00 of the Multilingual plane according to ISO/IEC 10646-1). This routine only handles characters in the range 0..255 since that is all that is required for demonstration of Baudot support.
546
+
547
+ Defined in: ucs\_functions.c
548
+
549
+ ### Input Variables:
550
+
551
+ ucsCode UCS code index, must be within the range 0...255
552
+
553
+ ### Return Value:
554
+
555
+ character (or '\0' if ucsCode is not valid)
556
+
557
+ ---
558
+
559
+ ```
560
+ void conv_encoder_exec(conv_encoder_t* ptr_state, Shortint* in, Shortint inbits, Shortint* out);
561
+ ```
562
+
563
+ Purpose: Execution of the convolutional encoder for error protection
564
+ Defined in: conv\_encoder.c
565
+
566
+ ### Input Variables:
567
+
568
+ in Vector with net bits
569
+ inbits Number of valid net bits in vector in
570
+
571
+ ### Output variables:
572
+
573
+ out Vector with the encoded gross bits. The gross bits are either 0 or 1. The vector out must have at least CHC\_RATE\*inbits elements.
574
+
575
+ ### Input/output variables:
576
+
577
+ \*ptr\_state state variable of the encoder
578
+
579
+ ---
580
+
581
+ ```
582
+ void conv_encoder_init(conv_encoder_t* ptr_state);
583
+ ```
584
+
585
+ Purpose: Initialization of the convolutional encoder
586
+ Defined in: conv\_encoder.c
587
+
588
+ ### Output Variables:
589
+
590
+ \*ptr\_state Initialized state variable of the encoder
591
+
592
+ ---
593
+
594
+ ```
595
+ void ctm_receiver(fifo_state_t* ptr_signal_fifo_state,
596
+ fifo_state_t* ptr_output_char_fifo_state,
597
+ Bool* ptr_early_muting_required,
598
+ rx_state_t* rx_state);
599
+ ```
600
+
601
+ Purpose: Runs the CTM Receiver for a block of (nominally) 160 samples. Due to the internal synchronization, the number of processed samples might vary between 156 and 164 samples. The input of the samples and the output of the decoded characters is handled via fifo buffers, which have to be initialized externally before using this function (see fifo.h for details).
602
+
603
+ Defined in: ctm\_receiver.c
604
+
605
+ #### input/output variables
606
+
607
+ ```
608
+ *ptr_signal_fifo_state fifo state for the input samples
609
+ *ptr_output_char_fifo_state fifo state for the output characters
610
+ *ptr_early_muting_required returns whether the original audio signal must not
611
+ be forwarded. This is to guarantee that the
612
+ preamble or resync sequence is detected only by the
613
+ first CTM device, if several CTM devices are
614
+ cascaded subsequently.
615
+ rx_state pointer to the variable containing the receiver
616
+ states
617
+ ```
618
+
619
+ ---
620
+
621
+ ```
622
+ void ctm_transmitter(ushortint ucsCode,
623
+ Shortint* txToneVec,
624
+ tx_state_t* tx_state,
625
+ Shortint *ptrNumBitsStillToModulate,
626
+ Bool sineOutput);
627
+ ```
628
+
629
+ Purpose: Runs the CTM Transmitter for a block of 160 output samples, representing 8 gross bits. The bits, which are modulated into tones, are taken from an internal fifo buffer. If the fifo buffer is empty, zero-valued samples are generated. The fifo buffer is filled with channel-encoded and interleaved bits, which are generated internally by coding the actual input character. With each call of this function one or less input characters can be coded. If there is no character to for transmission, one of the following codes has be used:
630
+
631
+ - 0x0016 <IDLE>: indicates that there is no character to transmit and that the transmitter should stay in idle mode, if it is currently already in idle mode. If the transmitter is NOT in idle mode, it might generate <IDLE> symbols in order to keep an active burst running. The CTM burst is terminated if five <IDLE> symbols have been generated consecutively.
632
+ - 0xFFFF: although there is no character to transmit, a CTM burst is initiated in order to signal to the far-end side that CTM is supported. The burst starts with the <IDLE> symbol and will be continued with <IDLE> symbols if there are no regular characters handed over during the next calls of this function. The CTM burst is terminated if five <IDLE> symbols have been transmitted consecutively.
