ES2946235T3 - Stereo audio signal encoder - Google Patents

Abstract

A method comprising: receiving at least two audio channel signals; determining, for a first frame, at least two parameters representing a difference between the at least two channel audio signals; scalar quantizing (451) the at least two parameters to generate at least two index values; determining an initial index map for reordering (453) one of the at least two index values, and determining at least one further index map for reordering at least one other of the at least two index values, wherein the at least a further index map is determined based on one of the at least two index values; reorder one of the at least two index values based on the initial index map; reordering the other of the at least two index values based on the at least one additional index map; encoding (455) the reordering of the at least two index values depending on an order position of the reordering of the at least two index values; encoding the most reordered of the at least two index values based on an order position of the most reordered of the at least two index values; generating a single channel representation of at least two audio channel signals based on at least two parameters; and encoding (456) the representation of a single channel. (Automatic translation with Google Translate, without legal value)

Description

DESCRIPTION

Stereo audio signal encoder

Field

The present application relates to a stereo audio signal encoder and, in particular, but not exclusively, to a stereo audio signal encoder for use in portable devices.

Background

Audio signals, such as speech or music, are encoded, for example, to allow efficient transmission or storage of the audio signals.

Audio encoders and decoders (also known as codecs) are used to represent audio-based signals such as music and environmental sounds (which in speech coding terms can be called background noise). These types of encoders typically do not use a speech model for the encoding process, instead they use processes to represent all kinds of audio signals, including speech. Speech encoders and decoders (codecs) can be thought of as audio codecs that are optimized for speech signals and can operate at a fixed or variable bit rate.

An audio codec can also be configured to operate with variable bit rates. At lower bit rates, such an audio codec can be optimized to work with speech signals at a coding rate equivalent to a pure speech codec. At higher bit rates, the audio codec can encode any signal, including music, background noise, and speech, with superior quality and performance. A variable rate audio codec can also implement an integrated scalable coding bitstream and structure, where additional bits (a specific number of bits often referred to as a layer) improve coding at lower rates, and where the bits of a higher rate can be truncated to obtain the bit stream of a lower rate encoding. Such an audio codec may use a codec designed solely for speech signals as the core layer or lower bit rate coding.

An audio codec is designed to maintain high (perceptual) quality while improving the compression ratio. Therefore, instead of waveform coincidence coding, it is common to employ various parametric schemes to reduce the bit rate. For multi-channel audio, such as stereo signals, it is common to use a greater amount of the available bit rate in a mono channel representation and encode the stereo or multi-channel information taking advantage of a parametric approach that uses relatively few bits.

Current voice and audio standardization efforts in the Third Generation Partnership Project (3GPP) aim to increase the quality of the encoded signal through coding efficiency, bandwidth, as well as the number of channels. A stereo/binaural extension is in preparation for the Enhanced Voice Services (EVS) voice and audio codec candidate. The coding efficiency for this approach is important, especially for lower codec bit rates. As the addition of a large bitrate extension would diminish the benefits of having an extension, if the total bitrate equals or exceeds the bitrate of a dual mode.

The proposed stereo/binaural extension is composed of encoded stereo parameters. Increasing the encoding efficiency for these parameters means lowering the bitrate of the extension and using the 'saved' bits for better encoding of the mono downmix. This is particularly useful at low bit rates where the quality of the encoded downmix is more sensitive to bit rate.

By addressing the coding efficiency of the stereo parameters, significant bit savings can be achieved. The coding efficiency of the stereo parameters has involved quantization of the values (levels), followed by entropy coding to further reduce the bit rate. A previously proposed method for encoding stereo parameters described in EP2856776 uses an adaptive version of Golomb Rice coding.

US2016/027445 discloses an apparatus comprising a mapper configured to map an instance of a parameter according to a first mapping to generate a first mapped instance; a remapper configured to remap the first mapped instance depending on the frequency distribution of mapped instances to generate a remapped instance with an associated order position; and an encoder configured to encode the remapped instance depending on an order position of the remapped instance.

US2015/194160 discloses an audio encoder for encoding coefficient slices, the coefficient slices representing different time or frequency resolutions of a sampled audio signal. The audio encoder includes a processor for deriving an encoding context for a currently encoded coefficient of a current segment based on a previously encoded coefficient of a previous segment, the previously encoded coefficient representing a different time or frequency resolution than the currently encoded coefficient. The audio encoder further includes an entropy encoder for entropy encoding the current coefficient based on the encoding context to obtain an encoded audio stream.

US2005/015249 discloses an audio encoder that performs adaptive entropy coding of audio data. For example, an audio encoder switches between variable dimension vector Huffman encoding of direct levels of quantized audio data and string level encoding of string lengths and levels of quantized audio data. The encoder may use, for example, context-based arithmetic encoding to encode string lengths and levels. The encoder can determine when to switch between encoding modes by counting consecutive coefficients that have a predominant value (eg, zero). An audio decoder performs the corresponding adaptive entropy decoding.

Document WO2014/013294 presents an apparatus comprising: a channel analyzer configured to determine at least one set of parameters that define a difference between at least two audio signal channels; a value analyzer configured to analyze the at least one set of parameters to determine an initial trend; a mapper configured to map instances of the at least one parameter set according to a first mapping to generate mapped instances with associated order position instances based on the initial trend; and an encoder configured to encode the mapped instances based on the order position of the mapped instances.

Summary

According to a first aspect, there is provided an apparatus as defined in claim 1.

According to a second aspect, there is provided a method as claimed in claim 10.

According to a third aspect, there is provided a method as defined in claim 13.

According to a second aspect, there is provided a method as defined in claim 14.

Brief description of the drawings

For a better understanding of the present invention, reference will now be made by way of example to the accompanying drawings, in which:

Figure 1 schematically shows an electronic device employing some embodiments;

Figure 2 schematically shows an audio codec system according to some embodiments;

Figure 3 schematically shows an encoder as shown in Figure 2 according to some embodiments; Figure 4 schematically shows a channel analyzer as shown in Figure 3 in more detail according to some embodiments;

Figure 5 schematically shows a stereo channel encoder as shown in Figure 3 in more detail according to some embodiments;

Figure 6 shows a flowchart illustrating the operation of the encoder shown in Figure 2 according to some embodiments;

Figure 7 shows a flowchart illustrating the operation of the channel analyzer as shown in Figure 4 according to some embodiments;

Figure 8 shows a flowchart illustrating the operation of the channel encoder as shown in Figure 5 according to some embodiments;

Figure 9 schematically shows the decoder as shown in Figure 2 according to some embodiments; and Figure 10 shows a flowchart illustrating the operation of the decoder as shown in Figure 9 according to some embodiments.

Description of some embodiments of the application

Possible stereo and multi-channel audio and speech codecs are described in more detail below, including layered or scalable variable rate audio and speech codecs. As discussed above, a previously proposed method of encoding stereo parameters described in EP2856776 uses an adaptive version of Golomb Rice coding.

