EP2115741B1 - Fortgeschrittene kodierung/dekodierung von digitalen tonsignalen - Google Patents

Fortgeschrittene kodierung/dekodierung von digitalen tonsignalen Download PDF

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EP2115741B1
EP2115741B1 EP08762010A EP08762010A EP2115741B1 EP 2115741 B1 EP2115741 B1 EP 2115741B1 EP 08762010 A EP08762010 A EP 08762010A EP 08762010 A EP08762010 A EP 08762010A EP 2115741 B1 EP2115741 B1 EP 2115741B1
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subband
signal
band
coding
masking threshold
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EP2115741A1 (de
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Stéphane RAGOT
Cyril Guillaume
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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
    • 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
    • G10L19/02—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
    • 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
    • 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
    • G10L19/002—Dynamic bit allocation
    • 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
    • G10L19/0212—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 orthogonal transformation
    • 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
    • G10L19/038—Vector quantisation, e.g. TwinVQ audio
    • 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 sound data processing.
  • This processing is adapted in particular to the transmission and / or storage of digital signals such as audio-frequency signals (speech, music, or other).
  • the LTP long-term prediction parameters including the pitch period, represent the fundamental vibration of the speech signal (when it is voiced), while the LPC short-term prediction parameters represent the spectral envelope. of this signal.
  • all of these LPC and LTP parameters thus resulting from a speech coding, can be transmitted in blocks to a peer decoder, via one or more telecommunication networks, to then restore the initial speech signal.
  • the encoder In conventional speech coding, the encoder generates a fixed rate bit stream. This flow constraint simplifies the implementation and use of the encoder and the decoder. Examples of such systems are the ITU-T G.711 64 kbit / s standard coding, the 8 kbit / s ITU-T G.729 coding or the 12.2 kbit / s GSM-EFR coding.
  • variable rate bit stream In some applications (such as mobile telephony or VoIP for "Internet Protocol"), it is best to generate a variable rate bit stream. Flow values are taken in a predefined set. Such a coding technique, called “multi-rate” is therefore more flexible than a fixed rate coding technique.
  • Hierarchical coding having the capacity to provide varied bit rates, is described below by distributing the information relating to an audio signal to be coded in hierarchical subsets, so that this information can be used in order of importance. in terms of audio rendering quality.
  • the criterion taken into account for determining the order is a criterion for optimizing (or rather reducing) the quality of the coded audio signal.
  • Hierarchical coding is particularly suited to transmission over heterogeneous networks or having variable available rates over time, or to transmission to terminals with varying capacities.
  • Hierarchical audio coding (or " scalable ) can be described as follows.
  • the bit stream includes a base layer and one or more enhancement layers.
  • the base layer is generated by a low-bandwidth codec (fixed), called a "codec heart", guaranteeing the minimum coding quality.
  • This layer must be received by the decoder to maintain an acceptable level of quality. Improvement layers are used to improve quality. However, they may not all be received by the decoder.
  • the main advantage of hierarchical coding is that it allows an adaptation of the bit rate simply by " truncation of the bit stream ".
  • the number of layers i.e., the number of possible truncations of the bitstream
  • the number of layers defines the granularity of the coding.
  • scalable scalability and bandwidth encoding techniques with a CELP heart-coder, a telephone band, and one or more broadband enhancement layer (s).
  • An example of such systems is given in the ITU-T G.729.1 8-32 kbit / s fine grain standard.
  • the G.729.1 coding / decoding algorithm is summarized below.
  • the G.729.1 encoder is an extension of the ITU-T G.729 coder. It is a modified G.729 heart-coded core encoder producing a bandwidth ranging from narrowband (50-4000 Hz) to wideband (50-7000 Hz) at a rate of 8 to 32 kbit / s for conversational services. This codec is compatible with existing VoIP devices (most of which are equipped according to G.729). Finally, it should be noted that G.729.1 was approved in May 2006.
  • the G.729.1 coder is schematized on the figure 1 .
  • the broadband input signal s wb sampled at 16 kHz, is first decomposed into two subbands by QMF (for "Quadrature Mirror Filter") filtering.
  • the low band (0-4000 Hz) is obtained by LP low-pass filtering (block 100) and decimation (block 101), and the high band (4000-8000 Hz) by HP high-pass filtering (block 102) and decimation (block 103).
