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EP2951813B1 - Correction perfectionnée de perte de trame au décodage d'un signal - Google Patents

Correction perfectionnée de perte de trame au décodage d'un signal Download PDF

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Publication number
EP2951813B1
EP2951813B1 EP14705848.1A EP14705848A EP2951813B1 EP 2951813 B1 EP2951813 B1 EP 2951813B1 EP 14705848 A EP14705848 A EP 14705848A EP 2951813 B1 EP2951813 B1 EP 2951813B1
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EP
European Patent Office
Prior art keywords
signal
segment
frame
spectral components
synthesis
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EP14705848.1A
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German (de)
English (en)
French (fr)
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EP2951813A1 (fr
Inventor
Julien Faure
Stéphane RAGOT
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Orange SA
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Orange 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
    • G10L19/02—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
    • G10L19/005—Correction of errors induced by the transmission channel, if related to the coding algorithm
    • 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
    • G10L19/06—Determination or coding of the spectral characteristics, e.g. of the short-term prediction coefficients
    • 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
    • G10L19/093—Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters using sinusoidal excitation models
    • 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
    • G10L2019/0001—Codebooks
    • G10L2019/0016—Codebook for LPC parameters

Definitions

  • the present invention relates to a signal correction, in particular in a decoder, in the event of loss of frame upon reception of the signal by this decoder.
  • the signal is in the form of a succession of samples, divided into successive frames and "frame" is then understood to mean a signal segment composed of one or more samples (an embodiment where a frame comprises a single sample being possible if the signal is in the form of a series of samples, for example in codecs according to ITU-T Recommendation G.711).
  • the invention lies in the field of digital signal processing, in particular but not exclusively in the field of coding / decoding of an audio signal.
  • Frame loss occurs when communication (either real-time transmission or storage for later transmission) using an encoder and a decoder is disturbed by channel conditions (because of radio problems, access network congestion, etc.).
  • the decoder uses frame loss correction mechanisms (or “masking”) to try to substitute the missing signal with a reconstituted signal, using the information available within the decoder (for example the already decoded signal or parameters received in previous frames). This technique maintains a good quality of service despite degraded channel performance.
  • Frame loss correction techniques are most often very dependent on the type of coding used.
  • the frame loss correction exploits in particular the CELP model.
  • the solution to replace a lost frame is to prolong the use of a long-term prediction gain by attenuating it, and to extend the use of each ISF parameter (for "Imittance Spectral Frequency") by making them tend towards their respective averages.
  • the pitch of the speech signal (or "pitch", parameter designated “LTP lag") is also repeated.
  • the decoder of the random values of parameters characterizing "innovation" (excitation in CELP coding).
  • the most used technique for correcting frame loss in the case of transform coding is to repeat the decoded spectrum in the last received frame.
  • the modulated lapped transform is equivalent to a modified discrete cosine transform (MDCT) with a 50% overlap and sinusoidal analysis / synthesis windows, ensures a transition (between the last lost frame and the repeated frame) which is slow enough to erase the artifacts related to the simple repetition of the spectrum; typically, if more than one frame is lost, the repeated spectrum is set to zero.
  • this masking method does not require additional delay since it exploits the overlap-addition between the reconstituted signal and the passed signal to achieve a sort of "crossfade” (with time folding due to the MLT transform). This is a very inexpensive technique in terms of resources.
  • the present invention improves the situation.
  • frame is understood to mean a block of at least one sample. In most codecs, these frames consist of several samples. However, in particular codecs of the PCM (for "Pulse Code Modulation") type, for example according to the G.711 recommendation, the signal consists simply of a succession of samples (a "frame” within the meaning of the invention with only one sample). The invention can then also be applied to this type of codecs.
  • PCM for "Pulse Code Modulation”
  • the valid signal may consist of the last valid frames received before the frame loss.
  • the samples of the valid signal that are used can be directly those of the frames, and possibly those which correspond to the memory of the transform and which typically contain a folding (or "aliasing") in the case of a transforming decoding with type MLT or MDCT overlay.
  • the invention then provides an advantageous solution to the loss of frame correction (s), especially in the case where an additional delay to the decoder is prohibited, for example when using a decoder transform with windows not allowing have a sufficiently large overlap between the substitution signal and the signal resulting from the temporal unfolding (typical case of low delay windows for an MDCT or an MLT, as shown in FIG. Figure 1B ).
  • the invention offers a particular advantage for an overlap, because of the use of the spectral components on the last valid frames received to construct a synthesis signal comprising the spectral coloring of these last valid frames. Nevertheless, the invention applies of course to any type of coding / decoding (by transform, CELP, PCM, or other).
  • the method comprises searching, by correlation in the valid signal, of a repetition period, the duration of the aforementioned segment then comprising at least one repetition period.
