EP2080194B1 - Attenuation du survoisement, notamment pour la generation d'une excitation aupres d'un decodeur, en absence d'information - Google Patents

Attenuation du survoisement, notamment pour la generation d'une excitation aupres d'un decodeur, en absence d'information Download PDF

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Publication number
EP2080194B1
EP2080194B1 EP07858612A EP07858612A EP2080194B1 EP 2080194 B1 EP2080194 B1 EP 2080194B1 EP 07858612 A EP07858612 A EP 07858612A EP 07858612 A EP07858612 A EP 07858612A EP 2080194 B1 EP2080194 B1 EP 2080194B1
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Prior art keywords
samples
signal
blocks
period
block
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EP07858612A
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German (de)
English (en)
French (fr)
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EP2080194A2 (fr
Inventor
David Virette
Balazs Kovesi
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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
    • G10L25/00—Speech or voice analysis techniques not restricted to a single one of groups G10L15/00 - G10L21/00
    • G10L25/90—Pitch determination of speech signals
    • 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
    • G10L19/08—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
    • G10L19/08—Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters
    • G10L19/09—Long term prediction, i.e. removing periodical redundancies, e.g. by using adaptive codebook or pitch predictor

Definitions

  • the present invention relates to the processing of digital audio signals, such as speech signals in telecommunications, in particular to the decoding of such signals.
  • 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 speech coding, are transmitted in blocks to a peer decoder, via one or more telecommunication networks, to then restore the initial speech signal.
  • block is understood to mean a succession of signal data which may be, for example, a frame in radiomobile communication, or else a packet for example in communication over IP (for " Internet Protocol "), or others.
  • most predictive synthesis coding techniques and in particular CELP coding (for " Code Excited Liner Predictive "), propose solutions for recovering erased frames.
  • the decoder is informed of the occurrence of an erased frame, for example by transmitting frame erase information from the channel decoder.
  • the purpose of recovering erased frames is to extrapolate the parameters of the erased frame from one or more previous frames considered valid.
  • Some parameters manipulated or coded by the predictive coders have a strong correlation between frames. These are typically long-term LTP prediction parameters, for voiced sounds for example, and LPC short-term prediction parameters. Because of this correlation, it is much more advantageous to reuse the parameters of the last valid frame to synthesize the erased frame, than to use random or even erroneous parameters.
  • the LPC parameters of a frame to be reconstructed are obtained from the LPC parameters of the last valid frame, by simple copy of the parameters or with introduction of a certain damping (technique used for example in the standardized encoder G723.1). Then, a voicing or non-voicing in the speech signal is detected to determine a degree of harmonicity of the signal at the erased frame.
  • an excitation signal can be generated randomly (by drawing a codeword from the past excitation, by a slight damping of the gain of past excitation, by random selection in the past excitation, or by still using transmitted codes which can be totally erroneous).
  • the pitch period (also called “ LTP delay ”) is generally the one calculated for the previous frame, possibly with a slight “jitter” (increase of the value of the LTP delay for consecutive error frames, the gain LTP being taken very close to 1 or equal to 1).
  • the excitation signal is therefore limited to the long-term prediction made from a past excitation.
  • the means for hiding erased frames, at decoding are generally strongly related to the structure of the decoder and may be common to modules of this decoder, such as for example the signal synthesis module. These means also use intermediate signals available within the decoder, such as the excitation signal passed and stored during the processing of valid frames preceding the erased frames.
  • Certain techniques used to conceal the errors produced by lost packets during the transport of coded data in time-type coding often use waveform substitution techniques. Such techniques seek to reconstruct the signal by selecting portions of the decoded signal before the lost period and do not use synthesis models. Smoothing techniques are also used to avoid the artifacts produced by the concatenation of the different signals.
  • the techniques for reconstructing erased frames generally rely on the coding structure used. Some techniques aim at regenerating the transformed coefficients lost from the values taken by these coefficients before the erasure.
  • the invention advantageously applies to the case where the digital audio signal is a voiced speech signal, and, more particularly, slightly voiced because the simple copy of pitch period gives poor results in this case.
  • a degree of voicing is detected in the speech signal and steps a) to d) are applied if the signal is at least slightly voiced.
  • the operation a1) may consist in detecting a voicing and the operation a2) would aim, if the speech signal is voiced, to select a number of samples which extends over a whole pitch period (inverse of a fundamental frequency of a voice tone).
  • this embodiment may also target a signal other than a speech signal, in particular a musical signal, if a fundamental frequency specific to a global tone of music can be detected therein.
