EP2168121A1 - Quantification after linear conversion combining audio signals of a sound scene, and related encoder - Google Patents
Quantification after linear conversion combining audio signals of a sound scene, and related encoderInfo
- Publication number
- EP2168121A1 EP2168121A1 EP08806144A EP08806144A EP2168121A1 EP 2168121 A1 EP2168121 A1 EP 2168121A1 EP 08806144 A EP08806144 A EP 08806144A EP 08806144 A EP08806144 A EP 08806144A EP 2168121 A1 EP2168121 A1 EP 2168121A1
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- quantization
- components
- function
- audio signals
- module
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- 238000011002 quantification Methods 0.000 title abstract description 6
- 238000006243 chemical reaction Methods 0.000 title abstract description 4
- 238000000034 method Methods 0.000 claims abstract description 29
- 230000000873 masking effect Effects 0.000 claims abstract description 19
- 238000012360 testing method Methods 0.000 claims abstract description 5
- 238000013139 quantization Methods 0.000 claims description 90
- 230000009466 transformation Effects 0.000 claims description 41
- 238000012545 processing Methods 0.000 claims description 4
- 238000004590 computer program Methods 0.000 claims description 2
- 238000012804 iterative process Methods 0.000 claims description 2
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- 230000003595 spectral effect Effects 0.000 description 16
- 239000013598 vector Substances 0.000 description 14
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Classifications
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- 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
- G10L19/02—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
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04S—STEREOPHONIC SYSTEMS
- H04S3/00—Systems employing more than two channels, e.g. quadraphonic
- H04S3/008—Systems employing more than two channels, e.g. quadraphonic in which the audio signals are in digital form, i.e. employing more than two discrete digital channels
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04S—STEREOPHONIC SYSTEMS
- H04S3/00—Systems employing more than two channels, e.g. quadraphonic
- H04S3/02—Systems employing more than two channels, e.g. quadraphonic of the matrix type, i.e. in which input signals are combined algebraically, e.g. after having been phase shifted with respect to each other
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04S—STEREOPHONIC SYSTEMS
- H04S2420/00—Techniques used stereophonic systems covered by H04S but not provided for in its groups
- H04S2420/01—Enhancing the perception of the sound image or of the spatial distribution using head related transfer functions [HRTF's] or equivalents thereof, e.g. interaural time difference [ITD] or interaural level difference [ILD]
Definitions
- the present invention relates to audio signal coding devices, intended in particular to take place in applications for transmission or storage of digitized and compressed audio signals.
- the invention relates more specifically to the quantization modules included in these audio coding devices.
- a 3D sound scene also called spatialized sound, comprises a plurality of audio channels each corresponding to monophonic signals.
- a signal coding technique for a sound stage used in the "MPEG Audio Surround” encoder includes the extraction and coding of spatial parameters from the set of monophonic audio signals on the different channels. These signals are then mixed to obtain a monophonic or stereophonic signal, which is then compressed by a conventional mono or stereo encoder (for example of the MPEG-4 AAC, HE-AAC type, etc.). At the level of the decoder, the synthesis of the rendered 3D sound scene is made from the spatial parameters and the decoded mono or stereo signal.
- the coding of the multichannel signals in certain cases requires the introduction of a transformation (KLT, Ambiophonic, DCT, etc.) making it possible to better take into account the interactions that may exist between the different signals of the sound scene to be encoded.
- KLT KLT, Ambiophonic, DCT, etc.
- the invention proposes a method for quantifying components, at least some of these components being each determined according to a plurality of audio signals of a scene. sound and calculable by applying a linear transformation on said audio signals.
- a quantization function is determined to be applied to said components in a given frequency band by testing a condition relating to at least one audio signal and depending at least on a comparison made between a psychoacoustic masking threshold relative to the audio signal. in the given frequency band, and a value determined according to the inverse linear transformation and quantization errors of the components by said function on the given frequency band.
- Such a method therefore makes it possible to determine a quantization function which makes it possible to mask, in the playback listening field, the noise introduced with respect to the audio signal of the initial sound scene.
- the sound scene restored after the coding and decoding operations therefore presents a better audio quality.
- the introduction of a multichannel transform transforms the real signals into a new domain different from the listening domain.
