EP1395981B1 - Einrichtung und verfahren zur verarbeitung eines audiosignals - Google Patents

Einrichtung und verfahren zur verarbeitung eines audiosignals Download PDF

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Publication number
EP1395981B1
EP1395981B1 EP02743323A EP02743323A EP1395981B1 EP 1395981 B1 EP1395981 B1 EP 1395981B1 EP 02743323 A EP02743323 A EP 02743323A EP 02743323 A EP02743323 A EP 02743323A EP 1395981 B1 EP1395981 B1 EP 1395981B1
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Prior art keywords
windows
processing
audio signal
stage
segmentation
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Expired - Lifetime
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EP02743323A
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English (en)
French (fr)
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EP1395981A1 (de
Inventor
Franck Bietrix
Hubert Cadusseau
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Sierra Wireless SA
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Wavecom 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
    • 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
    • G10L19/02—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 spectral analysis, e.g. transform vocoders or subband vocoders
    • G10L19/0212—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 spectral analysis, e.g. transform vocoders or subband vocoders using orthogonal transformation
    • 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
    • G10L2021/02082—Noise filtering the noise being echo, reverberation of the speech

Definitions

  • the present invention relates to the field of audio signal processing.
  • the invention relates, in particular, the reduction or cancellation of noise in an audio signal processed by a digital communication device, for example of the digital telephone type and / or radio mobile phones type hands-free.
  • this problem is remedied by inserting noise attenuators or cancellers, acting on the signal picked up by a microphone, before a specific processing of the audio signal.
  • a cancellation and echo or noise reduction device is inserted between a microphone intended to pick up an audio signal and a device for processing the audio signal.
  • This device improves the useful signal to noise ratio or reduces the echo so that the signal can be processed later under optimized conditions.
  • this technique of the prior art requires a specific dedicated device, which has the disadvantage of resulting in additional costs and increased complexity of use.
  • the noise reduction function based on the use of a Fast Fourier Transform (FFT) applied to a continuous stream of voice samples, is integrated in the digital communication device.
  • FFT Fast Fourier Transform
  • the sample flow is split into 256 sample windows obtained by applying a formatting window, with the windows overlapping by half (the first 128 samples of a window corresponding to the last 128 samples from the previous window).
  • An FFT is applied to each window then the result of the FFT is processed by a function of cancellation or reduction of noise or echo.
  • the result of this function is processed by an inverse fast Fourier transform (or IFFT) to reconstruct a flow of voice samples that can be processed by a voice processing function.
  • IFFT inverse fast Fourier transform
  • the invention in its various aspects is intended to overcome these disadvantages of the prior art.
  • an object of the invention is to provide a method and an audio processing device in a device that allows a reduction in the complexity of a processing based on a mathematical transformation applying to blocks of data while optimizing audio processing applying to audio frames.
  • Another object of the invention is to optimize the integration of processing based on a mathematical transformation and audio processing.
  • An object of the invention is also to optimize the delays of these treatments.
  • Another object of the invention is to reduce the computing power necessary for these treatments.
  • the audio processing steps can be implemented sequentially or in a multitasking environment. Moreover, this implementation is facilitated by the use of memory with a predictable, accurate and economical dimensioning.
  • the method is remarkable in that the second segmentation windows are successive frames.
  • the method is remarkable in that the last sample of a first sequence is also the last sample, after the first step, of the corresponding second sequence.
  • the second stage of audio processing is performed without unnecessary waiting to optimize the overall time of audio processing.
  • the first intermediate window being adapted to the mathematical transformation or transformations (it is notably an attenuation of the second lobe of the window relatively strong while the main lobe remains flat), the quality of the corresponding treatment is optimized.
  • the second intermediate window being rectangular, the corresponding sample processing is simple and effective.
  • the method is remarkable in that the predetermined processing sub-step comprises a reduction or cancellation of noise in the audio signal.
  • the method advantageously combines treatments such as reduction and / or cancellation of noise and / or echo and / or voice recognition in a device (for example of the telephone, personal computer or remote control type) which allows a reduction of the complexity while optimizing the effectiveness of these treatments and / or a strong integration of the device (which allows, as a consequence, a reduction in costs and energy consumption this which is relatively important especially for battery-operated communications devices).