The concept, as expressed in the embodiments described herein below, attempts to better capture and exploit intra-frame value mapping and, as a consequence, further reduce bit rate consumption for encoding stereo parameters.

As such, the implementations explicitly store the ordering of the first-order probabilities of the symbols to be encoded (instead of adaptively ordering them). In other words, for a single data frame, based on a previously encoded symbol, an array of integers maintains the ordering of probabilities for each symbol. In other words, 0 if it is the most likely, 1 if it is the second most likely, and so on. The probability order value is then encoded with an adaptive GR code.

In this regard, reference is first made to Figure 1, which shows a schematic block diagram of an illustrative electronic device or apparatus 10, which may incorporate a codec according to one embodiment of the application.

The apparatus 10 can be, for example, a mobile terminal or user equipment of a wireless communication system. In other embodiments, apparatus 10 may be an audio-video device, such as a video camera, television (TV) receiver, audio recorder, or audio player, such as an mp3 recorder/player, a media recorder (also known as mp4 recorder/player), or any computer suitable for audio signal processing.

Electronic device or apparatus 10 in some embodiments comprises a microphone 11, which is connected via an analog-to-digital converter (ADC) 14 to a processor 21. Processor 21 is further connected via a digital-to-digital converter 32 (DAC) to the speakers 33. The processor 21 is further connected to a transceiver (RX/TX) 13, a user interface (UI) 15 and a memory 22.

Processor 21 can be configured in some embodiments to execute various program codes. Program codes implemented in some embodiments comprise a multi-channel or stereo encoding or decoding code as described herein. The implemented program codes 23 may be stored in some embodiments, for example, in memory 22 for retrieval by processor 21 whenever necessary. The memory 22 could further provide a section 24 for storing data, eg data that has been encoded according to the application.

The encoding and decoding code in embodiments may be implemented in hardware and/or firmware.

The user interface 15 allows a user to enter commands into the electronic device 10, eg via a keyboard, and/or obtain information from the electronic device 10, eg via a display. In some embodiments, a touch screen may provide both input and output functions for the user interface. The apparatus 10 in some embodiments comprises a transceiver 13 suitable to allow communication with other apparatus, for example, through a wireless communication network.

It should again be understood that the structure of the apparatus 10 could be supplemented and varied in many ways.

A user of apparatus 10, for example, can use microphone 11 to input speech or other audio signals to be transmitted to some other apparatus or stored in data section 24 of memory 22. The user can activate a corresponding application in some embodiments for this purpose via the user interface 15 . This application in these embodiments can be performed by processor 21, it causes processor 21 to execute the encryption code stored in memory 22.

Analog-to-digital converter (ADC) 14 in some embodiments converts the input analog audio signal to a digital audio signal and provides the digital audio signal to processor 21. In some embodiments, microphone 11 may comprise an integrated function microphone and ADC and provide digital audio signals directly to the processor for processing.

The processor 21 in such embodiments then processes the digital audio signal in the same manner as described with reference to the system shown in Figure 2, the encoder shown in Figures 2 to 8 and the decoder shown in Figures 9 and 10. .

The resulting bit stream may in some embodiments be provided to transceiver 13 for transmission to other apparatus. Alternatively, the encoded audio data in some embodiments may be stored in the data section 24 of the memory 22, for example, for later transmission or for later presentation by the same apparatus 10.

Apparatus 10 in some embodiments may also receive a correspondingly encoded data bit stream from another apparatus via transceiver 13. In this example, processor 21 may execute the decoding program code stored in memory 22. Processor 21 in such embodiments, it decodes the received data and provides the decoded data to a digital-to-analog converter 32. The digital-to-analog converter 32 converts the digital decoded data to analog audio data and, in some embodiments, outputs the audio analog through the speakers 33. The execution of the decoding program code in some embodiments may also be triggered by an application called by the user through the user interface 15.

Received encoded data in some embodiment may also be stored instead of immediate presentation through speakers 33 in data section 24 of memory 22, for example, for later decoding and presentation or decoding and forwarding to further apparatus. .

It would be appreciated that the schematic structures depicted in Figures 3, 5, 7 and 9, and the process steps shown in Figures 4, 6, 8 and 10 represent only part of the operation of an audio codec and specifically part of an encoder/decoder apparatus or method as shown illustratively implemented in the apparatus shown in Figure 1.

The general operation of the audio codecs as employed by the embodiments is shown in Figure 2. General audio encoding/decoding systems comprise both an encoder and a decoder, as schematically illustrated in Figure 2. However, it would be understood that some embodiments may implement one of both the encoder and the decoder, or both the encoder and the decoder. Illustrated in Figure 2 is a system 102 with an encoder 104 and, in particular, a stereo encoder 151, a storage or media channel 106 and a decoder 108. It would be understood that, as described above, some embodiments may comprise or implement one of encoder 104 or decoder 108 or both encoder 104 and decoder 108.

Encoder 104 compresses an input audio signal 110 producing a bit stream 112, which in some embodiments may be stored or transmitted via media channel 106. Encoder 104 may further comprise a stereo encoder 151 as part of the overall encoding operation. It should be understood that the stereo encoder may be part of the overall encoder 104 or a separate encoding module. Encoder 104 may also comprise a multi-channel encoder that encodes more than two audio signals.

Bit stream 112 can be received into decoder 108. Decoder 108 decompresses bit stream 112 and produces an output audio signal 114 . Decoder 108 may comprise a stereo decoder as part of the overall decoding operation. It should be understood that the stereo decoder may be part of the overall decoder 108 or a separate decoding module. Decoder 108 may also comprise a multi-channel decoder that decodes more than two audio signals. The bit rate of the bit stream 112 and the quality of the output audio signal 114 relative to the input signal 110 are the main characteristics that define the performance of the encoding system 102.

Referring to Figure 3, an illustrative encoder 104 is shown according to some embodiments.

The encoder 104 in some embodiments comprises a frame splitter/transformer 201. Frame splitter/transformer 201 is configured to receive the left and right input audio signals (or more generally any multi-channel audio representation) and generate frequency domain representations of these audio signals for analysis and encoding. These frequency domain representations may be passed to the channel parameter determiner 203.

In some embodiments, the slicer/frame transformer 201 may be configured to slice or segment the audio signal data into sections or frames suitable for frequency domain transformation. Frame splitter/transformer 201 in some embodiments may be further configured to window these frames or sections of audio signal data according to any suitable windowing function. For example, frame splitter/transformer 201 may be configured to generate 20 ms frames that overlap the preceding and following frames by 10 ms each.

In some embodiments, frame splitter/transformer 201 may be configured to perform any suitable time-to-frequency domain transformation on the audio signal data. For example, the transformation from time to frequency domain can be a Discrete Fourier Transform (DFT), a Fast Fourier Transform (FFT), a Modified Discrete Cosine Transform (MDCT). The following examples use a Fast Fourier Transform (FFT). Furthermore, the output of the time domain to frequency domain transformer may be further processed to generate separate frequency band domain representations (subband representations) of each input channel audio signal data. These bands may be arranged in any suitable manner. For example, these bands may be linearly spaced, or perceptually or psychoacoustically assigned. In some embodiments, the frequency domain representations are passed to a channel analyzer 203.

