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5.7.6.5.2 Signaling interleaved waveform coding
The use of frequency-interleaved waveform coding in an A-SPX interval is indicated by the bitstream element aspx_fic_present. The presence of a frequency-interleaved waveform component in a particular subband group is indicated by the bitstream element aspx_fic_used_in_sfb. If a channel pair stereo coded, i.e. aspx_bal...
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5.7.6.5.3 Interleaving WCC and SEC
The WCC components are contained in the matrix QinASPX. The SEC are contained in the matrix Y. For frequency-interleaved waveform coding, the WCC are added to the SEC as follows: . {/012!"#$, ) = {/34!"#$, −5%&'() +  6,  {/012!"#$, ) = {/34!"#$, −5%&'() + {6, ) For time-interleaved waveform coding, t...
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5.7.7 Advanced Coupling tool - A-CPL
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5.7.7.1 Introduction
The Advanced Coupling tool is a coding tool for improved coding of signals with more than one channel. The tool operates in one of the following configurations: • Advanced Coupling in the channel pair element: For improved stereo coding. This configuration is described in clause 5.7.7.5. • Advanced Coupling in the mult...
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5.7.7.2 Parameter band to QMF subband mapping
Advanced Coupling parameters are transmitted per parameter band. Parameter bands are groupings of QMF subbands and have lower frequency resolution than the QMF subbands. The mapping of parameter bands to QMF subbands is shown below in table 191. The number of parameter bands - 7, 9, 12, or 15 - is given by the variable...
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5.7.7.3 Interpolation
Decoded and dequantized advanced coupling parameters carried in the bitstream are time-interpolated to calculate values that are applied to the input of the decorrelator and to the ducked output of the decorrelator. Parameter sets are transmitted either once or twice per frame, which is determined by the variable acpl_...
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5.7.7.4 Decorrelator and transient ducker
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5.7.7.4.1 Introduction
The decorrelation filters consist of a number of all-pass infinite impulse response (IIR) filter sections preceded by a constant frequency-dependent delay. The frequency axis of the QMF time/frequency matrix is divided into three different regions as shown in table 192, which correspond to the QMF split frequencies giv...
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5.7.7.4.2 Decorrelator IIR filtering
The decorrelator filters for the different frequency regions given in table 192 are implemented by a delay and a subsequent IIR filter. Pseudocode inputSignalModification(x) { for (sb = 0; sb < num_qmf_subbands; sb++) { for (ts = 0; ts < num_qmf_timeslots; ts++) { b[0] = a[filterLength(sb)]; y[ts][sb] = b[0]*x[n-delay(...
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5.7.7.4.3 Transient ducker
To be able to handle transients and other fast time-envelopes, the output of the decorrelator all-pass filter has to be attenuated and this is performed according to the pseudocode below: Pseudocode alpha = 0.76592833836465; alpha_smooth = 0.25; gamma = 1.5; epsilon = 1.0e-9; acpl_max_num_param_bands = 15; for (pb = 0;...
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5.7.7.5 Advanced coupling in the channel pair element
For Advanced Coupling in the channel pair element, two input channels are present, left (L) and right (R). Their elements are addressed as follows: • Left input channel x0(ts,sb) QinACPL,L • Right input channel x1(ts,sb) QinACPL,R The output channels are addressed as • Left input channel z0(ts,sb) QoutACPL,L • Right in...
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5.7.7.6 Advanced coupling in the multichannel element
5.7.7.6.1 5_X_codec_mode ∈ {ASPX_ACPL_1, ASPX_ACPL_2} For the configuration used when 5_X_codec_mode = ASPX_ACPL_1 or ASPX_ACPL_2, two parallel Advanced Coupling modules are used. There five input channels present. Their elements are addressed as: • Left input channel (L) x0(ts,sb) QinACPL,L • Right input channel (R) x...
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5.7.7.6.2 5_X_codec_mode =ASPX_ACPL_3
For the configuration used when 5_X_codec_mode = ASPX_ACPL_3 three parallel Advanced Coupling modules are used. There five input channels present: Their elements are addressed as: • Left input channel (L) x0(ts,sb) QinACPL,L • Right input channel (R) x1(ts,sb) QinACPL,R • Center channel (C) x2(ts,sb) QinACPL,C • Left s...
