EP1521242A1 - Procédé de codage de la parole avec réduction de bruit au moyen de la modification du gain du livre de code - Google Patents

Procédé de codage de la parole avec réduction de bruit au moyen de la modification du gain du livre de code Download PDF

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Publication number
EP1521242A1
EP1521242A1 EP03022249A EP03022249A EP1521242A1 EP 1521242 A1 EP1521242 A1 EP 1521242A1 EP 03022249 A EP03022249 A EP 03022249A EP 03022249 A EP03022249 A EP 03022249A EP 1521242 A1 EP1521242 A1 EP 1521242A1
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EP
European Patent Office
Prior art keywords
signal
time interval
noise
fixed gain
gain
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.)
Withdrawn
Application number
EP03022249A
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German (de)
English (en)
Inventor
Christophe Dr. Beaugeant
Nicolas Dütsch
Herbert Dr. Heiss
Hervé Dr. Taddei
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.)
Siemens AG
Original Assignee
Siemens AG
Priority date (The priority date 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 date listed.)
Filing date
Publication date
Application filed by Siemens AG filed Critical Siemens AG
Priority to EP03022249A priority Critical patent/EP1521242A1/fr
Priority to PCT/EP2004/051712 priority patent/WO2005031708A1/fr
Publication of EP1521242A1 publication Critical patent/EP1521242A1/fr
Withdrawn legal-status Critical Current

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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
    • G10L21/00—Speech or voice signal processing techniques to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
    • G10L21/02—Speech enhancement, e.g. noise reduction or echo cancellation
    • G10L21/0208—Noise filtering
    • 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
    • G10L21/00—Speech or voice signal processing techniques to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
    • G10L21/02—Speech enhancement, e.g. noise reduction or echo cancellation
    • G10L21/0208—Noise filtering
    • G10L21/0216—Noise filtering characterised by the method used for estimating noise
    • G10L2021/02168—Noise filtering characterised by the method used for estimating noise the estimation exclusively taking place during speech pauses
    • 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
    • G10L21/00—Speech or voice signal processing techniques to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
    • G10L21/02—Speech enhancement, e.g. noise reduction or echo cancellation
    • G10L21/0208—Noise filtering
    • G10L21/0216—Noise filtering characterised by the method used for estimating noise
    • G10L21/0232—Processing in the frequency domain

Definitions

  • the invention refers to a speech coding method applying noise reduction
  • noise reduction methods have been developed in speech processing. Most of the methods are performed in the frequency domain. They commonly comprise three major components:
  • the suppression rule modifies only the spectral amplitude, not the phase. It has been shown, that there is no need to modify the phase in speech enhancement processing. Nevertheless, this approximation is only valid for a Signal to Noise Ratio (SNR) greater than 6dB. However, this condition is supposed to be satisfied in the majority of the noise reduction algorithms.
  • SNR Signal to Noise Ratio
  • FIG. 1 A scheme of a treatment of a speech signal with noise reduction is depicted in Fig. 1.
  • the speech component s(p), where p denotes a time interval is superimposed with a noise component n(p).
  • n(p) This results in the total signal y(p).
  • the total signal y(p) undergoes a FFT.
  • the result are Fourier components Y(p, f k ), where f k denotes a quantized frequency.
  • the noise reduction NR is applied, thus producing modified Fouriercomponents S(p, S and (p,f k ). This leads after an IFFT to a clean speech signal estimate s and (p).
  • a problem of any spectral weighting noise reduction method is its computational complexity, e.g. if the following steps have to be performed successively:
  • a method for transmitting speech data said speech data are encoded by using an analysis through synthesis method.
  • a synthesised signal is produced for approximating the original signal.
  • the production of the synthesised signal is performed by using at least a fixed codebook with a respective fixed gain and optionally an adaptive codebook and a adaptive gain. The entries of the codebook and the gain are chosen such, that the synthesised signal resembles the original signal.
  • Parameters describing these quantities will be transmitted from a sender to a receiver, e.g. from a near-end speaker to a far-end speaker or vice versa.
  • the invention is based on the idea of modifying the fixed gain determined for the signal containing a noise component and a speech component. Objective of this modification is to obtain a useful estimate of the fixed gain of the speech component or clean signal.
  • the modification is done using a modification factor, which is determined on basis of an estimate of the signal to noise ratio.
  • This signal to noise ratio is calculated consecutively using also the past of this quantity. Thereby the noise component is represented by its fixed gain.
  • One advantage of this procedure is its low computational complexity, particularly if the speech enhancement through noise reduction is done independently from an encoding / decoding unit, e.g. in a certain position within a network, where according to a noise reduction method in the time domain all the steps of decoding, FFT, speech enhancement , IFFT and encoding would have to be performed one after the other. This is not necessary for a noise reduction method according based on modification of parameters
  • Another advantage is that by using the parameters for any modification, a repeated encoding and decoding process, the so called “tandeming" can be avoided, because the modification takes place in the parameter itself. Any tandeming decreases the speech quality. Furthermore the delay due to the additional encoding/decoding, which is e.g. in GSM typically 5 ms can be avoided.
  • the procedure is furthermore also applicable within a communications network.
  • An encoding apparatus set up for performing the above described encoding method includes at least a processing unit.
  • the encoding apparatus may be part of a communications device, e.g. a cellular phone or it may be also situated in a communication network or a component thereof.
  • the codec consists of a multi-rate, that is, the AMR codec can switch between the following bit rates: 12.2, 10.2, 7.95, 7.40, 6.70, 5.90, 5.15 and 4.75 kbit/s, speech codec, a source-controlled rate scheme including a Voice Activity Detection (VAD), a comfort noise generation system and an error concealment mechanism to compensate the effects of transmission errors.
  • VAD Voice Activity Detection
  • Fig. 2 shows the scheme of the AMR encoder. It uses a LTP (long term prediction) filter. It is transformed to an equivalent structure called adaptive codebook. This codebook saves former LPC filtered excitation signals. Instead of subtracting a long-term prediction as the LTP filter does, an adaptive codebook search is done to get an excitation vector from further LPC filtered speech samples. The amplitude of this excitation is adjusted by a gain factor g a .
  • the encoder transforms the speech signal to parameters which describe the speech.
  • these parameters namely the LSF (or LPC) coefficients, the lag of the adaptive codebook, the index of the fixed codebook and the codebook gains, as "speech coding parameters”.
  • the domain will be called “(speech) codec parameter domain” and the signals of this domain are subscripted with frame index $k$.
  • Fig. 3 shows the signal flow of the decoder.
  • the decoder receives the speech coding parameters and computes the excitation signal of the synthesis filter. This excitation signal is the sum of the excitations of the fixed and adaptive codebook scaled with their respective gain factors. After the synthesis-filtering is performed, the speech signal is post-processed.
  • a (total) signal containing clean speech or a speech component and a noise component is encoded.
  • a fixed gain g y (m) of the total signal is calculated.
  • This fixed gain g y (m) of the total signal is subject to a gain modification which bases on a noise gain estimation.
  • an estimate of the fixed gain g and n ( m ) is determined, which is used for the gain modification.
  • the result of the gain modification is an estimate of the fixed gain g and s ( m ) of the clean speech or the speech component.
  • This parameter is transmitted from a sender to a receiver. At the receiver side it is decoded. This procedure will now be described in detail:

