EP1944761A1 - Störreduktion in der digitalen Signalverarbeitung - Google Patents

Störreduktion in der digitalen Signalverarbeitung Download PDF

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Publication number
EP1944761A1
EP1944761A1 EP07000716A EP07000716A EP1944761A1 EP 1944761 A1 EP1944761 A1 EP 1944761A1 EP 07000716 A EP07000716 A EP 07000716A EP 07000716 A EP07000716 A EP 07000716A EP 1944761 A1 EP1944761 A1 EP 1944761A1
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EP
European Patent Office
Prior art keywords
signal
lpc
perturbation
speech
coefficients
Prior art date
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Withdrawn
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EP07000716A
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English (en)
French (fr)
Inventor
Christophe Beaugeant
Herve Taddei
Emmanuel Rossignol Thepie Fapi
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.)
Nokia Solutions and Networks GmbH and Co KG
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Siemens Networks GmbH and Co KG
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Publication date
Application filed by Siemens Networks GmbH and Co KG filed Critical Siemens Networks GmbH and Co KG
Priority to EP07000716A priority Critical patent/EP1944761A1/de
Priority to PCT/EP2007/063598 priority patent/WO2008086920A1/en
Publication of EP1944761A1 publication Critical patent/EP1944761A1/de
Withdrawn legal-status Critical Current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
    • 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
    • G10L19/06—Determination or coding of the spectral characteristics, e.g. of the short-term prediction coefficients