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5.7.7.7 Dequantization
The dequantized values acpl_alpha_1_dq, acpl_alpha_2_dq, acpl_beta1_dq, and acpl_beta2_dq are obtained using tables 197 and 198 if the quantization mode is set to fine, and using tables 199 and 200 if the quantization mode is set to coarse. Tables 197 and 199 are used to calculate dequantized alpha values, for fine and...
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5.7.8 Dialog Enhancement
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5.7.8.1 Introduction
The Dialog Enhancement tool is a tool to increase intelligibility of the dialog in an audio scene encoded in AC-4. The underlying algorithm uses metadata encoded in the bitstream to boost the dialog in the scene. This technology is especially beneficial for the hearing impaired. Dialog Enhancement supports enhancement ...
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5.7.8.2 Processed Channels
Each of the three decoding methods described below operates, by default, on the QMF domain time-frequency samples of a subset of the three front channels in one AC-4 frame: 7,  ∈/34)*,+,,-., 0 ≤+ℎ & ≤3 As the dialog may be limited to either, all, or a combination of the three front channels, de_channel_...
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5.7.8.3 Dequantization
The parameter index data shall be dequantized using the quantization vectors in tables 203 and 204.  0 ,  = : _ [ ][ ] , 0 < < __  Table 203: Quantization vector for channel-independent enhancement mode Parameter index Parameter value 0 0 1 0,1 2 0,2 3 0,3 4 0,4 5 0,5 6 0,6 7 0,7 8 0,8 9 0,9 10 1,0 11 ...
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5.7.8.4 Parameter Bands
The parametric enhancement methods rely on segmenting the num_qmf_subbands QMF subbands into de_nr_bands = 8 dialog enhancement parameter bands. The mapping of QMF subbands to parameter bands is described in clause 4.3.14.5.1. Hence, the entries of a parameter subset for channel i:  0 ,  = _ [ ][ ], 0 < < _...
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5.7.8.5 Rendering
In modes with cross-channel dialog enhancement the parametrically generated dialog enhancing signal shall be mixed into the audio scene according to the mixing parameters transmitted in the bitstream. When de_nr_channels indicates one involved channel, the dialog enhancement signal will be added to this channel only. T...
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5.7.8.6 Interpolation
For a smooth transition between parameter sets, interpolation is applied between successive frames. Clause 5.7.8.7 through clause 5.7.8.9 describe the parameter-based processing for each frame , in terms of a matrix operation. y = ; 23(,) ∙m dc The application of the dialog enhancement parameters shall facilitate cha...
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5.7.8.7 Parametric Channel Independent Enhancement
In this enhancement method, each channel i which has been indicated for processing is processed separately. A set of parameters  is transmitted for each. Additionally, a maximum overall gain % , defined in clause 4.3.14.3.2, limits the user-defined gain g to a range determined at encoding stage. Each time-frequency...
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5.7.8.8 Parametric Cross-channel Enhancement
This method focuses on enhancing the dialog using a combination of up to three input channels. The formula for reconstructing the dialog is noted as follows: • 1 processed channel 9 =  +  ∙# ∙ ∙ • 2 processed channels A9 9B = A B + A B ∙# ∙  ∙A B • 3 processed channels ? 9 991 @ = ?  1 @ + ?  ...
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5.7.8.9 Waveform-Parametric Hybrid
The hybrid dialog enhancement modes complement the parametric enhancement parameters with transmission of up to three waveforms. The encoder also transmits a trade-off parameter, $, derived from de_signal_contribution, that indicates how the enhancement contributions should be divided over the two approaches. The resu...
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5.7.9 Dynamic range control (DRC) tool
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5.7.9.1 Introduction
This clause describes a DRC (dynamic range control) processing unit that is configured by the AC-4 bitstream. For an explanation of how DRC can be used in the scope of an audio codec please refer to document [i.1], clause 6.7.1.1. DRC is operated in the QMF domain and is executed before QMF synthesis. Data and Control ...
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5.7.9.2 DRC Profiles
Compressor operation is controlled by DRC modes. Modes are configured either through a compression curve or by direct gains to be applied, similar to [i.1], clause 6.7.1. AC-4 supports four default modes: • Home theatre mode • Flat-panel mode • Portable mode - Speakers • Portable mode - Headphones Additional modes may ...