- the quantization of the components resulting from this transform according to the methods of the state of the art, based on a perceptual criterion (ie respecting the masking threshold on the latter), does not guarantee a minimum distortion on the real signals restored in the listening domain.
- the calculation of the quantization function according to the invention makes it possible to guarantee that the quantization noises induced on the real signals by the quantization of the transformed components are minimal in the sense of a perceptual criterion. The condition of a maximum improvement of the perceptual quality of the signals in the listening domain is then verified.
- the condition is relative to several audio signals and depends on several comparisons, each comparison being made between a psychoacoustic masking threshold relative to a respective audio signal in the given frequency band, and a value determined according to the inverse linear transformation and quantization errors of the components by said function. This arrangement further enhances the audio quality of the restored sound stage.
- the determination of the quantization function is repeated when updating the values of the components to be quantized. This arrangement also makes it possible to increase the audio quality of the restored sound scene, by adapting the quantization over time according to the characteristics of the signals.
- B ⁇ (s) represents a parameter the quantization function s in the band on the j th component
- ⁇ ⁇ (s) is the expected value in the strip s of the square root of the j-th component.
- a quantization function is determined to apply components in the given frequency band using an iterative process generating at each iteration a parameter of the candidate quantization function satisfying the condition and associated with a corresponding flow rate, the iteration being stopped when the flow rate is below a given threshold.
- Such an arrangement thus makes it possible to simply determine a quantization function based on the determined parameters, allowing the noise to be masked in the playback listening domain while reducing the coding bit rate below a given threshold.
- the linear transformation is an ambiophonic transformation.
- the linear transformation is an ambiophonic transformation (called “ambisonic").
- ambisonic ambiophonic transformation
- This arrangement makes it possible on the one hand to reduce the number of data to be transmitted since, in general, the N signals can be very satisfactorily described by a reduced number of ambiophonic components (for example, a number equal to 3 or 5). , which is smaller than N.
- This arrangement also allows coding adaptability to any type of sound rendering system, since it is sufficient at the decoder level to apply an inverse surround transform of size Q'x (2p '+ 1). , (where Q 'is equal to the number of loudspeakers of the sound rendering system used at the output of the decoder and 2p' + 1 the number of received surround components), to determine the signals to be supplied to the sound rendering system.
- the invention can be implemented with any linear transformation, for example the DCT or the KLT (in English "Karhunen Loeve Transform") transform which corresponds to a decomposition on principal components in a space representing the statistics of the signals and allows to distinguish the most energetic components from the least energy components.
- any linear transformation for example the DCT or the KLT (in English "Karhunen Loeve Transform") transform which corresponds to a decomposition on principal components in a space representing the statistics of the signals and allows to distinguish the most energetic components from the least energy components.
- the invention proposes a quantization module adapted to quantify components, at least some of these components being each determined according to a plurality of audio signals of a sound scene and calculable by application of a transformation. linearly on said audio signals, said quantization module being adapted to implement the steps of a method according to the first aspect of the invention.
- the invention provides an audio coder adapted to encode an audio scene comprising a plurality of respective signals into an output bit stream, comprising: a transform module adapted to calculate by applying a linear transformation on said audio signals, components at least some of which are determined each according to a plurality of audio signals of a sound scene; and a quantization module according to the second aspect of the invention adapted to determine at least one quantization function over at least a given frequency band and for quantizing the components on the given frequency band as a function of at least the determined quantization function; the audio coder being adapted to constitute a bit stream according to at least quantization data delivered by the quantization module.
- the invention proposes a computer program to be installed in a quantization module, said program comprising instructions for implementing the steps of a method according to the first aspect of the invention during execution. of the program by means of processing said module.
- the invention proposes coding data, determined following the implementation of a quantization method according to the first aspect of the invention.
- FIG. 1 shows an encoder in an embodiment of the invention
- FIG. 2 represents a decoder in one embodiment of the invention
- Fig. 3 is a flowchart showing steps of a method in one embodiment of the invention.
- Figure 1 shows an audio coder 1 in one embodiment of the invention. It relies on the technology of perceptual audio coders, for example MPEG-4 AAC type.
- the encoder 1 comprises a time / frequency transformation module 2, a linear transformation module 3, a quantization module 4, a Huffman entropy coding module 5 and a masking curve calculation module 6, for transmission.
