EP3577649A1 - Codeur de signal audio stéréo - Google Patents

Codeur de signal audio stéréo

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Publication number
EP3577649A1
EP3577649A1 EP18747600.7A EP18747600A EP3577649A1 EP 3577649 A1 EP3577649 A1 EP 3577649A1 EP 18747600 A EP18747600 A EP 18747600A EP 3577649 A1 EP3577649 A1 EP 3577649A1
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EP
European Patent Office
Prior art keywords
index
map
reordered
index values
reordering
Prior art date
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Granted
Application number
EP18747600.7A
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German (de)
English (en)
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EP3577649B1 (fr
EP3577649A4 (fr
Inventor
Adriana Vasilache
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Nokia Technologies Oy
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Nokia Technologies Oy
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Publication of EP3577649A4 publication Critical patent/EP3577649A4/fr
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Classifications

    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/008—Multichannel audio signal coding or decoding using interchannel correlation to reduce redundancy, e.g. joint-stereo, intensity-coding or matrixing
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/0017—Lossless audio signal coding; Perfect reconstruction of coded audio signal by transmission of coding error
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/04—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
    • G10L19/16—Vocoder architecture
    • G10L19/18—Vocoders using multiple modes
    • G10L19/24—Variable rate codecs, e.g. for generating different qualities using a scalable representation such as hierarchical encoding or layered encoding
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/02—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using spectral analysis, e.g. transform vocoders or subband vocoders
    • G10L19/032—Quantisation or dequantisation of spectral components
    • G10L19/035—Scalar quantisation