633
+
634
+ In order to avoid an overflow of the internal fifo buffer, the
635
+
636
+ variable \*ptrNumBitsStillToModulate should be checked before calling this function.
637
+
638
+ Defined in: ctm\_transmitter.c
639
+
640
+ input variables:
641
+ ucsCode UCS code of the character or one of the code 0x0016 or 0xFFFF
642
+ sineOutput must be false in regular mode; if true, a pure sine output signal is generated
643
+
644
+ output variables:
645
+ txToneVec output signal (vector of 160 samples)
646
+
647
+ input/output variables:
648
+ tx\_state pointer to the variable containing the transmitter states
649
+
650
+ ---
651
+
652
+ ```
653
+ void diag_deinterleaver(Shortint *out,
654
+ Shortint *in,
655
+ Shortint num_valid_bits,
656
+ interleaver_state_t *intl_state);
657
+ ```
658
+
659
+ Purpose: Corresponding deinterleaver to diag\_interleaver. An arbitrary number of bits can be interleaved, depending of the length of the vector "in". The vector "out", which must have the same length than "in", contains the interleaved samples. All states (memory etc.) of the interleaver are stored in the variable \*intl\_state. Therefore, a pointer to this variable must be handled to this function. This variable initially has to be initialized by the function init\_interleaver, which offers also the possibility to specify the dimensions of the deinterleaver matrix.
660
+
661
+ Defined in: diag\_deinterleaver.c
662
+
663
+ ---
664
+
665
+ ```
666
+ void diag_interleaver(Shortint *out,
667
+ Shortint *in,
668
+ Shortint num_bits,
669
+ interleaver_state_t *intl_state);
670
+ ```
671
+
672
+ Purpose: Diagonal (chain) interleaver, based on block-by-block processing. An arbitrary number of bits can be interleaved, depending of the value num\_bits. The vector "out", which must have the same length than "in", contains the interleaved samples. All states (memory etc.) of the interleaver are stored in the variable \*intl\_state. Therefore, a pointer to this variable must be handled to this function. This variable initially has to be initialized by the function init\_interleaver(), which offers also the possibility to specify the dimensions of the interleaver matrix.
673
+
674
+ Defined in: diag\_interleaver.c
675
+
676
+ ---
677
+
678
+ ```
679
+ void diag_interleaver_flush(Shortint *out,
680
+ ```
681
+
682
+ ```
683
+ Shortint *num_bits,
684
+ interleaver_state_t *intl_state);
685
+ ```
686
+
687
+ Purpose: Execution of the diagonal (chain) interleaver without writing in new samples. The number of calculated output samples is returned via the value \*num\_bits.
688
+ Defined in: diag\_interleaver.c
689
+
690
+ ---
691
+
692
+ ```
693
+ void generate_resync_sequence(Shortint *sequence);
694
+ ```
695
+
696
+ Purpose: Generation of the sequence for resynchronization. The length of the sequence is defined by the global constant RESYNC\_SEQ\_LENGTH. The vector sequence must be allocated accordingly before calling this function.
697
+ Defined in: wait\_for\_sync.c
698
+
699
+ ---
700
+
701
+ ```
702
+ void generate_scrambling_sequence(Shortint *sequence, Shortint length);
703
+ ```
704
+
705
+ Purpose: Generation of the sequence used for scrambling. The sequence consists of 0 and 1 elements. The sequence is stored into the vector \*sequence and the length of the sequence is specified by the variable length.
706
+ Defined in: init\_interleaver.c
707
+
708
+ ---
709
+
710
+ ```
711
+ void init_baudot_tonedemod(baudot_tonedemod_state_t* state);
712
+ ```
713
+
714
+ Purpose: Initialization of the demodulator for Baudot Tones
715
+ Defined in: baudot\_functions.c
716
+
717
+ Input/Output Variables:
718
+
719
+ state Pointer to the initialized state variable (must be allocated before calling init\_baudot\_tonedemod())
720
+
721
+ ---
722
+
723
+ ```
724
+ void init_baudot_tonemod(baudot_tonemod_state_t* state);
725
+ ```
726
+
727
+ Purpose: Initialization of the modulator for Baudot Tones
728
+ Defined in: baudot\_functions.c
729
+
730
+ Input/Output Variables:
731
+
732
+ state Pointer to the initialized state variable (must be allocated before calling init\_baudot\_tonemod())
733
+
734
+ ---
735
+
736
+ ```
737
+ void init_deinterleaver(interleaver_state_t *intl_state,
738
+ Shortint B, Shortint D);
739
+ ```
740
+
741
+ Purpose: Initialization of the deinterleaver.
