EP1989705B1 - Verfahren zur begrenzung der adaptiven erregungsverstärkung bei einem auidodecoder - Google Patents

Verfahren zur begrenzung der adaptiven erregungsverstärkung bei einem auidodecoder Download PDF

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Publication number
EP1989705B1
EP1989705B1 EP07731604A EP07731604A EP1989705B1 EP 1989705 B1 EP1989705 B1 EP 1989705B1 EP 07731604 A EP07731604 A EP 07731604A EP 07731604 A EP07731604 A EP 07731604A EP 1989705 B1 EP1989705 B1 EP 1989705B1
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gain
adaptive excitation
long
error indication
decoder
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French (fr)
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EP1989705A2 (de
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Balazs Kovesi
David Virette
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Orange SA
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France Telecom SA
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    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/04—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
    • G10L19/08—Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters
    • 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
    • G10L19/083—Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters the excitation function being an excitation gain
    • 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

Definitions

  • the present invention relates to a method of limiting adaptive excitation gain in a decoder of an audio signal. It also relates to a decoder of an audio signal encoded by means of an encoder comprising a long-term predictive filter.
  • the invention finds an advantageous application in the field of coding and decoding of digital signals such as audio-frequency signals.
  • the invention is particularly well suited to the transmission of speech and / or audio signals over packet networks, of the VoIP type, for example, to provide an acceptable quality during decoding after a loss of packets, in particular avoiding the saturation of the data.
  • CELP encoder is the G.729 system recommended in the ITU-T, designed for voiceband speech between 300 and 3400 Hz sampled at 8 kHz and transmitted at a fixed rate of 8 kbit / s with frames of 10 ms.
  • the detailed operation of this encoder is specified in the article of A: Salami, C. Laflamme, JP Adoul, A. Kataoka, S. Hayashi, T. Moriya, C. Lamblin, D. Massaloux, S. Proust, P. Kroon and Y. Shoham.
  • CS-ACELP a toll quality 8 kb / s speech coder
  • FIG. 1 (a) is represented a high level view of a G.729 encoder.
  • This figure shows a high-pass filtering 101 of pre-processing for removing frequency signals below 50 Hz.
  • the speech signal S (n) thus filtered is then analyzed by the block 102 to determine a filter ( z) Linear Prediction Coding (LPC), which is transmitted to the multiplexer 104 in the form of an index indexing the quantized vector (QV) in a dictionary.
  • LPC Linear Prediction Coding
  • the original signal S (n) filtered by the filter ⁇ (z) is processed by the block 103 in order to extract the parameters mentioned in the table of the figure 2 . These parameters are then coded and then transmitted to the MUX multiplexer 104.
  • the excitation thus decoded is shaped by the synthesis filter 120 LPC 1 / ⁇ (z) whose coefficients are decoded by the block 119 in the domain of the spectral line pairs (LSF) and interpolated by subframe 5. ms.
  • the reconstructed signal is then processed by an adaptive post-filter 121 and a post-processing high-pass filter 122.
  • the decoder of the Figure 1 (c) therefore relies on the source-filter model to synthesize the signal.
  • the CELP type encoders In the case of excitation from the long-term prediction LTP filter, and in order to generate an excitation signal capable of rapidly following the signal attacks, the CELP type encoders generally allow the choice of a gain g p pitch greater than 1. As a result, the decoder is locally unstable. However, this instability is controlled by the synthesis analysis model which permanently minimizes the difference between the LTP excitation signal and the original target signal.
  • this instability can lead to significant degradation due to the offset between the encoder and the decoder.
  • the gain value g p of pitch not received in a frame is generally replaced by the value of g p in the previous frame, and although the variable nature of the speech signal consists of an alternation periods of voices with a pitch gain close to 1 and unvoiced with a pitch gain of less than 1 allows, in general, to limit the potential problems related to this local instability, it remains nonetheless true that, for certain signals, especially the voiced signals, transmission errors in periodic stationary zones can cause significant damage when, for example, the gain g p of replacement is higher than the actual gain and the affected frame is followed by high gain frames, as happens during attacks. This situation can then quickly lead to a saturation of the LTP filter by cumulative effect linked to the recursive character of long-term predictive filtering.
  • a first solution to this problem is to limit the gp pitch gain to 1, but this constraint has the effect of degrading the performance of the CELP coders for attacks.
  • the technical problem to be solved by the object of the present invention is to propose a method for limiting adaptive excitation gain in a decoder of an audio signal encoded by means of an encoder comprising a long-term predictive filter. , following a loss of transmission frame between said encoder and said decoder, which would limit the adaptive excitation g p gain, or pitch gain, only in the case where an instability LTP filter is actually noted, and to ensure the best possible compromise between the quality of the decoding and its robustness vis-à-vis the frame loss.
  • frame loss is used here to mean the non-reception of a frame as well as the transmission errors in a frame.
  • said arbitrary value is equal to a value of the adaptive excitation gain determined during said lost frame by an error concealment algorithm.