Definitions

  • 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 apparatus.
  • Audio signals like speech or music, are encoded for example to enable efficient transmission or storage of the audio signals.
  • Audio encoders and decoders are used to represent audio based signals, such as music and ambient sounds (which in speech coding terms can be called background noise). These types of coders typically do not utilise a speech model for the coding process, rather they use processes for representing all types of audio signals, including speech. Speech encoders and decoders (codecs) can be considered to be audio codecs which are optimised for speech signals, and can operate at either a fixed or variable bit rate.
  • An audio codec can also be configured to operate with varying bit rates. At lower bit rates, such an audio codec may be optimized to work with speech signals at a coding rate equivalent to a pure speech codec. At higher bit rates, the audio codec may code any signal including music, background noise and speech, with higher quality and performance.
  • a variable-rate audio codec can also implement an embedded scalable coding structure and bitstream, where additional bits (a specific amount of bits is often referred to as a layer) improve the coding upon lower rates, and where the bitstream of a higher rate may be truncated to obtain the bitstream of a lower rate coding. Such an audio codec may utilize a codec designed purely for speech signals as the core layer or lowest bit rate coding.
  • An audio codec is designed to maintain a high (perceptual) quality while improving the compression ratio.
  • waveform matching coding it is common to employ various parametric schemes to lower the bit rate.
  • multichannel audio such as stereo signals
  • the proposed stereo/binaural extension is composed of encoded stereo parameters. Increasing the coding efficiency for these parameters means reducing 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 the bitrate.
  • Coding efficiency of stereo parameters has involved quantization of the values (levels), followed by entropy encoding to reduce further the bitrate.
  • a previously proposed method for encoding the stereo parameters disclosed in EP2856776 uses an adaptive version of the Golomb Rice coding.
  • 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 quantising 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 at least one further index map for reordering at least one further of the at least two index values, wherein the at least one further index map is determined based on the one of the at least two index values; reordering the one of the at least two index values based on the initial index map; reordering the further of the at least two index values based on the at least one further index map; encoding the reordered one of the at least two index values dependent on an order position of the reordered one of the at least two index values; encoding the reordered further of the at least two index values based on an order position of the reordered further of the at least
  • Scalar quantising the parameter may further comprise ordering the scalar quantized output according to a predetermined map.
  • Encoding the reordered one and further index values dependent on an order position of the reordered index one and further index values may comprise applying a Golomb-Rice encoding to the reordered one and further index values dependent on an order position of the reordered index one and further index values.
  • Determining, for a first frame, at least two parameters may comprise determining at least three parameters; scalar quantising the at least two parameters may comprise scalar quantising 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 further index value and a second further index value; and determining at least one further index map may comprise: determining a first further index map for reordering the first further index value, wherein the first further index map is determined based on the first index value; and determining a second further index map for reordering the second further index value, wherein the second further index map may be determined based on the first further index value.
  • Determining the first further index map for reordering the first further index value may comprise selecting, from a first array of index maps, the first further index map based on the first index value.
  • Determining the second further index map for reordering the second further index value may comprise selecting, from a second array of index maps, the second further index map based on the first further index value.
  • the second array of index maps may be the first array of index maps.
  • Determining the at least one further index map for reordering at least one further of the at least two index values may comprise selecting, from an array of index maps, the at least one further index map based on the one of the at least two index values.
  • Determining the at least one further index map for reordering at least one further of the at least two index values may comprise generating, from a compressed array of index maps, the at least one further index map based on the one of the at least two index values.
  • the at least one further index map may be further determined based on a further one of the at least two index values.
  • Encoding the single channel representation may comprise: determining the number of bits used for encoding the reordered further of the at least two index values; and encoding the single channel representation based on the determined number of bits.
  • a method comprising: decoding 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 an encoded multichannel audio signal; reordering a first of the at least two parameter index values based on a first determined reordering to generate a first reordered index value; reordering a second of the at least two parameter index values based on a second determined reordering to generate a second reordered index value, wherein the second determined reordering is based on the first reordered index value; and dequantizing the first and the second reordered index value to generate the at least two parameters.
  • Decoding from a first part of a signal may comprise decoding a first part of a signal using a Golomb-Rice decoding.
  • Reordering a first of the at least two parameter index values based on a first determined reordering to generate a first reordered index value may comprise: determining an inverse ordering; and applying the inverse ordering.
  • Reordering a second of the at least two parameter index values based on a first determined reordering to generate a first reordered index value may comprise: determining a second inverse ordering based on the first reordered index value; and applying the second inverse ordering.
  • the method may further comprise: receiving from a further part of a signal an encoded downmix channel signal; determining a number of bits used in the first part of the signal; and decoding the encoded downmix channel signal based on the number of bits used in the first part of the signal.