  • a device for example of the telephone, personal computer or remote control type
  • the invention advantageously makes it possible to use one or more mathematical transformations adapted to the first audio processing, these transformations applying to blocks of a size different from the size of the second segmentation windows.
  • the method is remarkable in that the source audio signal is a voice signal.
  • the invention is thus well suited to the second audio processing when it is specific to speech such as, for example, voice coding ("vocoding") and / or voice compression for storage and / or remote transmission.
  • speech such as, for example, voice coding ("vocoding") and / or voice compression for storage and / or remote transmission.
  • the invention relates to a computer program product, which is remarkable in that the program comprises instruction sequences adapted to the implementation of an audio processing method as described above when the program is executed on a computer.
  • FFT and IFFT deal with windows with a power of 2 samples (typically 128 or 256).
  • the speech coding takes into account windows that do not have the same size (typically the voice processing in the context of the GSM considers windows of 160 samples).
  • the voice signal is sampled at a frequency of 8 kHz before being transmitted in 20 ms form. compressed to a recipient.
  • the noise reduction and / or echo cancellation and / or cancellation device processes a window of length 256 which can intersect up to three windows. length 160. It is, among other things, the asynchronism inherent in this state-of-the-art technique that makes these processes complex and requires an oversizing of the memories and the computing power and / or the clock.
  • a DSP Signal Processing Processor
  • the two types of processing are synchronized by systematically matching the end of a cancellation or noise reduction and / or echo window with a voice processing frame and preferably with the end of a voice processing frame.
  • the noise reduction or cancellation windows have a size equal to 256 samples and if the speech processing frames have a size equal to 160 samples, a reduction or echo cancellation window will contain the a full voice processing frame and 96 samples (ie 256 minus 160) from the previous window.
  • Such a window is, for example, obtained by the convolution of a Hanning window of width 97 (denoted Hanning (97)) with a rectangular window of width 160 (denoted Rect (160)).
  • a 256-point FFT is then applied to each 256-sample window synchronized to frames of 160 samples.
  • the implementation of FFT is well known to those skilled in the art and is particularly detailed in the book "Numerical Recipes in C, 2 nd edition” (or in French “Digital Recipes in C language, 2nd edition") written by Press WH, Teukolsky SA, Vetterling WT and Flannery BP and published in 1992 by Cambridge University Press.
  • Blocks of 256 samples are thus treated successively.
  • the first 96 processed samples of the current window are added to the last 96 processed samples from the previous window.
  • the first 160 samples of the current window are transmitted to the vocoder to be processed according to the speech coding methods known per se, in accordance, where appropriate, with the applicable standard.
  • Figure 1 schematically illustrates a general block diagram of a radiotelephone, according to the invention according to a preferred embodiment.
  • register designates in each of the memories mentioned, as well a low capacitance memory area (a few binary data) a large capacity memory area (for storing a program whole or all of a sequence of transaction data).
  • the DSP is particularly suitable for processing Fourier transform and speech coding. It may be used, for example, a DSP core manufactured by the company "DSP GROUP” (registered trademark) under the reference “OAK” (registered trademark).
  • Figure 2 illustrates the successive processing carried out by the radiotelephone of Figure 1, on a voice signal.
  • the noisy signal picked up by the microphone 107 is delivered to the Analog / Digital converter 204 where it is converted into a series of digital samples during a step 204.
  • the sampling is typically done at frequency equal to 8kHz.
  • a step 205 the digital sample sequence is processed.
  • L 'frames (160) of processed samples are encoded by a vocoder according to a method known per se (typically as specified in the GSM standard).
  • "vocoded" frames are formatted by the unit 112 to be transmitted by the radio module 111 according to techniques known per se (for example, according to the GSM standard).
  • FIG. 3 presents a cancellation or noise reduction algorithm implemented in the processing step 205 of FIG. 2.
  • the DSP 104 initializes in the RAM 106, a first block of 96 zero samples corresponding to the last samples received as well as all the variables necessary for the proper operation of the processing 205.
  • the DSP 104 stores in the RAM 106 following the previously received samples a sequence of 160 incoming samples from the converter 108.
  • the DSP 104 applies a segmentation window of length 256 to the sequence of the last 256 received samples. (Note that this window is illustrated below with reference to Figure 7)
  • a mathematical transformation of 256-point FFT type is then applied to the sequence obtained by applying the segmentation window.
  • a noise reduction type processing (specified later with reference to FIG. 8) is applied to the sequence resulting from the mathematical transformation.
  • a transformation inverse to that of the step 302, of the IFFT type is applied to the processed sequence.