In some embodiments, encoder 104 may comprise a channel analyzer 203. The channel analyzer 203 may be configured to receive the subband filtered representations of the multi-channel or stereo input. Channel analyzer 203 may further be configured in some embodiments to analyze audio signals in the frequency domain and determine the parameters associated with each subband with respect to differences from stereo or multi-channel audio signals. Furthermore, the channel analyzer 203 can use these parameters and generate a mono channel.

The stereo parameters and mono parameters/signals may then be output to a mono quantizer/encoder processor 205.

In some embodiments, encoder 104 comprises a mono quantizer/encoder processor 205. The mono quantizer/encoder processor 205 may be configured to receive the stereo parameters (difference) determined by the channel analyzer 203 . The mono quantizer/encoder processor 205 may then be configured in some embodiments to perform quantization on the parameters and further encode the parameters so that they can be output (either to be stored in the apparatus or passed to additional apparatus). The mono quantizer/encoder processor 205 can be further configured to receive the mono parameters/channel and further encode the mono parameters/channel using any suitable encoding and further based on the number of bits used to encode the stereo parameters. In other words, first the stereo parameters are encoded and then the downmix signal is encoded. The bits that are saved using entropy coding for the stereo parameters can be used to down-code the mixed signal.

In some embodiments, the encoder comprises a signal output 207. The signal output, as shown in Figure 3, represents an output configured to pass the encoded stereo parameters to be stored or transmitted to an additional device.

With respect to Figure 4, a summary of the encoding process according to some embodiments and the operation of the encoder 104 shown in Figure 3 is shown as a flowchart.

The operation of generating audio frame band frequency domain representations is shown in Figure 4 by step 501.

The operation of determining the stereo parameters is shown in Figure 4 by step 502.

The operation of generating the mono channel (downmix) parameters is shown in Figure 4 by step 503. The operation of quantizing the stereo (multichannel) parameters and encoding the quantized stereo (multichannel) parameters is shown in Figure 4 by step 504.

The operation of encoding the mono channel parameters (downmix) based on the bit usage of the optimized quantized stereo parameters is shown in Figure 4 by step 505.

The output of the encoded quantized stereo (multichannel) parameters and the encoded mono (downmix) parameters/signal is shown in Figure 4 by step 507.

With respect to Figure 5, an illustrative channel analyzer 203 according to some embodiments is described in more detail. In some embodiments, the channel analyzer 203 comprises a channel difference parameter determiner 301. The channel difference parameter determiner 301 is configured to determine the various channel difference parameters. In the following examples, the input audio signals are left and right audio signals. In some embodiments, this can be generalized as j and j+1 audio channels of a multi-channel audio system.

For example, channel difference parameter determiner 301 may be configured to receive the following parameters from frame splitter/transformer 201,

i component of the DFT of the right channel,

- component i of the DFT of the left channel.

Furthermore, these can be represented as real and imaginary parts, such as for the right channel and

- real part of the component of order i of the DFT of the right channel,

- imaginary part of the component of order i of the DFT of the channel.

From these components, the channel difference determinator can be configured to generate channel power parameters, for example:

- energy of the right channel,

- right channel subband b energy,

- left channel energy,

- left channel subband b energy,

- geometric mean of the left and right energies,

- real dot product,

- imaginary scalar product,

In addition, the channel difference determiner can be configured to determine difference (stereo) parameters according to the following equations:

secondary gain for subband b

non-normalized residual prediction gain for subband b

- residual prediction gain (normalized with downmix power).

In addition, in some embodiments, the channel difference determiner may be configured to generate for non-voice signals additional parameters such as:

For speech signals and for the higher subbands, the channel difference determiner can be configured to generate:

- inter-channel phase difference for subband b (for higher subbands, this value can be set to 0).

Difference parameters such as interchannel phase difference, secondary gain, and residual prediction gain parameter values can be passed to the mono channel generator and as stereo channel parameters to the quantizer processor.

In some embodiments, encoder 104 (or, as shown in Figure 5, channel analyzer 203) comprises a mono channel generator 305. The mono channel generator is configured to receive the channel analyzer values, such as the secondary gains and inter-channel phase differences from the channel difference determiner 301. Additionally, in some embodiments, the mono channel generator/encoder 305 may be configured to additionally receive the input multi-channel audio signals. The mono channel generator 305 may be configured in some embodiments to generate a 'lined' or down-mixed channel that is representative of the audio signals. In other words, the mono channel generator 305 can generate a mono (or down-mixed) channel signal that represents an aligned multi-channel audio signal. For example, in some embodiments where there is a left channel audio signal and a right channel audio signal, one of the left or right channel audio signals is delayed relative to the other by a given delay difference and, then the delayed channel signals and audio from another channel are averaged to produce a mono channel signal. However, it would be understood that in some embodiments any suitable mono channel generation method may be implemented.

Then the mono channel signal/parameters can be output. In some embodiments, the mono channel signal is output to mono quantizer/encoder processor 205 to be encoded.

With respect to Figure 6, a summary of the analysis process (as described in Figure 4 by steps 502 and 503) according to some embodiments and the operation of the channel analyzer 203 shown in Figure 6 is shown as a flowchart. Figure 5.

The operation of receiving the multi-channel audio signal frequency components is shown in Fig. 6 by step 551.

The operation of determining intermediate parameters (eg, power parameters for audio signal channels) is shown in Figure 6 by step 552.

The step of determining the difference parameters (eg, secondary gain, interface difference, residual prediction gain) that are generated at least partially from the intermediate parameters is shown in Figure 6 by step 553.

The operation of generating a mono channel (downmix) signal/parameters from a stereo (multi-channel) signal is shown in Figure 6 by step 555.

Referring to Figure 7, an illustrative mono quantizer/encoder processor 205 is shown in greater detail.

In some embodiments, the mono quantizer/encoder processor 205 comprises a scalar quantizer 451. The scalar quantizer 451 is configured to receive the stereo parameters of the channel analyzer 203.

The scalar quantizer can be configured to perform scalar quantization on these values. For example, scalar quantizer 451 can be configured to quantize values with quantization partition regions defined by the following matrix.

Q= {-10000.0, -8.0, -5.0, -3.0, 0.0, 3.0, 5.0, 8.0, 100000.0}

Therefore, the scalar quantizer 451 may generate an index value symbol associated with the region within the quantization partition region within which the level difference value occurs. For example, an initial quantization index value output might be as follows:

The index values may be issued in some embodiments to a 453 remapper.