  • said arbitrary value is equal to the value of the adaptive excitation gain for the non-lost frame preceding said lost frame.
  • said arbitrary value is defined from a voicing detection of the previous frame.
  • said arbitrary value is equal to 1, otherwise the arbitrary value is equal to 0.
  • the excitation is composed of a random noise.
  • the method according to the invention has the advantage of modifying the gain g p of pitch only when a possible instability of the LTP filter is detected at the decoder itself and not at the encoder as in known techniques.
  • the method of the invention takes into account both the actual state of the decoder and the exact information on the transmission errors reached.
  • the method, object of the invention can be used autonomously, that is to say in coding structures that do not provide for limiting the pitch gain at the coder.
  • the invention provides that said adaptive excitation gain is supplied to said decoder by an encoder equipped with a gain limitation device.
  • the method according to the invention can therefore also be used in combination with a known taming technique, installed at the encoder.
  • the prior technique makes it possible to limit the long sequences of pitch gains greater than 1. Indeed, such sequences cause a large propagation of the error, forcing the method of the invention to modify the signal over long periods.
  • a too low trigger threshold of the "taming" technique a priori degrades the signal.
  • the invention thus makes it possible to reduce the number of triggers of the "taming" technique a priori by increasing the threshold, because even if this technique does not detect the risk of explosion, the posterior method according to the invention detects it and cure it.
  • the order N of the LTP filter can be taken as 1.
  • the gain g p of adaptive excitation of a long-term predictive filter of order 1 is limited to the value 1 if said parameter of indication of error is greater than said given threshold.
  • the invention provides that a correction factor is applied to the gains g i of adaptive excitation of a long-term predictive filter of order greater than 1 if said error indication parameter is greater than said given threshold.
  • said at least one adaptive excitation gain is limited by a linear function of said given threshold if said error indication parameter is greater than said threshold.
  • the invention also relates to a program comprising instructions recorded on a computer-readable medium for carrying out the steps of the method according to the invention, when said program is executed on a computer.
  • the invention finally relates to a decoder of an audio signal encoded by means of an encoder comprising the features of claim 11.
  • Adaptive excitation depends solely on the past excitation and makes it possible to efficiently model the periodic signals, especially voiced signals, where the excitation itself is repeated almost periodically.
  • the fixed part c (n) brings the innovation in the total excitation to model the difference between the periods, that is to say to correct the error between the adaptive excitation and the prediction residue.
  • this excitation signal is optimized to the encoder using the technique of synthesis analysis.
  • the synthesis filtering of this excitation is thus performed with the quantized filter to check the result that will be obtained at the decoder.
  • This explains why it is possible to use a locally unstable long-term filtering, that is with a value of g p greater than 1, to model a signal attack because the increase in energy due to this instability is controlled. On the other hand, this control is disturbed by the possible frame losses.
  • the error concealment algorithm uses an estimated excitation signal from the past excitation signal.
  • the decoder comprises a processing line of the excitation signal from the demultiplexer 112 constituted by the blocks 211 to 215.
  • This processing line of the decoder thus described also serves to illustrate the main steps of the adaptive excitation gain limiting method according to the invention.
  • a module 213 calculates from the values of the function x t (n) provided by the module 212, an error indication parameter S t .
  • a comparator 214 checks whether the parameter S t does not exceed a certain threshold S 0 . In case of overshoot and if the decoded pitch gain g p is greater than 1, the value of g p is limited, because in this case there is a risk of saturation of the LTP filter.
  • the error indication parameter S t may be the sum of the values of the function x t (n) , or the maximum value, the average or the sum of the squares of these values.
  • the comparator 214 is followed by a discriminator 215 able to determine the value g ' t of the pitch gain to be applied to the block 117 for the current frame, namely the decoded pitch value g p or a limited value.
  • the LTP, P and g p parameters are transmitted for each 5 ms subframe containing 40 samples.
  • the treatment to avoid saturation of the LTP filter, which is the subject of the invention, is also carried out at the rate of the subframes.
  • the error indication parameter S t for example the sum of the function x t (n) , is calculated for each subframe.
  • the memory of the signal x t (n) is updated with the new value g ' t .
  • the pitch gain of the current sub-frame is less than 1 or the value of S t is less than 80, corresponding to a cumulative error in the long-term weak synthesis filter, the value is not modified.
  • the decoded pitch gain and g ' t g t .
  • the long-term filter of the encoder is a filter of order 1.
  • the pseudo-LTP filter used to define the error indication function may be the equivalent filter of order 1 or more advantageously a filter identical to that used in the encoder, in particular of the same order.
  • the equivalent filter of order 1 is always used.
  • the gain g ' t can be calculated in the same way as for a filter of order 1. It then applies the corrective factor g ' t / g e to the gains g i of the higher order filter.