Definitions

  • the invention relates to disturbance reduction in digital signal processing.
  • Digital telecommunication systems include speech coding. Speech codecs are definitely perturbed by the presence of noise and echo. Indeed, they are optimized to handle single speech signals.
  • LPC Linear Prediction Coefficients
  • Noise reduction and echo cancellation are historically built as pre-processing before coding the speech.
  • Many solutions reducing the perturbations on PCM signals are available.
  • a state of the art overview can be found in " Combined Noise and Echo Reduction in Hands-Free systems: A Survey” by R. Le Bouquin Jeannippo, P. Scalart, G. Faucon, C. Beaugeant; IEEE Trans. On Speech and Audio Processing; vol.9; Nov 2001; pp 808-820 .
  • Such solutions are efficient when the PCM data is available, so typically if the problem are solved within the terminal itself, before the encoding of the signal.
  • LPC Linear Prediction Coefficients
  • the digital signal y(n) comprises a useful signal s(n) and a perturbation signal p(n).
  • the perturbation signal p(n) derives e.g. from noise or echo and includes everything of y(n) that is not part of the useful signal s(n).
  • the bitstream y e (n) is derived from y(n) by LPC-encoding.
  • LPC Linear Prediction Coefficients
  • Other parameters of the bitstream y e (n) may also be received, like the fixed gain or the adaptive gain of the bitstream y e (n).
  • the complete bitstream y e (n) is received.
  • the autocorrelation matrix ⁇ s of the useful signal s(n), of the autocorrelation matrix ⁇ p , of the perturbation signal p(n) and the LPC A p of the perturbation signal p(n) are estimated.
  • a modified LPC A s is calculated. It is calculated from A y and the estimated ⁇ s , ⁇ p , A p .
  • a modified data stream y e '(n) including the modified LPC A s is output. This data stream can be received by a decoder which decodes the original signal y(n).
  • Codecs for transmission of speech are optimized for speech signals.
  • the addition of noise or of echo to the useful speech signal leads to sub-optimal behaviour of the codecs, which means additive artefacts on the decoded signal and lower quality.
  • the use of LPC coefficients that are influenced by the noise signal makes the quality of the received speech worse. Accordingly, noise and echo are not only adding undesired information to the useful signal, they also lead to sub-optimal behaviour of speech codecs, decreasing all the quality of telecommunication.
  • the steps of estimating ⁇ s , ⁇ p , A p can be based on the residual signal of y e (n) and A y .
  • the residual signal is the signal that is obtained after the LPC filtering.
  • y e (n) comprises of the residual signal and the LPC coefficients.
  • the estimations of ⁇ s , ⁇ p , A p can be done by classical methods, e.g. by frequency analysis of the encoded signal y e (n).
  • the method also comprises a step of a noise reduction on the residual signal of the encoded signal y e (n).
  • a noise reduction technique on residual signals is described in the above-mentioned "Compressed Domain Noise Reduction and Echo Suppression for Network Speech".
  • the invention described here provides a solution to achieve a reduction of perturbation, like noise and echo, by modifying the LPC coefficients computed during LPC analysis.
  • the Linear Prediction Cofficients (LPC) A y of the signal y e (n) are not received, but calculated from the digital sample signal y(n).
  • the encoding and modifying the LPC coefficients is done only once. Therefore, the residual signal does not need to be encoded and output twice. This improves the speed for encoding and modifying the LPC.
  • the invention may be used with any system based on model of Eq (1) where additive perturbation disturbed the coefficients ⁇ a y ( k ) ⁇ .
  • the method is applicable in a broad range of applications in signal processing.
  • One possible application where the LPC modification would be useful is earthquake detection.
  • the method is especially qualified for signal transmission in telecommunication. Because the different signal characteristics of voice and noise signals, the autocorrelation matrix of the perturbation signal can be estimated relatively precisely. This ensures that the cleaning of the LPC parameters is made successful.
  • the invention also relates to a digital signal transmission apparatus that performs the inventive method.
  • Such an apparatus comprises means for receiving the LPC coefficients A y , for the estimation of ⁇ s , ⁇ p , A p , for the calculation and output of A p .
  • DSP digital signal processor
  • a corrupted signal y(n) is the sum of a useful signal s(n) with a perturbation p ( n ).
  • a s A y + ⁇ s - 1 . ⁇ p ⁇ A y - A p
  • the present invention proposes a method based on formula Eq. (20)-(21) or on any formula derived from this equation to obtain the LPC coefficients of the useful signal ( A s ), when the LPC coefficients of the perturbed signal ( A y ) are available.
  • Eq. (20)/(21) require to know the LPC coefficients of the perturbation A p , the correlation matrix of the perturbation ⁇ p and the inverse of the correlation matrix of the useful signal ⁇ s -1 .
  • these entities are not directly available as we place our problem in a scheme where only the perturbed coefficients A y are available. Accordingly A p , ⁇ p and ⁇ s -1 need to be estimated. It results that the invention can be seen as the generic process described below:
  • the LPC A y are available.
  • This process can be applied on speech codec bitstream by applying the following steps:
  • Figure 1 shows an embodiment of a telecommunication system 1 in a signal transmission with modified LPC coefficients.
  • the sender 2 generates the useful signal s(t) by talking. Perturbations generate a perturbation signal p(t) with is added to the useful signal resulting in the signal y(t).
  • the signal y(t) is digitalized in the Analog-Digital-Converter (AD-Converter) 3 which generates a digital signal y(n).