742
+ Defined in: init\_interleaver.c
743
+
744
+ ---
745
+
746
+ ```
747
+ void init_ctm_receiver(rx_state_t* rx_state);
748
+ ```
749
+
750
+ Purpose:            Initialization of the CTM Receiver.
751
+ Defined in:        ctm\_receiver.c
752
+
753
+ output variables:
754
+
755
+ rx\_state                                pointer to a variable of rx\_state\_t containing the
756
+                                          initialized states of the receiver
757
+
758
+ ---
759
+
760
+ ```
761
+ void init_ctm_transmitter(tx_state_t* tx_state);
762
+ ```
763
+
764
+ Purpose:            Initialization of the CTM Transmitter
765
+ Defined in:        ctm\_transmitter.c
766
+
767
+ input/output variables
768
+
769
+ tx\_state                                pointer to a variable of tx\_state\_t containing
770
+                                          initialized states of the transmitter
771
+
772
+ ---
773
+
774
+ ```
775
+ void init_interleaver(interleaver_state_t *intl_state,
776
+                       Shortint B, Shortint D,
777
+                       Shortint num_sync_lines1, Shortint num_sync_lines2);
778
+ ```
779
+
780
+ Purpose:            Function for initialization of diag\_interleaver and
781
+                       diag\_deinterleaver, respectively. The dimensions of the
782
+                       interleaver must be specified:
783
+                       B = (horizontal) blocklength, D = (vertical distance)
784
+                       According to this specifications, this function initializes a
785
+                       variable of type interleaver\_state\_t.
786
+                       Additionally, this function adds two types of sync information
787
+                       to the bitstream. The first sync info is for the demodulator
788
+                       and consists of a sequence of alternating bits so that the
789
+                       tones produced by the modulator are not the same all the time.
790
+                       This is essential for the demodulator to find the transitions
791
+                       between adjacent bits. The bits for this demodulator
792
+                       synchronization simply precede the bitstream.
793
+                       The second sync info is for synchronizing the deinterleaver
794
+                       and of a m-sequence with excellent autocorrelation properties.
795
+                       These bits are positioned at the locations of the dummy bits,
796
+                       which are not used by the interleaver. In addition, even more
797
+                       bits for this can be spent by inserting additional sync bits,
798
+                       which precede the interleaver's bitstream. This is indicated
799
+                       by choosing num\_sync\_lines2>0.
800
+ Defined in:        init\_interleaver.c
801
+
802
+ ---
803
+
804
+ ```
805
+ void init_tonedemod(demod_state_t *demod_state);
806
+ ```
807
+
808
+ Purpose:            Initialization of one instance of the Tone Demodulator. The
809
+                       argument must contain a pointer to a variable of type
810
+
811
+ demod\_state\_t, which contains all the memory of the tone demodulator. Each instance of tonedemod must have its own variable.
812
+ Defined In: tonedemod.c
813
+
814
+ ---
815
+
816
+ ```
817
+ void init_wait_for_sync(wait_for_sync_state_t *ptr_wait_state,
818
+ interleaver_state_t intl_state);
819
+ ```
820
+
821
+ Purpose: Initialization of the synchronization detector. The dimensions of the corresponding interleaver at the TX side must be specified by the variables B, D, and num\_sync\_lines2.
822
+ Defined In: wait\_for\_sync.c
823
+
824
+ ### Input Variables:
825
+
826
+ B (horizontal) blocklength
827
+ D (vertical) interlace factor
828
+ num\_Sync\_line2 number of interleaver lines with additional sync bits (see description of init\_interleaver())
829
+
830
+ ### Output Variables:
831
+
832
+ ptr\_wait\_state pointer to the state variable of the sync detector
833
+
834
+ ---
835
+
836
+ ```
837
+ int main(int argc, const char** argv)
838
+ ```
839
+
840
+ Purpose: main function of the signal adaptation Module
841
+ Defined in: adaptation\_switch.c
842
+
843
+ ---
844
+
845
+ ```
846
+ Bool mutingRequired(Shortint actualIndex,
847
+ Shortint *mute_positions,
848
+ Shortint length_mute_positions);
849
+ ```
850
+
851
+ Purpose: Determines whether the actual bit has to be muted, i.e. whether it is contained in the vector mute\_positions.
