EP3138095B1 - Correction de perte de trame perfectionnée avec information de voisement - Google Patents

Correction de perte de trame perfectionnée avec information de voisement Download PDF

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Publication number
EP3138095B1
EP3138095B1 EP15725801.3A EP15725801A EP3138095B1 EP 3138095 B1 EP3138095 B1 EP 3138095B1 EP 15725801 A EP15725801 A EP 15725801A EP 3138095 B1 EP3138095 B1 EP 3138095B1
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Prior art keywords
signal
frame
components
decoding
voicing
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German (de)
English (en)
French (fr)
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EP3138095A1 (fr
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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
    • G10L19/005—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
    • G10L19/02—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
    • G10L19/028—Noise substitution, i.e. substituting non-tonal spectral components by noisy source
    • 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
    • G10L25/00—Speech or voice analysis techniques not restricted to a single one of groups G10L15/00 - G10L21/00
    • G10L25/78—Detection of presence or absence of voice signals
    • G10L25/81—Detection of presence or absence of voice signals for discriminating voice from music
    • 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
    • G10L19/20—Vocoders using multiple modes using sound class specific coding, hybrid encoders or object based coding
    • 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
    • G10L25/00—Speech or voice analysis techniques not restricted to a single one of groups G10L15/00 - G10L21/00
    • G10L25/93—Discriminating between voiced and unvoiced parts of speech signals
    • G10L2025/932—Decision in previous or following frames
    • 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
    • G10L25/00—Speech or voice analysis techniques not restricted to a single one of groups G10L15/00 - G10L21/00
    • G10L25/93—Discriminating between voiced and unvoiced parts of speech signals