  • An apparatus may be configured to perform the method of encoding as described herein.
  • An apparatus may be configured to perform the method of decoding as described herein.
  • an apparatus comprising: a parameter determiner configured to determine, for a first frame, at least two parameters representing a difference between the at least two channel audio signals; a scalar quantizer configured to scalar quantise the at least two parameters to generate at least two index values; a map determiner configured to determine an initial index map for reordering one of the at least two index values, and determine at least one further index map for reordering at least one further of the at least two index values, wherein the at least one further index map is determined based on the one of the at least two index values; a reorderer configured to reorder the one of the at least two index values based on the initial index map and further configured to reorder the further of the at least two index values based on the at least one further index map; an encoder configured to encode the reordered one of the at least two index values dependent on an order position of the reordered one of the at least two index values, and encode the reordered further
  • the scalar quantizer may be further configured to order the scalar quantized output according to a predetermined map.
  • the encoder may be configured to applying a Golomb-Rice encoding to the reordered one and further index values dependent on an order position of the reordered index one and further index values.
  • the parameter determiner may be configured to determine at least three parameters
  • the scalar quantizer may be configured to scalar quantise 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 further index value and a second further index value
  • the map determiner may be configured to: determine a first further index map for reordering the first further index value, wherein the first further index map is determined based on the first index value; and determine a second further index map for reordering the second further index value, wherein the second further index map is determined based on the first further index value.
  • the map determiner may be configured to select, from a first array of index maps, the first further index map based on the first index value.
  • the map determiner may be configured to select, from a second array of index maps, the second further index map based on the first further index value.
  • the second array of index maps may be the first array of index maps.
  • the map determiner may be configured to select, from an array of index maps, the at least one further index map based on the one of the at least two index values.
  • the map determiner may be configured to generate, from a compressed array of index maps, the at least one further index map based on the one of the at least two index values.
  • the map determiner may be configured to determine the at least one further index map based on a further one of the at least two index values.
  • the mono channel encoder may be configured to: determine the number of bits used for encoding the reordered further of the at least two index values; and encode the single channel representation based on the determined number of bits.
  • an apparatus comprising: a decoder configured to decode 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 an encoded multichannel audio signal; a reorderer configured to reorder a first of the at least two parameter index values based on a first determined reordering to generate a first reordered index value and further configured to reorder a second of the at least two parameter index values based on a second determined reordering to generate a second reordered index value, wherein the second determined reordering is based on the first reordered index value; and a dequantizer configured to dequantize the first and the second reordered index value to generate the at least two parameters.
  • the decoder may be configured to decode a first part of a signal using a
  • the reorderer may be configured to: determine an inverse ordering; and apply the inverse ordering.
  • the reorderer configured to reorder a second of the at least two parameter index values based on a first determined reordering to generate a first reordered index value may be configured to: determine a second inverse ordering based on the first reordered index value; and apply the second inverse ordering.
  • the apparatus may further comprise a mono/downmix decoder configured to: receive from a further part of a signal an encoded downmix channel signal; determine a number of bits used in the first part of the signal; and decode the encoded downmix channel signal based on the number of bits used in the first part of the signal.
  • a mono/downmix decoder configured to: receive from a further part of a signal an encoded downmix channel signal; determine a number of bits used in the first part of the signal; and decode the encoded downmix channel signal based on the number of bits used in the first part of the signal.
  • the parameters representing a difference between the at least two channel audio signals may be at least one of: a side gain, an interphase difference, a residual prediction gain.
  • a computer program product may cause an apparatus to perform the method as described herein.
  • An electronic device may comprise apparatus as described herein.
  • a chipset may comprise apparatus as described herein.
  • Figure 1 shows schematically an electronic device employing some embodiments
  • FIG. 2 shows schematically an audio codec system according to some embodiments
  • Figure 3 shows schematically an encoder as shown in Figure 2 according to some embodiments
  • Figure 4 shows schematically a channel analyser as shown in Figure 3 in further detail according to some embodiments
  • Figure 5 shows schematically a stereo channel encoder as shown in Figure 3 in further detail according to some embodiments
  • Figure 6 shows a flow diagram illustrating the operation of the encoder shown in Figure 2 according to some embodiments
  • Figure 7 shows a flow diagram illustrating the operation of the channel analyser as shown in Figure 4 according to some embodiments
  • Figure 8 shows a flow diagram illustrating the operation of the channel encoder as shown in Figure 5 according to some embodiments
  • Figure 9 shows schematically the decoder as shown in Figure 2 according to some embodiments.
  • Figure 10 shows a flow diagram illustrating the operation of the decoder as shown in Figure 9 according to some embodiments.
  • Description of Some Embodiments of the Application The following describes in more detail possible stereo and multichannel speech and audio codecs, including layered or scalable variable rate speech and audio codecs.
