WO2006056100A1 - Procede et dispositif de codage/decodage utilisant la redondance des signaux intra-canal - Google Patents

Procede et dispositif de codage/decodage utilisant la redondance des signaux intra-canal Download PDF

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WO2006056100A1
WO2006056100A1 PCT/CN2004/001349 CN2004001349W WO2006056100A1 WO 2006056100 A1 WO2006056100 A1 WO 2006056100A1 CN 2004001349 W CN2004001349 W CN 2004001349W WO 2006056100 A1 WO2006056100 A1 WO 2006056100A1
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integer
transform
channel
coefficients
klt
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PCT/CN2004/001349
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Chinese (zh)
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Xingde Pan
Lei Wang
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Beijing E-World Technology Co., Ltd
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Priority to CN200480044452.4A priority Critical patent/CN101065796A/zh
Priority to PCT/CN2004/001349 priority patent/WO2006056100A1/fr
Publication of WO2006056100A1 publication Critical patent/WO2006056100A1/fr

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    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; 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

Definitions

  • the present invention relates to the field of audio codec technology, and in particular to a method and apparatus for encoding and decoding using inter-channel redundancy. Background technique
  • the digital audio signal is audio-encoded or audio-compressed for storage and transmission.
  • the purpose of encoding an audio signal is to achieve a transparent representation of the audio signal with as few bits as possible, such as little difference between the originally input audio signal and the encoded output audio signal.
  • CDs represented the many advantages of digitally representing audio signals, such as high fidelity, large dynamic range, and robustness.
  • these advantages are at the expense of high data rates.
  • the digitization of a CD-quality stereo signal requires a sampling rate of 44.1 kHz, and each sample value is uniformly quantized with 16 bits, so that the uncompressed data rate reaches 1.41 Mb/s, so high.
  • the data rate brings great inconvenience to the transmission and storage of data, especially in the case of multimedia applications and wireless transmission applications, and is limited by bandwidth and cost.
  • new network and wireless multimedia digital audio systems are required to reduce the rate of data without compromising the quality of the audio.
  • FIG. 1 shows a block diagram of an MPEG-2 AAC encoder including a gain controller 101, a modified discrete pre-transform (MDCT) module 102, a time domain noise shaping module 103, an intensity/coupling module 104, a psychoacoustic model, a second order backward adaptive predictor 105, a / difference stereo module 106, a bit allocation and quantization encoding module 107, and a bitstream multiplexing module 108, wherein the bit allocation and quantization encoding module 107 further includes a compression ratio/distortion processing controller, Scale factor module, non-uniform quantizer and entropy coding module.
  • MDCT modified discrete pre-transform
  • the audio signal After the audio signal passes through the gain controller 101, it enters the modified discrete cosine transform module 102, performs time-frequency transform according to different signals, and then processes the spectral coefficients output by the modified discrete cosine transform module 102 through the time domain noise shaping module 103, and time domain noise.
  • the shaping technique performs linear prediction analysis on the spectral coefficients in the frequency domain, and then controls the shape of the quantization noise in the time domain according to the above analysis, thereby achieving the purpose of controlling the pre-echo.
  • the intensity/coupling module 104 is used for stereo encoding of signal strength, since for a high frequency band (greater than 2 kHz) the sense of direction of the hearing is related to the change in signal strength (signal envelope), independent of the waveform of the signal, ie The constant envelope signal has no effect on the sense of direction of the hearing, so this feature and related information between multiple channels can be used to combine several channels into one common channel for encoding.
  • the second-order backward adaptive predictor 105 is used to eliminate redundancy of the steady state signal and improve coding efficiency.
  • the and difference stereo (M/S) module 106 is used to operate the channel pair, which refers to two channels such as a left channel or a left and right surround channel in a two-channel signal or a multi-channel signal.
  • the M/S module 106 utilizes the correlation between the two channels of the channel pair to achieve the effect of reducing the code rate and improving the coding efficiency.
  • the bit allocation and quantization coding module 107 is implemented by a nested loop process in which the non-uniform quantizer performs lossy coding, and the entropy coding module performs lossless coding, which removes redundancy and reduces correlation.