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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)

Claims (11)

  1. Verfahren zum Begrenzen der adaptiven Erregungsverstärkung in einem Decodierer eines Audiosignals, das mittels eines Codierers codiert ist, der ein langfristiges prädiktives Filter enthält, als Folge eines Übertragungsrahmenverlusts zwischen dem Codierer und dem Decodierer, dadurch gekennzeichnet, dass das Verfahren die folgenden Schritte umfasst, die im Decodierer darin bestehen:
    - zu detektieren, ob ein Rahmen korrekt empfangen worden ist oder nicht,
    - eine Fehlerangabefunktion mit der folgenden Form zu erstellen: x t n = e t n + ∑ i g it . x t ⁢ n - P + i i ∈ - N - 1 / 2 , N - 1 / 2
    Figure imgb0012
    wobei:
    - N die Ordnung des langfristigen prädiktiven Filters ist,
    - die Verstärkungen git gleich den adaptiven Erregungsverstärkungen des langfristigen prädiktiven Filters für die empfangenen Rahmen oder gleich den adaptiven Erregungsverstärkungen des langfristigen prädiktiven Filters in dem vorhergehenden Rahmen für die verlorenen Rahmen sind,
    - et(n) für die empfangenen Rahmen 0 ist und für die verlorenen Rahmen 1 ist,
    - P die Periode der adaptiven Erregung ist, die für die empfangenen Rahmen decodiert wurde oder für die verlorenen Rahmen zuletzt korrekt decodiert wurde,
    - im Verlauf der Decodierung Werte der Fehlerangabefunktion zu berechnen,
    - einen Fehlerangabeparameter anhand der Werte der Fehlerangabefunktion zu berechnen,
    - den Fehlerangabeparameter mit wenigstens einem gegebenen Schwellenwert zu vergleichen,
    - eine Beschränkung auf wenigstens eine adaptive Erregungsverstärkung im Fall eines positiven Vergleichs anzuwenden, falls eine Verstärkung, die zu der wenigstens einen adaptiven Erregungsverstärkung äquivalent ist, größer als ein gegebener Wert ist.
  2. Verfahren nach Anspruch 1, dadurch gekennzeichnet, dass die äquivalente Verstärkung die adaptive Erregungsverstärkung gp eines langfristigen prädiktiven Filters der Ordnung 1 ist.
  3. Verfahren nach Anspruch 1, dadurch gekennzeichnet, dass die äquivalente Verstärkung die äquivalente Verstärkung ge eines langfristigen prädiktiven Filters mit einer Ordnung größer als 1 ist.
  4. Verfahren nach einem der Ansprüche 1 bis 3, dadurch gekennzeichnet, dass der Fehlerangabeparameter ein Parameter ist, der die Energie der Fehlerangabefunktion repräsentiert.
  5. Verfahren nach Anspruch 4, dadurch gekennzeichnet, dass der repräsentative Parameter durch die Summe der Werte der Fehlerangabefunktion gegeben ist.
  6. Verfahren nach einem der Ansprüche 1 bis 5, dadurch gekennzeichnet, dass die adaptive Erregungsverstärkung gp eines langfristigen prädiktiven Filters der Ordnung 1 auf den Wert 1 begrenzt wird, falls der Fehlerangabeparameter größer als der gegebene Schwellenwert ist.
  7. Verfahren nach einem der Ansprüche 1 bis 5, dadurch gekennzeichnet, dass auf die adaptiven Erregungsverstärkungen gi eines langfristigen prädiktiven Filters der Ordnung größer als 1 ein Korrekturfaktor angewendet wird, falls der Fehlerangabeparameter größer als der gegebene Schwellenwert ist.
  8. Verfahren nach einem der Ansprüche 1 bis 5, dadurch gekennzeichnet, dass die wenigstens eine adaptive Erregungsverstärkung durch eine lineare Funktion des gegebenen Schwellenwerts begrenzt wird, falls der Fehlerangabeparameter größer als der Schwellenwert ist.
  9. Verfahren nach einem der Ansprüche 1 bis 8, dadurch gekennzeichnet, dass die adaptive Erregungsverstärkung an den Decodierer durch einen Codierer geliefert wird, der mit einer Verstärkungsbegrenzungsvorrichtung ausgerüstet ist.
  10. Programm, das Befehle enthält, die in einem von einem Rechner lesbaren Träger aufgezeichnet sind, um die Schritte des Verfahrens nach den Ansprüchen 1 bis 9 auszuführen, wenn das Programm auf einem Rechner ausgeführt wird.
  11. Decodierer eines Audiosignals, das mittels eines Codierers codiert ist, der ein langfristiges prädiktives Filter enthält, dadurch gekennzeichnet, dass der Decodierer Folgendes umfasst:
    - einen Block (211) für die Detektion von Übertragungsrahmenverlusten,
    - ein Modul (222) zum Berechnen von Werten einer Fehlerangabefunktion, wobei die Fehlerangabefunktion die folgende Form besitzt: x t n = e t n + ∑ i g it . x t ⁢ n - P + i i ∈ - N - 1 / 2 , N - 1 / 2
    Figure imgb0013
    wobei:
    - N die Ordnung des langfristigen prädiktiven Filters ist,
    - die Verstärkungen git gleich den adaptiven Erregungsverstärkungen des langfristigen prädiktiven Filters für die empfangenen Rahmen oder gleich den adaptiven Erregungsverstärkungen des langfristigen prädiktiven Filters im vorhergehenden Rahmen für die verlorenen Rahmen sind,
    - et(n) für die empfangenen Rahmen 0 ist und für die verlorenen Rahmen 1 ist,
    - P die Periode der adaptiven Erregung ist, die für die empfangenen Rahmen decodiert worden ist oder für die verlorenen Rahmen zuletzt korrekt decodiert worden ist,
    - ein Modul (213) zum Berechnen eines Fehlerangabeparameters anhand der Werte der Fehlerangabefunktion,
    - einen Komparator (214) des Fehlerangabeparameters mit wenigstens einem gegebenen Schwellenwert,
    - einen Diskriminator (215), der dafür ausgelegt ist, als Funktion des von dem Komparator (214) gelieferten Ergebnisses einen Wert wenigstens einer adaptiven Erregungsverstärkung, der von dem Decodierer zu verwenden ist, zu bestimmen.
EP07731604A 2006-02-28 2007-02-13 Verfahren zur begrenzung der adaptiven erregungsverstärkung bei einem auidodecoder Not-in-force EP1989705B1 (de)

