EP1316087A1 - Übertragungsfehler-verdeckung in einem audiosignal - Google Patents

Übertragungsfehler-verdeckung in einem audiosignal

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Publication number
EP1316087A1
EP1316087A1 EP01969857A EP01969857A EP1316087A1 EP 1316087 A1 EP1316087 A1 EP 1316087A1 EP 01969857 A EP01969857 A EP 01969857A EP 01969857 A EP01969857 A EP 01969857A EP 1316087 A1 EP1316087 A1 EP 1316087A1
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EP
European Patent Office
Prior art keywords
signal
samples
valid
synthesis
decoder
Prior art date
Legal status (The legal status is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the status listed.)
Granted
Application number
EP01969857A
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English (en)
French (fr)
Other versions
EP1316087B1 (de
Inventor
Balazs Kovesi
Dominique Massaloux
David Deleam
Current Assignee (The listed assignees may be inaccurate. Google has not performed a legal analysis and makes no representation or warranty as to the accuracy of the list.)
Orange SA
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France Telecom SA
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Application granted granted Critical
Publication of EP1316087B1 publication Critical patent/EP1316087B1/de
Anticipated expiration legal-status Critical
Expired - Lifetime legal-status Critical Current

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Classifications

    • 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

Definitions

  • the present invention relates to techniques for concealing consecutive transmission errors in transmission systems using any type of digital coding of the speech and / or sound signal.
  • time coders which carry out the compression of samples of digitized signal sample by sample
  • the coded values are then transformed into a binary train which will be transmitted on a transmission channel.
  • disturbances can affect the transmitted signal and produce errors on the bit stream received by the decoder. These errors can occur in isolation in the bitstream but very frequently occur in bursts. It is then a packet of bits corresponding to a complete portion of signal which is erroneous or not received. This type of problem is encountered for example for transmissions on mobile networks. It is also encountered in transmissions on packet networks and in particular on internet-type networks.
  • a general object of the invention is to improve, for any system of speech and sound compression, the subjective quality of the speech signal restored to the decoder when, due to poor quality of the transmission channel or due to the loss or non-reception of a packet in a packet transmission system, a set of consecutive coded data has been lost.
  • Most predictive coding algorithms offer techniques for recovering erased frames ([GSM-FR], [REC G.723.1A], [SALAMI], [HONKA EN], [COX-2], [CHEN- 2], [CHEN-3], [CHEN-4], [CHEN-5], [CHEN-6], [CHEN-7], [KROON-2], [WATKINS]).
  • the decoder is informed of the occurrence of a frame erased in one way or another, for example in the case of radio mobile systems by the transmission of the frame erasure information coming from the channel decoder.
  • the purpose of the devices for recovering erased frames is to extrapolate the parameters of the erased frame from the last previous frame (s) considered to be valid.
  • Some parameters manipulated or coded by predictive coders have a strong inter-frame correlation (case of short-term prediction parameters, also called “Lear” of "Linear Predictive Coding” (see [RABINER]) which represent the spectral envelope, and long-term prediction settings for voiced sounds, for example). Because of this correlation, it is much it is more advantageous to reuse the parameters of the last valid frame to synthesize the erased frame than to use erroneous or random parameters.
  • the LPC filter is obtained from the LPC parameters of the last valid frame either by copying the parameters or with the introduction of a certain damping (cf. encoder G723.1 [REC G.723.1A]).
  • the voicing is detected to determine the degree of harmonicity of the signal at the level of the erased frame ([SALAMI], this detection taking place as follows:
  • the procedures for concealing erased frames are strongly linked to the decoder and use modules of this decoder, such as the signal synthesis module. They use also intermediate signals available within this decoder such as the excitation signal passed and stored during the processing of valid frames preceding the erased frames.
  • the techniques for reconstructing erased frames are also based on the coding structure used: algorithms, such as [PICTEL, MAHIEUX-2], aim to regenerate the lost transformed coefficients from the values taken by these coefficients before erasure.
  • a prior art which can be considered as closest to the present invention is that which is described in [COMBESCURE], which proposed a method for concealing erased frames equivalent to that used in CELP coders for a transform coder.
  • the disadvantages of the proposed method were the introduction of audible spectral distortions
  • the invention allows for the concealment of erased frames without marked distortion at higher error rates and / or for longer erased intervals.
  • the energy of the synthesis signal thus generated is controlled using a gain calculated and adapted sample by sample.
  • the gain for controlling the synthesis signal is advantageously calculated as a function of at least one of the following parameters: energy values previously stored for the samples corresponding to valid data, fundamental period for the voiced sounds, or any parameter characterizing the frequency spectrum.