Definitions

  • the present invention relates to the field of telecommunication coding / decoding, and more particularly that of decoding frame loss correction.
  • a "frame” is understood to mean an audio segment composed of at least one sample (so that the invention applies equally well to the loss of one or more samples coded according to the G.711 standard, as well as to a loss of one or more sample packets coded according to G.723, G.729, etc.).
  • the loss of audio frames occurs when a real-time communication using an encoder and a decoder is disturbed by the conditions of a telecommunication network (radio frequency problems, congestion of the access network, etc.).
  • the decoder uses frame loss correction mechanisms to try to substitute the missing signal with a reconstructed signal by using information available to the decoder (for example the audio signal already decoded for one or more past frames). This technique can maintain quality of service despite degraded network performance.
  • Frame loss correction techniques are most often very dependent on the type of coding used.
  • CELP coding it is common to repeat certain parameters decoded at the previous frame (spectral envelope, pitch, dictionary gains), with adjustments such as a modification of the spectral envelope to converge towards a medium envelope or the use of a random fixed dictionary.
  • a Modulated Lapped Transform (or MLT) with 50% overlap addition and sinusoidal windows provide 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 frame in the case of a single lost frame.
  • MLT Modulated Lapped Transform
  • this realization does not require additional delay since it exploits the existing delay and the time folding of the MLT transform to make an overlay with the reconstituted signal.
  • the document FR 1350845 proposes a hybrid method that combines the advantages of the two methods by allowing phase continuity in the transformed domain.
  • the present invention falls within this framework. A detailed description of the solution that is the subject of this document FR 1350845 is described later with reference to the figure 1 .
  • the invention improves the situation.
  • the invention aims to improve the state of the art within the meaning of the document FR 1350845 by modifying different stages of the processing presented in this document (pitch search, selection of components, noise injection) but nevertheless according in particular to the characteristics of the original signal.
  • These characteristics of the original signal may be encoded as particular information in the data stream to the decoder (or "bitstream") depending on the classification of the speech and / or music, and the case of the speech class. in particular.
  • Such an embodiment can be implemented in an encoder for determining the voicing information, and more particularly in a decoder, particularly in the case of frame loss. It can be implemented as software in a coded / decoded implementation for Enhanced Voice Services ("EVS") specified by the 3GPP (SA4) group.
  • EVS Enhanced Voice Services
  • the present invention is also directed to a computer program as in claim 13.
  • An example of a flow chart of such a program is presented in the following detailed description with reference to FIG. figure 4 for decoding in the sense of the invention and with reference to the figure 3 for coding useful for the invention.
  • the present invention also relates to a device for decoding a digital audio signal as in claim 14 comprising a succession of samples distributed in successive frames.
  • a succession of N audio samples denoted b (n) below, is stored in a buffer memory of the decoder (or "buffer"). These samples correspond to samples already decoded and are therefore accessible for the decay field decoder correction. If the first sample to be synthesized is the sample N, the audio buffer corresponds to the previous samples 0 to N-1.
  • the audio buffer corresponds to the samples at the previous frame, and are not modifiable because this type of coding / decoding does not provide for delay in the return of the signal, so that it It is not planned to perform a crossfade of sufficient duration to cover a frame loss.
  • Fc separation frequency
  • This filtering is preferably a filtering without delay.
  • this filtering step may be optional, the following steps being carried out in full band.
  • the next step S3 consists of searching in the low band for a loopback point and a segment p (n) corresponding to the fundamental period (or "pitch” hereinafter) within the buffer b (n) resampled to the frequency Fc.
  • This realization makes it possible to take into account the continuity of the pitch in the frame (s) lost (s) to be reconstructed.
  • Step S4 consists of breaking down the segment p (n) into a sum of sinusoidal components.
  • the discrete Fourier transform (DFT) of the signal p (n) can be computed over a period corresponding to the length of the signal. The frequency, the phase and the amplitude of each of the sinusoidal components (or "peaks") that make up the signal are thus obtained.
  • Other transforms than DFT are possible. For example, transforms of DCT, MDCT or MCLT type can be implemented.
  • Step S5 is a step of selecting K sinusoidal components so as to keep only the most important components.
  • the selection of the components corresponds firstly to selecting the amplitudes A (n) for which A (n)> A (n-1) and A (n)> A (n + 1) with not ⁇ 0 ; P ' 2 - 1 , which ensures that the amplitudes correspond to spectral peaks.
  • Fourier transform analysis FFT is therefore more efficient over a length that is a power of 2, without changing the actual pitch period (due to the interpolation).
  • ⁇ ( k ) FFT ( p' (n)); and, from the FFT transform, we obtain directly the phases ⁇ ( k ) and amplitudes A ( k ) of the sinusoidal components, the normalized frequencies between 0 and 1 being given here by: f k - 2 ⁇ kP ' P 2 k ⁇ 0 ; P ' 2 - 1
  • the step S6 sinusoidal synthesis consists in generating a segment s (n) of length at least equal to the size of the lost frame (T).
  • Step S7 consists of "injecting noise” (filling the spectral zones corresponding to the unselected lines) so as to compensate for the energy loss linked to the omission of certain frequency peaks in the low band.
  • Step S8 applied to the high band may simply consist of repeating the past signal.
  • step S9 the signal is synthesized by resampling the low band at its original frequency fc, after being mixed in step S8 to the high band filtered (simply repeated in step S11).
  • Step S10 is an overlay addition that provides continuity between the signal before the frame loss and the synthesized signal.
  • signaling information of the signal before loss of frame, transmitted to at least one encoder bit rate, is used at decoding (step DI-1) to quantitatively determine a proportion of noise to be added to the synthesis signal replacing one or more lost frames .
  • the decoder uses the voicing information, to decrease, as a function of the voicing, the general amount of noise mixed with the synthesis signal (by assigning a gain G (res) lower to the noise signal r '(k) from a residue in step DI-3, and / or selecting more amplitude components A (k) to be used for construction of the synthesis signal in step DI-4).
  • the decoder can further adjust its parameters, including pitch search, to optimize the compromise quality / complexity of the treatment, according to the information of voicing. For example, for the pitch search, if the signal is voiced, the pitch search window Nc may be larger (at step DI-5), as will be seen later with reference to the figure 5 .
  • This "flatness" data Pi of the spectrum can be received on several bits at the decoder at the optional step DI-10 of the figure 2 , then compared to a threshold in step DI-11, which amounts to determining in steps DI-1 and DI-2 whether the voicing is greater or less than a threshold, and deducing from it the appropriate treatments, in particular for the selection of peaks and for the choice of duration of the pitch search segment.