  • the LP and HP filters are of length 64.
  • the low band is pretreated with a high-pass filter eliminating the components below 50 Hz (block 104), to obtain the signal S LB , before CELP coding in narrow band (block 105) at 8 and 12 kbit / s.
  • This high-pass filtering takes into account that the Useful band is defined as covering the range 50-7000 Hz.
  • the narrow-band CELP coding is a cascaded CELP coding comprising as a first stage a modified G.729 coding without pre-processing filter and as a second stage an additional fixed CELP dictionary.
  • the high band is first pretreated (block 106) to compensate for the folding due to the high-pass filter (block 102) combined with the decimation (block 103).
  • the high band is then filtered by a low pass filter (block 107) eliminating the components between 3000 and 4000 Hz from the high band (that is, the components between 7000 and 8000 Hz in the original signal) to obtain the signal S HB .
  • a band extension (block 108) is then performed.
  • the error signal d LB of the low band is calculated (block 109) from the output of the CELP coder (block 105) and a predictive coding by transform (for example of the TDAC type for "Time Domain Aliasing Cancellation" in the G.729.1) is carried out at block 110.
  • a predictive coding by transform for example of the TDAC type for "Time Domain Aliasing Cancellation" in the G.729.1
  • Additional parameters can be transmitted by the block 111 to a homologous decoder, this block 111 performing a so-called “FEC” treatment for "Frame Erasure Concealment", in order to reconstitute possible erased frames.
  • the different bitstreams generated by the coding blocks 105, 108, 110 and 111 are finally multiplexed and structured into a hierarchical bit stream in the multiplexing block 112.
  • the coding is performed by 20 ms sample blocks (or frames). 320 samples per frame.
  • the G.729.1 homologous decoder is illustrated on the figure 2 .
  • the bits describing each frame of 20 ms are demultiplexed in block 200.
  • the bit stream of the 8 and 12 kbit / s layers is used by the CELP decoder (block 201) to generate the narrow-band synthesis (0-4000 Hz).
  • the portion of the bit stream associated with the 14 kbit / s layer is decoded by the tape extension module (block 202).
  • the portion of the bit stream associated with data rates greater than 14 kbit / s is decoded by the TDAC module (block 203).
  • Pre-echo and post-echo processing is performed by blocks 204 and 207 as well as enrichment (block 205) and aftertreatment of the low band (block 206).
  • the broadband output signal ⁇ wb sampled at 16 kHz, is obtained via the QMF synthesis filter bank (blocks 209, 210, 211, 212 and 213) incorporating the inverse folding (block 208).
  • the TDAC type transform coding in the G.729.1 encoder is illustrated on the figure 3 .
  • This spectrum is divided into eighteen sub-bands, a sub-band j being assigned a number of coefficients noted nb_coef ( j ).
  • the subband splitting is specified in Table 1 below.
  • a subband j comprises the coefficients Y ( k ) with sb_bound ( j ) ⁇ k ⁇ sb_bound ( j + 1).
  • Table 1 TDAC Encoding Boundary Limits and Size J sb_bound ( j ) nb_coef ( j ) 0 0 16 1 16 16 2 32 16 3 48 16 4 64 16 5 80 16 6 96 16 7 112 16 8 128 16 9 144 16 10 160 16 11 176 16 12 192 16 13 208 16 14 224 16 15 240 16 16 256 16 17 272 8 18 280 -
  • the spectral envelope is coded at a variable rate in block 305.
  • This quantized value rms _index ( j ) is transmitted to the bit allocation block 306.
  • the number of bits allocated to each sub-band for its quantization is determined in block 306 from the quantized spectral envelope from block 305.
  • the allocation of the bits performed minimizes the squared error while respecting the constraint of a number of integer bits allocated per subband and a maximum number of bits not to be exceeded.
  • the spectral content of the subbands is then encoded by spherical vector quantization (block 307).
  • the different bit streams generated by the blocks 305 and 307 are then multiplexed and structured into a hierarchical bit stream at the multiplexing block 308.
  • the TDAC type transform decoding step in the G.729.1 decoder is illustrated on the figure 4 .
  • the decoded spectral envelope (block 401) makes it possible to find the allocation of the bits (block 402).
  • each of the subbands is found by inverse spherical vector quantization (block 403).