  • Nested loops include inner loops and outer loops, where inner loops are tuned The step size of the non-uniform quantizer is used until the provided bits are used up, and the outer loop uses the ratio of quantization noise to the masking threshold to estimate the encoding quality of the signal.
  • the last encoded signal forms an encoded audio stream output through bitstream multiplexing module 108.
  • the input signal simultaneously performs four-band multi-phase filter bank (PQF) to generate four equal-bandwidth bands, and each band uses MDCT to generate 256 spectral coefficients, for a total of 1024.
  • PQF multi-phase filter bank
  • each band uses MDCT to generate 256 spectral coefficients, for a total of 1024.
  • a gain controller 101 is used in each frequency band.
  • the high frequency PQF band can be ignored to obtain a low sampling rate signal.
  • FIG. 2 shows a block diagram of the corresponding MPEG-2 AAC decoder.
  • the decoder includes a bitstream demultiplexing module 201, a lossless decoding module 202, an inverse quantizer 203, a scale factor module 204, and/or a difference stereo (M/S) module 205, a prediction module 206, an intensity/coupling module 207, and a time.
  • the encoded audio stream is demultiplexed by the bitstream demultiplexing module 201 to obtain a corresponding data stream and control stream.
  • IMDCT inverse modified discrete cosine transform
  • the inverse quantizer 203 is a set of non-uniform quantizers implemented by a companding function for converting integer quantized values into reconstructed spectra. Since the scale factor module in the encoder differentiates the current scale factor from the previous scale factor and then uses the Huffman code for the difference value, the scale factor module 204 in the decoder performs Huffman decoding to obtain the corresponding difference value, and then recovers. A true scale factor.
  • the M/S module 205 converts the sum/difference channel into left and right channels under the control of side information.
  • the prediction decoding is performed by the prediction module 206 in the decoder.
  • the intensity/coupling module 207 performs intensity/coupling decoding under the control of the side information, and then outputs it to the time domain noise shaping module 208 for time domain noise shaping decoding, and finally performs frequency-time conversion by the inverse modified discrete cosine transform module 209.
  • the sampling frequency is scalable, the high frequency PQF band can be ignored by the gain control module 210 to obtain a low sampling rate signal.
  • the Dolby AC_3 encoder Similar to MPEG AAC, the Dolby AC_3 encoder also uses inter-channel intensity combining to improve multi-channel signal encoding efficiency.
  • the international patent application with the international application number PCT/IB02/01595 proposes to quantize a plurality of channels when encoding an audio signal of more than one channel.
  • the coefficients use an integer discrete cosine transform (INT DCT) method to remove inter-channel redundancy.
  • INT DCT integer discrete cosine transform
  • This method is proposed for the shortcomings of the current multi-channel coding method, but does not solve the problem of two-channel stereo coding efficiency.
  • the method of integer discrete cosine transform employed in the method of the patent application is not an optimal solution for the redundancy removal between quantized coefficients (considering the time variation of the source). At the same time, this method also inevitably increases the computational complexity of encoding and decoding. Summary of the invention
  • the object of the present invention is to provide a method and apparatus for encoding and decoding using inter-channel redundancy in order to solve the shortcomings of the prior art, to solve stereo in any stereo and multi-channel audio codec in the prior art. Codec low efficiency and poor quality.
  • the present invention provides a method for encoding using inter-channel redundancy, including the following steps:
  • Step 1 Transform a linear PCM (Pulse Code Modulation) signal into the frequency domain, and calculate a masking threshold of the scale factor band;
  • PCM Pulse Code Modulation
  • Step 2 The frequency domain coefficients of the region are quantized according to the masking threshold of the scale factor band, and the integer coefficients of each channel are obtained;
  • Step 3 Organizing the integer coefficients according to the principle of maximizing the coding gain, and obtaining channel pairs/groups of specific regions of time-frequency;
  • Step 4 Perform matrix transformation on the quantized integer coefficients of the channel, and perform the entropy coding and code stream multiplexing output on the transformed channel pair/group integer coefficients.