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5.7.9.3 Decoding process
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5.7.9.3.1 Compression Curves
When the DRC profile specifies a compression curve (drc_compression_curve_flag == 1), the DRC processor, as depicted in figure 9, comprises the following functions: • Level computation. This function calculates the current signal level. • Gain computation. This function maps the current level into a gain, controlled by...
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5.7.9.3.2 Directly transmitted DRC Gains
For profiles where DRC gains are transmitted (drc_compression_curve_flag == 0), there are four configurations supported for transmitting gain values in the AC-4 bitstream. • Single wideband gain for all channels • Wideband gain for each channel group • Multiband for each channel group, 2 bands ETSI ETSI TS 103 190 V1.1...
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5.7.9.3.3 Application of gain values
As a last step, the DRC tool shall map the dialnorm level to the output level Lout, and apply the gains that have either been calculated per clause 5.7.9.3.1.2 or transmitted as per clause 5.7.9.3.2. Qout,, = 10   ∙, , ∙Qin1,,
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5.7.9.4 Transcoding to a AC-3 or E-AC-3 format
When an AC-4 bitstream is transcoded into a format such as AC-3 or E-AC-3, no DRC processing should be applied. Instead, the encoder or transcoder should be configured with the compression curve indicated by the field drc_eac3_transcode_curve in table 154.
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6 Decoding the AC-4 bitstream
6.1 Introduction Clause 4 specifies the details of the AC-4 bitstream syntax and clause 5 specifies the algorithmic details of the tools used within AC-4. This clause describes the complete AC-4 decoding process and by this makes a connection between the bitstream elements and the AC-4 tools. Figure 10 shows a flow dia...
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6.2.1 Input bitstream
The input bitstream will typically come from a transmission or storage system. To decode an AC-4 bitstream all raw_ac4_frame elements shall be extracted according to the used transport format and presented to the AC-4 decoder.
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6.2.2 Structure of the bitstream
An AC-4 bitstream is roughly structured in several containers, namely: • Raw AC-4 Frame The actual codec frame, consisting of a table of contents plus several byte aligned substreams. • Table Of Contents (TOC) Signaling information on how to treat the content of the substream container of the AC-4 frame. • Substream En...
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6.2.2.1 Raw AC-4 Frame
Each raw AC-4 frame contains a table of contents (TOC) and at least one byte aligned substream. The TOC can be considered as the bitstream inventory where all information that is important for the overlaying system resides. Substreams are decodable units that represent a specific channel configuration (mono, stereo, 3....
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6.2.2.2 Table of Contents (TOC)
The Table of Contents contains the TOC data and at least one presentation as well as a substream index table. As the bitstream inventory the TOC provides information about and access to individual substreams. The TOC data contains stream specific metadata, such as: • Bitstream version Signals if the bitstream is decoda...
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6.2.2.3 Presentation information
Typically, a presentation in AC-4 describes a set of substreams that is to be decoded and presented simultaneously. Mixing of decoded content will only occur within one presentation. That means only one presentation has to be decoded at a time. On the other hand, one substream can be part of several presentations. Exam...
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6.2.2.4 Substream
A substream contains the actual audio data, as well as the corresponding metadata. Additionally, the size of the audio data is given in the header of the substream, such that the metadata can be accessed without the need of parsing the audio data. The ac4_substream () element is specified in clause 4.3.4.
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6.2.3 Selecting and decoding a presentation
To successfully select and decode an appropriate presentation, a decoder system has to execute the following steps. 1) Create a list of presentations available in the bitstream Initially, the number of presentations, n_presentations, has to be derived from the bitstream elements b_single_presentation and b_more_present...
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6.2.4 Buffer model
Decoders implemented in accordance with the present document shall provide a minimum input buffer of size N bytes where N is set according to: = ×    and  are the total bitrate and the frame_rate, respectively, and =  6,  ≤60 12,  . See table 205 for typical values of N. NOTE: VBR strea...
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6.2.5 Decoding of a substream
As described in clause 4.2.4 there are three different types of AC-4 substreams: • AC-4 substream • AC-4 high sampling frequency extension substream • Universal metadata payloads substream
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6.2.5.1 Decoding of an AC-4 substream
A decoder shall decode the AC-4 substream's audio_data() and metadata(). Both bitstream elements can be accessed independently of each other by using the variable audio_size. Decoding of the channel elements within audio_data() is described in clause 6.2.6 and following. The bitstream elements contained in metadata() s...