- a bit stream ⁇ representing the signals supplied at the input of the encoder 1.
- a 3D sound scene comprises N channels on each a respective audio signal S 1 , ..., S N is delivered.
- Figure 2 shows an audio decoder 100 in one embodiment of the invention.
- the decoder 100 comprises a bit sequence reading module 101, an inverse quantization module 102, an inverse linear transformation module 103, a frequency / time transformation module 104.
- the decoder 100 is adapted to receive as input the bitstream ⁇ transmitted by the encoder 1 and to output Q 'signals S ⁇ , ..., S' Q. for supplying the respective loudspeakers H1, H2 ..., HQ 'of a sound rendering system 105.
- the time / frequency conversion module 2 of the encoder 1 receives as input the N signals S 1 ,... S N of the 3D sound scene to be encoded, in the form of successive blocks.
- Each block m received has N time frames each indicating different values taken over time by a respective signal.
- the time / frequency transformation module 2 On each time frame of each of the signals, the time / frequency transformation module 2 performs a time / frequency transformation, in this case a modified discrete cosine transform (MDCT).
- MDCT modified discrete cosine transform
- the coding of multichannel signals comprises in the case considered a linear transformation, making it possible to take into account the interactions between the different audio signals to be coded, before the monophonic coding, by the quantization module 4, of the components resulting from the linear transformation.
- the linear transformation module 3 is adapted to perform a linear transformation of the coefficients of the spectral representations (X t ⁇ ⁇ ⁇ N provided, in one embodiment it is adapted to perform a spatial transformation, and it determines the spatial components of the signals ⁇ x, ⁇ ⁇ ⁇ N in the frequency domain, resulting from the projection on a spatial referential depending on the order of the transformation
- the order of a spatial transformation is related to the angular frequency according to which it "scrutinizes" The sound field.
- the surround components are determined as follows:
- R is the ambiophonic transformation matrix
- Each of the ambiophonic components is therefore determined according to several signals (S 1 ) ⁇ N.
- the masking curve calculation module 6 is adapted to determine the spectral masking curve of each frame of a signal Si considered individually in the block m, using its spectral representation Xi and a psychoacoustic model.
- the masking curve calculation module 6 thus calculates a masking threshold M TM (s, i) relative to the frame of each signal (S t ) 1 ⁇ n ⁇ N in the block m, for each frequency band s considered during the quantification.
- Each frequency band s is part of a set of frequency bands including for example the bands as normalized for the MPEG-4 AAC encoder.
- the masking thresholds M TM (s, i) for each signal S 1 and each frequency band s are delivered to the quantization module 4.
- the quantization module 4 is adapted to quantify the components ⁇ Y ⁇ ) ⁇ r that are input to it, so as to reduce the bit rate required for transmission. Respective quantization functions are determined by the quantization module 4 on each frequency band s.
- the quantization module 4 quantizes each spectral coefficient (Y ] t ) 1]] ⁇ r such that the frequency F t is an element of the
- k takes the values of the set +1) is equal to the number of spectral coefficients to be quantized in the s-band for all the surround components.
- O ⁇ t ⁇ M-1 signals takes the following form, according to MPEG-4 AAC
- Arr is a rounding function that delivers an integer value.
- Arr (x) is for example the function providing the integer closest to the variable x, or the function "integer part" of the variable x, etc.
- the quantization module 4 is adapted to determine a quantization function to be applied on a frequency band, verifying that the masking threshold M TM (s, i) of each signal S 1 in the listening domain, with 1 ⁇ i ⁇ N, is greater than the power of the error made, on an audio signal restored in the listening domain corresponding to the channel i (and not in the linear transformation domain), by the quantization errors made to the ambiophonic components.
- the quantization module 4 is therefore adapted to determine, during the processing of a block m of signals, the quantization function defined using the scale parameters ⁇ Bf is)) ⁇ ⁇ r relative to each band s, such that, for all i, 1 ⁇ i ⁇ N, the error introduced on the signal S 1 in the band s by the quantization of the ambiophonic components is less than the mask threshold M TM (s, i) of the signal S 1 on the band s.