In some embodiments, mono quantizer/encoder processor 205 comprises a remapper 453. Remapper 453 may be configured in some embodiments to receive the output of scalar quantizer 451, in other words, an index value associated with the quantization partition region within of which the stereo or difference parameter is found, and then mapping or ordering the index value according to a defined mapping. In some embodiments, the index (re)mapping (or reordering) is based on an adaptive map selected from a defined range of maps. The defined maps may be maps that are determined from training data or any other suitable form that takes advantage of intra-frame correlation. For example, these maps can take advantage of the correlation between adjacent symbols representing adjacent subband parameters. As such, the first symbol within a frame may be mapped according to a predetermined or defined map. The second symbol within a frame mapped according to a map that is selected based on the first symbol, and so on.

For example, a first symbol can be remapped according to the table

The next (second) symbol can be remapped based on a map that depends on the previous (first) symbol. For example, the reordering or remapping of the second symbol can be defined as

Where the previous (first) symbol = 0

Where the previous (first) symbol =1

Where the previous symbol (first) =2

Where the previous symbol (first) =3

Where the previous (first) symbol = 4

Where the previous symbol (first) =5

Where the previous (first) symbol = 6

and where the previous symbol (first) =7

These mappings can be stored as an array of mappings, such as, for example,

shortmaps[]=

{

7, 6, 4, 3, 1, 0, 2, 5,

7, 6, 4, 3, 1, 0, 2, 5,

7, 5, 3, 1, 0, 2, 4, 6,

7, 5, 3, 1, 0, 2, 4, 6,

7, 6, 4, 1, 0, 2, 3, 5,

7, 6, 4, 2, 0, 1, 3, 5,

7, 6, 4, 1, 0, 2, 3, 5,

7, 6, 5, 4, 3, 2, 1, 0, };

Where if the previous symbol was '0' then the first line of the previous two-dimensional array is used as the map, if the previous symbol was '1' then the second line and so on.

In the example above, the matrix of reordering or remapping functions is the same for each symbol. In some embodiments, each symbol may have a separate array of reordering or remapping functions. For example

the second symbol can have an array

short mapsSymbol2[]= {...};

the third symbol can have an array

short mapsSymbol3[]= {...};

and so on up to the eighth symbol array

short mapsSymbol8[]= {...};

where each matrix can be different.

This can provide the ability to tune coding efficiency with respect to specific subband to subband mappings at the cost of requiring additional matrices to be stored in the encoder and decoder.

Furthermore, in some embodiments, the matrix may be defined or selected from more than first order relations. For example, the matrix mapping function may be determined based on more than one previously determined symbol (subband) within the frame. This can also provide the ability to tune encoding efficiency at the cost of requiring additional matrices to be stored in the encoder and decoder.

Also, in some embodiments, the matrix mapping function can be determined based on a previous time symbol. For example, the mapping function can take advantage of any frame-to-frame mapping. The implementation of subband and time-based adaptive mapping causes the table ROM to increase significantly. For 8 symbols, the mapping table will have 64 lines instead of 8 lines. In some embodiments and depending on the data, only interframe could be used instead of intraframe. In some examples, interframe correlation is exploited by applying GR encoding to the difference between the current and previous frame. The numbers 0,1,-1,2 ,-2 ,. they map to 0,1,2,3,4. and are then encoded with GR of order 0 or 1, whichever is better.

The output of the remapper 453 is then sent to the Golomb-Rice encoder 455.

In some embodiments, the mono quantizer/encoder processor 205 may comprise a map selector (or the next symbol map selector) 454. The map selector 454 or map determiner may be configured to select or determine the map or ordering. to be applied by the remapper 453. The map selector 454 can therefore receive a symbol index value or parameter of the scalar quantizer and, from this value, determine the map. In some embodiments, as described in detail herein, the selection or determination may be based on a lookup table implementation. However, in some embodiments, the selection or determination may be performed at least partially algorithmically.

The mono quantizer/encoder processor 205 may comprise in some embodiments a Golomb-Rice encoder 455. The Golomb-Rice encoder (GR encoder) 455 is configured to receive the remapped index values or symbols generated by the remapper and encode the index values according to the Golomb-rice encoding method. The Golomb-Rice 455 encoder in such embodiments therefore outputs a code word representing the current and previous index values.

An example of a Golomb-Rice integer code for the first symbol is one where the output is as follows.

It would be understood that any suitable entropy encoding may be used in place of the GR integer code described herein.

The GR 455 encoder can then output the stereo codewords. In some embodiments, the code words are passed to a multiplexer for mixing with the encoded mono channel audio signal. However, in some embodiments, the stereo codewords may in some embodiments be passed to be stored or passed to additional apparatus as a separate stream.

The encoding procedure can be used for DFT parameters within a parametric stereo audio encoder. In some embodiments, the parameters to be encoded are secondary gains, residual prediction gains, and inter-channel phase differences.

For an illustrative super-wideband case for an audio data frame, there may be

12 secondary gain values that need to be transmitted, corresponding to the first 12 subbands; 5 residual prediction gains and

8 interchannel phase differences

The values of all parameters can be scalarly quantized and their index is encoded with adaptive GR. In some implementations there may be 31 (0 to 30) values for the secondary gains (quantized using 5 bits), 8 values for the residual prediction gains (quantized using 3 bits), and 8 values for the first 7 interchannel phase differences ( quantized using 3 bits) and 4 values for the last component of inter-channel phase differences (quantized using 2 bits).

An example of the encode function written in C can be written as:

shorts

adapt_GR_sg(short * in, /* (i) input parameters to encode */

short len, /* (i) number of parameters */

short no_symb, /* (i) number of possible values for the parameters */

short * nbits, /* (o) used number of bits */

short * in_enc, /* (o) values to be encoded */

short no_GR_ord)

{

short *map;

short nbits0, nbits1, nbits2, symb, map_symb, i, nb0, nb1, nb2;

if (no_symb==NO_SYMB_GR_SIDE_G) /* for secondary gains */

{

map = &map0[15*no_symb];

}

else /* for ipd and residual prediction gain */

{

map = &(map0[8*no_symb]);

}

set_s(in_enc,0,len);

nbits1 = 0;

nbits0 = 0;

nbits2 = 0;

for(i=0;i<len;i++)

{

symb = in[i];

map_symb = map[symb];

if (i==0)

{

/* GR order is 1 for the first band */

nb0 = (map_symb>>1) 1;

nb2 = nb0-2;

}else

{

nb0 = map_symb;

nb2 = ((map_symb)>>2);

}

nb1 = ((map_symb)>>1);

in_enc[i] = map_symb;

nbits0 = nb0; /* command GR 0 */

nbits1 = nb1; /* command GR 1 */

nbits2 = nb2; /* command GR 2 */

map = &(map0[symb*no_symb]); /* select map for next subband prm */ }

if (no_GR_ord==2) /* only test 0 and 1 as GR orders */

{

nbits0+=len+1;

nbits1+=2*len+1;

if((nbits0<=nbits1))

{

*nbits = nbits0;

return 0;

}

else

{

*nbits = nbits1;

return 1;

}

}

else

{ /* 0, 1,2 and two are tested as GR commands */

nbits0 = len+2;

nbits1 = 2*len+1;

nbits2 = 3*len+2;

if((nbtis1<=nbits2) && (nbits1 <= nbits0))

