EP1989706B1 - Dispositif de ponderation perceptuelle en codage/decodage audio - Google Patents

Dispositif de ponderation perceptuelle en codage/decodage audio Download PDF

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Publication number
EP1989706B1
EP1989706B1 EP07731586A EP07731586A EP1989706B1 EP 1989706 B1 EP1989706 B1 EP 1989706B1 EP 07731586 A EP07731586 A EP 07731586A EP 07731586 A EP07731586 A EP 07731586A EP 1989706 B1 EP1989706 B1 EP 1989706B1
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Prior art keywords
perceptual weighting
band
filter
gain compensation
signal
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EP07731586A
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German (de)
English (en)
French (fr)
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EP1989706A2 (fr
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Stéphane RAGOT
Romain Trilling
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Orange SA
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France Telecom SA
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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
    • G10L19/04—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
    • 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
    • G10L19/0204—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 using subband decomposition
    • G10L19/0208—Subband vocoders
    • 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
    • G10L19/08—Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters
    • G10L19/12—Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters the excitation function being a code excitation, e.g. in code excited linear prediction [CELP] vocoders
    • 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

Definitions

  • the present invention relates to a perceptual weighting device for encoding / decoding an audio signal in a given frequency band. It also relates to a hierarchical audio encoder and decoder comprising a coding / decoding device according to the invention.
  • the invention finds a particularly advantageous application in the field of transmission and storage of digital signals, such as audio-frequency signals of speech, music, etc.
  • the invention is more particularly directed to coding methods of the "Predictive Transform Coding" type incorporating CELP coding and transform coding techniques.
  • the coder generates a fixed rate bit stream.
  • This fixed rate constraint simplifies the implementation and use of the encoder and decoder, commonly referred to together as the "codec". Examples of such systems are: ITU-T G.711 coding at 64 kbit / s, ITU-T G.729 coding at 8 kbit / s or GSM-EFR at 12.2 kbit / s.
  • the invention is of interest here more particularly to hierarchical coding.
  • the bit stream comprises a base layer, or core, and one or more enhancement layers.
  • the base layer is generated by a fixed low rate codec, termed a "core codec", guaranteeing the minimum quality of the coding; this layer must be received by the decoder to maintain an acceptable level of quality.
  • Improvement layers are used to improve the quality; it may happen that they are not all received by the decoder.
  • the main advantage of hierarchical coding is that it allows an adaptation of the bit rate by simple truncation of the bit stream.
  • the number of layers namely the number of possible truncations of the bitstream, defines the granularity of the coding: speaks of coding with high granularity if the bit stream comprises few layers (of the order of 2 to 4), while a fine granular coding for example allows a step of the order of 1 kbit / s.
  • the invention relates to scalable bandwidth and bandwidth encoding techniques with a CELP heart-coder in a telephone band and one or more band-enhanced enhancement layer with respect to the actual telephone band.
  • Examples of such systems are given in the article by H. Taddei et al, Scalable Three Bitrate (8, 14.2 and 24 kbit / s) Audio Coder; 107th Convention AES, 199, with a high granularity of 8, 14.2 and 24 kbit / s, and with fine granularity of 6.4 to 32 kbit / s in the article by B. Kovesi et al supra.
  • G.729EV EV for Embedded Variable Bitrate
  • the objective of the G.729EV standardization is to obtain a G.729 core hierarchical encoder, producing a signal whose band extends from the narrow band (300-3400 Hz) to the broadband (50-7000 Hz). ) at a rate of 8 to 32 kbit / s for conversational services.
  • This encoder is inherently interoperable with Recommendation G.729, which ensures compatibility with existing VoIP devices.
  • This is a three-layer coding comprising cascaded CELP coding, full band linear predictive coding (LPC) bandwidth, and transform predictive coding.
  • LPC linear predictive coding
  • TDAC Time Domain Aliasing Cancellation
  • MDCT Modified Discrete Cosine Transform
  • the transform predictive coding layer uses a full-band perceptual weighting filter ⁇ WB (z) .
  • perceptual weighting filtering noise is explained in WB's work. Kleijn et al supra. In essence, perceptual weighting filtering is used to shape the coding noise by attenuating the signal at frequencies where its intensity is strong and where the noise can be more easily masked.
  • the most common perceptual weighting filters used in narrow-band CELP coding are of the form ⁇ (z / ⁇ 1 ) / ⁇ (z / ⁇ 2 ) where 0 ⁇ ⁇ 2 ⁇ ⁇ 1 ⁇ 1 and ⁇ (z) represents the LPC spectrum of a signal segment of length 5 to 30 ms.
  • the synthesis analysis in CELP coding thus amounts to minimizing the quadratic error in a signal domain perceptually weighted by this type of filter.
  • the patent application WO 01/73759 discloses a method of reducing noise in an audio signal by defining a subband gain factor, wherein the gain factor is determined to obtain a better signal to noise ratio.