- a problem to be solved by the quantization module 4 is therefore to determine, on each band s, the set of scaling coefficients ( ⁇ j (S)) ⁇ satisfying the following formula (1):
- B ⁇ (s) represents a parameter characterizing the quantization function s in the band on the j-th component.
- the choice of B ⁇ (s) determines in a bijective manner the quantization function used.
- This arrangement has the effect that the noise brought into the listening domain by the quantization on the components resulting from the linear transformation remains masked by the signal in the listening domain, which contributes to a better quality of the signals restored in the listening domain.
- e TM (k) are the quantization errors introduced on the (k max s - k ⁇ an + ls + l) spectral coefficients of ambiophonic components corresponding to frequencies in the band s.
- the quantization errors e TM (k) are independent random variables equi-distributed according to the index k; the quantization errors e TM (k) are random variables according to the index i; the number of samples in a band s is large enough; the coder 1 works at high resolution.
- the power P e m (s, i) of the quantization error, in a subband s and for a signal S 1 tends, when the number of coefficients in a band s increases, to a Gaussian whose mean m um / e ⁇ and the variance ⁇ um , e ⁇ are given by the following formulas:
- e R the rounding error specific to the rounding function Arr. For example, if Arr (x) is the function providing the integer closest to the variable x, e R is equal to 0.5. If Arr (x) is the function "integer part" of the variable x, e R is equal to 1.
- This last equation represents a sufficient condition for the noise corresponding to the channel i to be masked at the output in the listening domain.
- the quantization module 4 is adapted to determine using the latter equation, for a block m of current frames, scale coefficients [BJ (s)) ⁇ r guaranteeing that the noise in the listening domain is hidden.
- the quantization module 4 is adapted to determine, for a block m of current frames, scaling coefficients [BJ (s)) ⁇ ensuring that the noise in the d domain listening is masked and further to respect a flow constraint.
- the conditions to be respected are the following:
- D TM (s) is the bit rate assigned to the surround component Y 1 in the s band.
- bit rate assigned to an ambiophonic component in a band s is a logarithmic function of the scale coefficient, ie:
- the resolution of this constrained optimization problem is for example carried out using the Lagrangian method.
- the Lagrangian function is written in the following form:
- the iterative relative gradient method (see in particular the Derrien document) is used to solve this system.
- the vector m is chosen equal to:
- the quantization module 4 is adapted to implement the steps of the method described below with reference to FIG. 3 on each quantization band s during the quantization of a block m of signals ( S t ) 1 ⁇ N .
- the method is based on an iterative algorithm comprising instructions for implementing the steps described below during the execution of the algorithm on calculation means of the quantization module 4.
- the steps of the iterative loop for a (k + 1) th iteration, with k integer greater than or equal to 0, are as follows.
- a step d / the value of the function F is calculated on the band s, representing the corresponding bit rate for the band s:
- a step e / the calculated value F (s) is compared with the given threshold D.
- a step g / the index k is incremented by one unit and the steps b /, c /, d / and e / are repeated.
- the quantization function thus determined for the respective s-bands and respective surround components is then applied to the spectral coefficients of the surround components.
- the quantization indices as well as definition elements of the quantization function are provided to the Huffman coding module.
- the coding data delivered by the Huffman coding module 5 is then transmitted as a bit stream ⁇ to the decoder 100. Operations performed at the decoder:
- the bit sequence reading module 101 is adapted to extract coding data present in the stream ⁇ received by the decoder and to deduce, in each band s, quantization indices i (k) and scale coefficients (B TM (s)) ⁇ ] ⁇ r .
- the inverse quantization module 102 is adapted to determine the spectral coefficients, relative to the band s, of the corresponding ambiophonic components as a function of the quantization indices i (k) and the scale coefficients (B TM (s)) ⁇ ] ⁇ r in each band s.
- Ambiophonic decoding is then applied to the decoded surround components, so as to determine the signals S'i, S ' 2 , ..., S'Q ⁇ for the Q' speakers H1, H2 ..., HQ .
- the quantization noise at the output of the decoder 100 is a constant which depends only on the transform R used and the quantization module 4 because the psychoacoustic data used during the coding do not take into consideration the processing performed during the rendering by the processor. decoder. Indeed, the psychoacoustic model does not take into account the acoustic interactions between the different signals, but calculates the masking curve of a signal as if it were the only one listened to. The error calculated on this signal therefore remains constant and masked for any surround decoding matrix used. This surround decoding matrix will simply change the distribution of the error on the different speakers output.