  • the DSP 104 adds, if necessary (that is to say after a first iteration), the last 96 samples of the preceding processed sequence to the first 96 processed samples of the current sequence. .
  • the sequence or frame formed of the first 160 processed current samples is transmitted to the vocoder.
  • the 160 received samples corresponding to the 160 samples transmitted during the step 305 are erased from the memory 106.
  • step 301 is repeated.
  • FIG. 4 presents a coding of the speech, implemented in step 206 of FIG.
  • the DSP 104 initializes in the RAM 106, all the variables necessary for the proper functioning of the coding 206.
  • the DSP 104 stores in the RAM 106 a frame of 160 samples transmitted during the step 307.
  • the DSP 104 applies speech coding processing to the frame of 160 samples according to a technique known per se.
  • the coded frame is formatted and transmitted to the unit 102 to be transmitted to a recipient.
  • Figure 5 depicts a window of the sample sequences as performed by the treatments of Figures 3 and 4.
  • the curve 500 of the intensity 504 of the signal processed in step 205 is shown as a function of time t 502.
  • the segmentation of the signal is such that the windows 505 (respectively 506) and 507 (respectively 502) are perfectly synchronous.
  • the windows 505 (respectively 506) and 507 (respectively 502) end on the same sample before or after treatment (according to the steps 303, 304 and 305).
  • Figure 6 illustrates a shaping window known per se.
  • the graph giving the amplitude 602 of a window as a function of the rank of a sample 601 is represented by Hanning windows 603 and 604 of length 256 with an overlap of 128.
  • the windowing can in no way be synchronous with a frame segmentation of 160 samples.
  • FIG. 7 illustrates shaping windows 700 and 701, optimized according to the invention (corresponding to the windows 505 and 506 respectively of FIG. 5, but represented more precisely).
  • the graph gives the amplitude 602 of a window as a function of the rank of a sample 601.
  • windows 700 and 701 are Hanning windows obtained by convolution of an intermediate Hanning window of length 97 with a rectangular window of length 160. Thus, with the successive offsets of the windows, 160 parallels are obtained. windows with perfect reconstruction.
  • FIG. 8 specifies step 303 of noise reduction type processing as illustrated with reference to FIG. 3.
  • a frame 801 having 256 spectral components corresponding to a noisy speech signal is processed according to the treatment 303 described hereinafter.
  • the DSP 104 converts the components of the rectangular coordinate frame 801 to polar coordinates to separate the phase of the spectral amplitude.
  • 2 (to which a correction value may be added to improve the speed of convergence of the estimate); P xk m ⁇ P xk ⁇ m - 1 + 1 - ⁇
  • an improved noise reduction algorithm is used. Nevertheless, the introduction of additional delay in this algorithm required a larger memory size for the storage of spectral components with complex values.
  • the coefficient ⁇ is an overestimation factor of the noise that is introduced to obtain better performances of the noise reduction algorithm.
  • ⁇ f corresponds to a spectral value floor.
  • ⁇ f limits the attenuation of the noise reduction filter to a positive value to leave a minimum noise in the signal.
  • the DSP 104 multiplies the amplitude
  • g k (m) .
  • the DSP 104 constructs the signal 809 with reduced noise from the amplitude
  • the signal 809 is then processed according to the inverse Fourier transformation step 304.
  • a person skilled in the art can provide any variant in the application of the invention which is not limited to mobile telephony (in particular GSM, UMTS, IS95, etc.) but extends to any type of device comprising audio coding after or before a mathematical transformation on an incoming audio signal.
  • mobile telephony in particular GSM, UMTS, IS95, etc.
  • any type of device comprising audio coding after or before a mathematical transformation on an incoming audio signal.
  • the invention applies not only to the processing of voice source signals but extends to any type of audio processing.
  • the mathematical transformation applied is in particular of any type applying to blocks of samples of a particular length which is not equal to the size of the frames processed according to an audio processing or which is not a multiple or divider adjacent to this frame size.
  • the invention extends to the case where the size of the audio frames is equal to 160 or more generally is not a power of 2 and where a mathematical transformation applies to block sizes of length 256, 128, 512 or more generally 2 "(where n represents an integer) in particular an FFT, an FHT (of the English" Fast Hadamard Transform “or, in French” Fast Hadamard Transform ”) or a DCT (of the English" Discrete Cosine Transform: “or, in French,” transformed into discrete cosine ”) or variants of these transformations (obtained, for example, by combining one or more of these transformations with one or more other transformations) ...
  • the invention applies to any type of processing associated with the mathematical transformation and performed before or after a speech coding step, particularly in the case of voice recognition or cancellation and / or reduction. echo.
  • the invention is not limited to a purely material implantation but that it can also be implemented in the form of a sequence of instructions of a computer program or any form mixing a material part and a part software.
  • the corresponding instruction sequence can be stored in a removable storage means (such as for example a floppy disk, a CD-ROM or a DVD-ROM) or no, this storage means being partially or completely readable by a computer or a microprocessor.