  • a previously proposed method for encoding the stereo parameters disclosed in EP2856776 uses an adaptive version of the Golomb Rice coding.
  • the concept as expressed in the embodiments described hereafter is one which attempts to better capture and exploit intraframe value correlation and as a consequence further reduce bitrate consumption for encoding the stereo parameters.
  • the embodiments explicitly store the order of first order probabilities of the symbols to be encoded (instead of having them adaptively sorted). In other words, for a single data frame, based on a previously encoded symbol, an array of integers keeps the order of probabilities for each symbol. In other words 0 if it is most probable, 1 , if is the second most probable and so on. The probability order value is then encoded with an adaptive GR code.
  • Figure 1 shows a schematic block diagram of an exemplary electronic device or apparatus 10, which may incorporate a codec according to an embodiment of the application.
  • the apparatus 10 may for example be a mobile terminal or user equipment of a wireless communication system.
  • the apparatus 10 may be an audio-video device such as video camera, a Television (TV) receiver, audio recorder or audio player such as a mp3 recorder/player, a media recorder (also known as a mp4 recorder/player), or any computer suitable for the processing of audio signals.
  • an audio-video device such as video camera, a Television (TV) receiver, audio recorder or audio player such as a mp3 recorder/player, a media recorder (also known as a mp4 recorder/player), or any computer suitable for the processing of audio signals.
  • TV Television
  • mp3 recorder/player such as a mp3 recorder/player
  • media recorder also known as a mp4 recorder/player
  • the electronic device or apparatus 10 in some embodiments comprises a microphone 1 1 , which is linked via an analogue-to-digital converter (ADC) 14 to a processor 21 .
  • the processor 21 is further linked via a digital-to-analogue (DAC) converter 32 to loudspeakers 33.
  • the processor 21 is further linked to a transceiver (RX/TX) 13, to a user interface (Ul) 15 and to a memory 22.
  • the processor 21 can in some embodiments be configured to execute various program codes.
  • the implemented program codes in some embodiments comprise a multichannel or stereo encoding or decoding code as described herein.
  • the implemented program codes 23 can in some embodiments be stored for example in the memory 22 for retrieval by the processor 21 whenever needed.
  • the memory 22 could further provide a section 24 for storing data, for example data that has been encoded in accordance with the application.
  • the encoding and decoding code in embodiments can be implemented in hardware and/or firmware.
  • the user interface 15 enables a user to input commands to the electronic device 10, for example via a keypad, and/or to obtain information from the electronic device 10, for example via a display.
  • 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 for enabling communication with other apparatus, for example via a wireless communication network.
  • a user of the apparatus 10 for example can use the microphone 1 1 for inputting speech or other audio signals that are to be transmitted to some other apparatus or that are to be stored in the data section 24 of the memory 22.
  • a corresponding application in some embodiments can be activated to this end by the user via the user interface 15. This application in these embodiments can be performed by the processor 21 , causes the processor 21 to execute the encoding code stored in the memory 22.
  • the analogue-to-digital converter (ADC) 14 in some embodiments converts the input analogue audio signal into a digital audio signal and provides the digital audio signal to the processor 21 .
  • the microphone 1 1 can comprise an integrated microphone and ADC function 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 way as described with reference to the system shown in Figure 2, the encoder shown in Figures 2 to 8 and the decoder as shown in Figures 9 and 10.
  • the resulting bit stream can in some embodiments be provided to the transceiver 13 for transmission to another apparatus.
  • the coded audio data in some embodiments can be stored in the data section 24 of the memory 22, for instance for a later transmission or for a later presentation by the same apparatus 10.
  • the apparatus 10 in some embodiments can also receive a bit stream with correspondingly encoded data from another apparatus via the transceiver 13.
  • the processor 21 may execute the decoding program code stored in the memory 22.
  • the processor 21 in such embodiments decodes the received data, and provides the decoded data to a digital-to-analogue converter 32.
  • the digital-to-analogue converter 32 converts the digital decoded data into analogue audio data and can in some embodiments output the analogue audio via the loudspeakers 33.
  • Execution of the decoding program code in some embodiments can be triggered as well by an application called by the user via the user interface 15.
  • the received encoded data in some embodiment can also be stored instead of an immediate presentation via the loudspeakers 33 in the data section 24 of the memory 22, for instance for later decoding and presentation or decoding and forwarding to still another apparatus.
  • FIG. 2 The general operation of audio codecs as employed by embodiments is shown in Figure 2.
  • General audio coding/decoding systems comprise both an encoder and a decoder, as illustrated schematically in Figure 2. However, it would be understood that some embodiments can implement one of either the encoder or decoder, or both the encoder and decoder. Illustrated by 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 can comprise or implement one of the encoder 104 or decoder 108 or both the encoder 104 and decoder 108.
  • the encoder 104 compresses an input audio signal 1 10 producing a bit stream 1 12, which in some embodiments can be stored or transmitted through a media channel 106.
  • the encoder 104 furthermore can comprise a stereo encoder 151 as part of the overall encoding operation. It is to be understood that the stereo encoder may be part of the overall encoder 104 or a separate encoding module.
  • the encoder 104 can also comprise a multi-channel encoder that encodes more than two audio signals.
  • the bit stream 1 12 can be received within the decoder 108.
  • the decoder 108 decompresses the bit stream 1 12 and produces an output audio signal 1 14.
  • the decoder 108 can comprise a stereo decoder as part of the overall decoding operation. It is to be understood that the stereo decoder may be part of the overall decoder 108 or a separate decoding module.
  • the decoder 108 can also comprise a multi-channel decoder that decodes more than two audio signals.
  • the bit rate of the bit stream 1 12 and the quality of the output audio signal 1 14 in relation to the input signal 1 10 are the main features which define the performance of the coding system 102.