  • the gain applied to the synthesis signal decreases progressively as a function of the duration during which the synthesis samples are generated.
  • the contents of the memories used for the decoding processing are updated as a function of the synthesis samples generated.
  • a coding analogous to that implemented at the transmitter is implemented at least partially on the synthesized samples possibly followed by a decoding operation (possibly partial), the data obtained serving to regenerate the memories of the decoder.
  • this optionally partial coding-decoding operation can be advantageously used to regenerate the first erased frame because it makes it possible to exploit the content of the memories of the decoder before the cut, when these memories contain information not provided by the last valid samples. decoded (for example in the case of addition-overlap transformers, see paragraph 5.2.2.2.1 point 10).
  • an excitation signal is generated at the input of the short-term prediction operator which, in the neighboring zone, is the sum of a harmonic component and a weakly harmonic component or non harmonic, and in the voiced zone limited to the non harmonic component.
  • the harmonic component is advantageously obtained by implementing a filtering by means of the long-term prediction operator applied to a residual signal calculated by implementing reverse short-term filtering on the stored samples.
  • the other component can be determined with the idea of a long-term prediction operator to which pseudo-random disturbances (for example gain or period disturbances) are applied.
  • pseudo-random disturbances for example gain or period disturbances
  • the harmonic component represents the low frequencies of the spectrum, while the other component represents the high frequency part.
  • the long-term prediction operator is determined from the samples of valid stored frames, with a number of samples used for this estimation varying between a minimum value and a value equal to at least twice the estimated fundamental period for voiced sound.
  • the residual signal is advantageously modified by treatments of the non-linear type to eliminate amplitude peaks.
  • voice activity is detected by estimating noise parameters when the signal is considered to be non-active, and parameters of the synthesized signal are made to tend towards those of the estimated noise.
  • the spectral envelope of the noise of the decoded samples is estimated. valid and one generates a synthesized signal evolving towards a signal having the same spectral envelope.
  • the invention also provides a method for processing sound signals, characterized in that a discrimination is made between speech and musical sounds and when musical sounds are detected, a method of the aforementioned type is implemented without the estimation of a long-term prediction operator, the excitation signal being limited to a non-harmonic component obtained for example by generating uniform white noise.
  • the invention further relates to a device for concealing a transmission error in an audio-digital signal which receives as input a decoded signal which is transmitted to it by a decoder and which generates missing or erroneous samples in this decoded signal, characterized in that 'It includes processing means capable of implementing the above method.
  • It also relates to a transmission system comprising at least one encoder, at least one transmission channel, a module capable of detecting that transmitted data has been lost or is greatly erroneous, at least one decoder and an error concealment device which receives the decoded signal, characterized in that this error concealment device is a device of the aforementioned type.
  • FIG. 1 is a block diagram illustrating a transmission system according to a possible embodiment of the invention
  • FIG. 2 and Figure 3 are block diagrams illustrating an implementation according to a possible embodiment of the invention.
  • FIGS. 4 to 6 schematically illustrate the windows used with the error concealment method according to a possible embodiment of one invention
  • Figures 7 and 8 are schematic representations illustrating a possible embodiment of the invention in the case of musical signals.
  • FIG. 1 shows a device for coding and decoding the digital audio signal, comprising an encoder 1, a transmission channel 2, a module 3 making it possible to detect that the transmitted data has been lost or is strongly erroneous, a decoder 4, and a module 5 for concealing errors or lost packets in accordance with a possible embodiment of the invention.
  • this module in addition to the indication of erased data, receives the decoded signal in valid period and transmits signals used to update it to the decoder.
  • module 5 is based on:
  • the memory of the decoded samples is updated, containing a sufficient number of samples for the regeneration of any periods erased subsequently.
  • a signal of the order of 20 to 40 ms is stored.
  • the energy of the valid frames is also calculated and the energies corresponding to the last valid frames processed are stored in memory (typically of the order of 5 s).
  • a method for detecting voiced sounds (processing 12 in FIG. 3: V / NV detection, for "voiced / unvoiced") is used on the last stored data. For example one can use for that the normalized correlation ([KLEIJN]), or the criterion presented in the example of realization which follows.