  • Such a "repetition period” corresponds for example to a pitch period in the case of a voiced speech signal (inverse of the fundamental frequency of the signal).
  • the signal may also be derived from a music signal, for example, having a global tone associated with a fundamental frequency, as well as a fundamental period that could correspond to the aforementioned repetition period.
  • a repetition period related to the tone of the signal For example, we can build a first buffer (or "buffer" in French) of the last few validly received samples and search by correlation in a second buffer of larger size, the few samples of the second buffer that best fit in their succession to those of the first buffer.
  • the time difference between these identified samples of the second buffer and those of the first buffer may constitute a repetition period or a multiple of this period (depending on the fineness of the correlation search). It may be noted that the fact of taking a multiple of the repetition period does not degrade the implementation of the invention, because, in this case, the spectral analysis is simply done over a length covering several periods instead of only one, which helps to increase the fineness of the analysis.
  • the aforementioned repetition period corresponds to a duration for which the correlation exceeds a predetermined threshold value.
  • the duration of the signal is identified as soon as the correlation exceeds a predetermined threshold value for this duration.
  • the duration thus identified corresponds to one or more periods associated with a frequency of the above-mentioned overall tone.
  • the method further comprises a determination of the respective phases associated with these spectral components and the construction of the synthesis signal then comprises the phases of the spectral components.
  • the construction of the signal then integrates these phases, as will be seen later, for an optimization of the connection of the synthesis signal to the last valid frames and, in most natural cases, to the following valid frames.
  • the method further comprises a determination of respective amplitudes associated with the spectral components, and the construction of the synthesis signal comprises these amplitudes of the spectral components (for their inclusion in the construction of the synthesis signal).
  • the spectral components of the highest amplitudes may be those selected for the construction of the synthesis signal. It is also possible to select, in addition or alternatively, those whose amplitude forms a peak in the frequency spectrum.
  • noise is added to the synthesis signal to compensate for a loss of energy relative to spectral components not selected for the construction of the synthesis signal.
  • the aforementioned noise is obtained by a weighted (temporally) residual between the segment signal and the synthesis signal.
  • a weighted (temporally) residual between the segment signal and the synthesis signal may be weighted by overlapping windows, as in the case of overlap transformation encoding / decoding.
  • the spectral analysis of the segment comprises a Fast Fourier Transform (FFT) sinusoidal analysis, preferably of length 2 k, where k is greater than or equal to log 2 (P), where P is the number of samples in the signal.
  • FFT Fast Fourier Transform
  • P log 2
  • MCLT Modulated Complex Lapped Transform
  • the present invention finds an advantageous but in no way limiting application to the context of decoding by transform with overlap.
  • the synthesis signal may be constructed (repeated) over a period of at least two frames, so as to cover also the parts having an aliasing beyond one frame.
  • the synthesis signal can be constructed over two frame times and still an additional duration corresponding to a delay introduced by a resampling filter (in particular in the embodiment described above and where a resampling is planned).
  • the invention can then be applied under these conditions by adapting the duration of the synthesis signal.
  • the method further comprises a separation in a high frequency band and a low frequency band, of the signal coming from the valid frame (s), and the spectral components are selected in the band of low frequencies.
  • the present invention is also directed to a computer program comprising instructions for implementing the method (of which, for example, a general flow chart may be the general diagram of the figure 2 , and possibly specific flow charts of figures 5 and / or 8 in some embodiments).
  • Such a device can take the physical form of, for example, a processor and possibly a working memory, typically in a communication terminal.
  • a treatment according to the invention is illustrated on the figure 2 . It is implemented with a decoder.
  • the decoder can be of any type, the processing being generally independent of the nature of the coding / decoding. In the example described, the processing applies to a received audio signal. However, it can be applied more generally to any type of signal analyzed by time windowing and transformation, with harmonization to ensure with one or more replacement frames during a recovery-addition synthesis.
  • the audio buffer corresponds to the samples already decoded in the past frame (and are therefore non-modifiable). If the addition of an additional delay to the decoder is possible (for example of D samples), the buffer may contain only a part of the samples available at the decoder, leaving, for example, the last D samples for the recovery-addition ( from step S10 of the figure 2 ).
  • This filtering is preferably a filtering without delay.
  • the sliding segment, of search is prior to the target segment, as represented on the figure 3 .
  • the first sample of the target segment corresponds to the last sample of the search segment.
  • at least one pitch period (with the same sinusoidal intensity for example) flows between the point of temporal index mc and the temporal index sample mc + P.
  • at least one pitch period is passed between the sample of index mc + Ns (loopback point, of index pb) and the last sample of buffer N '.
  • a variant of this embodiment consists of an autocorrelation on the buffer, returning to find an average period P identified in the buffer.
  • the segment serving for the synthesis comprises the last P samples of the buffer.