  • the fragmentation of step b) is carried out in groups of two samples, and the positions of the samples of the same group are reversed with each other.
  • the pitch period (or more generally the inverse period of the fundamental frequency) comprises an even or odd number of samples.
  • the number of samples that comprises the period of the detected tone is an even number
  • an odd number of samples is advantageously added or removed from the samples of said period to form the selection of the step a).
  • the predetermined rules of inversion These rules, which can be chosen according to the characteristics of the signal received, in particular impose the number of samples in groups in step b) and the manner of inverting the samples in a group.
  • groups of two samples and a simple inversion of the respective positions of these two samples are provided.
  • other configurations are possible (groups comprising more than two samples and permutation of all the samples of such groups).
  • the inversion rules can also set the number of groups in which inversion is performed. One particular achievement is to randomize the sample inversion occurrences in each group and set a probability threshold to invert or not the samples of a group.
  • This probability threshold may have a fixed value or a variable value and advantageously depend on a correlation function relating to the pitch period. In this case, the formal determination of the pitch period, itself, is not necessary. Moreover, more generally, the treatment in the sense of the invention can be carried out also if the valid signal received is simply not voiced, in which case there is not really a detectable pitch period. In this case, it may be provided to set a given number of arbitrary samples (for example two hundred samples) and perform the treatment in the sense of the invention on this number of samples.
  • the lost blocks are replaced for example by a white noise, audible, called " comfort noise " 52, and the gain 61 of the samples of the blocks is adjusted thus reconstructed.
  • comfort noise 52 a white noise, audible, called " comfort noise " 52
  • the gain 61 of the samples of the blocks is adjusted thus reconstructed.
  • the voiced signals on the one hand and the weakly or unvoiced signals, on the other hand.
  • the advantage of this variant is that the generation of the unvoiced signal will be identical to the weakly voiced synthesis.
  • the "pitch period" used for the unvoiced signals is a random value, preferably quite large (for example two hundred samples).
  • the preceding signal is non-harmonic, by applying the processing in the sense of the invention to a sufficiently large period, it is ensured that the signal thus generated remains non-harmonic.
  • the nature of the signal will advantageously be preserved, which would not be the case using a randomly generated signal (for example a white noise).
  • the lost blocks are replaced by copying the pitch period T.
  • the pitch period T identified in the last still valid part of the received signal Se is determined ( by a technique 53 any that can be known per se).
  • the samples of this pitch period T are then copied into the lost blocks (reference 54).
  • An appropriate gain 61 is then applied to the samples thus replaced (for example to perform attenuation or "fading").
  • the method is applied in the sense of the invention (arrow M at the output of the test 51 on the degree of voicing).
  • the principle of the invention consists in collecting the samples of the last valid blocks received, in groups of at least two samples.
  • these samples were effectively grouped together in two.
  • they could be grouped by more than two samples, in which case the rules for inversion of samples per group and for taking into account the parity in the number of samples of the pitch period T, described in detail below, would be slightly adapted.
  • groups A, B, C, D of two samples in the last valid blocks received are copied and concatenated to the last samples received.
  • the values of the two samples in each group were inverted (or kept their value and inverted their respective positions).
  • the group A becomes the group A ', with its two samples reversed compared to the group A (according to the two arrows of the group A' on the figure 2 ).
  • Group B becomes group B ', with its two samples inverted with respect to group B, and so on.
  • the copy and concatenation of the groups A ', B', C ', D' is advantageously carried out while respecting the pitch period T.
  • the group A ' consisting of the inverted samples of the group A, is separated from the group A a number of samples corresponding to the duration of the pitch period T.
  • the group B ' is separated from the group B by a duration corresponding to the pitch period T, and so on.
  • the inversion of the samples by group is systematic.
  • the threshold p is set at 50% so that only two groups B ', C' out of four have their samples inverted. It may also be planned to make the probability threshold p variable, in particular to make it depend on a correlation function relating to the pitch period T, as will be seen below.
  • the first two samples 10 and 11 are inverted in the signal to be reconstructed, denoted S.
  • the third and fourth samples 12 and 13 are reversed. also, and so on.
  • the number of samples per periods T , T ', T " is equal to the same odd number (thirteen samples in the example shown), which makes it possible to obtain a gradual mixing of the samples as and when the reconstruction of the signal Ss, and from there, an effective attenuation of the over-harmonicity (or, in other words, the overwriting of the reconstructed signal).
  • This problem can be overcome by modifying the number of samples to be inverted per group (and for example taking an odd number of samples per group).