  • This information (whether in the form of a single bit or a multi-bit value) is received from the encoder (at at least one rate of the codec), in the example described here.
  • the input signal presented in the form of C1 frames is analyzed in step C2.
  • the analysis step consists in determining whether the audio signal of the current frame has characteristics that would require special treatment in the event of loss of frames to the decoder, as is the case, for example, on voiced speech signals.
  • a classification speech / music or other
  • a coder classification already makes it possible to adapt the technique used for the coding according to the nature of the signal (speech or music).
  • predictive coders such as, for example, the coder according to the G.718 standard also use a classification so as to adapt the parameters of the coder to the nature of the signal (voiced / unvoiced, transient, generic, inactive).
  • characterization for the loss of frame is reserved. It is added to the code stream (or bitstream) in step C3 to indicate whether the signal is a speech signal (voiced or generic). This bit is for example set to 1 or to 0 according to the case of the table below: • the decision of the speech / music classifier, • and additionally the decision of the classifier of the speech coding mode. Decision of the encoder classifier speech Music Characterization bit value for frame loss Decision of the classifier Coding mode: 0 Voiced 1 Not Voised 0 transient 0 Generic 1 Inactive 0
  • the information transmitted to the decoder in the coded stream is not binary but corresponds to a quantification of the ratio between the peak levels and the valley levels in the spectrum.
  • x (k) is the amplitude spectrum of size N resulting from the analysis of the current frame in the frequency domain (after FFT).
  • a sinusoidal analysis decomposing the signal to the sinusoidal component and noise encoder is available and the measure of flatness is obtained by ratio between the sinusoidal components and the overall energy on the frame.
  • step C3 (comprising the single-bit voicing information or the flatness measurement over several bits)
  • the audio buffer of the encoder is conventionally coded in a step C4 before possible subsequent transmission to the decoder.
  • the decoder reads the information contained in the coded stream, including the "characterization for frame loss” information in step D2 (at at least one rate of the codec). These are stored in memory so that they can be reused in case a next frame is missing. The decoder then continues the conventional decoding steps D3, etc. to obtain the SYN SYNTH synthesized output frame.
  • step D4, D5, D6, D7, D8 and D12 corresponding respectively to the steps S2, S3, S4, S5, are applied.
  • S6 and S11 of the figure 1 are applied.
  • steps S3 and S5 respectively to the steps D5 (search for a loopback point for the determination of the pitch) and D7 (selection of the sinusoidal components).
  • the noise injection at step S7 of the figure 1 is performed with a gain determination according to two steps D9 and D10 in the figure 4 decoder within the meaning of the invention.
  • the invention consists in modifying the processing of steps D5, D7 and D9-D10, as follows.
  • step D7 of the figure 4 we select sinusoidal components so as to keep only the most important components.
  • the first selection of components amounts to selecting the amplitudes A (n) for which A (n)> A (n-1) and A (n)> A (n + 1) with not ⁇ 0 : p ' 2 - 1 .
  • the signal that one seeks to reconstruct is a speech signal (voiced or generic) so with marked peaks and a low noise level.
  • the signal that one seeks to reconstruct is a speech signal (voiced or generic) so with marked peaks and a low noise level.
  • This modification notably makes it possible to lower the noise level (and in particular the level of noise injected in steps D9 and D10 presented below) with respect to the level of signal synthesized by sinusoidal synthesis in step D8, while maintaining a global level. sufficient energy not to cause audible artifacts related to energy fluctuations.
  • the voicing information is advantageously used here to attenuate the noise by applying a gain G to the step D10.
  • G may be a constant equal to 1 or 0.25 as a function of the voiced or unvoiced nature of the signal of the preceding frame, according to the table given below by way of example: Bit value of "characterization for frame loss" 0 1 Gain G 1 0.25
  • the "frame loss characterization" information has several discrete levels characterizing the flatness Pl of the spectrum.
  • the gain G can be expressed directly as a function of the value Pl. The same applies to the limit of the segment Nc for the pitch search and / or for the number of peaks An to be taken into account for the synthesis of the signal.
  • a treatment can be defined as follows.
  • the gain G is already defined directly as a function of the value PI: G ( Pl ) - 2 Pl
  • the value P1 is compared to a mean value -3dB, with the proviso that the value 0 corresponds to a flat spectrum, and -5 dB corresponds to a spectrum with pronounced peaks.
  • the duration of the segment of search for pitch Nc at 33 ms and select the peaks A (n) such that A (n)> A (n-1) and A (n)> A (n + 1), as well as the neighboring first peaks A (n) n-1) and A (n + 1).
  • the duration Nc can be chosen shorter, for example 25 ms and only A (n) peaks such as A (n)> A (n-1) and A (n)> A (n + 1) are selected.
  • the decoding can then be continued by mixing the noise, the gain of which is thus obtained, with the components thus selected to obtain the synthesis signal in the low frequencies at the D13 tab, which is added to the synthesis signal in the high frequencies obtained at step D14, to obtain in step D15 the synthesized overall signal.
  • a DECOD decoder (comprising, for example, software and hardware hardware such as a judiciously programmed MEM memory and a PROC processor cooperating with this memory, or alternatively a component such as a ASIC, or other, as well as a COM communication interface) implanted for example in a telecommunication device such as a TEL telephone, uses, for the implementation of the method of the figure 4 , a voicing information that it receives from a coder COD.
  • This encoder comprises, for example, software and hardware hardware such as a memory MEM 'judiciously programmed to determine the voicing information and a processor PROC' cooperating with this memory, or alternatively a component such as an ASIC, or other, as well as a communication interface COM '.
  • the coder COD is implanted in a telecommunication device such as a TEL 'telephone.
  • the information on voicing can take different forms that can be varied.
  • it may be a binary value on a single bit (voicing or not), or a value on several bits which may be relative to a parameter such as the flatness of the signal spectrum, or any other parameter to characterize (quantitatively or qualitatively) a voicing.
  • this parameter can be determined at decoding, for example according to the degree of correlation that can be measured during the identification of the pitch period.
  • an embodiment comprising a separation in a high frequency band and a low frequency band of the signal from previous valid frames, with in particular a selection of the spectral components in the first embodiment, has been presented as an example. low frequency band. Nevertheless, this embodiment is optional although advantageous in the sense that it reduces the complexity of the treatment.
  • the frame replacement method assisted by the voicing information in the sense of the invention can nevertheless be achieved by considering the entire spectrum of the valid signal, alternatively.
  • the aforementioned noise signal can be obtained by the residue (between the valid signal and the sum of the peaks) by weighting this residue temporally. For example, it can be weighted by overlapping windows, as in the usual framework of a transform coding / decoding with overlap.