  • a self-correlation calculation on a large segment can be complex and require more computing resource than a simple correlation of the type described above.
  • another variant of this embodiment consists in not necessarily seeking the maximum correlation over the entire search segment, but simply looking for a segment where the correlation with the target segment is greater than a chosen threshold (for example 70 %).
  • a chosen threshold for example 70 %.
  • Such an embodiment does not give precisely a single pitch period P (but possibly several successive periods), but nevertheless the complexity related to the processing of a long synthetic segment (of several pitch periods) requires as much or less resource, as the search for maximum correlation across the entire search segment.
  • the correlation search zone for example by shifting the search correlation (by starting it typically 20 ms after the start of the audio buffer as shown by way of example on the figure 4 , or performing the correlation search in a time zone beginning after the end of a transient).
  • the next step S4 consists of breaking down the segment p (n) into a sum of sines.
  • a conventional way of breaking down a signal into a sum of sines is to calculate the discrete Fourier transform (or DFT in English) of the signal over a duration corresponding to the length of the signal. This gives the frequency, the phase and the amplitude of each of the sinusoidal components that make up the signal.
  • this analysis is performed by a fast Fourier transform FFT, size 2 ⁇ k (with k greater than or equal to log 2 (P)).
  • the spectral component selection method is not limited to the examples presented above. It is susceptible of variants. It can in particular be based on any criterion making it possible to identify spectral components useful for the synthesis of the signal (for example subjective criteria related to masking, criteria related to the harmonicity of the signal, or others).
  • the next step S6 is a sinusoidal synthesis.
  • it consists in generating a segment s (n) of length at least equal to the size of a lost frame (T).
  • a length equal to 2 frames is generated so as to be able to perform a "cross-fade" sound mix (as a transition) between the synthesized signal (by loss correction). a frame) and the decoded signal to the next valid frame when such a frame is received again correctly.
  • Step S7 of the figure 2 consists in injecting noise so as to compensate for the energy loss associated with the omission of certain frequency components in the low frequency band.
  • This residue of size P is repeated so that it reaches a size 2 T + LF 2 .
  • the signal s (n) is then mixed (added with possibly weighting) to the signal r (n).
  • the noise generation method (to obtain a natural background noise) is not limited to the example above and admits variants.
  • step S8 consists of processing the high frequency band simply by repeating the signal. For example, it may be to repeat a frame length T.
  • Such an embodiment advantageously makes it possible to avoid audible artifacts by putting the intensities at the beginning and end of the frames at the same level.
  • the frame of size T ' may be weighted so as to avoid certain artifacts when the contents are particularly energetic in the high frequency band.
  • the weighting (denoted W on the figure 6 ) can for example take the form of a sinusoidal half-window of 1 ms at the beginning and at the end of the frame of size T / 2.
  • the successive frames can also be overlapped.
  • a step S9 the signal is synthesized by resampling the low frequency band at its original frequency Fc, and adding it to the signal from the repetition of step S8 in the high frequency band.
  • step S10 a recovery-addition is carried out which ensures a continuity between the signal before the loss of frame and the synthesized signal.
  • the L samples located between the beginning of the "aliased" portion (the remaining folded portion) of the MDCT transform are used. and three quarters of the size of the window (with for example a temporal folding axis of windows as usually in the context of an MDCT transform). With reference to the figure 7 these samples are already covered by the synthesis window W1 of the MDCT transform. In order to be able to apply a cover window W2, the samples are divided by the window W1 (which is already known to the decoder), then multiplied by the window W2.
  • S not The not W 3 not W 1 not + S not W 2 not not ⁇ 0 ,
  • the 2 and W 3 not 1 - W 2 not not ⁇ 0 ;
  • this delay time can be used to overlap with the synthesized portion, using any appropriate weighting for the overlay.
  • step S2 separation in high and low frequency bands in step S2 is optional.
  • the signal from the buffer (step S1) is not separated into two subbands and the steps S3 to S10 remain identical to those described above. Nevertheless, the processing of the spectral components in the low frequencies only advantageously makes it possible to limit their complexity.
  • the invention can be implemented in a conversational decoder, in the case of a loss of frame. Materially, it can be implemented in a circuit for decoding, typically in a telephony terminal.
  • a circuit for decoding typically in a telephony terminal.
  • a circuit CIR may comprise or be connected to a processor PROC, as illustrated on the figure 9 , and may include a MEM working memory, programmed with computer program instructions according to the invention to perform the above method.
  • the invention can be implemented in a real-time transform decoder.
  • the decoder sends requests to obtain an audio frame in a frame buffer (step S81). If the frame is available (OK output of the test), the decoder decodes the frame (S82) to obtain a signal in the transformed domain, operates an IMDCT inverse transform (S83) which then makes it possible to obtain "aliased" time samples, and proceeds to a final step S84 of windowing (through a synthesis window) and overlapping to obtain temporal samples free of aliasing that will then be sent to a digital-to-analog converter for playback.
  • IMDCT inverse transform S83
  • the decoder When a frame is missing (KO output of the test), the decoder then uses the already decoded signal as well as the "aliased" part of the previous frame (step S85), in the frame loss correction method according to the invention. 'invention.