  • the sub-bands not transmitted, due to a lack of "budget" of bits, are extrapolated (block 404) from the MDCT transform of the signal at the output of the band extension block (block 202 of FIG. figure 2 ).
  • IMDCT inverse MDCT transform
  • W LB inverse perceptual weighting
  • the subbands bit allocation (block 306 of FIG. figure 3 or block 402 of the figure 4 ).
  • nbit ( j ) arg min r ⁇ R nb_coef j ⁇ nb_coef j ⁇ ip j - ⁇ Opt - r where ⁇ opt is a parameter optimized by dichotomy.
  • the TDAC coding uses the perceptual weighting W LB ( z ) filter in the low band (block 300), as indicated above.
  • perceptual weighting filtering allows you to format the coding noise.
  • the principle of this filtering is to exploit the fact that it is possible to inject more noise in the frequency zones where the original signal has a high energy.
  • 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 a prediction spectrum linear (LPC).
  • LPC prediction spectrum linear
  • the fac factor makes it possible to ensure at the junction of the low and high bands (4 kHz) a gain of the filter at 1 to 4 kHz. It is important to note that in the G.729.1 TDAC coding, the coding is based on an energetic criterion only.
  • the joint coding of these two signals is carried out in the MDCT domain according to the criterion of the quadratic error.
  • the high band is coded according to energy criteria, which is suboptimal (in the "perceptual" sense of the term).
  • multi-band coding may be considered, a perceptual weighting filter being applied to the signal of at least one band in the time domain, and the set of subbands being coded together. by transform coding. If we want to apply the perceptual weighting in the frequency domain, then there is the problem of continuity and homogeneity of the spectra between subbands.
  • the present invention improves the situation.
  • the present invention therefore proposes to calculate a frequency perceptual weighting, using a masking threshold, on only a part of the frequency band (at least on the "second subband” mentioned above) and to ensure spectral continuity with at least another frequency band (at least the aforementioned "first sub-band”) by normalizing the masking threshold on the spectrum covering these two frequency bands.
  • the allocation of the bits for the second sub-band at least is determined furthermore according to a standardized masking curve calculation, applied at least to the second sub-band.
  • the application of the invention makes it possible to transmit the bits to the sub-bands that require the most bits according to a perceptual criterion.
  • perceptual frequency weighting is then applied by masking a part of the audio band, so as to improve the audio quality by optimizing in particular the distribution of bits between subbands according to criteria. perceptual.
  • the transformed signal in the second subband is weighted by a factor proportional to the square root of the normalized masking threshold for the second subband.
  • the normalized masking threshold is not used for the allocation of the bits to the subbands as in the first application mode above, but it can advantageously be used to directly weight the signal of the second sub-band at least in the transformed domain.
  • the present invention is advantageously, but not exclusively, applied to a TDAC-type transform coding in a global encoder according to the G.729.1 standard, the first subband being included in a low frequency band, whereas the second subband is included in a low frequency band, while the second subband is included in a low frequency band.
  • -band is included in a high frequency band that can extend up to 7000 Hz, or even more (typically up to 14 kHz) in band extension.
  • the application of the invention may then consist in providing a perceptual weighting for the high band while ensuring spectral continuity with the low band.
  • the signal from the core coding can be perceptually weighted and the implementation of the invention is advantageous in the sense that the entire spectral band can finally be perceptually weighted.
  • the signal from the core coding may be a signal representative of a difference between an original signal and a synthesis of this original signal (called “difference signal” or "error signal”).
  • difference signal or "error signal”
  • a first decoding application mode homologous to the first application mode of the coding defined above, aims at the allocation of bits to the decoding and a number of bits to be allocated to each subband is determined from a decoding spectral envelope.
  • the bit allocation for the at least second subband is further determined according to a normalized masking curve calculation applied at least to the second subband.
  • a second method of applying decoding within the meaning of the invention consists in weighting the transformed signal in the second subband by the square root of the normalized masking threshold. This embodiment will be described in detail with reference to the figure 10B .
  • the invention provides an improvement to the perceptual weighting performed in the transform coder by exploiting the masking effect known as "simultaneous masking" or "frequency masking".