  • a residual signal is calculated by reverse filtering LPC (processing 10) of the last stored samples. This signal is then used to generate an excitation signal from the LPC 11 synthesis filter (see below).
  • the synthesis of the replacement samples is carried out by introducing an excitation signal (calculated in 13 from the signal at the output of the inverse LPC filter) in the LPC synthesis filter 11 (l / A (z)) calculated in 1.
  • This excitation signal is generated in two different ways depending on whether the signal is voiced or unvoiced:
  • the excitation signal is the sum of two signals, one strongly harmonic component and the other less harmonic or not at all.
  • the strongly harmonic component is obtained by LTP filtering (processing module 14) using the parameters calculated in 2, of the residual signal mentioned in 3.
  • the second component can also be obtained by LTP filtering but made non-periodic by random modifications of the parameters, by generation of a pseudo-random signal.
  • the residual signal used for generating the excitation is processed to eliminate the amplitude peaks significantly above the average.
  • the energy of the synthesis signal is controlled using a gain calculated and adapted sample by sample. In the case where the erasure period is relatively long, it is necessary to gradually lower the energy of the synthesis signal.
  • the gain adaptation law is calculated according to different parameters: stored energy values before erasure (see in 1), fundamental period, and local stationarity of the signal at the time of cutting.
  • the system includes a module allowing the discrimination of stationary (like music) and non-stationary (like speech) sounds, different adaptation laws can also be used.
  • the first half of the memory of the last frame correctly received contains fairly precise information on the first half of the first lost frame (its weight in the addition-recovery is more important than that of the current frame). This information can also be used to calculate the adaptive gain.
  • the synthesis parameters can also be changed. If the system is coupled to a voice activity detection device with estimation of the noise parameters (such as [REC-G.723.1A], [SALAMI-2],
  • KLEIJN predictions
  • This information is normally available both to the coder, who must have done this for these preceding samples have a form of local decoding, and at the remote decoder present at the reception. As soon as the transmission channel is disturbed and the remote decoder no longer has the same information as the local decoder present at transmission, there is desynchronization between the encoder and the decoder.
  • this desynchronization can cause audible degradations which can last a long time or even increase over time if there are instabilities in the structure. In this case, it is therefore important to endeavor to resynchronize the coder and the decoder, that is to say to make an estimation of the memories of the decoder as close as possible to those of the coder.
  • resynchronization techniques depend on the coding structure used. One will be presented, the principle of which is general in this patent, but the complexity of which is potentially significant.
  • One possible method consists in introducing into the decoder on reception a coding module of the same type as that present on the transmission, making it possible to carry out the coding-decoding of the samples of the signal produced by the techniques mentioned in the preceding paragraph during the periods deleted. In this way the memories necessary to decode the following samples are completed with a priori similar data.
  • This update can be carried out at the time of production of the replacement samples, which distributes the complexity over the entire erasure zone, but is combined with the synthesis procedure described above.
  • the above procedure can also be limited to an intermediate zone at the start of the period of valid data succeeding an erased period, the updating procedure then being combined with the decoding operation. .
  • TDAC or TCDM ([MAHIEUX]) type transform coders are particularly addressed.
  • Broadband encoder (50-7000 Hz) at 24 kb / s or 32 kb / s. 20 ms frame (320 samples).
  • a binary frame contains the coded parameters obtained by the TDAC transformation on a window. After decoding these parameters, by doing the reverse transformation TDAC, we obtain an output frame of 20 ms which is the sum of the second half of the previous window and the first half of the current window.
  • the two parts of windows used for the reconstruction of the frame n have been marked in bold.
  • a lost binary frame disturbs the reconstruction of two consecutive frames (the current one and the next one, Figure 5).
  • FIG. 6 binary frame
  • the memory of the decoded samples is updated.
  • This memory is used for LPC and LTP analyzes of the signal passed in the event of erasure of a binary frame.
  • the LPC analysis is performed over a signal period of 20 ms (320 samples).
  • LTP analysis requires more samples to be stored.
  • the number of samples stored is equal to twice the maximum value of the pitch. For example, if the maximum value of the MaxPitch pitch is fixed at 320 samples (50 Hz, 20 ms), the last 640 samples will be memorized (40 ms of the signal).
  • MaxCorr 0.6
  • Tj the position of this maximum
  • MaxCorrL Corr (T] _) If ⁇ > MinPitch and MaxCorrL> 0.75 * MaxCorr, we choose i as the new fundamental period.
  • T p is less than MaxPitch / 2
  • we can check if it is really a voiced frame by looking for the local maximum of the correlation around 2 * TP (TPP) and checking if Corr (T PP )> 0.4. If Corr (T) ⁇ 0.4 and if the signal energy decreases, we set DiminFlag l and we decrease the value of MaxCorr, otherwise we look for the next local maximum between the current T P and MaxPitch.