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  • Engineering & Computer Science (AREA)
  • Physics & Mathematics (AREA)
  • Health & Medical Sciences (AREA)
  • Signal Processing (AREA)
  • Audiology, Speech & Language Pathology (AREA)
  • Human Computer Interaction (AREA)
  • Computational Linguistics (AREA)
  • Acoustics & Sound (AREA)
  • Multimedia (AREA)
  • Spectroscopy & Molecular Physics (AREA)
  • Compression, Expansion, Code Conversion, And Decoders (AREA)
  • Compression Or Coding Systems Of Tv Signals (AREA)
  • Error Detection And Correction (AREA)
EP14705848.1A 2013-01-31 2014-01-30 Correction perfectionnée de perte de trame au décodage d'un signal Active EP2951813B1 (fr)

Applications Claiming Priority (2)

Application Number Priority Date Filing Date Title
FR1350845A FR3001593A1 (fr) 2013-01-31 2013-01-31 Correction perfectionnee de perte de trame au decodage d'un signal.
PCT/FR2014/050166 WO2014118468A1 (fr) 2013-01-31 2014-01-30 Correction perfectionnée de perte de trame au décodage d'un signal

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EP2951813A1 EP2951813A1 (fr) 2015-12-09
EP2951813B1 true EP2951813B1 (fr) 2016-12-07

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US (1) US9613629B2 (ru)
EP (1) EP2951813B1 (ru)
JP (1) JP6426626B2 (ru)
KR (1) KR102398818B1 (ru)
CN (1) CN105122356B (ru)
BR (1) BR112015018102B1 (ru)
CA (1) CA2899438C (ru)
FR (1) FR3001593A1 (ru)
MX (1) MX350634B (ru)
RU (1) RU2652464C2 (ru)
WO (1) WO2014118468A1 (ru)

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BR112015018102B1 (pt) 2022-03-22
RU2652464C2 (ru) 2018-04-26
KR102398818B1 (ko) 2022-05-17
RU2015136540A (ru) 2017-03-06
CA2899438A1 (fr) 2014-08-07
CN105122356B (zh) 2019-12-20
MX2015009964A (es) 2016-06-02
US20150371647A1 (en) 2015-12-24
EP2951813A1 (fr) 2015-12-09
US9613629B2 (en) 2017-04-04
KR20150113161A (ko) 2015-10-07
MX350634B (es) 2017-09-12
BR112015018102A2 (pt) 2017-07-18
CN105122356A (zh) 2015-12-02
FR3001593A1 (fr) 2014-08-01
CA2899438C (fr) 2021-02-02
JP6426626B2 (ja) 2018-11-21
WO2014118468A1 (fr) 2014-08-07

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