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FR0704794 | 2007-07-03 | ||
PCT/FR2008/051220 WO2009007639A1 (en) | 2007-07-03 | 2008-07-01 | Quantification after linear conversion combining audio signals of a sound scene, and related encoder |
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Cited By (2)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
CN112634913A (en) * | 2015-03-09 | 2021-04-09 | 弗劳恩霍夫应用研究促进协会 | Audio encoder for encoding and audio decoder for decoding |
US11146903B2 (en) | 2013-05-29 | 2021-10-12 | Qualcomm Incorporated | Compression of decomposed representations of a sound field |
Families Citing this family (9)
Publication number | Priority date | Publication date | Assignee | Title |
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EP2469741A1 (en) * | 2010-12-21 | 2012-06-27 | Thomson Licensing | Method and apparatus for encoding and decoding successive frames of an ambisonics representation of a 2- or 3-dimensional sound field |
JP6267860B2 (en) * | 2011-11-28 | 2018-01-24 | 三星電子株式会社Samsung Electronics Co.,Ltd. | Audio signal transmitting apparatus, audio signal receiving apparatus and method thereof |
US9466305B2 (en) | 2013-05-29 | 2016-10-11 | Qualcomm Incorporated | Performing positional analysis to code spherical harmonic coefficients |
US9922656B2 (en) | 2014-01-30 | 2018-03-20 | Qualcomm Incorporated | Transitioning of ambient higher-order ambisonic coefficients |
US9489955B2 (en) | 2014-01-30 | 2016-11-08 | Qualcomm Incorporated | Indicating frame parameter reusability for coding vectors |
US10770087B2 (en) | 2014-05-16 | 2020-09-08 | Qualcomm Incorporated | Selecting codebooks for coding vectors decomposed from higher-order ambisonic audio signals |
US9620137B2 (en) | 2014-05-16 | 2017-04-11 | Qualcomm Incorporated | Determining between scalar and vector quantization in higher order ambisonic coefficients |
US9852737B2 (en) | 2014-05-16 | 2017-12-26 | Qualcomm Incorporated | Coding vectors decomposed from higher-order ambisonics audio signals |
US9747910B2 (en) | 2014-09-26 | 2017-08-29 | Qualcomm Incorporated | Switching between predictive and non-predictive quantization techniques in a higher order ambisonics (HOA) framework |
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WO1992012607A1 (en) * | 1991-01-08 | 1992-07-23 | Dolby Laboratories Licensing Corporation | Encoder/decoder for multidimensional sound fields |
US7548853B2 (en) * | 2005-06-17 | 2009-06-16 | Shmunk Dmitry V | Scalable compressed audio bit stream and codec using a hierarchical filterbank and multichannel joint coding |
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2008
- 2008-07-01 EP EP08806144.5A patent/EP2168121B1/en active Active
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Cited By (5)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
US11146903B2 (en) | 2013-05-29 | 2021-10-12 | Qualcomm Incorporated | Compression of decomposed representations of a sound field |
US11962990B2 (en) | 2013-05-29 | 2024-04-16 | Qualcomm Incorporated | Reordering of foreground audio objects in the ambisonics domain |
CN112634913A (en) * | 2015-03-09 | 2021-04-09 | 弗劳恩霍夫应用研究促进协会 | Audio encoder for encoding and audio decoder for decoding |
US11741973B2 (en) | 2015-03-09 | 2023-08-29 | Fraunhofer-Gesellschaft Zur Foerderung Der Angewandten Forschung E.V. | Audio encoder for encoding a multichannel signal and audio decoder for decoding an encoded audio signal |
CN112634913B (en) * | 2015-03-09 | 2024-04-09 | 弗劳恩霍夫应用研究促进协会 | Audio encoder for encoding and audio decoder for decoding |
Also Published As
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US8612220B2 (en) | 2013-12-17 |
US20100198585A1 (en) | 2010-08-05 |
WO2009007639A1 (en) | 2009-01-15 |
EP2168121B1 (en) | 2018-06-06 |
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