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  • Engineering & Computer Science (AREA)
  • Computational Linguistics (AREA)
  • Quality & Reliability (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)
EP03022249A 2003-10-01 2003-10-01 Procédé de codage de la parole avec réduction de bruit au moyen de la modification du gain du livre de code Withdrawn EP1521242A1 (fr)

Priority Applications (2)

Application Number Priority Date Filing Date Title
EP03022249A EP1521242A1 (fr) 2003-10-01 2003-10-01 Procédé de codage de la parole avec réduction de bruit au moyen de la modification du gain du livre de code
PCT/EP2004/051712 WO2005031708A1 (fr) 2003-10-01 2004-08-04 Procede de codage de la parole appliquant une reduction du bruit par une modification du gain de livre de codes

Applications Claiming Priority (1)

Application Number Priority Date Filing Date Title
EP03022249A EP1521242A1 (fr) 2003-10-01 2003-10-01 Procédé de codage de la parole avec réduction de bruit au moyen de la modification du gain du livre de code

Publications (1)

Publication Number Publication Date
EP1521242A1 true EP1521242A1 (fr) 2005-04-06

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EP03022249A Withdrawn EP1521242A1 (fr) 2003-10-01 2003-10-01 Procédé de codage de la parole avec réduction de bruit au moyen de la modification du gain du livre de code

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EP (1) EP1521242A1 (fr)
WO (1) WO2005031708A1 (fr)

Families Citing this family (2)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
EP3701523B1 (fr) * 2017-10-27 2021-10-20 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. Atténuation de bruit au niveau d'un décodeur
CN114023352B (zh) * 2021-11-12 2022-12-16 华南理工大学 一种基于能量谱深度调制的语音增强方法及装置

Citations (4)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US6026356A (en) * 1997-07-03 2000-02-15 Nortel Networks Corporation Methods and devices for noise conditioning signals representative of audio information in compressed and digitized form
WO2001002929A2 (fr) * 1999-07-02 2001-01-11 Tellabs Operations, Inc. Gestion du bruit du domaine code
US20020184010A1 (en) * 2001-03-30 2002-12-05 Anders Eriksson Noise suppression
EP1301018A1 (fr) * 2001-10-02 2003-04-09 Alcatel Méthode et appareille pour modifié un signal digital dons un domain codifié

Patent Citations (4)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US6026356A (en) * 1997-07-03 2000-02-15 Nortel Networks Corporation Methods and devices for noise conditioning signals representative of audio information in compressed and digitized form
WO2001002929A2 (fr) * 1999-07-02 2001-01-11 Tellabs Operations, Inc. Gestion du bruit du domaine code
US20020184010A1 (en) * 2001-03-30 2002-12-05 Anders Eriksson Noise suppression
EP1301018A1 (fr) * 2001-10-02 2003-04-09 Alcatel Méthode et appareille pour modifié un signal digital dons un domain codifié

Non-Patent Citations (2)

* Cited by examiner, † Cited by third party
Title
CHANDRAN R ET AL: "COMPRESSED DOMAIN NOISE REDUCTION AND ECHO SUPPRESSION FOR NETWORK SPEECH ENHANCEMENT", PROCEEDINGS OF THE 43RD. IEEE MIDWEST SYMPOSIUM ON CIRCUITS AND SYSTEMS. MWSCAS 2000. LANSING, MI, NEW YORK, NY: IEEE, US, vol. 1 OF 3, 8 August 2000 (2000-08-08) - 11 August 2000 (2000-08-11), pages 10 - 13, XP002951730, ISBN: 0-7803-6476-7 *
MARTIN R ET AL: "Optimized estimation of spectral parameters for the coding of noisy speech", IEEE INTERNATIONAL CONFERENCE ON ACOUSTICS, SPEECH AND SIGNAL PROCESSING, vol. 3, 5 June 2000 (2000-06-05), Istanbul, Turkey, pages 1479 - 1482, XP010507630 *

Also Published As

Publication number Publication date
WO2005031708A1 (fr) 2005-04-07

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