{

*nbits = nbits1;

return 1;

}

if((nbits0<=nbits2) && (nbits0 <= nbits1))

{

*nbits = nbits0;

return 0;

}

if((nbits2<=nbits0) && (nbits2 <= nbits1))

{

*nbits = nbits2;

return 2;

}

}

return -1;

}

The map arrays for the three-parameter type can be:

For secondary gain

shortmaps_sg[] =

{ 0, 1,2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16,

17, 18, 19, 20, 21,22, 23, 24, 25, 26, 27, 28, 29, 30,

1.0, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16,

17, 18, 19, 20, 21,22, 23, 24, 25, 26, 27, 28, 29, 30,

15, 4, 0, 1,2, 3, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 16,

17, 18, 19, 20, 21,22, 23, 24, 25, 26, 27, 28, 29, 30,

12, 9, 4, 1.0, 2, 3, 5, 6, 7, 8, 10, 11, 13, 14, 15, 16,

17, 18, 19, 20, 21,22, 23, 24, 25, 26, 27, 28, 29, 30,

16, 14, 8, 4, 2, 0, 1,3, 5, 6, 7, 9, 10, 11, 12, 13, 15,

17, 18, 19, 20, 21,22, 23, 24, 25, 26, 27, 28, 29, 30,

18, 16, 14, 10, 5, 0, 1,2, 3, 4, 6, 7, 8, 9, 11, 12, 13,

15, 17, 19, 20, 21,22, 23, 24, 25, 26, 27, 28, 29, 30,

21, 19, 17, 15, 8, 4, 2, 0, 1,3, 5, 6, 7, 9, 10, 11, 12,

13, 14, 16, 18, 20, 22, 23, 24, 25, 26, 27, 28, 29, 30,

21, 19, 17, 15, 12, 8, 4, 0, 1,2, 3, 5, 6, 7, 9, 10, 11,

18, 20, 22, 23, 24, 25, 26, 27, 28, 29, 30, 24, 22, 20, 18, 16, 14, 12, 9, 6, 0, 1,2, 3, 4, 5, 7, 8, 10, 11, 13, 15, 17, 19, 21,23, 25, 26, 27, 28, 29, 30, 25, 23, 21, 19, 17, 15, 13, 11,9, 6, 0, 1,2, 3, 4, 5, 7,13, 14, 16, 18, 20, 22, 23, 24, 25, 26, 27, 28, 29, 30, 21, 19, 17, 15, 13, 11.9, 3, 0, 1,2, 4, 5, 6, 7, 8, 10, 12, 14, 16,

18, 20, 22, 23, 24, 25, 26, 27, 28, 29, 30, 24, 22, 20, 18, 16, 14, 12, 9, 6, 0, 1,2, 3, 4, 5, 7, 8, 10, 11, 13, 15, 17, 19, 21,23, 25, 26, 27, 28, 29, 30, 25, 23, 21, 19, 17, 15, 13, 11, 9, 6, 0, 1,2, 3, 4, 5, 7,

8, 10, 12, 14, 16, 18, 20, 22, 24, 26, 27, 28, 29, 30,

27, 25, 23, 21, 19, 17, 15, 13, 11,8, 5, 0, 1,2, 3, 4, 6,

7, 9, 10, 12, 14, 16, 18, 20, 22, 24, 26, 28, 29, 30,

26, 24, 22, 20, 18, 16, 14, 12, 10, 8, 6, 4, 2, 1.0, 3, 5,

7, 9, 11, 13, 15, 17, 19, 21,23, 25, 27, 28, 29, 30,

28, 26, 24, 22, 20, 18, 16, 14, 12, 10, 8, 6, 4, 2, 0, 1,3, 5, 7, 9, 11, 13, 15, 17, 19, 21,23, 25, 27, 29, 30,

23, 21, 19, 17, 15, 13, 11,9, 7, 5, 3, 0, 1,2, 4, 6, 8, 10, 12, 14, 16, 18, 20, 22, 24, 26, 28, 30,29, 27, 25,

23, 21, 19, 17, 15, 13, 11,9, 7, 5, 3, 0, 1,2, 4, 6, 8, 10, 12, 14, 16, 18, 20, 22, 24, 26, 28, 30,

30, 28, 26, 24, 22, 20, 18, 16, 14, 12, 10, 8, 6, 4, 2, 0, I, 3, 5, 7, 9, 11, 13, 15, 17, 19, 21,23, 25, 27, 29,

30, 28, 26, 24, 22, 20, 18, 16, 14, 12, 10, 8, 6, 4, 2, 1,

0, 3, 5, 7, 9, 11, 13, 15, 17, 19, 21,23, 25, 27, 29,

30, 29, 27, 25, 23, 21, 19, 17, 15, 13, 11, 9, 7, 5, 3, 1,

0, 2, 4, 6, 8, 10, 12, 14, 16, 18, 20, 22, 24, 26, 28,

30, 29, 28, 27, 25, 23, 21, 19, 17, 15, 13, 11,9, 7, 5, 3,

0, 1,2, 4, 6, 8, 10, 12, 14, 16, 18, 20, 22, 24, 26,

30, 29, 28, 26, 24, 22, 20, 18, 16, 14, 12, 10, 9, 7, 6, 4,

3, 2, 1, 0, 5, 8, 11, 13, 15, 17, 19, 21,23, 25, 27,

30, 29, 28, 27, 26, 24, 22, 20, 18, 16, 14, 12, 10, 8, 7, 5,

4, 3, 2, 1.0, 6, 9, 11, 13, 15, 17, 19, 21,23, 25,

30, 29, 28, 27, 26, 25, 23, 21, 19, 17, 15, 13, 11, 10, 8, 7, 5, 4, 3, 2, 1.0, 6, 9, 12, 14, 16, 18, 20, 22, 24,

30, 29, 28, 27, 26, 25, 24, 23, 22, 20, 18, 16, 14, 12, 10, 8, 7, 6, 5, 4, 2, 1,0, 3, 9, 11, 13, 15, 17, 19, 21,

30, 29, 28, 27, 26, 25, 24, 23, 22, 20, 18, 16, 14, 13, 11, 10, 9, 7, 6, 5, 3, 2, 1,0, 4, 8, 12, 15, 17, 19, 21,