  • This property corresponds to the modification of the hearing threshold in the presence of a so-called “masking” sound. This phenomenon is observed typically when, for example, one tries to hold a discussion with ambient noise, for example in the street and that the noise of a vehicle comes to "hide” the voice of a speaker.
  • an approximate masking threshold is calculated for each spectrum line. This threshold is the one above which the line concerned is supposed to be audible.
  • the masking threshold is calculated from the convolution of the signal spectrum with a spreading function B ( v ) modeling the masking effect of a sound (sinusoid or filtered white noise) by another sound (sinusoid or noise filtered white).
  • the spreading function may be a function of the level of the line and / or the frequency of the masking line. A detection of "peaks" can also be implemented.
  • An application of the invention described hereinafter makes it possible to improve the TDAC coding of the encoder according to the G.729.1 standard, in particular by applying a perceptual weighting of the high band (4 to 7 kHz) while ensuring the continuity spectral between low and high bands for a satisfactory and joint coding of these two bands.
  • the input signal is sampled at 16 kHz, bandwidth 50 Hz to 7 kHz.
  • the encoder always operates at the maximum rate of 32 kbit / s, while the decoder can receive the core (8 kbit / s), as well as one or more enhancement layers (12 to 32 kbit / s per step). 2 kbit / s), as in G.729.1. Coding and decoding have the same architecture as that presented to figures 1 and 2 .
  • blocks 110 and 203 are modified as described in figures 6 and 7 .
  • the modified TDAC coder is identical to that of the figure 3 , except that the allocation of the bits following the squared error (block 306) is now replaced by a masking curve calculation and a modified bit allocation (blocks 606 and 607), the invention forming part of the calculation of the masking curve (block 606) and its use in the allocation of bits (block 607).
  • the modified TDAC decoder is presented on the figure 7 in this first embodiment.
  • This decoder is identical to that of the figure 4 , except that the allocation of the bits following the squared error (block 402) is replaced by a masking curve calculation and a modified bit allocation (blocks 702 and 703).
  • the invention relates to blocks 702 and 703.
  • An advantageous spreading function is that presented to the figure 5 . It is a triangular function whose first slope is + 27dB / Bark and -10dB / Bark for the second.
  • ⁇ 1 ( j ) and ⁇ 2 ( j ) can be pre-calculated and stored.
  • a first embodiment of the invention is described below for the allocation of bits in a hierarchical coder such as the G.729.1 encoder.
  • bit allocation criterion is based here on the signal-to-mask ratio given by: 1 2 ⁇ log 2 ⁇ ⁇ 2 j M j
  • the masking threshold is normalized by its value on the last subband of the low band.
  • the definition of the term ip ( j ) , j 10, ..., 17, is changed.
  • FIG 8 An illustration of the standardization of the masking threshold is given in figure 8 , showing the connection of the high band on which the masking (4-7 kHz) is applied to the low band (0-4 kHz).
  • the standardization of the masking threshold can be rather carried out from the value of the band.
  • these relations giving the normalization factor normfac or the masking threshold M ( j ) can be generalized to any number of sub-bands (different, in total, from eighteen) in high band (with a different number of eight), as in low band (with a different number of ten).
  • the normalized masking threshold is not used to weight the energy in the definition of the perceptual importance, as in the first embodiment described above, but it serves to directly weight the high band signal before TDAC coding.
  • This second embodiment is illustrated on the Figures 9A (for encoding) and 10A (for decoding).
  • a variant of this second mode, which is the object of the present invention, in particular for the decoding performed, is illustrated on the Figures 9B (for encoding) and 10B (for decoding).
  • the masking threshold is calculated (block 905 of the Figure 9A and block 906b of the Figure 9B ) from the unquantized spectral envelope.
  • the peer decoder is shown in the figure 10A .
  • the block 1002 is then performed as described in Ragot et al. supra.
  • Block 1004 also performs a function similar to that of block 405 of the figure 4 .
  • This second embodiment may be particularly advantageous, particularly in an implementation according to the 3GPP-AMR-WB + standard, which is the preferred context of the document Ragot et al. supra.
  • the coded information remains the envelope of energy (rather than the masking threshold itself as on the Figures 9A and 10A ).
  • the masking threshold is calculated and normalized (block 906b of the Figure 9B ) from the coded spectral envelope (block 905b).