  • an example encoder 104 is shown according to some embodiments.
  • the encoder 104 in some embodiments comprises a frame sectioner/transformer 201 .
  • the frame sectioner/transformer 201 is configured to receive the left and right (or more generally any multichannel audio representation) input audio signals and generate frequency domain representations of these audio signals to be analysed and encoded. These frequency domain representations can be passed to the channel parameter determiner 203.
  • the frame sectioner/transformer 201 can be configured to section or segment the audio signal data into sections or frames suitable for frequency domain transformation.
  • the frame sectioner/transformer 201 in some embodiments can further be configured to window these frames or sections of audio signal data according to any suitable windowing function.
  • the frame sectioner/transformer 201 can be configured to generate frames of 20ms which overlap preceding and succeeding frames by 10ms each.
  • the frame sectioner/transformer 201 can be configured to perform any suitable time to frequency domain transformation on the audio signal data.
  • the time to frequency domain transformation can be a discrete Fourier transform (DFT), Fast Fourier transform (FFT), modified discrete cosine transform (MDCT).
  • DFT discrete Fourier transform
  • FFT Fast Fourier transform
  • MDCT modified discrete cosine transform
  • FFT Fast Fourier Transform
  • the output of the time to frequency domain transformer can be further processed to generate separate frequency band domain representations (sub-band representations) of each input channel audio signal data. These bands can be arranged in any suitable manner. For example these bands can be linearly spaced, or be perceptual or psychoacoustically allocated.
  • the frequency domain representations are passed to a channel analyser 203.
  • the encoder 104 can comprise a channel analyser 203.
  • the channel analyser 203 can be configured to receive the sub-band filtered representations of the multichannel or stereo input.
  • the channel analyser 203 can furthermore in some embodiments be configured to analyse the frequency domain audio signals and determine parameters associated with each sub-band with respect to the stereo or multichannel audio signal differences. Furthermore the channel analyser 203 can use these parameters and generate a mono channel.
  • the stereo parameters and the mono parameters/signal can then be output to a quantizer processor/mono encoder 205.
  • the encoder 104 comprises a quantizer processor/mono encoder 205.
  • the quantizer processor/mono encoder 205 can be configured to receive the stereo (difference) parameters determined by the channel analyser 203.
  • the quantizer processor/mono encoder 205 can then in some embodiments be configured to perform a quantization on the parameters and furthermore encode the parameters so that they can be output (either to be stored on the apparatus or passed to a further apparatus).
  • the quantizer processor/mono encoder 205 may furthermore be configured to receive the mono parameters/channel and furthermore encode the mono parameters/channel using any suitable encoding and furthermore based on the number of bits used to encode the stereo parameters.
  • the stereo parameters are first encoded and then the downmixed signal is encoded.
  • the bits that are saved by using entropy encoding for the stereo parameters may be used to encode the downmixed signal.
  • 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 a further apparatus.
  • step 501 The operation of generating audio frame band frequency domain representations is shown in Figure 4 by step 501 .
  • the channel analyser 203 comprises a channel difference parameter determiner 301 .
  • the channel difference parameter determiner 301 is configured to determine the various channel difference parameters.
  • the input audio signals are left and right audio signals. In some embodiments this may be generalised as j'th and j+1 'th audio channels from an multichannel audio system.
  • the channel difference parameter determiner 301 may be configured to receive the following parameters from the frame sectioner/transformer 201 , component i of the DFT of the right channel, - component i of the DFT of the left channel.
  • the channel difference determiner may be configured to generate channel energy parameters, for example:
  • channel difference determiner may be configured determine difference (stereo) parameters according to the following equations: side gain for sub-band b
  • the channel difference determiner may be configured to generate:
  • this value may be set to 0
  • the difference parameters such as the interchannel phase difference, the side gain and the residual prediction gain parameter values can be passed to the mono channel generator and as stereo channel parameters to the quantizer processor.
  • the encoder 104 (or as shown in Figure 5, the channel analyser 203) comprises a mono channel generator 305.
  • the mono channel generator is configured to receive the channel analyser values such as the side gains and inter channel phase differences from the channel difference determiner 301 .
  • the mono channel generator/encoder 305 can be configured to further receive the input multichannel audio signals.
  • the mono channel generator 305 can in some embodiments be configured to generate an 'aligned' or downmixed channel which is representative of the audio signals. In other words the mono channel generator 305 can generate a mono (or downmixed) channel signal which represents an aligned multichannel audio signal.
  • one of the left or right channel audio signals are delayed with respect to the other according to a determined delay difference and then the delayed channel and other channel audio signals are averaged to generate a mono channel signal.
  • any suitable mono channel generating method can be implemented.
  • the mono channel parameters/signal can then be output.
  • the mono channel signal is output to the quantizer processor/mono encoder 205 to be encoded.
  • FIG. 6 a summary of the analysis process (such as described in Figure 4 by steps 502 and 503) according to some embodiments and the operation of the channel analyser 203 shown in Figure 5 is shown as a flow diagram.
  • step 551 The operation of receiving the multichannel audio signal frequency components is shown in Figure 6 by step 551 .
  • step 552 The operation of determining intermediate parameters (e.g. Energy parameters for the audio signal channels) is shown in Figure 6 by step 552.
  • step 553 The operation of determining the difference parameters (e.g. side gain, interphase difference, residual prediction gain) which are generated at least partially from the intermediate parameters is shown in Figure 6 by step 553.
  • difference parameters e.g. side gain, interphase difference, residual prediction gain