  • gain factors are however not calculated nor adopted to compensate for any spectral continuity between sub - bumps or between frequency repeats.
  • the technical problem to be solved by the object of the present invention is to propose a perceptual weighting device for encoding / decoding an audio signal in a given frequency band, which would make it possible to carry out a full perceptual weighting filtering. band, that is to say over the whole of said given frequency band, in particular the 0-8000 Hz wide band of a hierarchical audio coder, without this operation leading to long and expensive calculations of resources
  • the solution to the technical problem posed is, according to the present invention in that, said coding / decoding being performed in a plurality of adjacent subbands in said given frequency band, said device is as defined in claim 1. a gain compensation that ensures spectral continuity over the entire width of the frequency band. The invention therefore makes it possible to obtain a homogeneous band at the output of the perceptual weighting filtering even if the subbands that constitute it have been treated separately from this point of view.
  • each subband can be filtered or not by perceptual weighting.
  • the spectral continuity can therefore be ensured between a filtered sub-band and another unfiltered, or between two filtered subbands.
  • said gain-compensated perceptual weighting filter comprises a perceptual weighting filter and a gain compensation module.
  • the gain compensation module is disposed at the output of said perceptual weighting filter.
  • the gain compensation module is disposed at the input of said perceptual weighting filter.
  • said perceptual weighting filter with gain compensation comprises a perceptual weighting filter incorporating said gain compensation.
  • said perceptual weighting filter in the first subband is of the form ((z / ⁇ 1 ) / ((z / ⁇ 2 ) where ((z) represents a linear prediction filter.
  • only the first subband is subject to perceptual weighting filtering, the second subband not being filtered.
  • said gain-compensated perceptual weighting filter comprises a perceptual weighting filter in the first sub-band
  • the invention provides that said perceptual weighting filter in the first sub-band is of the form A 1 (z / ⁇ 1 ) l 1 (z / ⁇ 2 ) where ⁇ 1 (z) represents a linear prediction filter.
  • the signal from the perceptual weighting device in the first subband and the original signal in the second sub-bands are respectively applied to transform analysis modules, and said transform analysis modules are connected to a transform encoder in said frequency band.
  • said encoder also comprises a perceptual weighting device of the original signal in the second subband, comprising a perceptual weighting filter with gain compensation able to achieve the spectral continuity. between the output signal of said perceptual weighting filter with gain compensation and the output signal of the perceptual weighting device in the first subband.
  • said perceptual weighting filter with gain compensation comprises a perceptual weighting filter in the second band
  • said perceptual weighting filter in the second subband is of the form ⁇ 2 (z / ⁇ ' 1 ) / ⁇ 2 (z / ⁇ ' 2 ) where ⁇ 2 (z) represents a linear prediction filter.
  • the coefficients of said linear prediction filter are provided by a band extension module.
  • the signal from the perceptual weighting device in the first subband and the signal from the perceptual weighting device in the second subband are respectively applied to transform analysis modules, and said analysis modules to transformed are connected to a transform encoder in said frequency band.
  • the core coder is a linear prediction based coder, for example a CELP coder.
  • said inverse perceptual weighting device comprises a perceptual weighting filter with gain compensation, inverse of the perceptual weighting filter with gain compensation of the encoder in the first subband.
  • said decoder also comprises an inverse perceptual weighting device of the decoded signal in the second subband, comprising a perceptual weighting filter with gain compensation, inverse of the perceptual weighting filter with gain compensation of the encoder in the second subband.
  • said gain-compensated perceptual weighting filter comprises a perceptual weighting filter in the second band
  • said gain-compensated inverse perceptual weighting filter comprises an inverse perceptual weighting filter in the second band. subband.
  • said inverse perceptual weighting filter in the second subband is of the form ⁇ 2 (z / ⁇ ' 2 ) / ⁇ 2 (z / ⁇ ' 1 ).
  • the coefficients of the linear prediction filter ⁇ 2 (z) are provided by a band extension module.
  • the invention further relates to a perceptual weighting method for encoding an audio signal in a given frequency band, wherein said encoding is performed in a plurality of adjacent subbands in said given frequency band, said method comprises, in at least one subband, a step of perceptual weighting with gain compensation adapted to achieve the spectral continuity between the signal from said perceptual weighting step with gain compensation and the signals in the subbands adjacent to said sub-band.
  • the invention relates to a perceptual weighting method for decoding an audio signal encoded in a given frequency band in accordance with the perceptual weighting method for encoding said signal, which is remarkable in that said method comprises band, a perceptual weighting step with gain compensation, inverse of said perceptual weighting step with gain compensation.
  • FIG. 2 On the figure 2 is represented a hierarchical audio coder in subbands at rates ranging from 8 to 32 kbit / s. This figure gives the different steps of the corresponding coding method.
  • the input signal in a 50 to 7000 Hz, so-called “expanded" frequency band, sampled at 16 kHz, is first broken down into 2 adjacent subbands by Quadrature Mirror Filter (QMF). .