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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)
  • Spectroscopy & Molecular Physics (AREA)
  • Compression, Expansion, Code Conversion, And Decoders (AREA)
  • Transmission Systems Not Characterized By The Medium Used For Transmission (AREA)
EP15725801.3A 2014-04-30 2015-04-24 Correction de perte de trame perfectionnée avec information de voisement Active EP3138095B1 (fr)

Applications Claiming Priority (2)

Application Number Priority Date Filing Date Title
FR1453912A FR3020732A1 (fr) 2014-04-30 2014-04-30 Correction de perte de trame perfectionnee avec information de voisement
PCT/FR2015/051127 WO2015166175A1 (fr) 2014-04-30 2015-04-24 Correction de perte de trame perfectionnée avec information de voisement

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EP3138095B1 true EP3138095B1 (fr) 2019-06-05

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US (1) US10431226B2 (ru)
EP (1) EP3138095B1 (ru)
JP (1) JP6584431B2 (ru)
KR (3) KR20170003596A (ru)
CN (1) CN106463140B (ru)
BR (1) BR112016024358B1 (ru)
ES (1) ES2743197T3 (ru)
FR (1) FR3020732A1 (ru)
MX (1) MX368973B (ru)
RU (1) RU2682851C2 (ru)
WO (1) WO2015166175A1 (ru)
ZA (1) ZA201606984B (ru)

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EP3138095A1 (fr) 2017-03-08
KR20170003596A (ko) 2017-01-09
RU2682851C2 (ru) 2019-03-21
MX2016014237A (es) 2017-06-06
KR20220045260A (ko) 2022-04-12
US20170040021A1 (en) 2017-02-09
US10431226B2 (en) 2019-10-01
ZA201606984B (en) 2018-08-30
BR112016024358A2 (pt) 2017-08-15
FR3020732A1 (fr) 2015-11-06
JP2017515155A (ja) 2017-06-08
ES2743197T3 (es) 2020-02-18
WO2015166175A1 (fr) 2015-11-05
RU2016146916A (ru) 2018-05-31
KR20230129581A (ko) 2023-09-08
MX368973B (es) 2019-10-23
CN106463140B (zh) 2019-07-26
JP6584431B2 (ja) 2019-10-02
RU2016146916A3 (ru) 2018-10-26
BR112016024358B1 (pt) 2022-09-27
CN106463140A (zh) 2017-02-22

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