  • step 555 The operation of generating a mono (downmix) channel signal/parameters from a stereo (multichannel) signal is shown in Figure 6 by step 555.
  • the quantizer processor/mono encoder 205 comprises a scalar quantizer 451 .
  • the scalar quantizer 451 is configured to receive the stereo parameters from the channel analyser 203.
  • the scalar quantizer can be configured to perform a scalar quantization on these values.
  • the scalar quantizer 451 can be configured to quantize the values with quantisation partition regions defined by the following array.
  • the scalar quantizer 451 can thus output an index value symbol associated with the region within the quantization partition region the level difference value occurs within.
  • an initial quantisation index value output can be as follows:
  • the index values can in some embodiments be output to a remapper 453.
  • the quantizer processor/mono encoder 205 comprises a remapper 453.
  • the remapper 453 can in some embodiments be configured to receive the output of the scalar quantizer 451 , in other words an index value associated with the quantization partition region within which the stereo or difference parameter is found and then the map or order the index value according to a defined mapping.
  • the index (re)mapping is based on an adaptive map selected from a range of defined maps.
  • the defined maps may be maps which are determined from training data or any other suitable manner which exploit intraframe correlation. For example these maps may exploit the correlation between adjacent symbols representing adjacent sub-band parameters.
  • the first symbol within a frame may be mapped according to a default or defined map.
  • the second symbol within a frame mapped according to a map which is selected based on the first symbol, and so on.
  • next (second) symbol may then be remapped based on a map which depends on the previous (first) symbol.
  • the reordering or remapping of the second symbol may be defined as
  • mappings may be stored as an array of mappings, such as for example
  • each symbol may have a separate array of reordering or remapping functions.
  • each array may be different.
  • the array may be defined or selected from more than first order relationships.
  • the array mapping function may be determined based on more than one previously determined symbol (sub- band) within the frame. This may also provide the ability to tune the coding efficiency at the cost of requiring additional arrays to be stored at the encoder and decoder.
  • the array mapping function may be determined based on a time previous symbol.
  • the mapping function may exploit any frame to frame correlation.
  • the implementation of time and sub- band based adaptive mapping causes the table ROM to significantly increase.
  • the table with the mapping will have 64 lines instead of 8 lines.
  • interframe correlation is exploited by applying GR coding to the difference between the current and previous frame. The numbers 0,1 ,-1 ,2,-2,... are mapped to 0,1 ,2,3,4 ...and encoded then with GR of order 0 or 1 , whichever is best.
  • the output of the remapper 453, is then output to the Golomb-Rice encoder
  • the quantizer processor/mono encoder 205 may comprise a map selector (or next symbol map selector) 454.
  • the map selector 454 or map determiner may be configured to select or determine the map or ordering which is to be applied by the remapper 453.
  • the map selector 454 may therefore receive a symbol or parameter index value from the scalar quantizer and from this value determine the map.
  • the selection or determination may be based on a look-up-table implementation. However in some embodiments the selection or determination may be made at least partially algorithmically.
  • the quantizer processor/mono encoder 205 can in some embodiments comprise 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 encoder 455 in such embodiments therefore outputs a codeword representing the current and previous index values.
  • the GR encoder 455 can then output the stereo codewords.
  • the codewords are passed to a multiplexer to be mixed with the encoded mono channel audio signal.
  • the stereo codewords can in some embodiments be passed to be stored or passed to further apparatus as a separate stream.
  • the encoding method may be used for the DFT parameters within a parametric stereo audio encoder.
  • the parameters to be encoded are side gains, residual prediction gains and interchannel phase differences. For an example superwideband case for a frame of audio data there may be
  • the values of all parameters may be scalarly quantized and their index is encoded with the adaptive GR.
  • the maps arrays for the three parameters type may be: For the side gain
  • the 'maps' table is relatively large compared with the other 'maps' table.
  • the structure of the maps table is analysed and where there is any defined structure in the table that can be exploited then this can be used to compress the maps table.
  • this can be used to compress the maps table.
  • the analysis may enable the following data to be stored:
  • This data may be used such that, in order to obtain for instance the 5 th line of 'maps'
  • the 4 th pseudo-line line of sg_data1 tells that its data (in bold and underlined in the above line taken as example) is part of the corresponding line in 'maps'.
  • the data in sg_data2, 4 th pseudo-line states that there are 14 components in sg_data1 that should be copied in 'maps' (first parameter of sg_data2, line 4), and that starting with position 16 the corresponding 'maps' line will be automatically filled.
  • the automatic filling is such that the first consecutive number after the last value from sg_data1 pseudo-line will be at the beginning of the string right 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 at one end, then numbers are filled consecutively just on one side.
  • the quantizer processor/mono encoder 205 further comprises a mono (downmix) channel encoder 456.
  • the mono (downmix) channel encoder 456 may be configured to receive the mono (downmix) channel or parameters. Furthermore the mono (downmix) channel encoder 456 may be configured to receive an indication of the number of bits which have been used in the GR encoder for encoding the current frame. The mono (downmix) channel encoder 456 may then be configured to encode the mono (downmix) channel or parameters based on any suitable encoding method based on the knowledge of the number of bits used by the stereo parameter encoding.
  • the mono channel generator/encoder 456 can encode the generated mono channel audio signal using any suitable encoding format.
  • the mono channel audio signal can be encoded using an Enhanced Voice Service (EVS) mono channel encoded form, which may contain a bit stream interoperable version of the Adaptive Multi-Rate - Wide Band (AMR-WB) codec.
  • EVS Enhanced Voice Service
  • AMR-WB Adaptive Multi-Rate - Wide Band