  • the masking threshold is calculated and standardized (block 1011b of the figure 10B ) from the decoded spectral envelope (block 1001b), the decoding of the envelope making it possible to perform a level adjustment (block 1010b of the figure 10B ) from the quantized values rms_q ( j ).
  • a masking threshold is calculated for each sub-band, at least for the sub-bands of the high frequency band, this masking threshold being normalized to ensure spectral continuity between the subbands concerned.
  • the calculation of the masking threshold is particularly advantageous when the signal to be coded is not tonal, in the first mode, as in the second embodiment, described above.
  • the application of the spreading function B (v) results in a masking threshold very close to a tone a little more spread out in frequencies.
  • the allocation criterion minimizing the masked coding noise ratio then gives a bit of bit allocation.
  • the same is true for the direct weighting of the high band signal according to the second embodiment. It is therefore preferred, for a tonal signal, to use a bit allocation according to energy criteria.
  • the invention is applied only if the signal to be encoded is not tonal.
  • the bit relating to the mode of the coding of the spectral envelope indicates a "differential Huffman" mode or a "natural direct binary” mode.
  • This mode bit can be interpreted as a tone detection, since, in general, a tonal signal leads to envelope coding by the "natural direct binary” mode, while most non-tonal signals, having a spectral dynamic more limited, lead to envelope coding by the "Differential Huffman" mode.
  • the module 904 of the Figure 9A can, by calculating the spectral envelope, determine whether the signal is tonal or not and so Block 905 is bypassed if yes.
  • the module 904 can make it possible to determine whether the signal is tonal or not and so bypass the block 907 in the affirmative.
  • the figure 11 generalize the normalization of the masking curve (described in figure 8 ) in the case of super wide band coding.
  • the signals in this embodiment are sampled at a frequency of 32 kHz (instead of 16 kHz) for a useful band of 50 Hz - 14 kHz.
  • the masking curve log 2 [ M ( j )] is then defined at least for the sub-bands ranging from 7 to 14 kHz.
  • the spectrum covering the band 50 Hz - 14 kHz is coded by subbands and the allocation of bits to each subband is made from the spectral envelope as in the G.729.1 encoder.
  • a partial masking threshold can be calculated as previously described.
  • the standardization of the masking threshold as illustrated on the figure 11 , so also generalizes to the case where the high band has more subbands or covers a wider frequency area than in G.729.1.
  • a first T1 transform is applied to the time weighted difference signal.
  • a second transform T2 is applied to the signal on the first high band between 4 and 7 kHz and a third transform T3 is applied to the signal on the second high band between 7 and 14 kHz.
  • the invention is not limited to signals sampled at 16kHz. Its implementation is particularly advantageous also for signals sampled at higher frequencies, such as for the extension of the G.729.1 encoder to signals sampled not at 16 kHz but at 32 kHz, as described above. If the TDAC coding is generalized to such a frequency band (50 Hz - 14 kHz instead of 50 Hz - 7 kHz currently), the advantage provided by the invention will be really major.
  • the invention also aims to improve the TDAC coding, in particular by applying a perceptual weighting of the high-bandwidth (4-14 kHz) while ensuring the spectral continuity between bands, this criterion being important for a joint coding of the band.
  • first low band and the second high and extended band up to 14 kHz.
  • the perceptual weighting in the low band is not necessary for the application of the invention.
  • the present invention also relates to a first computer program, stored in a memory of an encoder of a telecommunication terminal and / or stored on a memory medium intended to cooperate with a reader of said encoder.
  • This first program then comprises instructions for implementing the coding method defined above, when these instructions are executed by an encoder processor.
  • the present invention also relates to an encoder comprising at least one memory storing this first computer program.
  • FIGS. 6 , 9A and 9B may constitute flowcharts of this first computer program, or further illustrate the structure of such an encoder, according to different embodiments and variants.
  • the present invention also relates to a second computer program, stored in a memory of a decoder of a telecommunication terminal and / or stored on a storage medium intended to cooperate with a reader of said decoder.
  • This second program then comprises instructions for implementing the decoding method defined above, when these instructions are executed by a processor of the decoder.
  • the present invention also relates to a decoder comprising at least one memory storing this second computer program.
  • FIG. 7 , 10A , 10B may constitute flowcharts of this second computer program, or further illustrate the structure of such a decoder, according to different embodiments and variants.