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6.2.5.2 Decoding of an AC-4 HSF extension substream
Decoding of the HSF substream can be achieved using the information derived from clause 5.4. Afterwards, the SAP tool and the IMDCT described in clauses 5.3 and 5.5 need to be employed. No QMF domain processing is required.
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6.2.5.3 Decoding of a UMD payloads substream
A decoder shall be able to skip a UMD substream, and may otherwise ignore its contents.
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6.2.6 Spectral frontend decoding
The channel element contained in the audio_data element of an AC-4 substream can be one of four types, signaled by the channel_mode, as indicated in clause 4.2.6. Within the channel elements, audio tracks are encoded as sf_data elements. These elements are associated with an sf_info element, as well as with a spectral ...
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6.2.6.1 Mono decoding
In case that channel_mode is mono, there is just one track in the bitstream stored as sf_data element within a mono_data element. The decoding of the sf_data element depends on the spectral frontend type and on the sf_info element and is done in 3 steps: 1) Parsing of sf_info which includes transform and psy informatio...
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6.2.6.2 Stereo decoding
In case that channel_mode is stereo, there are up to two audio tracks in the bitstream, stored as sf_data elements within a channel_pair_element. Similar to the mono decoding, the decoding of the sf_data element is done in 3 steps, see clause 6.2.6.1. If the b_enable_mdct_stereo_proc value is '1', a common sf_info elem...
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6.2.6.3 Multichannel audio decoding
Similar to stereo decoding, first all the sf_data elements are decoded. followed by routing the decoded tracks into the stereo and multichannel processing tool. This tool returns dedicated audio channels. The spectral lines of these channels shall subsequently be IMDCT processed. In case the channel mode is "5.1" or "7...
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6.2.6.4 Audio spectral frontend (ASF)
The asf_section_data and asf_scalefac_data elements of the sf_data element are parsed and helper elements are set up. See clauses 4.3.6.3 and 4.3.6.5. The asf_spectral_data is Huffman decoded using the helper elements. See clause 5.1.2 for details of the Huffman decoding process for the quantized spectral lines. Next, ...
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6.2.6.5 Speech spectral frontend (SSF)
The sf_data element contains just the ssf_data element if the spec_frontend value is '1'. Either one or two ssf_granule elements are present in the ssf_data element. Each ssf_granule element is decoded into a vector of spectral lines as specified in clause 5.2.
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6.2.7 Inverse transform (IMDCT) and window overlap/add
The decoding steps described above will result in a set of frame_length (reordered) spectral lines for each spectral frontend channel. The inverse MDCT transform converts the blocks of spectral lines into blocks of time samples. The individual blocks of time samples shall be windowed, and adjacent blocks shall be overl...
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6.2.8 QMF analysis
The time domain samples of each channel are transformed individually into the QMF domain by the use of a QMF analysis filter as described in clause 5.7.3. Depending on the frame_length, 32, 30, 24, 15, 12 or 6 QMF slots are the result of the QMF analysis. Each QMF slot is the result of one QMF filtering step with num_q...
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6.2.9 Companding
The companding tool mitigates pre- and post-echo artefacts. The encoder applies attenuation to signal parts with higher energy, and gain to signal parts with lower energy. The decoder applies the inverse process, which effectively shapes the coding noise by the signal energy. Whether or not a channel is processed by th...
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6.2.10 Advanced spectral extension - A-SPX
Spectral extension includes translating the spectral content of suitable low frequency regions of the signal to higher frequencies and adjusting them such that they perceptually match the high frequency content that is present in the original signal. Waveform coding may be employed in addition, either when critical sig...
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6.2.11 Advanced coupling - A-CPL
Advanced coupling extracts multichannel content from a downmix into fewer channels, using encoder controlled side information. As a special case, a stereo signal can be generated from a mono downmix. An AC-4 decoder shall process the QMF domain channels with the A-CPL tool after the A-SPX tool has been applied. A-CPL s...
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6.2.12 Dialog Enhancement - DE
Dialog Enhancement control metadata can be specified in the bitstream. When the decoder is configured to enable dialog enhancement, the dialog enhancement tool uses this metadata to apply a gain on the dialog in the signal according to a system-defined gain. Table 209 lists the input channels for the DE tool. Table 209...