  • FIG 8 a summary of the encoding process (such as described in Figure 4 by steps 505) according to some embodiments and the operation of the quantizer processor/mono encoder 205 shown in Figure 7 is shown as a flow diagram.
  • step 701 The operation of receiving the stereo parameters is shown in Figure 8 by step 701 .
  • the 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.
  • the mono channel decoder 801 can be configured to decode the encoded mono channel audio signal using the inverse process to the mono channel coder shown in the encoder.
  • the mono channel decoder 801 may be configured to receive an indicator from the stereo channel decoder 803 indicating the number of bits used for the stereo signal to assist the decoding of the mono channel.
  • the mono channel decoder 801 can be configured to output the mono channel audio signal to the stereo channel generator 809.
  • the decoder 108 can comprise a stereo channel decoder 803.
  • the stereo channel decoder 803 is configured to receive the encoded stereo parameters.
  • stereo channel decoder 803 can 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.
  • the decoder comprises a symbol map selector 805 (or map determiner or order determiner or order selector).
  • the symbol map selector 805 can be configured to receive the current frame stereo channel index values (decoded and reordered symbols) and select a symbol map to reverse the mapping used in the encoder.
  • the symbol map selector 805 is configured to determine a map based on a previously determined symbol decoded within a frame.
  • the (symbol) map can be output to the symbol reorderer 807.
  • the decoder 108 comprises a symbol reorderer 807.
  • the symbol or index reorderer 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.
  • the symbol reorderer 807 is configured to re-order the index values to the original order output by the scaler quantizer within the encoder.
  • the symbol reorderer 807 is configured to de- quantize the demapped or re-ordered index value into a parameter (such as the interaural time difference/correlation value; and interaural level difference/energy difference value) using the inverse process to that defined within the quantizer section of the quantizer processor within the encoder.
  • a parameter such as the interaural time difference/correlation value; and interaural level difference/energy difference value
  • 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 applying the level differences to the mono channel to generate a second channel.
  • step 901 The operation of receiving the encoded mono channel audio signal is shown in Figure 10 by step 901 .
  • step 907 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 outputting 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.
  • the map is selected from a stored table it is understood that in some embodiments the map for the current symbol may be determined algorithmically based on a function which receives as an input a previously determined symbol.
  • embodiments of the application operating within a codec within an apparatus 10, it would be appreciated that the invention as described below may be implemented as part of any audio (or speech) codec, including any variable rate/adaptive rate audio (or speech) codec.
  • embodiments of the application may be implemented in an audio codec which may implement audio coding over fixed or wired communication paths.
  • user equipment may comprise an audio codec such as those described in embodiments of the application above.
  • user equipment is intended to cover any suitable type of wireless user equipment, such as mobile telephones, portable data processing devices or portable web browsers.
  • PLMN public land mobile network
  • elements of a public land mobile network may also comprise audio codecs as described above.
  • the various embodiments of the application may be implemented in hardware or special purpose circuits, software, logic or any combination thereof.
  • some aspects may be implemented in hardware, while other aspects may be implemented in firmware or software which may be executed by a controller, microprocessor or other computing device, although the invention is not limited thereto.
  • firmware or software which may be executed by a controller, microprocessor or other computing device, although the invention is not limited thereto.
  • While various aspects of the application may be illustrated and described as block diagrams, flow charts, or using some other pictorial representation, it is well understood that these blocks, apparatus, systems, techniques or methods described herein may be implemented in, as non-limiting examples, hardware, software, firmware, special purpose circuits or logic, general purpose hardware or controller or other computing devices, or some combination thereof.
  • any blocks of the logic flow 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 to 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.
  • the data processors may be of any type suitable to 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 (ASIC), gate level circuits and processors based on multi-core processor architecture, as non-limiting examples.
  • Embodiments of the application may be practiced in various components such as integrated circuit modules.
  • the design of integrated circuits is by and large a highly automated process. Complex and powerful software tools are available for converting 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, California and Cadence Design, of San Jose, California automatically route conductors and locate components on a semiconductor chip using well established rules of design as well as libraries of pre-stored design modules.
  • the resultant design in a standardized electronic format (e.g., Opus, GDSII, or the like) may be transmitted to a semiconductor fabrication facility or "fab" for fabrication.
  • circuitry refers to all of the following: (a) hardware-only circuit implementations (such as implementations in only analog and/or digital circuitry) and
  • circuits and software and/or firmware
  • combinations of circuits and software 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(ies) that work together to cause an apparatus, such as a mobile phone or server, to perform various functions and
  • circuits such as a microprocessor(s) or a portion of a microprocessor(s), that require software or firmware for operation, even if the software or firmware is not physically present.
  • circuitry' applies to all uses of this term in this application, including any claims.
  • the term 'circuitry' would also cover an implementation of merely a processor (or multiple processors) or portion of a processor and its (or their) accompanying software and/or firmware.
  • the term 'circuitry' would also cover, for example and if applicable to the particular claim element, a baseband integrated circuit or applications processor integrated circuit for a mobile phone or similar integrated circuit in server, a cellular network device, or other network device.