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Claims (19)

  1. Verfahren zum Codieren eines Tonsignals in mehreren Unterbändern, bei dem mindestens ein erstes und ein zweites benachbartes Unterband transformationscodiert werden (601, 602; 901, 902), dadurch gekennzeichnet, dass das Verfahren zur Anwendung einer perzeptuellen Gewichtung im transformierten Bereich auf mindestens das zweite Unterband aufweist:
    - eine Bestimmung mindestens einer Frequenzmaskierungsschwelle (606; 905; 906b), die an das zweite Unterband anzuwenden ist, und
    - eine Normalisierung der Maskierungsschwelle, um eine spektrale Kontinuität zwischen dem ersten und dem zweiten Unterband zu gewährleisten.
  2. Verfahren nach Anspruch 1, bei dem eine Anzahl von jedem Unterband zuzuweisenden Bits ausgehend von einer spektral Hüllkurve bestimmt wird,
    dadurch gekennzeichnet, dass die Zuweisung der Bits (607) für mindestens das zweite Unterband außerdem in Abhängigkeit von einer Berechung einer normalisierten Maskierungskurve bestimmt wird, die mindestens an das zweite Unterband (606) angewendet wird.
  3. Verfahren nach Anspruch 2, bei dem die Codierung an mehr als zwei Unterbändern durchgeführt wird, wobei das erste Unterband in einem ersten Spektralband und das zweite Unterband in einem zweiten Spektralband enthalten ist, dadurch gekennzeichnet, dass die Anzahl von Bits pro Unterband nbit(j) für jedes Unterband mit dem Index j sich in Abhängigkeit von einer perzeptuellen Größe (ip(j) ergibt, die ausgehend von einer Beziehung des folgenden Typs berechnet wird:
    - ip j = 1 2 rms_index j ,
    Figure imgb0047
     wenn j ein Unterband-Index im ersten Band ist, und
    - ip j = 1 2 ⁢ rms_index j - log_mask j ,
    Figure imgb0048
     wenn j ein Unterband-Index im zweiten Band ist, mit log_mask(j) = log2(M(j))-normfac, wobei:
    - rms_index(j) aus der Codierung der Hüllkurve stammende quantisierte Werte für das Unterband j sind,
    - M(j) die Maskierungsschwelle für das Unterband mit dem Index j ist, und
    - normfac ein bestimmter Normalisierungsfaktor ist, um die spektrale Kontinuität zwischen dem ersten und dem zweiten Unterband zu gewährleisten.
  4. Verfahren nach Anspruch 1, dadurch gekennzeichnet, dass das transformierte Signal im zweiten Unterband mit einem Faktor proportional zur Quadratwurzel der normalisierten Maskierungsschwelle für das zweite Unterband gewichtet wird (905).
  5. Verfahren nach Anspruch 4, bei dem die Codierung an mehr als zwei Unterbändern durchgeführt wird, wobei das erste Unterband in einem ersten Spektralband und das zweite Unterband in einem zweiten Spektralband enthalten ist, dadurch gekennzeichnet, dass Gewichtungswerte von M j
    Figure imgb0049
     codiert werden (906), wobei M(j) die normalisierte Maskierungsschwelle für ein Unterband mit dem Index j ist, das im zweiten Spektralband enthalten ist.
  6. Verfahren nach einem der vorhergehenden Ansprüche, dadurch gekennzeichnet, dass die Transformationscodierung in einer oberen Schicht (110) eines hierarchischen Codierers stattfindet,
    - wobei das erste Unterband ein Signal d LB w
    Figure imgb0050
     aufweist, das von einer Kerncodierung (105) des hierarchischen Codierers stammt,
    - und das zweite Unterband ein originales Signal (SHB) aufweist.
  7. Verfahren nach Anspruch 6, dadurch gekennzeichnet, dass das von der Kerncodierung stammende Signal d LB w
    Figure imgb0051
     perzeptuell gewichtet wird (600; 900).
  8. Verfahren nach einem der Ansprüche 6 und 7, dadurch gekennzeichnet, dass das von der Kerncodierung stammende Signal d LB w
    Figure imgb0052
     ein für eine Differenz zwischen einem originalen Signal und einer Synthese dieses originalen Signals repräsentatives Signal ist.