  • step 4 performing matrix transformation on the quantized integer coefficients of the channel adopts an optimal transform mode, where the optimal transform mode is a determined number of integer transforms, KLT transforms, and KLTs.
  • the optimal transform mode is a determined number of integer transforms, KLT transforms, and KLTs.
  • the approximate transformation a transform having the largest coding gain is selected for encoding the quantized integer coefficients of the determined region.
  • the present invention also provides an apparatus for encoding using inter-channel redundancy, comprising a psychoacoustic module, a modified discrete cosine transform module, a quantizer, an entropy coding and a code stream multiplexing module, and a matrix transformation module, wherein the matrix The transform module is configured to organize the integer coefficients of each channel output from the quantizer according to the principle of maximizing the coding gain, and obtain channel pairs/groups of specific regions of time-frequency, for the channels Performing matrix transformation on the quantized integer coefficients of the group/group, and outputting the transformed channel pair/group integer coefficients to the entropy coding and code stream multiplexing module; the psychoacoustic module is configured to calculate the current according to the auditory characteristics of the human ear a masking curve of the frame signal, calculating a masking threshold of the specific time-frequency region according to the masking curve, for guiding quantization of the current frame signal; and the modified discrete cosine transform module for linear PCM (Pulse Code
  • the present invention further provides a method for decoding using inter-channel redundancy, comprising the following steps: Step 1. Perform inverse matrix transformation on integer coefficients of code stream demultiplexing and entropy decoding to obtain an integer quantization coefficient;
  • Step 2 Perform inverse quantization processing on the integer quantized coefficients to recover the frequency domain coefficients
  • Step 3 Perform inverse inverse cosine transform on the frequency domain coefficients to obtain a linear PCM signal.
  • the inverse matrix transformation in the step 1 adopts an optimal transformation mode, and the optimal transformation mode is in the side information in a certain number of integer transformation modes, KLT transformation modes, and KLT approximate transformation modes.
  • the present invention also provides an apparatus for decoding using inter-channel redundancy, which includes a code stream demultiplexing and entropy decoding module, an inverse quantizer, an inverse modified discrete cosine transform module, and an inverse matrix transform module, where
  • the inverse matrix transform module is configured to perform inverse matrix transform on integer coefficients output from the code stream demultiplexing and entropy decoding module to obtain integer quantized coefficients;
  • the code stream demultiplexing and entropy decoding module is configured to input Compressed bit stream demultiplexing and entropy decoding to obtain integer coefficients;
  • the inverse quantizer is configured to inverse quantize the integer quantized coefficients output from the inverse matrix transform module to recover frequency domain coefficients;
  • the inverse modified discrete cosine Transform module for frequency domain output from inverse quantizer
  • the coefficients are inversely modified by discrete cosine transform to obtain a linear PCM signal.
  • the invention adopts an optimal transform method in encoding and decoding, that is, can perform lossless de-duplication processing on the quantized multi-channel coefficients; and can be used for lossless two-channel and multi-channel encoding (Loss less Stereo and Mul t ichannel Audio Coding ) »
  • lossy coding for transformed (such as MDCT transform, QMF subband filtering and wavelet transform, etc.), frequency domain processing (such as predictive coding, noise shaping and differential stereo coding) and quantization
  • the post-spectrum coefficients including the transform coefficients and the filtered sub-band signals
  • the present invention can also be used to remove channel signals (such as time domain PCM samples, sub-band samples). Statistical redundancy between the frequency domain coefficients and the frequency domain.
  • the stereo codec efficiency and quality are improved for any stereo and multi-channel audio codec.
  • FIG. 1 is a schematic block diagram of an MPEG-2 AAC encoder in the prior art
  • FIG. 2 is a schematic block diagram of an MPEG-2 AAC decoder in the prior art
  • Figure 3 is a schematic block diagram of an encoder of the present invention.
  • FIG. 4 is a schematic block diagram of a decoder of the present invention. detailed description
  • a method of encoding using inter-channel redundancy includes the following steps:
  • Step 1 Transform the linear PCM signal into the frequency domain, and calculate the masking threshold of the scale factor band.