  • the first sub-band, or low band, from 0 to 4000 Hz is obtained by low-pass filtering L 300 and decimation 301, and the second sub-band, or high band, from 4000 to 8000 Hz by high-pass filtering H 302 and decimation 303.
  • the filters L 300 and H 302 are of length 64 and conform to those described in the J. Johnston, ICASSP, vol. 5, pp. 291 - 294, 1980 .
  • the first sub-band is pre-processed by a high-pass filter 304 eliminating the components below 50 Hz before encoding by a narrow-band CELP 305 core coder.
  • the high-pass filtering takes into account that the broadband is defined as covering the range 50-7000 Hz.
  • the narrow-band CELP coding corresponds to that described in FIG. figure 1 ; it is a cascaded CELP coding comprising as a first stage a modified G.729 coding (ITU-T G.729 Recommendation, Coding of Speech at 8 kbps using Conjugate Structure Algebraic Code Excited Linear Prediction (CS-ACELP ), March 1996) without a pre-treatment filter, and as a second stage an additional fixed dictionary.
  • CS-ACELP Conjugate Structure Algebraic Code Excited Linear Prediction
  • the residual signal e related to the error due to the CELP coding is calculated by the stage 306 and then perceptually weighted by a device 307 comprising a perceptual weighting filter to obtain the signal x lo in the time domain.
  • This signal is analyzed by Modified Discrete Cosine Transform (MDCT) 308 to obtain the discrete spectrum X lo in the frequency domain.
  • MDCT Modified Discrete Cosine Transform
  • the device 307 for perceptual weighting is shown in FIG. figure 3 .
  • This device W 1 (z) comprises a perceptual weighting filter ⁇ 1 (z / ⁇ 1 ) / ⁇ 1 (z / ⁇ 2 ) comprising the filtering stages 501 and 502 respectively by ⁇ 1 (z / y 1 ) and 1 / ⁇ 1 (z / ⁇ 2 ) .
  • the linear prediction filter ⁇ 1 (z) is derived from narrowband CELP coding.
  • the second subband, or high band is first unfolded spectrally 309 to compensate for the folding due to high pass filter 302 combined with decimation 303.
  • This high band is then pre-processed by a low pass filter 310 eliminating the components between 7000 and 8000 Hz in the original signal.
  • the resulting signal x hi in the time domain is transformed by MDCT 311 to obtain the discrete spectrum X hi in the frequency domain.
  • a band extension 312 is made from x hi and X hi .
  • the MDCT transformation is implemented by means of the algorithm of P. Duhamel, Y. Mahieux, JP Petit, A fast algorithm for the implementation of filter banks on 'time domain aliasing cancellation', ICASSP, vol. 3, pp.2209-2212, 1991 .
  • the low band MDCT and high band X lo and X hi spectra are encoded in the transform coding module 313.
  • the different bit streams generated by the coding modules 305, 312 and 313 are multiplexed and structured into a hierarchical bit stream in the multiplexer 314.
  • the coding is carried out in blocks of samples (or frames) of 20 ms, ie 320 samples.
  • the coding rate is 8, 12, 14 to 32 kbit / s.
  • the hierarchical audio decoder associated with the encoder which has just been described with regard to the figures 2 , 3 and 4 is represented at the figure 5 .
  • This figure illustrates the steps of decoding the signal encoded by said encoder.
  • the bits describing each frame of 20 ms are demultiplexed in the demultiplexer 700.
  • a decoding operation of 8 to 32 kbit / s is presented, although in practice the bit stream can be truncated to 8, 12, 14 or between 14 and 32 kbit / s.
  • the bit stream of the 8 and 12 kbit / s layers is used by the CELP decoder 701 to generate a first synthesis in the first subband, or narrow band, between 0 and 4000 Hz.
  • the portion of the bit stream associated with the layer at 14 kbit / s is decoded by the band extension module 702 and the signal obtained in the second subband, or high band, between 4000 and 7000 Hz is transformed by MDCT 703 into a spectrum X hi .
  • the MDCT decode 704 generates from the bit stream associated with the bit rates of 14 to 32 kbit / s a reconstructed spectrum X lo in low band and a reconstructed spectrum X hi in high band.
  • the extended band output signal is obtained via a bank of QMF synthesis filters which perform the oversampling operations 710 and 712, low-pass filtering 711 and high-pass filtering. 713 and addition 714.
  • the coefficients i i are held constant in each 5 ms subframe.
  • a variant of the embodiment of the coder of the figure 2 is represented on the figure 6 .
  • LPC Linear Prediction
  • the gain compensation in low and high bands by the factors fac 1 and fac 2 respectively ensure a continuity of the responses of the filters at 4 kHz. It is this continuity that then makes it possible to code the two discrete spectra X lo and X hi into a single vector X. Again, it is important to note that the value 0 dB used here to define the continuity between low and high bands n is indicative.
  • the hierarchical audio decoder corresponding to this variant is described in figure 7 .
  • the only difference consists in recovering the quantized LPC coefficients ⁇ 2 (z) used by the band extension module 1002 and applying a perceptual weighting filter.
  • the inverse filter W 2 (z) -1 in the high band is of type ⁇ 2 (z / ⁇ ' 2 ) / ⁇ 2 (z / ⁇ ' 1 ) followed by the gain compensation factor 1 / fac 2 where fac 2 has been defined above.
  • the invention furthermore covers a computer program comprising a sequence of instructions stored on a medium for execution by a computer or a dedicated device, which is remarkable in that, during the execution of these instructions, the latter executes the method of perceptual weighting object of the invention for coding and / or decoding.
  • the aforementioned computer program is for example a directly executable program implanted in a perceptual weighting device object of the invention.