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  • Engineering & Computer Science (AREA)
  • Physics & Mathematics (AREA)
  • Multimedia (AREA)
  • Health & Medical Sciences (AREA)
  • Audiology, Speech & Language Pathology (AREA)
  • Human Computer Interaction (AREA)
  • Signal Processing (AREA)
  • Acoustics & Sound (AREA)
  • Computational Linguistics (AREA)
  • Spectroscopy & Molecular Physics (AREA)
  • Quality & Reliability (AREA)
  • Mathematical Physics (AREA)
  • Compression, Expansion, Code Conversion, And Decoders (AREA)
  • Electrophonic Musical Instruments (AREA)
  • Stereo-Broadcasting Methods (AREA)
  • Signal Processing Not Specific To The Method Of Recording And Reproducing (AREA)

Abstract

La présente invention concerne un procédé comprenant : la réception d'au moins deux signaux de canal audio ; la détermination, pour une première trame, d'au moins deux paramètres représentant une différence entre les au moins deux signaux de canal audio ; la quantification scalaire (451) des au moins deux paramètres pour générer au moins deux valeurs d'index ; la détermination d'une carte d'index initiale pour réordonner (453) l'une des au moins deux valeurs d'index, et la détermination d'au moins une carte d'index supplémentaire pour réordonner au moins une autre des au moins deux valeurs d'index, l'au moins une carte d'index supplémentaire étant déterminée sur la base de l'une des au moins deux valeurs d'index ; le réordonnancement de l'une des au moins deux valeurs d'index sur la base de la carte d'index initiale ; le réordonnancement de l'autre desdites au moins deux valeurs d'index sur la base de l'au moins une carte d'index supplémentaire ; le codage (455) de la valeur d'index réordonnée parmi les au moins deux valeurs d'index en fonction d'une position d'ordre de l'une des au moins deux valeurs d'index ; le codage de la valeur d'index supplémentaire réordonnée des au moins deux valeurs d'index sur la base d'une position d'ordre de la valeur réordonnée supplémentaire des au moins deux valeurs d'index ; la génération d'une représentation de canal unique des au moins deux signaux de canal audio en fonction des au moins deux paramètres ; et le codage (456) de la représentation de canal unique.
EP18747600.7A 2017-01-31 2018-01-11 Codeur de signal audio stéréo Active EP3577649B1 (fr)

Applications Claiming Priority (2)

Application Number Priority Date Filing Date Title
GB1701594.2A GB2559199A (en) 2017-01-31 2017-01-31 Stereo audio signal encoder
PCT/FI2018/050018 WO2018142018A1 (fr) 2017-01-31 2018-01-11 Codeur de signal audio stéréo

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EP3577649A1 true EP3577649A1 (fr) 2019-12-11
EP3577649A4 EP3577649A4 (fr) 2020-11-11
EP3577649B1 EP3577649B1 (fr) 2023-05-10

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MX2019005147A (es) * 2016-11-08 2019-06-24 Fraunhofer Ges Forschung Aparato y metodo para codificar o decodificar una se?al multicanal usando una ganancia lateral y una ganancia residual.
GB2580899A (en) * 2019-01-22 2020-08-05 Nokia Technologies Oy Audio representation and associated rendering

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EP1523863A1 (fr) * 2002-07-16 2005-04-20 Koninklijke Philips Electronics N.V. Codage audio
US7433824B2 (en) * 2002-09-04 2008-10-07 Microsoft Corporation Entropy coding by adapting coding between level and run-length/level modes
FI3573056T3 (fi) * 2008-07-11 2022-11-30 Audiokooderi ja audiodekooderi
US9799339B2 (en) * 2012-05-29 2017-10-24 Nokia Technologies Oy Stereo audio signal encoder
WO2013185857A1 (fr) * 2012-06-14 2013-12-19 Telefonaktiebolaget L M Ericsson (Publ) Procédé et dispositif pour codage/décodage évolutif de faible complexité
EP2875510A4 (fr) * 2012-07-19 2016-04-13 Nokia Technologies Oy Codeur de signal audio stéréo
US9530422B2 (en) * 2013-06-27 2016-12-27 Dolby Laboratories Licensing Corporation Bitstream syntax for spatial voice coding
TWI579831B (zh) * 2013-09-12 2017-04-21 杜比國際公司 用於參數量化的方法、用於量化的參數之解量化方法及其電腦可讀取的媒體、音頻編碼器、音頻解碼器及音頻系統

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EP3577649B1 (fr) 2023-05-10
GB2559199A (en) 2018-08-01
WO2018142018A1 (fr) 2018-08-09
ES2946235T3 (es) 2023-07-14
EP3577649A4 (fr) 2020-11-11
GB201701594D0 (en) 2017-03-15

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