30, 29, 28, 27, 26, 25, 24, 23, 22, 20, 18, 16, 14, 13, 12,

I I , 10, 9, 7, 6, 5, 3, 1,0, 2, 4, 8, 15, 17, 19, 21,

30, 29, 28, 27, 26, 25, 24, 23, 22, 21,20, 19, 17, 15, 13,

12, 11, 9, 8, 7, 6, 4, 3, 2, 1,0, 5, 10, 14, 16, 18,

30, 29, 28, 27, 26, 25, 24, 23, 22, 21,20, 19, 18, 17, 15,

13, 12, 11, 10, 9, 7, 6, 5, 3, 1.0, 2, 4, 8, 14, 16,

30, 29, 28, 27, 26, 25, 24, 23, 22, 21,20, 19, 18, 17, 16,

15, 14, 13, 11, 10, 8, 7, 6, 5, 3, 2, 0, 1,4, 9, 12,

30, 29, 28, 27, 26, 25, 24, 23, 22, 21,20, 19, 18, 17, 16,

14, 13, 12, 11, 10, 9, 8, 7, 6, 5, 3, 2, 1,0, 4, 15,

30, 29, 28, 27, 26, 25, 24, 23, 22, 21,20, 19, 18, 17, 16,

15, 14, 13, 12, 11, 10, 9, 8, 7, 6, 5, 4, 3, 2, 0, 1,

30, 29, 28, 27, 26, 25, 24, 23, 22, 21,20, 19, 18, 17, 16,

15, 14, 13, 12, 11, 10, 9, 8, 7, 6, 5, 4, 3, 2, 1.0,

};

For residual prediction gain

short maps_rμg[] = {

0, 1, 2, 3, 4, 5, 6, 7,

2, 0, 1, 3, 4, 5, 6, 7,

6, 2, 0, 1, 3, 4, 5, 7,

7, 5, 2, 0, 1, 3, 4, 6,

7, 6, 4, 3, 1, 0, 2, 5,

7, 6, 5, 3, 2, 1, 0, 4,

7, 6, 5, 4, 3, 2, 0, 1,

7, 6, 5, 4, 3, 2, 0, 1,

6, 5, 4, 3, 1, 0, 2, 7,

};

and for the interface differences

short maps_ipd[]=

{

7, 6, 4, 3, 1, 0, 2, 5,

7, 6, 4, 3, 1, 0, 2, 5,

7, 5, 3, 1, 0, 2, 4, 6,

7, 5, 3, 1, 0, 2, 4, 6,

7, 6, 4, 1, 0, 2, 3, 5,

7, 6, 4, 2, 0, 1, 3, 5,

7, 6, 4, 1, 0, 2, 3, 5,

7, 6, 5, 4, 3, 2, 1, 0,

6, 5, 4, 2, 0, 1, 3, 7

};

As shown above, since there are 31 symbols for the secondary gains (the secondary gains are first scalarly quantized using 5 bits), the 'map' table is relatively large compared to the other 'map' table.

In some embodiments, the structure of the map table is parsed and where there is a structure defined in the table that can be exploited, then this can be used to compress the map table. For example, in the illustrative map table sg defined above, the analysis may allow the following data to be stored:

short sg_data1[] = {1,0,2,

4, 0, 1, 2, 3, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14,

9, 4, 1, 0, 2, 3, 5, 6, 7, 8, 10, 11,

8, 4, 2, 0, 1,3, 5, 6, 7, 9, 10, 11, 12, 13,

10, 5, 0, 1,2, 3, 4, 6, 7, 8, 9, 11, 12, 13,

8, 4, 2, 0, 1,3, 5, 6, 7, 9, 10, 11, 12, 13, 14,

12, 8, 4, 0, 1,2, 3, 5, 6, 7, 9, 10, 11, 13, 14,

3, 0, 1, 2, 4, 5, 6, 7, 8,

9, 6, 0, 1,2, 3, 4, 5, 7, 8, 10, 11,

6, 0, 1, 2, 3, 4, 5, 7, 8,

8, 5, 0, 1,2, 3, 4, 6, 7, 9, 10,

2, 1, 0, 3,

0, 1,

0, 1, 2,

0};

short sg_data2[] =

{3,3,

15,16,

12, 13,

14, 16,

14, 17,

15, 19,

15, 19,

9, 16,

12, 19,

9, 18,

11.20,

4, 16,

2, 16,

3, 17,

1, 16};

This data can be used in such a way that to get, for example, the 5th line of 'maps'

16, 14, 8, 4, 2, 0, 1,3, 5, 6, 7, 9, 10, 11, 12, 13, 15, 17, 18, 19, 20, 21,22, 23, 24, 25, 26, 27, 28, 29, 30,

The 4th line of the sg_data1 pseudo-line indicates that its data (bold and underlined in the example line above) is part of the corresponding line in 'maps'. The data in sg_data2, the 4th pseudo line states that there are 14 components in sg_data1 that should be copied to 'maps' (first parameter of sg_data2, line 4), and that starting with position 16, the corresponding 'maps' line will be filled automatically . The autofill is such that the first consecutive number after the last value of the sg_data1 pseudoline will be at the beginning of the string just before 8, i.e. the value 14, then 15 will be at the other end, 16 at the beginning and so on. If there is no possibility to continue on one end, then the numbers are filled in consecutively on one side only.

An illustrative function that can be used to recreate the 'maps' array for secondary gains is:

void

fill_in_map_sg(short * maps_line, short * lens, short * data)

{

short i, k, diff, last;

diff = lens[1]-lens[0];

for (i=diff;i<lens[1];i++)

{

maps_line[i] = data[i-diff];

}

if (diff>0)

{

/* fill both sides */

last = data[lens[0]-1]+1;

k = lens[1];

for(i=diff-1 ;i>=0;i--)

{

maps_line[i] = last;

last = 1;

maps_line[k++] = last;

last=1;

}

for (i=k;i<NO_SYMB_GR_SIDE_G;i++)

{

maps_line[i] = last;

last++;

}

}

else

{

for (i=lens[1];i<NO_SYMB_GR_SIDE_G;i++)

{

maps_line[i] = i;

}

}

}

void

mvs2s_rev(short * x, short * res, short len)

{

shorti;

for (i=0;i<len;i++)

{

res[len-1-i] = x[i];

}

}

void

init_map_sg(short * maps_sg)

{

short i, s;

for (i=0;i<NO_SYMB_GR_SIDE_G;i++)

{

maps_sg[i] = i;

}

s = 0;

for(i=0;i<15;i++)

{

fill_in_map_sg(&maps_sg[(i+1)*NO_SYMB_GR_SIDE_G], &sg_data2[i*2], &sg_datal[s]; s = sg_Data2[i*2];

}

for(i=0;i<15;i++)

{

mvs2s_rev(&maps_sg[(14-i)*NO_SYMB_GR_SIDE_G], &maps_sg[(16+i)*NO_SYMB_GR_SIDE_G], NO_SYMB_GR_SIDE_G);

}

In some embodiments, the mono quantizer/encoder processor 205 further comprises a mono channel (downmix) encoder 456 . The mono channel (downmix) encoder 456 can be configured to receive the mono channel or parameters (downmix). In addition, the mono channel (downmix) encoder 456 may be configured to receive an indication of the number of bits that have been used in the GR encoder to encode the current frame. The mono channel (downmix) encoder 456 may then be configured to encode the mono channel (downmix) or the parameters based on any suitable encoding method based on knowledge of the number of bits used for encoding the stereo parameters. The mono channel generator/encoder 456 may encode the generated mono channel audio signal using any suitable encoding format. For example, in some embodiments, the mono channel audio signal may be encoded using a mono channel encoded form of Enhanced Voice Service (EVS), which may contain a bitstream interoperable version of the Adaptive Multi-Rate codec - Broadband (AMR-WB).