852
+ Defined in: init\_interleaver.c
853
+
854
+ ---
855
+
856
+ ```
857
+ void m_sequence(Shortint *sequence, Shortint length);
858
+ ```
859
+
860
+ Purpose: Calculates one period of an m-sequence (binary pseudo noise). The sequence is stored in the vector sequence, which must have a of $(2^r)-1$ , where r is an integer number between 2 and 10. Therefore, with this release of m\_sequence, sequences of length 3, 7, 15, 31, 63, 127, 255, 511, or 1023 can be generated. The resulting sequence is bipolar, i.e. it has values -1 and +1.
861
+ Defined in: m\_sequence.c
862
+
863
+ ---
864
+
865
+ ```
866
+ void polynomials(Shortint rate, Shortint k,
867
+ Shortint* polya, Shortint* polyb,
868
+ Shortint* polyc, Shortint* polyd);
869
+ ```
870
+
871
+ Purpose: Returns the polynomials for the convolutional encoder and the Viterbi decoder for various rates and constraint lengths. The following parameters are supported:
872
+ rate = {2, 3, or 4}
873
+ k = {3, 4, 5, 6, 7, 8, 9}
874
+ Defined in: conv\_poly.c
875
+
876
+ ### Input Variables:
877
+
878
+ rate Rate of the convolutional encoder (2, 3, or 4)
879
+ k Constraint length (length of the impulse response of the encoder)
880
+
881
+ ### Output Variables:
882
+
883
+ poly\_a Vector with polynomials #1
884
+ poly\_b Vector with polynomials #2
885
+ poly\_c Vector with polynomials #3 (only if rate > 2)
886
+ poly\_d Vector with polynomials #4 (only if rate > 3)
887
+
888
+ ---
889
+
890
+ ```
891
+ void reinit_deinterleaver(interleaver_state_t *intl_state);
892
+ ```
893
+
894
+ Purpose: Re-Initialization of the deinterleaver.
895
+ Defined in: init\_interleaver.c
896
+
897
+ ---
898
+
899
+ ```
900
+ void reinit_interleaver(interleaver_state_t *intl_state);
901
+ ```
902
+
903
+ Purpose: Re-initialization of the deinterleaver
904
+ Defined in: init\_interleaver.c
905
+
906
+ ---
907
+
908
+ ```
909
+ void reinit_wait_for_sync(wait_for_sync_state_t *ptr_wait_state);
910
+ ```
911
+
912
+ Purpose: Reinitialization of synchronization detector. This function is used in case that a burst has been finished and the transmitter has switched into idle mode. After calling reinit\_wait\_for\_sync(), the function wait\_for\_sync() inhibits the transmission of the demodulated bits to the deinterleaver, until the next synchronization sequence can be detected.
913
+ Defined In: wait\_for\_sync.c
914
+
915
+ ---
916
+
917
+ ```
918
+ void shift_deinterleaver(Shortint shift,
919
+ Shortint *insert_bits,
920
+ interleaver_state_t *ptr_state);
921
+ ```
922
+
923
+ Purpose: Shift of the deinterleaver buffer by <shift> samples.
924
+ shift>0 -> shift to the right
925
+
926
+ shift<0 -> shift to the left
927
+ The elements from <insert\_bits> are inserted into the
928
+ resulting space. The vector <insert\_bits> must have at least
929
+ abs(shift) elements.