  • Step 2. Quantize the frequency domain coefficients of the region according to the masking threshold of the scale factor band, and obtain the integer coefficients of each channel;
  • Step 4 Perform matrix transformation on the channel pair/group quantized integer coefficients, and perform the entropy coding and code stream multiplexing output on the transformed channel pair/group integer coefficients.
  • the channel coefficients (including the time domain, the frequency domain, and the sub-bands processed by the present invention, whether it is lossy coding or lossless coding, for convenience of description, the following processing time domain samples, sub-band samples, and frequency domain coefficients are collectively referred to as "Coefficients".) are all integers and are treated in much the same way. Therefore, in the following description, “lossy coding” and “lossless coding” are not distinguished.
  • FIG. 3 A block diagram of a device for encoding with inter-channel redundancy is shown in FIG. 3.
  • the linear PCM signals are input to a modified discrete cosine transform module 301 and a psychoacoustic model 305, respectively, and the modified discrete cosine transform module 301 converts the PCM signals into the frequency domain.
  • the modified discrete cosine transform window function and block length can be switched according to signal characteristics to ensure sufficient time-frequency resolution and effectively remove intra-channel time domain redundancy.
  • the psychoacoustic model 305 is used to calculate a masking curve of the current frame signal according to the auditory characteristics of the human ear, and the masking threshold of the specific time-frequency region can be calculated according to the masking curve for guiding the quantization of the current frame signal.
  • the frequency domain coefficients obtained by the modified discrete cosine transform module 301 are sent to the quantizer 302.
  • the quantizer is composed of a set of sub-quantizers, and each sub-quantizer separately quantizes the mask according to a masking threshold of a specific time-frequency region.
  • the frequency domain coefficient of a region which is usually referred to as a scale factor band.
  • the quantizer has a bit allocation mechanism that controls the number of bits that each sub-quantizer can utilize, such that the number of bits taken to quantize the frequency domain coefficients of the current frame does not exceed the allowed bit limit and minimizes quantization distortion.
  • the bit allocation strategy described herein can employ general common strategies, such as the rate control method of MPEG AAC.
  • the quantizer described here can use a scalar quantizer and a vector quantizer, such as MPEG AAC. Linear scalar quantizer, and vector quantizer for MPEG TwinVQ.
  • the integer coefficients are sent to the matrix transformation module 303.
  • the matrix conversion module 303 organizes the quantized integer coefficients of the respective channels obtained by the quantization according to the principle of maximizing the coding gain to obtain channel pairs/groups of the specific regions of the time-frequency. Also, the channel pair/group of different time-frequency regions (time segments for time domain samples, frequency segments for frequency domain coefficients, and time-frequency regions for sub-band samples) may be different.
  • the correlation between the left channel (L) and the right channel (R) is high, as well as the left surround channel (LS) and the right surround channel (RS). The correlation is high, and the L/R pair and the LS/RS pair are often obtained.
  • the channel-to-organization information needs to be encoded as control information.
  • the following channel groups often appear when organized according to the channel group: Left channel/Right channel/Center channel, Left front channel/Right front channel/Left center channel/Right center channel/Central sound Road, left surround/right surround/back surround, and more.
  • the so-called optimal transform means that one of the determined number of integer transforms, KLT transforms, and any transforms used for the approximate LT transform is selected, and the coding gain is maximum.
  • the LIFTING algorithm is used to transform the integer coefficient to the integer coefficient.
  • the so-called maximum coding gain means that the number of bits used is the least when encoding a specific signal at a specific quality.
  • the integer transform refers to a transform in which each coefficient of the transform matrix is an integer, and there is an inverse matrix (each coefficient is an integer) such that I is a unit matrix.
  • each coefficient is an integer
  • I is a unit matrix.
  • Z and ? to represent the two channel integer coefficients of the channel pair (this , and 7? represent any channel that may appear in the encoding, and should not be interpreted as merely "left channel, and "right channel”)
  • £ and the quantized integer coefficient, ⁇ and integer transform The resulting integer coefficients, for each channel pair, are integer-scaled for the channel-to-integer coefficients within a certain resolution scale (eg, using the so-called "scale factor band”):
  • the number of bits used for encoding is less than the number of bits used for ⁇ ' encoding.