With respect to Figure 8, a summary of the encoding process (as described in Figure 4 by steps 505) according to some embodiments and the operation of the quantizer/encoder processor 205 shown in Figure 8 is shown as a flow chart. 7.

The operation of receiving the stereo parameters is shown in Figure 8 by step 701.

The operation of quantizing the stereo parameters to generate index values or symbols is shown in Figure 8 by step 703.

The operation of retrieving a map based on at least one previous value or index symbol (within the frame) is shown in Figure 8 by step 704.

The operation of reordering or remapping the symbol or index value based on the retrieved map is shown in Figure 8 by step 705.

The operation of generating codewords according to the Golomb-Rice coding system from the remapped symbol values is shown in Figure 8 by step 707.

The operation of outputting stereo codewords is shown in Figure 8 by step 709.

Also, the operation of receiving the mono parameters is shown in Fig. 8 by step 702.

The operation of encoding the mono parameters/channel based on the use of Golomb-Rice encoding bits is shown in Figure 8 by step 708.

The operation of issuing mono codewords is shown in Figure 8 by step 710.

To fully show codec operations, Figures 9 and 10 show a decoder and decoder operation according to some embodiments.

In some embodiments, decoder 108 comprises a mono channel decoder 801. The mono channel decoder 801 is configured in some embodiments to receive the encoded mono channel signal.

In addition, the mono channel decoder 801 may be configured to decode the encoded mono channel audio signal using the reverse process to the mono channel encoder shown in the encoder. In some embodiments, the mono channel decoder 801 may be configured to receive a flag from the stereo channel decoder 803 indicating the number of bits used for the stereo signal to aid decoding of the mono channel.

In some embodiments, mono channel decoder 801 may be configured to output the mono channel audio signal to stereo channel generator 809.

In some embodiments, decoder 108 may comprise a stereo channel decoder 803. The stereo channel decoder 803 is configured to receive the encoded stereo parameters.

In addition, the stereo channel decoder 803 may be configured to decode the stereo channel signal parameters from the entropy code to a symbol value.

The stereo channel decoder 803 is further configured to output the decoded index values to a symbol reorderer (demapper) 807.

In some embodiments, the decoder comprises a symbol map selector 805 (or map determiner or order determiner or order selector). The symbol map selector 805 may be configured to receive the current frame stereo channel index values (decoded and reordered symbols) and select a symbol map to invert the mapping used in the encoder. In other words, the symbol map selector 805 is configured to determine a map based on a previously decoded determined symbol within a frame.

The (symbol) map may be output to the symbol reorderer 807.

In some embodiments, decoder 108 comprises a symbol reorderer 807. The symbol or index reorderer (demapper) in some embodiments is configured to receive the symbol map from the map selector 805 and reorder the decoded symbols received from the stereo channel decoder 803 according to the selected map. In other words, the symbol reorderer 807 is configured to reorder the index values to the original order output by the scaling quantizer within the encoder. In addition, in some embodiments, the symbol reorderer 807 is configured to dequantize the unmapped or reordered index value into a parameter (such as the interaural time difference/correlation value; and the interaural level difference/energy difference value). ) using the reverse process to that defined within the quantizer section of the quantizer processor within the encoder.

In some embodiments, the decoder comprises a stereo channel generator 809 configured to receive the reordered decoded symbols (the stereo parameters) and the decoded mono channel and regenerate the stereo channels, in other words, apply the level differences to the mono channel to generate a second channel. With respect to Figure 10, a summary of the decoding process according to some embodiments and the operation of the decoder 108 shown in Figure 9 is shown as a flowchart.

The operation of receiving the encoded mono channel audio signal is shown in Figure 10 by step 901. The operation of receiving the encoded stereo parameters is shown in Figure 10 by step 902.

The operation of decoding the mono channel (based on the number of bits used by the stereo channel) is shown in Figure 10 by step 903.

The operation of decoding the stereo parameters is shown in Figure 10 by step 904.

The operation of reordering and dequantizing the decoded symbols to generate dequantized (regenerated) stereo parameters for each frame is shown in Figure 10 by step 906.

The operation of selecting the map for a next symbol based on a current symbol value is shown in Figure 10 by step 907.

The output of the stereo parameters to the stereo channel generator is shown in Figure 10 by step 908. The operation of generating the stereo channels from the mono channel stereo parameters is shown in Figure 10 by step 909.

Although in the above examples the map is selected from a stored table, it is understood that in some embodiments the map for the current symbol may be algorithmically determined based on a function that receives a previously determined symbol as input.

Although the above examples describe application embodiments operating within a codec within an appliance 10, it would be appreciated that the invention as described below can be implemented as part of any audio (or speech) codec, including any audio codec (or voice) variable rate/adaptive rate. Thus, for example, the application embodiments may be implemented in a video codec that may implement audio encoding over fixed or wireline communication paths.

Therefore, the user equipment may comprise an audio codec such as those described in the application embodiments above.

It will be appreciated that the term user equipment is intended to cover any suitable type of wireless user equipment, such as mobile phones, portable data processing devices or portable web browsers.

In addition, the elements of a public land mobile network (PLMN) may also comprise audio codecs as described above.

In general, the various embodiments of the application may be implemented in hardware or special purpose circuitry, software, logic, or any combination thereof. For example, some aspects may be implemented in hardware, while other aspects may be implemented in firmware or software that may be executed by a controller, microprocessor, or other computing device, although the invention is not limited thereto. Although various aspects of the application may be illustrated and described as block diagrams, flowcharts, or using some other graphical representation, it is understood that these blocks, apparatus, systems, techniques, or procedures described herein may be implemented, as examples not limited, in special purpose hardware, software, firmware, circuitry or logic, general purpose hardware or controller or other computing devices, or any combination thereof.

Embodiments of this application may be implemented by computer software executable by a mobile device data processor, such as in the processor entity, or by hardware, or by a combination of software and hardware. Furthermore, in this regard, it should be noted that any logic flow blocks as in the Figures may represent program steps, or interconnected logic circuits, blocks and functions, or a combination of program steps and logic circuits, blocks and functions.

The memory may be of any type suitable for the local technical environment and may be implemented using any suitable data storage technology, such as semiconductor-based memory devices, magnetic memory devices and systems, optical memory devices and systems, fixed memory and removable memory. Data processors may be of any type suitable for the local technical environment, and may include one or more of general purpose computers, special purpose computers, microprocessors, digital signal processors (DSPs), application specific integrated circuits (application-specific integrated circuits - ASICs), gate-level circuits, and processors based on multi-core processor architecture, as non-limiting examples.

The application embodiments can be implemented in various components such as integrated circuit modules. Integrated circuit design is, to a large extent, a highly automated process. There are complex and powerful software tools available to convert a logic level design into a semiconductor circuit design ready to be etched and formed on a semiconductor substrate.