930
+ Defined in: diag\_deinterleaver.c
931
+
932
+ Shortint sin\_fip(Shortint phase\_value);
933
+
934
+ Purpose: Fixed Point sine function, returns the following value:
935
+
936
+ $$\sin\_fip(phase\_value) = \text{round}(32767 * \sin(2 * \pi * 50 / 8000 * phase\_value))$$
937
+ phase\_value must be within the range [0..159]. This function
938
+ can be used for calculating sine waveforms of frequencies that
939
+ are integer-multiples of 50 Hz
940
+ Defined in: sin\_fip.c
941
+
942
+ void tonedemod(Shortint \*bits\_out,
943
+ Shortint \*rx\_tone\_vec,
944
+ Shortint num\_in\_samples,
945
+ Shortint \*ptr\_sampling\_correction,
946
+ demod\_state\_t \*demod\_state);
947
+
948
+ Purpose: Tone Demodulator for the CTM using one out of four tones for
949
+ coding two bits in parallel within a frame of 40 samples (5
950
+ ms).
951
+ The function has to be called for every frame of 40 samples of
952
+ the received tone sequence. However, in order to track a
953
+ non-ideal of the transmitter's and the receiver's clock
954
+ frequencies, one frame might be shorter (only 39 samples) or
955
+ longer (41 samples). The length of the following frame is
956
+ indicated by the variable \*sampling\_correction, which is
957
+ calculated and returned by this function.
958
+
959
+ Defined in: tonedemod.c
960
+
961
+ ### input variables:
962
+
963
+ bits\_out contains the 39, 40 or 41 actual samples of the
964
+ received tones; the bits are soft bits, i.e. they
965
+ are in the range between -1.0 and 1.0, where the
966
+ magnitude serves as reliability information
967
+ num\_in\_samples number of valid samples in bits\_out
968
+
969
+ ### output variables:
970
+
971
+ bits\_out contains the two actual decoded soft bits
972
+ sampling\_correction is either -1, 0, or 1 and indicates whether the
973
+ next frame shall contain 39, 40, or 41 samples.
974
+ demod\_state contains all the memory of tonedemod. Must be
975
+ initialized using the function init\_tonedemod()
976
+
977
+ void tonemod(Shortint \*tones\_out,
978
+ Shortint \*bits\_in,
979
+ Shortint num\_samples\_tones\_out,
980
+
981
+ ```
982
+ Shortint num_bits_in,
983
+ mod_state_t *mod_state);
984
+ ```
985
+
986
+ Purpose:            Modulator for the CTM. The input vector bits\_in must contain the bits that have to be transmitted. The length of bits\_in must be even because always two bits are coded in parallel. Bits are either unipolar (i.e. {0, 1}) or bipolar (i.e. {-1, +1}). The length of the output vector tones\_out must be 20 times longer than the length of bits\_in, since each pair of two bits is coded within a frame of 40 audio samples.
987
+
988
+ Defined In:        tonemod.c
989
+
990
+ ```
991
+ void transformUCS2UTF(UShortint        ucsCode,
992
+                               fifo_state_t* ptr_octet_fifo_state);
993
+ ```
994
+
995
+ Purpose:            Transformation from UCS code into UTF-8. UTF-8 is a sequence consisting of 1, 2, 3, or 5 octets (bytes). See ISO/IEC 10646-1 Annex G.
996
+ This routine only handles UCS codes in the range 0...0xFF since that is all that is required for the demonstration of Baudot support.
997
+
998
+ Defined In:        ucs\_functions.c
999
+
1000
+ ### Input Variables:
1001
+
1002
+ ucsCode                    UCS code index
1003
+
1004
+ ### Output Variables:
1005
+
1006
+ ptr\_octet\_fifo\_state    pointer to the output fifo state buffer for the UTF-8 octets.
1007
+
1008
+ ```
1009
+ Bool transformUTF2UCS(UShortint        *ptr_ucsCode,
1010
+                               fifo_state_t* ptr_octet_fifo_state)
1011
+ ```
1012
+
1013
+ Purpose:            Transformation from UTF-8 into UCS code.
1014
+
1015
+ This routine only handles UTF-8 sequences consisting of one or two octets (corresponding to UCS codes in the range 0...0xFF) since that is all that is required for the demonstration of Baudot support.
1016
+
1017
+ Defined In:        ucs\_functions.c
1018
+
1019
+ ### Input/Output Variables:
1020
+
1021
+ ptr\_octet\_fifo\_state    pointer to the input fifo state buffer for the UTF-8 octets.