  • Another voicing criterion consists in checking whether at least in 2/3 of the cases the signal delayed by the fundamental period has the same sign as the non-delayed signal.
  • the voicing decision also takes into account the signal energy: if the energy is strong, the value of MaxCorr is increased, so it is more likely that the frame is decided voiced. On the other hand, if the energy is very low, the value of MaxCorr is reduced.
  • this vector of Tp samples is processed.
  • the method used in our example is as follows: "We calculate the mean MeanAmpl of the absolute values of the last Tp samples of the residual signal.
  • the excitation signal is the sum of two signals, a strongly harmonic component limited in band at the low frequencies of the excb spectrum and another less harmonic limited to the highest frequencies exch.
  • the coefficients [0.15, 0.7, 0.15] correspond to a low pass FIR filter of 3 dB attenuation at Fs /.
  • the second component is also obtained by an LTP filtering made non-periodic by the random modification of its fundamental period Tph.
  • Tph is chosen as the integer part of a random real value Tpa.
  • the initial value of Tpa is equal to Tp then it is modified sample by sample by adding a random value in [-0.5, 0.5].
  • the voiced excitation is then the sum of these 2 components:
  • the excitation signal exe is also obtained by LTP filtering of order 3 with the coefficients [0.15, 0.7, 0.15] but it is made non-periodic by increasing the fundamental period d 'a value equal to 1 every 10 samples, and inversion of the sign with a probability of 0.2.
  • the memory of the decoder is updated for decoding the next frame (synchronization of the encoder and the decoder, see paragraph 5.1.4).
  • the addition-recovery technique makes it possible to check whether the synthesized voiced signal corresponds well to the original signal or not because for the first half of the first frame lost the weight of the last window memory correctly received is greater ( figure 6). So by taking the correlation between the first half of the first synthesized frame and the first half of the frame obtained after the TDAC g reverse TDAC operations, we can estimate the similarity between the lost frame and the replacement frame. A weak correlation ( ⁇ 0.65) indicates that the original signal is enough different from that obtained by the replacement method, and it is better to decrease the energy of the latter quickly to the minimum level.
  • points 1-6 relate to the analysis of the decoded signal preceding the first erased frame and allowing the construction of a synthesis model (LPC and possibly LTP) of this signal.
  • LPC synthesis model
  • the analysis is not repeated, the replacement of the lost signal is based on the parameters (LPC coefficients, pitch, MaxCorr, ResMem) calculated during the first erased frame.
  • Such processing implements the following steps for the music synthesis module, illustrated in FIG. 8:
  • the synthesis of the replacement samples is carried out by introducing an excitation signal into the LPC synthesis filter (l / A (z)) calculated in step 19.
  • This excitation signal - calculated in a step 20 - is a white noise whose amplitude is chosen to obtain a signal having the same energy as that of the last N samples stored in valid period.
  • the filtering step is referenced by 21.
  • Example of the control of the amplitude of the residual signal If the excitation is presented as a uniform white noise multiplied by a gain, one can calculate this gain G as follows:
  • Durbin's algorithm gives the energy of the residual signal. Knowing also the energy of the signal to be modeled, the gain G ⁇ c: of the LPC filter is estimated as the ratio of these two energies. Calculation of the target energy:
  • the target energy is estimated equal to the energy of the last N samples stored in a valid period (N is typically ⁇ the length of the signal used for the LPC analysis).
  • the energy of the synthesized signal is the product of the energy of white noise by G 2 and G ⁇ ⁇ . We choose G so that this energy is equal to the target energy.
  • the energy of the synthesis signal is controlled at using a gain calculated and adapted sample by sample. In the case where the erasure period is relatively long, it is necessary to gradually lower the energy of the synthesis signal.
  • the gain adaptation law can be calculated as a function of various parameters such as the energy values memorized before erasure, and local stationarity of the signal at the time of cutting. 6. Evolution of the synthesis procedure over time:
  • the synthesis parameters can also be changed. If the system is coupled to a device for detecting voice activity or musical signals with estimation of the noise parameters (such as [REC-G.723.1A],
  • the technique which has just been described has the advantage of being usable with any type of coder; in particular it makes it possible to remedy the problems of lost bit packets for time or transform coders, on speech and music signals with good performance: indeed in the present technique, the only signals memorized during periods when the data transmitted are valid are the samples from the decoder, information that is available regardless of the coding structure used.
  • AT&T DA Kapilo, RV Cox
  • FEC frame erasure concealment
  • GSM-FR GSM Recommendation 06.11. "Substitution and muting of lost frames for full rate speech traffic channels”. ETSI / TC SMG, ver. : 3.0.1. , February 1992.