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6.2.13 Dynamic range control - DRC
Dynamic range control metadata can be specified in the bitstream. The DRC tool uses this metadata to apply gains to the channels in the QMF domain in order to adjust the dynamic range as specified in clause 5.7.9. The DRC tool takes two inputs per channel: the audio signal which is output by a preceding tool (usually d...
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6.2.14 QMF synthesis
A QMF synthesis filter shall transform each audio channel from the QMF domain back to the time domain as specified in clause 5.7.4.
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6.2.15 Sampling rate converter
For all values of the frame_rate_index, the decoder may be operated at external sampling frequencies of 48 kHz, 96 kHz, or 192 kHz. For frame_rate_index = 13, 44,1 kHz is also a permissible external sampling frequency. To adjust the sampling frequency to the external sampling frequency a sampling rate converter shall b...
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6.2.16 Mixing substream outputs
This clause describes how to mix several substreams contained within one presentation. This includes: • Mixing main + associated audio • Mixing dry main + dialog Dialog enhancement is not part of this clause. The decoding of a presentation which contains multiple substreams, as indicated by a b_single_substream value o...
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6.2.16.1 Mixing dry main and dialog
Before mixing the dialog channels, a single user agent provided gain g_dialog [-∞, g_dialog_max] dB should be applied to all dialog channels. This allows for a user-controlled adjustment of the relative dialog level. The user agent default value for g_dialog shall be 0 dB. Each channel which is only present in the dry ...
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6.2.16.2 Mixing main and associate
Before mixing the associate channels, a single user agent provided gain g_assoc [-∞, 0] dB should be applied to all associate channels. This allows for a user-controlled adjustment of the relative associate level. The user agent default value for g_assoc shall be 0 dB. The extended_metadata part of the associate substr...
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6.2.16.3 Mixing dry main, dialog and associate
The mixing of dry main, dialog and associate substream outputs is done in two stages. First, the dry main and the dialog channels are mixed as specified in clause 6.2.16.1. The resulting main signal is mixed with the associate signal as described in clause 6.2.16.2.
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6.2.17 Rendering a presentation
When the number of loudspeakers does not match the number of encoded audio channels either downmixing or upmixing is required to render the complete audio programme. It is important that the equations for downmixing and upmixing be standardized, so that programme providers can be confident of how their programme will b...
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6.2.17.1 Generalized rendering matrix and equation
The generalized rendering matrix, R, is as shown below. ' , , ⋯ ,  , ,, ⋯ ,  ⋮ ⋮ ⋱ ⋮ , , ⋯ ,  ( where n is the number of channels listed in table D.1. The elements r0,0 to rn-1,n-1 are defined for each combination of available input and output channels. The rendering equation is as follows: ' _"...
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6.2.17.2 Downmixing from two channels into one channel
The rendering matrix R2:1 used to downmix two channels (L,R) into one channel (C) has the elements r1,0 and r1,2 set to 1 and all other elements of the matrix set to 0.
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6.2.17.3 Downmixing from three channels into two channels
The rendering matrix R3:2 used to downmix from three channels into two channels is dependent on the bitstream element preferred_dmx_method, described in clause 4.3.12.2.19. The non-zero entries of the rendering matrix R3:2 are given in table 210. Table 210: Matrix entries for 3- to 2-channel downmix matrix preferred_dm...
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6.2.17.4 Downmixing from five channels into two channels
The rendering matrix R5:2 used to downmix from five channels into two channels is dependent on the bitstream element preferred_dmx_method, described in clause 4.3.12.2.19. The non-zero entries of the rendering matrix R5:3 are given in table 211. Table 211: Matrix entries for 5- to 2-channel downmix matrix preferred_dmx...
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6.2.17.5 Downmixing from seven channels into five channels
The rendering matrix R7:5 used to downmix seven channels into five channels is dependent on the channel mode, as well as on the bitstream element add_ch_base. Center channel (r1,1 = 1) and LFE (if present) are routed through (r18,18 = 1). The remaining non-zero entries of the rendering matrix R7:5 are given in table 21...
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6.2.17.6 Upmixing from one channel into two channels
The rendering matrix R1:2 used to upmix from a mono channel to stereo has the elements r1,0 and r1,2 set to 0,707 and all other elements of the matrix set to 0.