1022
+
1023
+ ### Output Variables:
1024
+
1025
+ \*ptr\_ucsCode            UCS code index
1026
+
1027
+ ### Return Value:
1028
+
1029
+ | | |
1030
+ |--------|--------------------------------------------------------------------------------------------------------------------------------------------------------------|
1031
+ | true, | if conversion was successful |
1032
+ | false, | if the input fifo buffer didn't contain enough octets for a conversion into UCS code. The output variable *ptr_ucsCode doesn't contain a value in this case. |
1033
+
1034
+ ---
1035
+
1036
+ ```
1037
+ void viterbi_exec(Shortint* inputword, Shortint length_input,
1038
+ Shortint* out, Shortint* num_valid_out_bits,
1039
+ viterbi_t* viterbi_state);
1040
+ ```
1041
+
1042
+ Purpose: Execution of the Viterbi decoder
1043
+ Defined in: viterbi.c
1044
+
1045
+ ### Input Variables:
1046
+
1047
+ | | |
1048
+ |--------------|----------------------------------------------------------------------------------------------------------|
1049
+ | inputword | Vector with gross bits |
1050
+ | length_input | Number of valid gross bits in vector inputword.<br>length_input must be an integer multiple of CHC_RATE. |
1051
+
1052
+ ### Output variables:
1053
+
1054
+ | | |
1055
+ |---------------------|-------------------------------------------------------------------|
1056
+ | out | Vector with the decoded net bits. The net bits are either 0 or 1. |
1057
+ | *num_valid_out_bits | Number of valid bits in vector out. |
1058
+
1059
+ ### Input/output variables:
1060
+
1061
+ | | |
1062
+ |----------------|-------------------------------|
1063
+ | *viterbi_state | state variable of the decoder |
1064
+ |----------------|-------------------------------|
1065
+
1066
+ ---
1067
+
1068
+ ```
1069
+ void viterbi_init(viterbi_t* viterbi_state);
1070
+ ```
1071
+
1072
+ Purpose: Initialization of the Viterbi decoder
1073
+ Defined in: viterbi.c
1074
+
1075
+ ### Output Variables:
1076
+
1077
+ | | |
1078
+ |----------------|-------------------------------------------|
1079
+ | *viterbi_state | Initialized state variable of the decoder |
1080
+ |----------------|-------------------------------------------|
1081
+
1082
+ ---
1083
+
1084
+ ```
1085
+ void viterbi_reinit(viterbi_t* viterbi_state);
1086
+ ```
1087
+
1088
+ Purpose: Re-Initialization of the Viterbi decoder. This function should be used for re-setting a Viterbi decoder that has already been initialized. In contrast to init\_viterbi(), this reinit function does not calculate the values of all members of viterbi\_state that do not change during the execution of the Viterbi algorithm.
1089
+ Defined in: viterbi.c
1090
+
1091
+ ### Output Variables:
1092
+
1093
+ | | |
1094
+ |----------------|-------------------------------------------|
1095
+ | *viterbi_state | Initialized state variable of the decoder |
1096
+ |----------------|-------------------------------------------|
1097
+
1098
+ ---
1099
+
1100
+ ```
1101
+ Bool wait_for_sync(Shortint *out_bits,
1102
+ Shortint *in_bits,
1103
+ ```
1104
+
1105
+ ```
1106
+
1107
+ Shortint num_in_bits,
1108
+ Shortint num_received_idle_symbols,
1109
+ Shortint *ptr_num_valid_out_bits,
1110
+ Shortint *ptr_wait_interval,
1111
+ Shortint *ptr_resync_detected,
1112
+ Bool *ptr_early_muting_required,
1113
+ wait_for_sync_state_t *ptr_wait_state);
1114
+
1115
+ ```
1116
+
1117
+ **Purpose:** This function shall be inserted between the demodulator and the deinterleaver. The function searches the synchronization bitstream and cuts all received heading bits. As long as no sync is found, this function returns \*ptr\_num\_valid\_out\_bits=0 so that the main program is able to skip the deinterleaver as long as no valid bits are available. If the sync info is found, the complete internal shift register is copied to out\_bits so that wait\_for\_sync can be transparent and causes no delay for future calls. \*ptr\_wait\_interval returns a value of 0 after such a synchronization indicating that this was a regular synchronization.