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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)
  • Detection And Prevention Of Errors In Transmission (AREA)
  • Transmission Systems Not Characterized By The Medium Used For Transmission (AREA)
  • Mobile Radio Communication Systems (AREA)
  • Input Circuits Of Receivers And Coupling Of Receivers And Audio Equipment (AREA)
  • Automobile Manufacture Line, Endless Track Vehicle, Trailer (AREA)
  • Arrangements For Transmission Of Measured Signals (AREA)
EP01969857A 2000-09-05 2001-09-05 Übertragungsfehler-verdeckung in einem audiosignal Expired - Lifetime EP1316087B1 (de)

Applications Claiming Priority (3)

Application Number Priority Date Filing Date Title
FR0011285 2000-09-05
FR0011285A FR2813722B1 (fr) 2000-09-05 2000-09-05 Procede et dispositif de dissimulation d'erreurs et systeme de transmission comportant un tel dispositif
PCT/FR2001/002747 WO2002021515A1 (fr) 2000-09-05 2001-09-05 Dissimulation d'erreurs de transmission dans un signal audio

Publications (2)

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EP1316087A1 true EP1316087A1 (de) 2003-06-04
EP1316087B1 EP1316087B1 (de) 2008-01-02

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US (2) US7596489B2 (de)
EP (1) EP1316087B1 (de)
JP (1) JP5062937B2 (de)
AT (1) ATE382932T1 (de)
AU (1) AU2001289991A1 (de)
DE (1) DE60132217T2 (de)
ES (1) ES2298261T3 (de)
FR (1) FR2813722B1 (de)
HK (1) HK1055346A1 (de)
IL (2) IL154728A0 (de)
WO (1) WO2002021515A1 (de)

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WO2015044609A1 (fr) 2013-09-30 2015-04-02 Orange Re-echantillonnage d'un signal audio pour un codage/decodage a bas retard
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US20040010407A1 (en) 2004-01-15
AU2001289991A1 (en) 2002-03-22
EP1316087B1 (de) 2008-01-02
US20100070271A1 (en) 2010-03-18
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DE60132217T2 (de) 2009-01-29
US7596489B2 (en) 2009-09-29
ATE382932T1 (de) 2008-01-15
US8239192B2 (en) 2012-08-07
IL154728A0 (en) 2003-10-31
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