  • the digital signal y(n) is encoded in the encoder to the signal y e (n).
  • the encoding is done with the help of an LPC analysis.
  • the encoded signal y e (n) is transmitted via the transmission block 5 to the decoder 6.
  • the decoder 6 receives the signal y e ' (n) from the transmission block and decodes y e ' (n) to a digital signal y d (n).
  • y e ' (n) is either equal or unequal to y e (n).
  • the transmission block 5 is e.g. a telephone switch, a router or a simple wire.
  • y d (n) is finally DA-converted by the DA-converter to y a (t) which is received as an analog signal by the receiver 8.
  • the modification of the LPC parameters is done in the transmission block 5, whereas in the embodiment of Figure 4 , the encoder 4 directly modifies the LPC parameters.
  • FIG. 2 is a flow chart for the modification of LPC coefficients within the transmission block 5.
  • y e (n) is a bitstream including LPC coefficients. If the encoder uses the AMR codec, the LPC coefficients are transmitted as Line Spectral Pair (LSP). The frames of y e (n) also comprise the parameters pitch delay, fixed codebook index, fixed gain and adaptive gain. The bitstream is computed by the analysis of successive frames, each, each comprising a defined number of samples (thoughy 160). If the signal y e (t) is sampled at a frequency of 8 MHz, in the so-called narrow band, the number of LPC coefficients is chosen to 8 or 10 in current standardized codecs (AMR, EFR, FR). In other words, the codec uses 8th respectively a 10th order linear prediction filter. In Eq. (1) k runs from 1 to 9 respectively from 1 to 11.
  • the sampling frequency is 16 kHz and the number of coefficients is preferably chosen to 16 in current standardized codecs (AMR-WB).
  • the Figure 2 shows the flow chart where the LPC coefficients are extracted from the bitstream y e (n).
  • the bitstream is divided in the LPC coefficients and the rest of the bitstream, including the information needed to decode the residual waveform. Then the estimations of A p , ⁇ p and ⁇ s -1 are applied, taking into account the LPC coefficient as well as eventually additive information from the bitstream.
  • a p is generated by the help of a Voice Activity Detection (VAD).
  • VAD Voice Activity Detection
  • the output of the VAD generates zero if no voice signal is detected in the bitsteam, else the VAD outputs a one.
  • the bitstream of the encoded signal has to be decoded to make the estimation of ⁇ p , ⁇ s .
  • the estimation of this matrices and vectors can also be done on the basis of the codec parameters of the signal y e (n) by interpreting the fixed gain and the adaptive gain.
  • the clean LPC coefficients A s are generated by one of the equations 21 or 22. It should be noticed that the calculated clean LCP A s are an estimation of a LPC of the useful signal s(n). Accordingly, the calculated LPC A s are as good as the estimations for A s , ⁇ p , ⁇ s are.
  • the filter on the LPC A y coefficients is applied to get the clean LPC A s and finally the LPC are replaced in the bitstream by changing each frame by the use of the clean LPC parameters A s .
  • the frames are modified sequentially and sent to the decoder as signals y e '(n).
  • This method of improving the speech signal quality can be done anywhere in the path between the encoder and decoder.
  • the method can be applied in the terminal of the sender, in the terminal of the receiver or in one of the routers telephone switches or gateways between different networks.
  • Figure 3 shows a comparison of transfer functions with non-modified LPC versus modified LPC coefficients.
  • the synthesis LPC filter function can be described by the filter transfer function H(f) in the frequency domain.
  • the graph of Fig. 3 shows a functions H(f) dependent on the frequency f for a non-noisy LPC function and, with the dashed line, for a noisy LPC filter.
  • the transfer function of the noisy LPC filter in our case a non-modified LPC, has more energy but is smoother.
  • Using a LPC that was generated on the basis of a noisy signal worsens the quality of the speech.
  • the modification of the LPC to a clean LPC make it easier for the receiver to understand the received speech and enhance the clarity of the speech.
  • the flow chart of Figure 2 may be extended by an additional step which reduces noise on the rest bitstream. This noise reduction is performed after the estimation of A p , ⁇ p and ⁇ s -1 and before the generation the new frames of the bitstream of the signal.
  • One examplary noise reduction technique for the rest bitstream is the method for reducing noise on the codec parameters pitch gain and codebook gain described in the above-mentioned "Compressed Domain Noise Reduction and Echo Suppression for Network Speech".
  • Figure 4 shows a second embodiment of the modification of LPC coeffients.
  • the function is included in the encoder 4.
  • the LPC coefficient are computed by an analysis of successive weighted frames.
  • the Levinson-Durbin algorithm permits to get the LPC coefficient from the sample y(n) of the analysis frame.
  • Our method maybe placed as a postfilter of the computation blocks of the LPC analysis. In this scenario, it enhances the LPC coefficient by reducing the influence of the noise.
  • the needed estimations ( A p , ⁇ p and ⁇ s -1 ) may be done by using the LPC coefficients but also some additive information of sample y(n) as depicted in Fig 4 .
  • the filter of Eq (20)/(21) is applied on the perturbed coefficients to get the enhanced ones.
  • the method of improving the speech quality is performed within the encoder.
  • the encoder receives the samples from the A/D-converter. The samples are organized as frames. After windowing a frame, the LPC analysis outputs the LCP coefficients A y .
  • the parameters A p , ⁇ p and ⁇ s -1 are estimated like one of the embodiments described above.
  • LPC coefficients A s are calculated by one of the equations (21) and (22). The encoding of the frame is done with A s .