  • the pitch period comprises an even number of samples and when the inversions aim at even numbers of samples per group, to add an odd number of samples to the pitch period of the signal to be reconstructed.
  • the last detected pitch period T comprises twelve samples 31, 32, ..., 42.
  • a sample is then added to the pitch period and a period T + 1 having an odd number of samples is obtained. So, in the example shown on the figure 3c the sample 30 becomes the first sample of the memory from which the two-by-two sample inversion is applied as shown in FIG. figure 2 (or the figure 3a ).
  • the pitch period T is determined on the last samples of the signal Is validly received (by a technique 56 which can be known per se). It is detected whether the number of samples in the pitch period T is even or odd. If this number is odd (arrow N at the output of the test 57), the sample inversion is applied directly two by two (step 58) as described above with reference to the figure 3a .
  • step 59 a sample is added to the pitch period T (step 59) and then the sample inversion is applied two by two. (step 58), in accordance with the treatment described above with reference to the figure 3c . Then, a chosen gain 61 is optionally applied to the succession of samples thus obtained to form the finally reconstructed signal Ss.
  • the pitch period is firstly calculated from one or a few previous frames. Then the reduced harmonic excitation is generated as illustrated on the figure 2 , with systematic inversion. However, in the variant illustrated on the figure 1 it can be generated with random inversion. This irregular inversion of the samples of the voiced excitation advantageously makes it possible to attenuate the over-harmonicity. This advantageous embodiment is described below.
  • the voiced excitation is calculated by group of two samples and with random inversion according to the treatment below.
  • the length of this memory L mem (in number of stored samples) must be at least twice the maximum value of the pitch period duration (in number of samples).
  • the number of samples to be stored can be of the order of 300, for a low sampling rate in narrow band, and more than 300 for higher sampling rates.
  • the prior determination of the pitch period is not necessary to build the groups of samples to be reversed.
  • the determination of the pitch period T 0 can be carried out together with the constitution of the groups within the meaning of the invention, by application of formula (5) above.
  • the probability p will be very large, and the voicing will be preserved according to the calculation according to the formula (1). If, on the other hand, the voicing of the signal Se is not too marked, the probability p will be lower and the equations (2) and (3) will advantageously be used.
  • equation (1) For highly voiced classes, equation (1) will be used instead.
  • equations (2) and (3) For moderately or weakly voiced classes, equations (2) and (3) will be used instead.
  • equations (2) and (3) For unvoiced classes, no harmonic excitation is generated and the excitation can then be generated from a white noise.
  • equations (2) and (3) will also be used with a sufficiently large arbitrary pitch period.
  • the present invention is not limited to the embodiments described above by way of example; it extends to other variants.
  • the generation of excitation in predictive synthesis coding CELP aims to avoid overwriting in the context of the concealment of frame transmission errors. Nevertheless, it is possible to use the principles of the invention for band extension. It is then possible to use the generation of an expanded band excitation in a band extension system (with or without information transmission), based on a CELP type model (or CELP subband). The excitation of the high band can then be calculated as previously described, which then limits the over-harmonicity of this excitation.
  • the implementation of the invention is particularly suited to the transmission of signals over packet networks, or even by packets, for example " voice over IP " (for " Internet Protocol ”) packets, so as to provide acceptable quality when losing such packets over IP, while still ensuring limited complexity.
  • packets for example " voice over IP " (for " Internet Protocol ") packets, so as to provide acceptable quality when losing such packets over IP, while still ensuring limited complexity.
  • the inversion of the samples can be carried out on groups of samples larger than two.
  • the present invention also relates to a computer program intended to be stored in memory of a device for synthesizing a digital audio signal.
  • This program then comprises instructions for implementing the method within the meaning of the invention, when it is executed by a processor of such a synthesis device.
  • the figure 4 described above can illustrate a flowchart of such a computer program.
  • the synthesis device SYN within the meaning of the invention comprises means such as a working memory MEM (or storage of the aforementioned computer program) and a PROC processor cooperating with this memory MEM, for the implementation of the method within the meaning of the invention, and thus to synthesize the current block from at least one of the preceding blocks of the signal Se.
  • a working memory MEM or storage of the aforementioned computer program
  • PROC processor cooperating with this memory MEM, for the implementation of the method within the meaning of the invention, and thus to synthesize the current block from at least one of the preceding blocks of the signal Se.
  • the present invention also provides an apparatus for receiving a digital audio signal consisting of a succession of blocks, such as a decoder of such a signal for example. Still referring to the figure 5 this apparatus may advantageously comprise an invalid block detector DET, as well as the device SYN within the meaning of the invention for synthesizing invalid blocks detected by the detector DET.