Programs, such as those provided by Synopsys, Inc. of Mountain View, Calif., and Cadence Design of San Jose, Calif., automatically route leads and locate components on a semiconductor chip using well-established design rules as well as module libraries. previously stored layout. Once the design for a semiconductor circuit has been completed, the resulting design, in a standardized electronic format (eg, Opus, GDSII, or similar) can be transmitted to a semiconductor manufacturing facility or "fab" for processing. manufacturing.

As used in this application, the term "circuitry" refers to all of the following:

(a) hardware-only circuit implementations (such as analog-only and/or digital-only circuitry implementations), and

(b) to combinations of circuitry and software (and/or firmware), such as: (i) to a combination of processor(s) or (ii) to portions of processor(s)/software (including digital signal processor(s)) , software and memory or memories that work together to make a device, such as a mobile phone or server, perform various functions and

(c) to circuits, such as a microprocessor or microprocessors or a portion of a microprocessor or microprocessors, that require software or firmware for their operation, even if the software or firmware is not physically present.

This definition of 'circuitry' applies to all uses of this term in this application, including any claim. As a further example, as used in this application, the term 'circuitry' will also cover an implementation of as little as one processor (or multiple processors) or a portion of a processor and its software and/or firmware. attach. The expression 'circuitry' would also cover, for example and if applicable to the particular element of the claim, a baseband integrated circuit or an application processor integrated circuit for a mobile phone or a similar integrated circuit in a server, a cellular network device or other network device.

Claims (14)

1. An apparatus comprising: means for receiving at least two audio channel signals; means for determining (301), for a first frame, at least two parameters representing a difference between the at least two channel audio signals; means for scalar quantization (451) of the at least two parameters to generate at least two index values; means for determining an initial index map for reordering one of the at least two index values, and means for determining at least one additional index map for reordering at least one additional of the at least two index values, wherein the at at least one additional index map is determined based on one of the at least two index values; means for reordering the one of the at least two index values based on the initial index map; means for reordering the additional one of the at least two index values based on the at least one additional index map; means for entropy encoding the reordering of the at least two index values based on an order position of the reordering of the at least two index values; means for entropy encoding the further reordering of the at least two index values based on an order position of the further reordering of the at least two index values; means for generating a single channel representation (305) of the at least two audio channel signals that depend on the at least two parameters; and means for encoding (456) the single channel representation. The apparatus according to claim 1, wherein the means for scalar quantization further comprises ordering the scalar quantized output according to a predetermined map. The apparatus according to any one of claims 1 to 2, wherein the means for entropy encoding the reordered and additional index values based on an order position of the reordered index and additional index values comprises means (455) for applying a Golomb-Rice encoding of the additional and reordered index values based on an order position of the additional and reordered index values. The apparatus according to any of claims 1 to 3, wherein the means for determining, for a first frame, at least two parameters comprises means for determining at least three parameters; the means (451) for scalarly quantizing the at least two parameters, comprising means for scalarly quantizing the at least three parameters to generate at least three index values, the at least three index values comprising a first index value, a first value additional index value and a second additional index value; and the means for determining at least one additional index map comprise: means for determining a first additional index map for reordering the first additional index value, wherein the first additional index map is determined depending on the first index value; and means for determining a second additional index map for reordering the second additional index value, wherein the second additional index map is determined depending on the first additional index value. The apparatus according to claim 4, wherein the means for determining the first additional index map for reordering the first additional index value comprises means for selecting, from a first array of index maps, the first additional index map based on at the first index value. The apparatus according to claim 5, wherein the means for determining the second additional index map for reordering the second additional index value comprises means for selecting, from a second array of index maps, the second additional index map based on at the first additional index value. The apparatus according to claim 6, wherein the second index map array is the first index map array. The apparatus according to any of claims 1 to 4, wherein the means for determining the at least one additional index map for reordering the at least one additional of the at least two index values comprises means for selecting, from an array of index maps, the at least one additional index map based on the one of the at least two index values. The apparatus according to claim 8, wherein the means for determining the at least one additional index map for reordering the at least one additional of the at least two index values comprises means for generating, from a compressed array of maps index, the at least one additional index map based on the one of the at least two index values. 10. An apparatus comprising: means for entropy decoding (803) from a first part of a signal at least two parameter index values, wherein the parameters represent a difference between at least two channel audio signals, and wherein the signal is a encoded multi-channel audio signal; means for reordering a first of the at least two parameter index values based on a determined first reordering to generate a first reordered index value; means for reordering a second of the at least two parameter index values based on a determined second reordering to generate a second reordered index value, wherein the determined second reordering is based on the first reordered index value; means for dequantizing the first and second reordered index values to generate the at least two parameters; means for receiving from a further part of the signal an encoded single channel representation signal; means for determining a number of bits used in the first part of the signal; means for decoding the encoded single channel representation signal based on the number of bits used in the first part of the signal; and means for generating the multi-channel audio signal based on applying the at least two parameters to the encoded and decoded single channel representation signal to generate a second channel. The apparatus according to claim 10, wherein the means for reordering a first of the at least two parameter index values based on a determined first reordering to generate a first reordered index value comprises: means for determining a reverse ordering; and means for applying the reverse ordering. The apparatus according to claim 11, wherein the means for reordering a second of the at least two parameter index values based on a determined first reordering to generate a first reordered index value comprises: means for determining a second reverse ordering based on the first reordered index value; and means for applying the second reverse ordering. 13. A procedure comprising: receive at least two audio channel signals; determining (502), for a first frame, at least two parameters that represent a difference between the at least two channel audio signals; quantizing (703) scalarly the at least two parameters to generate at least two index values; determining an initial index map for reordering one of the at least two index values, and determining (704) at least one additional index map for reordering at least one additional of the at least two index values, wherein the at least a further index map is determined based on one of the at least two index values; reordering (705) the one of the at least two index values based on the initial index map; reordering (705) the additional of the at least two index values based on the at least one additional index map; entropy encoding (707) the reordering of the at least two index values based on an order position of the reordering of the at least two index values; entropy encoding (707) the further reordering of the at least two index values based on an order position of the further reordering of the at least two index values; generating a single channel representation (503) of the at least two audio channel signals that depend on the at least two parameters; and encoding (505) the single channel representation. 14. A procedure comprising: entropy decoding (904) from a first part of a signal at least two parameter index values, wherein the parameters represent a difference between at least two channel audio signals, and wherein the signal is a channel audio signal. scrambled multichannel audio; reordering a first of the at least two parameter index values based on a determined first reordering to generate a first reordered index value; reordering a second of the at least two parameter index values based on a determined second reordering to generate a second reordered index value, wherein the determined second reordering is based on the first reordered index value; dequantizing the first and second reordered index values to generate the at least two parameters. receiving (901) from a further part of the signal an encoded mono channel signal; determining a number of bits used in the first part of the signal; decoding (903) the encoded mono channel signal based on the number of bits used in the first part of the signal; and generating (909) the multi-channel audio signal based on applying the at least two parameters to the encoded and decoded mono channel to generate a second channel.