1118
+
1119
+ Regularly, the initial preamble of each burst is used as sync info. In addition, the resynchronization sequences, which occur periodically during a running burst, are used as "back-up" synchronization in order to avoid loosing all characters of a burst, if the preamble was not detected.
1120
+
1121
+ If the receiver is already synchronized on a running burst and the resynchronization sequence is detected, \*ptr\_resync\_detected returns a non-negative value in the range 0...num\_in\_bits-1 indicating at which bit the resynchronization sequence has been detected. If no resynchronization has been detected, \*ptr\_resync\_detected is -1. If the receiver is NOT synchronized and the resynchronization sequence is detected, the resynchronization sequence is used as initial synchronization. \*ptr\_wait\_interval returns a value of 32 in this case due to the different alignments of the synchronizations based on the preamble or the resynchronization sequence, respectively.
1122
+
1123
+ In order to carry all bits, the minimum length of out\_bits must be
1124
+ in\_bits.size()-1 + ptr\_wait\_state->shift\_reg\_length
1125
+
1126
+ **Defined In:** wait\_for\_sync.c
1127
+
1128
+ ### **InputVariables:**
1129
+
1130
+ | | |
1131
+ |-------------|------------------------------------------------------------------------------------------------------------------------------------------------------------------------|
1132
+ | in_bits | Vector with bits from the demodulator. The vector's length can be arbitrarily chosen, i.e. according to the block length of the signal processing of the main program. |
1133
+ | num_in_bits | length of vector in_bits |
1134
+
1135
+ ### **Output Variables:**
1136
+
1137
+ | | |
1138
+ |---------------------------|---------------------------------------------------------------------------------------------------------------|
1139
+ | num_received_idle_symbols | Number if idle symbols received coherently |
1140
+ | out_bits | Vector with bits for the deinterleaver. The number of the valid bits is indicated by *ptr_num_valid_out_bits. |
1141
+ | *ptr_num_valid_out_bits | returns the number of valid output bits |
1142
+ | *ptr_wait_interval | returns either 0 or 32 |
1143
+ | *ptr_resync_detected | returns a value -1, 0,...num_in_bits |
1144
+
1145
+ `*ptr_early_muting_required` returns whether the original audio signal must not be forwarded. This is to guarantee that only the first CTM device will detect the preamble or resync sequence, if several CTM devices are cascaded subsequently.
1146
+
1147
+ Input/Output Variables:
1148
+
1149
+ `ptr_wait_state` state information. This variable must be initialized with `init_wait_for_sync()`.
1150
+
1151
+ # Annex A (informative): Change history
1152
+
1153
+ | Change history | | | | | | | |
1154
+ |----------------|---------|-----------|-----|-----|------------------------------------------------------------------------|--------|--------|
1155
+ | Date | TSG SA# | TSG Doc. | CR | Rev | Subject/Comment | Old | New |
1156
+ | 12-2000 | 10 | SP-000570 | | | Specification approved for Release 4 | | 4.0.0 |
1157
+ | 03-2001 | 11 | SP-010108 | 001 | | Bug fix in source code of the CTM receiver | 4.0.0 | 5.0.0 |
1158
+ | 05-2001 | | | | | Correct source code CTM attached | 5.0.0 | 5.0.1 |
1159
+ | 07-2004 | | | | | Removed copyright terms and conditions in the source code CTM attached | 5.0.1 | 5.0.2 |
1160
+ | 12-2004 | 26 | | | | Version for Release 6 | 5.0.2 | 6.0.0 |
1161
+ | 06-2007 | 36 | | | | Version for Release 7 | 6.0.0 | 7.0.0 |
1162
+ | 03-2008 | 39 | SP-080006 | 002 | 1 | Bug fix to baudot_tonemod function in baudot_functions.c | 7.0.0 | 7.1.0 |
1163
+ | 12-2008 | 42 | | | | Version for Release 8 | 7.1.0 | 8.0.0 |
1164
+ | 12-2009 | 46 | | | | Version for Release 9 | 8.0.0 | 9.0.0 |
1165
+ | 03-2011 | 51 | | | | Version for Release 10 | 9.0.0 | 10.0.0 |
1166
+ | 09-2012 | 57 | | | | Version for Release 11 | 10.0.0 | 11.0.0 |