  • KLT transform refers to a signal adaptation matrix whose row vector is the eigenvector of the multi-channel coefficient covariance matrix. Since the KLT transformation matrix is an orthogonal matrix, it can be decomposed into a GIVENS matrix and approximated by the LIFTING algorithm, and an integer result can be obtained.
  • the covariance matrix ⁇ of the signal is calculated according to the time domain signal.
  • the calculation methods of covariance matrix ⁇ and orthogonal matrix Q are introduced in signal processing and linear algebra books, such as "Digital Signal Processing: Theory, Algorithm and Implementation", Tsinghua University Press, edited by Hu Guangshu, 1997.
  • the KLT transform needs to be approximated using the so-called LIFTING algorithm.
  • the LIFTING algorithm described herein can be referred to related documents such as "Factor ing Wavelet Transforms into Lifting Steps" (I. Daubechies, W. Sweldens, Tech. Rep., Bel l Labora tories, Lucent Technologies, 1996).
  • Orthogonal matrix 0 happens to be a GIVENS rotation matrix, so it can be decomposed into the following form According to the LIFTING algorithm, after each transformation, the coefficients can be rounded and do not affect the complete reversibility of the system.
  • the KLT transformation matrix and the LIFTING algorithm are similar to the channel pair method.
  • the approximate transformation of the KLT transformation refers to the transformation method used to approximate the KLT transformation under certain premise (such as source statistical properties and computational complexity). Since the KLT transform is the optimal transform in the sense of mean square error, the calculation amount and sideband information are large. Therefore, other transform methods can be used to approximate the KLT transform to reduce the computational amount and/or sideband information, such as DFT (Discrete Fourier). Transform), DCT (discrete Cosine transform), DST (discrete sine transform), etc.
  • DFT Discrete Fourier
  • DCT discretrete Cosine transform
  • DST discretrete sine transform
  • the so-called optimal transformation means that in a certain number of integer transformations, KLT transformations (LIFTING implementation) and KLT approximate transformations (LIFTING implementation), the transformation with the largest coding gain is selected for encoding the determined region.
  • the matrix transformation module includes a determined number of integer transform units, KLT transform units, and KLT approximate transform units, and the matrix transformation manner includes selecting a certain number of integer transform modes, KLT transform modes, and KLT approximate transforms. Ways (such as DFT, DCT, DST, etc.). For example, you can select M integer conversion methods, set the code to 4, ⁇ 2, which is not less than
  • each channel pair it can be handled as follows to reduce the number of bits required for encoding.
  • the code numbers are ⁇ , . 4 and A are two integer transformation methods, which is the KLT transformation method. among them
  • the value of 0 is as shown in equations (4) and (5).
  • the channel The integer coefficient after the quantization is not processed; when the transform 4 is adopted, the quantized integer coefficient of the first channel of the channel pair is unchanged, and the integer coefficient of the second channel obtained by the transform is the original first
  • the quantized integer coefficients of the channels are reduced by the difference of the quantized integer coefficients of the second channel; when the transform A is used, the KLT transform is used to achieve redundancy cancellation between the channel coefficients, in addition to coding transformation In addition to the code of the way, it is also necessary to encode (9 values).
  • the decision switch 306 employing the transformation matrix may be used to select an optimal transformation mode among a determined number of integer transformation units or KLT transformation units in the matrix transformation module or an approximate transformation unit of the KLT, and the selected optimal transformation mode
  • the code number is encoded as side information.
  • the matrix transformation type adopted may be selected according to the scale factor band, and the selected matrix is transformed.
  • the serial number is encoded.
  • the transformation mode A is adopted, that is, the channel internal coefficient is not changed.
  • O and O the integer transformation method is used. In other cases, the transformation method 4 is used.
  • the selected transform mode A, A or ⁇ is written as a side information into the compressed bit stream to control the decoder to accurately decode.
  • the integer coefficients are sent to the entropy coding and code stream multiplexing module 304.
  • the statistical redundancy of the integer coefficients can be removed by the effective entropy coding, and then the entropy coding result is multiplexed with the other control information into the compressed bit stream, and output to the transmission.