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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)
  • Quality & Reliability (AREA)
  • Spectroscopy & Molecular Physics (AREA)
  • Compression, Expansion, Code Conversion, And Decoders (AREA)
EP07000716A 2007-01-15 2007-01-15 Störreduktion in der digitalen Signalverarbeitung Withdrawn EP1944761A1 (de)

Priority Applications (2)

Application Number Priority Date Filing Date Title
EP07000716A EP1944761A1 (de) 2007-01-15 2007-01-15 Störreduktion in der digitalen Signalverarbeitung
PCT/EP2007/063598 WO2008086920A1 (en) 2007-01-15 2007-12-10 Disturbance reduction in digital signal processing

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Application Number Priority Date Filing Date Title
EP07000716A EP1944761A1 (de) 2007-01-15 2007-01-15 Störreduktion in der digitalen Signalverarbeitung

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Cited By (1)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
WO2017050972A1 (en) * 2015-09-25 2017-03-30 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. Encoder and method for encoding an audio signal with reduced background noise using linear predictive coding

Families Citing this family (1)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CA2958360C (en) 2010-07-02 2017-11-14 Dolby International Ab Audio decoder

Citations (2)

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Publication number Priority date Publication date Assignee Title
WO2002054744A1 (en) * 2000-12-29 2002-07-11 Nokia Corporation Audio signal quality enhancement in a digital network
WO2002080149A1 (en) * 2001-03-30 2002-10-10 Telefonaktiebolaget Lm Ericsson Noise suppression

Patent Citations (2)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
WO2002054744A1 (en) * 2000-12-29 2002-07-11 Nokia Corporation Audio signal quality enhancement in a digital network
WO2002080149A1 (en) * 2001-03-30 2002-10-10 Telefonaktiebolaget Lm Ericsson Noise suppression

Non-Patent Citations (1)

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Title
CHANDRAN R ET AL: "Compressed domain noise reduction and echo suppression for network speech enhancement", CIRCUITS AND SYSTEMS, 2000. PROCEEDINGS OF THE 43RD IEEE MIDWEST SYMPOSIUM ON AUGUST 8-11, 2000, PISCATAWAY, NJ, USA,IEEE, vol. 1, 8 August 2000 (2000-08-08), pages 10 - 13, XP010558066, ISBN: 0-7803-6475-9 *

Cited By (5)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
WO2017050972A1 (en) * 2015-09-25 2017-03-30 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. Encoder and method for encoding an audio signal with reduced background noise using linear predictive coding
KR20180054823A (ko) * 2015-09-25 2018-05-24 프라운호퍼 게젤샤프트 쭈르 푀르데룽 데어 안겐반텐 포르슝 에. 베. 선형 예측 코딩을 사용하여 감소된 배경 잡음을 갖는 오디오 신호를 인코딩하기 위한 인코더 및 방법
CN108352166A (zh) * 2015-09-25 2018-07-31 弗劳恩霍夫应用研究促进协会 使用线性预测编码以使背景噪声减小的方式对音频信号进行编码的编码器和方法
US10692510B2 (en) 2015-09-25 2020-06-23 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. Encoder and method for encoding an audio signal with reduced background noise using linear predictive coding
CN108352166B (zh) * 2015-09-25 2022-10-28 弗劳恩霍夫应用研究促进协会 使用线性预测编码对音频信号进行编码的编码器和方法

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