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  • Engineering & Computer Science (AREA)
  • Physics & Mathematics (AREA)
  • Computational Linguistics (AREA)
  • Signal Processing (AREA)
  • Health & Medical Sciences (AREA)
  • Audiology, Speech & Language Pathology (AREA)
  • Human Computer Interaction (AREA)
  • Acoustics & Sound (AREA)
  • Multimedia (AREA)
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  • Spectroscopy & Molecular Physics (AREA)
  • Compression, Expansion, Code Conversion, And Decoders (AREA)
EP07731586A 2006-02-14 2007-02-07 Dispositif de ponderation perceptuelle en codage/decodage audio Not-in-force EP1989706B1 (fr)

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FR0650538 2006-02-14
PCT/FR2007/050760 WO2007093726A2 (fr) 2006-02-14 2007-02-07 Dispositif de ponderation perceptuelle en codage/decodage audio

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EP1989706B1 true EP1989706B1 (fr) 2011-10-26

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US (1) US8260620B2 (ja)
EP (1) EP1989706B1 (ja)
JP (1) JP5117407B2 (ja)
KR (1) KR101366124B1 (ja)
CN (1) CN101385079B (ja)
AT (1) ATE531037T1 (ja)
WO (1) WO2007093726A2 (ja)

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EP1989706A2 (fr) 2008-11-12
WO2007093726A3 (fr) 2007-10-18
KR101366124B1 (ko) 2014-02-21
JP5117407B2 (ja) 2013-01-16
KR20080093450A (ko) 2008-10-21
ATE531037T1 (de) 2011-11-15
CN101385079A (zh) 2009-03-11
CN101385079B (zh) 2012-08-29
JP2009527017A (ja) 2009-07-23
US8260620B2 (en) 2012-09-04
US20090076829A1 (en) 2009-03-19
WO2007093726A2 (fr) 2007-08-23

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