  • the entropy coding may employ an encoding method such as Huffman coding, run length coding, and arithmetic coding.
  • the present invention also discloses a decoding method and apparatus using inter-channel redundancy, as shown in FIG. 4, including a code stream demultiplexing and decoding module, an inverse matrix transform module, an inverse quantizer, and an inverse.
  • a modified discrete cosine transform module the method comprising the following steps:
  • Step 1 The compressed bit stream is demultiplexed and entropy decoded by the code stream demultiplexing and entropy decoding module. To the integer coefficient and the edge information used to determine which inverse matrix transformation method is used; Step 2, the integer coefficient is inverse matrix transformed by the inverse matrix transformation module to obtain an integer quantization coefficient after the inverse matrix transformation;
  • Step 3 The integer quantization coefficient transformed by the inverse matrix is inverse quantized by the inverse quantizer to recover the frequency domain coefficient;
  • Step 4 The frequency domain coefficient is subjected to inverse modified discrete cosine transform by an inverse modified discrete cosine transform module to obtain a linear PCM signal.
  • the inverse matrix transformation in the step 2 is determined by the conversion mode code in the side information obtained from the step 1, which one of the above conversion methods is employed.
  • the integer transform may be used to restore the integer quantized coefficients.
  • Step la obtaining a covariance matrix or corresponding parameters from the code stream (such as step lb in equation (4), calculating a KLT transformation matrix according to the covariance matrix or corresponding parameters;
  • Step lc For the LT transformation matrix, use the LIFTING algorithm to restore the channel-to-integer quantization coefficient.
  • the integer coefficients and the side information for determining which inverse matrix transform method is used are obtained, and the integer coefficients are sent to the inverse matrix transform module 402.
  • the matrix transformation of the three matrix transformation modes of equation (6) is performed, the corresponding inverse matrix is transformed into
  • the inverse matrix transform module 402 selects which inverse matrix transform method is used to recover the integer quantized coefficients at the time of encoding based on the side information obtained from 401.
  • the integer quantized coefficients obtained by the inverse matrix transform are sent to the inverse quantization module 403 for inverse quantization processing.
  • the recovered frequency domain coefficients are fed to an inverse modified discrete cosine transform 404 to obtain a linear PCM audio signal.
  • the inverse matrix transform module includes an integer transform unit, a KLT transform unit, and an approximate transform unit of the KLT, wherein the matrix transform code in the side information is used to select which inverse matrix transform method is used to demultiplex the code stream and
  • the integer coefficients output by the entropy decoding module are inverse matrix transformed, and the transformed integer quantized coefficients are output to an inverse quantizer.

Abstract

Cette invention concerne un procédé et un dispositif de codage/décodage utilisant la redondance des signaux intra-canal, ce procédé consistant: à transformer les signaux PCM linéaires dans le domaine fréquentiel par transformée en cosinus discrète modifiée (MDCT), et à calculer les seuils de masquage des bandes des facteurs d'échelle au moyen du module psycho-acoustique; à quantifier les coefficients du domaine fréquentiel dans la zone au moyen du quantificateur sur la base des seuils de masquage des bandes des facteurs d'échelle, afin d'obtenir les coefficients intégraux de chaque canal; à transformer la matrice de ces coefficients intégraux de chaque canal au moyen du module de transformation de matrice, et à émettre les coefficients intégraux ainsi transformés des paires de canaux au moyen du codeur d'entropie et du multiplexeur de trains de codes. Cette invention concerne également le dispositif correspondant à ce procédé de codage, et les procédé et dispositifs de décodage correspondant à ces procédés et dispositifs de codage. Grâce à cette invention, on améliore l'efficacité de codage des signaux audio pour le codage avec perte, et on supprime la redondance statistique des signaux intra-canal pour le codage sans perte, à des fins de compression des signaux. On peut également améliorer l'efficacité et la qualité de codage/décodage pour tout codeur/décodeur audio stéréo et pour tout codeur/décodeur audio multicanal.
PCT/CN2004/001349 2004-11-24 2004-11-24 Procede et dispositif de codage/decodage utilisant la redondance des signaux intra-canal WO2006056100A1 (fr)

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