  9. Verfahren nach einem der Ansprüche 6 bis 8, dadurch gekennzeichnet, dass die Transformationscodierung vom Typ TDAC in einem globalen Codierer gemäß der Norm G.729.1 ist, und dass das erste Unterband in einem Band niederer Frequenzen (T1) enthalten ist, während das zweite Unterband in einem Band hoher Frequenzen enthalten ist.
  10. Verfahren nach Anspruch 9, dadurch gekennzeichnet, dass das Band hoher Frequenzen sich bis 7000 Hz (T2) erstreckt, mindestens (T3).
  11. Verfahren nach einem der vorhergehenden Ansprüche, bei dem eine spektrale Hüllkurve berechnet wird (604; 904), dadurch gekennzeichnet, dass die Maskierungsschwelle für ein Unterband definiert wird durch eine Faltung zwischen:
    - einem Ausdruck der spektralen Hüllkurve und
    - einer Spreizfunktion, die eine zentrale Frequenz des Unterbands benutzt.
  12. Verfahren nach einem der vorhergehenden Ansprüche, bei dem eine Information (305) erhalten wird, gemäß der das zu codierende Signal ein tonales Signal ist oder nicht, dadurch gekennzeichnet, dass die perzeptuelle Gewichtung des zweiten Unterbands mit der Bestimmung der Maskierungsschwelle und der Normalisierung nur durchgeführt werden, wenn das Signal nicht tonal ist.
  13. Verfahren zur Decodierung eines Tonsignals in mehreren Unterbändern, bei dem mindestens ein erstes und ein zweites benachbartes Unterband transformationsdecodiert werden (709, 711; 1007, 1009), dadurch gekennzeichnet, dass das Verfahren zur Anwendung einer perzeptuellen Gewichtung im transformierten Bereich auf mindestens das zweite Unterband aufweist:
    - eine Bestimmung mindestens einer Frequenzmaskierungsschwelle (702; 1001; 1011b), die ausgehend von einer decodierten spektralen Hüllkurve an das zweite Unterband anzuwenden ist, und
    - eine Normalisierung der Maskierungsschwelle, um eine spektrale Kontinuität zwischen dem ersten und dem zweiten Unterband zu gewährleisten.
  14. Verfahren nach Anspruch 13, bei dem eine Anzahl von jedem Unterband (703) zuzuweisenden Bits ausgehend von einer Decodierung der spektralen Hüllkurve (701) bestimmt wird,
    dadurch gekennzeichnet, dass die Zuweisung der Bits (703) für mindestens das zweite Unterband außerdem in Abhängigkeit von einer Berechung einer normalisierten Maskierungskurve (702) bestimmt wird, die mindestens an das zweite Unterband angewendet wird.
  15. Verfahren nach Anspruch 13, dadurch gekennzeichnet, dass das transformierte Signal im zweiten Unterband mit einem Faktor proportional zur Quadratwurzel der normalisierten Maskierungsschwelle für das zweite Unterband gewichtet wird (1004).
  16. EDV-Programm, das in einem Speicher eines Codierers eines Telekommunikations-Endgeräts gespeichert ist und/oder auf einem Speicherträger gespeichert ist, der dazu bestimmt ist, mit einem Lesegerät des Codierers zusammenzuwirken,
    dadurch gekennzeichnet, dass es Anweisungen zur Durchführung des Codierverfahrens nach einem der Ansprüche 1 bis 12 aufweist, wenn die Anweisungen von einem Prozessor des Codierers ausgeführt werden.
  17. Codierer, dadurch gekennzeichnet, dass er mindestens einen Speicher aufweist, der ein EDV-Programm nach Anspruch 16 speichert.
  18. EDV-Programm, das in einem Speicher eines Decodierers eines Telekommunikations-Endgeräts gespeichert ist und/oder auf einem Speicherträger gespeichert ist, der dazu bestimmt ist, mit einem Lesegerät des Decodierers zusammenzuwirken,
    dadurch gekennzeichnet, dass es Anweisungen zur Durchführung des Decodierverfahrens nach einem der Ansprüche 13 bis 15 aufweist, wenn die Anweisungen von einem Prozessor des Decodierers ausgeführt werden.
  19. Decodierer, dadurch gekennzeichnet, dass er mindestens einen Speicher aufweist, der ein EDV-Programm nach Anspruch 18 speichert.
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