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  • Engineering & Computer Science (AREA)
  • Computational Linguistics (AREA)
  • Signal Processing (AREA)
  • Health & Medical Sciences (AREA)
  • Audiology, Speech & Language Pathology (AREA)
  • Human Computer Interaction (AREA)
  • Physics & Mathematics (AREA)
  • Acoustics & Sound (AREA)
  • Multimedia (AREA)
  • Compression, Expansion, Code Conversion, And Decoders (AREA)
EP07858612A 2006-10-20 2007-10-17 Attenuation du survoisement, notamment pour la generation d'une excitation aupres d'un decodeur, en absence d'information Active EP2080194B1 (fr)

Applications Claiming Priority (2)

Application Number Priority Date Filing Date Title
FR0609225 2006-10-20
PCT/FR2007/052188 WO2008047051A2 (fr) 2006-10-20 2007-10-17 Attenuation du survoisement, notamment pour la generation d'une excitation aupres d'un decodeur, en absence d'information

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EP2080194A2 EP2080194A2 (fr) 2009-07-22
EP2080194B1 true EP2080194B1 (fr) 2011-12-07

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US (1) US8417520B2 (ja)
EP (1) EP2080194B1 (ja)
JP (1) JP5289319B2 (ja)
KR (1) KR101409305B1 (ja)
CN (1) CN101573751B (ja)
AT (1) ATE536613T1 (ja)
BR (1) BRPI0718423B1 (ja)
ES (1) ES2378972T3 (ja)
MX (1) MX2009004212A (ja)
RU (1) RU2437170C2 (ja)
WO (1) WO2008047051A2 (ja)

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IL196146A (en) * 2008-12-23 2014-01-30 Elta Systems Ltd Signal transmission system and method back to the source of transmission
GB0920729D0 (en) * 2009-11-26 2010-01-13 Icera Inc Signal fading
CN105976830B (zh) * 2013-01-11 2019-09-20 华为技术有限公司 音频信号编码和解码方法、音频信号编码和解码装置
FR3004876A1 (fr) * 2013-04-18 2014-10-24 France Telecom Correction de perte de trame par injection de bruit pondere.
CA2916150C (en) 2013-06-21 2019-06-18 Fraunhofer-Gesellschaft Zur Foerderung Der Angewandten Forschung E.V. Apparatus and method realizing improved concepts for tcx ltp
ES2805744T3 (es) 2013-10-31 2021-02-15 Fraunhofer Ges Forschung Decodificador de audio y método para proporcionar una información de audio decodificada usando un ocultamiento de errores en base a una señal de excitación de dominio de tiempo
KR101940740B1 (ko) 2013-10-31 2019-01-22 프라운호퍼 게젤샤프트 쭈르 푀르데룽 데어 안겐반텐 포르슝 에. 베. 시간 도메인 여기 신호를 변형하는 오류 은닉을 사용하여 디코딩된 오디오 정보를 제공하기 위한 오디오 디코더 및 방법
EP2980798A1 (en) * 2014-07-28 2016-02-03 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. Harmonicity-dependent controlling of a harmonic filter tool

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EP0712218B1 (en) * 1994-11-10 1999-07-14 Telefonaktiebolaget Lm Ericsson A method and an arrangement for sound reconstruction during erasures
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FR2813722B1 (fr) * 2000-09-05 2003-01-24 France Telecom Procede et dispositif de dissimulation d'erreurs et systeme de transmission comportant un tel dispositif
EP1217613A1 (fr) * 2000-12-19 2002-06-26 Koninklijke Philips Electronics N.V. Reconstitution de trames manquantes ou mauvaises en téléphonie cellulaire
US7711563B2 (en) * 2001-08-17 2010-05-04 Broadcom Corporation Method and system for frame erasure concealment for predictive speech coding based on extrapolation of speech waveform
DE10219133B4 (de) * 2002-04-29 2007-02-22 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. Vorrichtung und Verfahren zum Verschleiern eines Fehlers
JP4445328B2 (ja) * 2004-05-24 2010-04-07 パナソニック株式会社 音声・楽音復号化装置および音声・楽音復号化方法
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Publication number Publication date
US8417520B2 (en) 2013-04-09
MX2009004212A (es) 2009-07-02
WO2008047051A3 (fr) 2008-06-12
KR101409305B1 (ko) 2014-06-18
KR20090090312A (ko) 2009-08-25
BRPI0718423A2 (pt) 2013-11-12
BRPI0718423B1 (pt) 2020-03-10
JP2010507120A (ja) 2010-03-04
RU2009118918A (ru) 2010-11-27
RU2437170C2 (ru) 2011-12-20
CN101573751A (zh) 2009-11-04
CN101573751B (zh) 2013-09-25
WO2008047051A2 (fr) 2008-04-24
US20100324907A1 (en) 2010-12-23
JP5289319B2 (ja) 2013-09-11
EP2080194A2 (fr) 2009-07-22
ES2378972T3 (es) 2012-04-19
ATE536613T1 (de) 2011-12-15

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