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FR0650688A FR2897977A1 (fr) 2006-02-28 2006-02-28 Procede de limitation de gain d'excitation adaptative dans un decodeur audio
PCT/FR2007/050779 WO2007099244A2 (fr) 2006-02-28 2007-02-13 Procede de limitation de gain d'excitation adaptative dans un decodeur audio

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EP1989705A2 EP1989705A2 (de) 2008-11-12
EP1989705B1 true EP1989705B1 (de) 2012-08-15

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EP (1) EP1989705B1 (de)
JP (1) JP4988774B2 (de)
KR (1) KR101372460B1 (de)
CN (1) CN101395659B (de)
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CN101138174B (zh) * 2005-03-14 2013-04-24 松下电器产业株式会社 可扩展解码装置和可扩展解码方法
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WO2007000988A1 (ja) * 2005-06-29 2007-01-04 Matsushita Electric Industrial Co., Ltd. スケーラブル復号装置および消失データ補間方法

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EP1989705A2 (de) 2008-11-12
JP2009528563A (ja) 2009-08-06
WO2007099244A2 (fr) 2007-09-07
KR20080102262A (ko) 2008-11-24
JP4988774B2 (ja) 2012-08-01
FR2897977A1 (fr) 2007-08-31
KR101372460B1 (ko) 2014-03-11
CN101395659B (zh) 2012-11-07
CN101395659A (zh) 2009-03-25
US20090204412A1 (en) 2009-08-13
US8180632B2 (en) 2012-05-15
WO2007099244A3 (fr) 2007-10-25

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