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6.2.17.7 Upmixing from two channels into three channels
The rendering matrix R2:3 used to upmix from two channels to three channels has the elements r0,0, and r2,2 set to 1 and all other elements of the matrix set to 0.
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6.2.17.8 Upmixing from three channels into five channels
The rendering matrix R3:5 used to upmix from three channels to five channels has the elements r0,0, r1,1, and r2,2 set to 1and all other elements of the matrix set to 0.
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6.2.17.9 Upmixing from five channels into seven channels
The rendering matrix R5:7 used to upmix from five channels into seven channels has the elements r0,0, r1,1, r2,2, r3,3, and r4,4 set to 1. The LFE channel (if present) is routed through (r18,18 = 1) as well. All other elements of the matrix are set to 0.
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6.2.18 Decoding audio in sync with video
In order to achieve A/V sync, a synchronization mechanism, usually employing presentation timestamps, needs to be defined in application standards referencing the present document. If such a timestamp is defined and applies to a certain audio frame, it should indicate the presentation time of the sample at the referenc...
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6.2.19 Switching streams while decoding
AC-4 has been designed to allow for a seamless switching of AC-4 streams during the run-time of an AC-4 decoder. When the switch happens at an I-frame boundary, which is the case when the first frame of the new stream is an I- frame, the AC-4 decoder shall produce a flawless output. When the switch does not happen at a...
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1 Scope
The present document specifies a set of end-to-end speech and video quality test extensions to TS 186 011-1 [4], TS 102 901 [2] and TS 103 029 [3]. In particular, it describes end-to-end tests based on instrumental assessment of speech samples, end-to-end tests based on instrumental assessment of video samples, other e...
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2 References
References are either specific (identified by date of publication and/or edition number or version number) or non-specific. For specific references, only the cited version applies. For non-specific references, the latest version of the referenced document (including any amendments) applies. Referenced documents which a...
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2.1 Normative references
The following referenced documents are necessary for the application of the present document. [1] ETSI ES 201 168: "Speech Processing, Transmission and Quality Aspects (STQ); Transmission characteristics of digital Private Branch eXchanges (PBXs) for interconnection to private networks, to the public switched network o...
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2.2 Informative references
The following referenced documents are not necessary for the application of the present document but they assist the user with regard to a particular subject area. [i.1] ETSI EG 201 377-1: "Speech and multimedia Transmission Quality (STQ);Specification and measurement of speech transmission quality; Part 1: Introductio...
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3 Abbreviations
For the purposes of the present document, the following abbreviations apply: 3GPP 3rd Generation Partnership Project AGC Automatic Gain Control AMR-WB Adaptive Multi-Rate Wideband ASL Active Speech Level CIF Common Intermediate Format CS Composite Source CSS Composite Source Signal CTI (ETSI) Centre for Testing and Int...
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4.1 Tests cases described in TS 186 011-2
4.1.1 User-initiated VoIP call setup and release (TS 186 011-2, clause 4.4.2) After step 33 of the functional test UC_02_I (A calls B) and after step 48 of the functional test UC_02_R (A call B, B is roaming) the following speech quality tests are carried out: Table 1: Summary of speech quality tests realized within th...
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4.1.2 Addition of media stream (TS 186 011-2, clause 4.4.11)
After steps 63A and 63B of the functional test UC_13 the quality tests defined in clause 4.1.1 (voice media stream) or clause 4.1.4 (video media stream) are carried out. After the quality tests are completed the functional test continues with step 64.
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4.1.3 Ad-hoc Conferencing service (TS 186 011-2, clause 4.4.13)
After step 59 of the functional test UC_16 the following speech quality tests are carried out: ETSI ETSI TS 103 189 V1.2.1 (2014-04) 13 Table 2: Summary of speech quality tests realized within the functional test UC_16 QoS_Voice_ac_01 QoS_Voice_ac_02 QoS_Voice_ac_03 QoS_Voice_ac_04 QoS_Voice_ac_05 QoS_Voice_ac_06 QoS_V...
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4.1.4 IPTV service (TS 186 011-2, clause 4.4.14)
After step 7 of the functional test UC_19 the following video quality tests are carried out: Table 3: Summary of video quality tests realized within the functional test UC_19 QoS_Video_filebased_01 QoS_Video_func_01 QoS_Video_np_01 ETSI ETSI TS 103 189 V1.2.1 (2014-04) 14 After the video quality tests are completed the...
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4.1.5 IMS-to-PSTN call (TS 186 011-2, clause 4.4.16.1)
After step 27 of the functional test UC_23 the following speech quality tests are carried out: Table 4: Summary of speech quality tests realized within the functional test UC_23 QoS_Voice_ac_01 QoS_Voice_ac_02 QoS_Voice_ac_03 QoS_Voice_ac_04 QoS_Voice_ac_05 QoS_Voice_ac_06 QoS_Voice_ac_07 QoS_Voice_ac_08 QoS_Voice_ac_0...
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4.1.6 PSTN-to-IMS call (TS 186 011-2, clause 4.4.16.2)
After step 28 of the functional test UC_24 the following speech quality tests are carried out: Table 5: Summary of speech quality tests realized within the functional test UC_24 QoS_Voice_ac_01 QoS_Voice_ac_02 QoS_Voice_ac_03 QoS_Voice_ac_04 QoS_Voice_ac_05 QoS_Voice_ac_06 QoS_Voice_ac_07 QoS_Voice_ac_08 QoS_Voice_ac_0...
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4.2 Tests cases described in TS 102 901
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4.2.1 TD_IMS_SHARE_0001 (TS 102 901, clause 4.5.4.1.1)
After step 30 of the functional test TD_IMS_SHARE_0001 the following video quality tests are carried out: Table 6: Summary of video quality tests realized within the functional test TD_IMS_SHARE_0001 QoS_Video_filebased_01 QoS_Video_func_01 QoS_Video_np_01 After the video quality tests are completed the functional test...
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4.2.2 TD_IMS_SHARE_0002 (TS 102 901, clause 4.5.4.1.2)
After step 45 of the functional test TD_IMS_SHARE_0002 the following video quality tests are carried out: Table 7: Summary of video quality tests within the functional test TD_IMS_SHARE_0002 QoS_Video_filebased_01 QoS_Video_func_01 QoS_Video_np_01 After the video quality tests are completed the functional test continue...
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4.3 Tests and test sessions
The assessment of end-to-end quality shall be an integral part of the interoperability tests. In order to allow an adaptation to the practical arrangements of a test event, the selection of quality parameters from the present document is up to the test house and shall be documented together with the results. Due to the...
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4.4 Test Reports
Test reports should contain as many details of the equipment under test, such as SW and HW versions, settings, type of codec and other signal enhancement procedures activated during the test. The media sequences injected and those retrieved shall be stored in a database.
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5 Test overview
This clause provides an overview of end-to-end quality test suite realized as a part of VoLTE and RCS Interop Events. It starts with a general test description and carries on with basic descriptions of all tests associated with the end-to-end quality test suite.
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5.1 General Test Description
The tests are based on the ETSI TC INT specifications as described in the latest versions of TS 186 011-1 (Test purposes for IMS NNI Interoperability) [4], TS 186 011-2 (Test descriptions for IMS NNI Interoperability) [5], TS 102 901 (IMS NNI interoperability test descriptions for RCS) [2], TS 103 029 (IMS & EPC Intero...
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5.4 Other end-to-end tests in the voice channel
The following parameters shall be tested in the end-to-end voice channel: • End-to-end frequency response: Non-linearities in the overall end-to-end frequency response will be perceived as quality degradation. Although the main influence on this parameter comes from the terminals, there might be undesired influences ca...
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5.5 End-to-end Tests based on functional parameters
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5.5.1 Voice Service
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5.5.1.1 Telephony Call Setup Time
Abstract Definition: The telephony setup time describes the time period between sending the complete address information and receiving the call set-up notification. Abstract Equation: ( )[s] t - t = [s] Time Setup Telephony on UE button send presses user d establishe connect where tconnectestablished is a time when a c...
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5.5.2 Video Streaming Service
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5.5.2.1 Streaming Service Access Time
Abstract Definition: The parameter Streaming Service Access Time describes the duration of a service access from requesting the stream at the portal until the reception of the first stream data packet at the UE. The first data packet refers to the RTP protocol. Abstract Equation: ( )[s] t - t = [s] Time Access Service ...