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  • Engineering & Computer Science (AREA)
  • Physics & Mathematics (AREA)
  • Acoustics & Sound (AREA)
  • Signal Processing (AREA)
  • Health & Medical Sciences (AREA)
  • Audiology, Speech & Language Pathology (AREA)
  • Human Computer Interaction (AREA)
  • Computational Linguistics (AREA)
  • Multimedia (AREA)
  • Quality & Reliability (AREA)
  • Spectroscopy & Molecular Physics (AREA)
  • Telephone Function (AREA)
  • Compression, Expansion, Code Conversion, And Decoders (AREA)
  • Signal Processing Not Specific To The Method Of Recording And Reproducing (AREA)
  • Input Circuits Of Receivers And Coupling Of Receivers And Audio Equipment (AREA)
  • Stereo-Broadcasting Methods (AREA)
  • Noise Elimination (AREA)
  • Soundproofing, Sound Blocking, And Sound Damping (AREA)
  • Stereophonic System (AREA)

Claims (11)

  1. Verfahren zur Verarbeitung eines Audiosignals, umfassend:
    - einen ersten Schritt (205) zur Verarbeitung eines Quellenaudiosignals, der mindestens eine mathematische Transformation einsetzt, die auf erste Mustersequenzen angewandt wird, welche durch Anwendung von ersten Segmentierungsfenstern (505,506,700,701) auf dieses Quellenaudiosignal erhalten werden, und
    - einen zweiten Schritt (206) zur Audioverarbeitung, der auf zweite Mustersequenzen angewandt wird, die durch Anwendung von zweiten Segmentierungsfenstern (507, 508) auf das vom ersten Schritt ausgegebene Signal erhalten werden, wobei die Länge der zweiten Segmentierungsfenster von der Länge der ersten Segmentierungsfenster verschieden ist,
    dadurch gekennzeichnet, dass zwei aufeinander folgende erste Fenster und/oder zwei aufeinander folgende zweite Fenster sich überschneiden, wobei diese Überschneidungen dergestalt sind, dass die Segmentierungen synchron sind und dass die Segmentierungen auf das Ende des ersten und des zweiten Fensters synchronisiert sind.
  2. Verfahren nach Anspruch 1,
    dadurch gekennzeichnet, dass die zweiten Segmentierungsfenster aufeinander folgende Rahmen sind.
  3. Verfahren nach einem der Ansprüche 1 und 2,
    dadurch gekennzeichnet, dass das letzte Muster einer ersten Sequenz nach dem ersten Schritt auch das letzte Muster der entsprechenden zweiten Sequenz ist.
  4. Verfahren nach einem der Ansprüche 1 bis 3,
    dadurch gekennzeichnet, dass das erste Segmentierungsfenster (700,701) ein Fenster mit vollkommener Rekonstruktion ist, die erreicht wird durch Konvolution:
    - eines ersten Zwischenfensters mit vollkommener Rekonstruktion, das spektrale Eigenschaften besitzt, die für die mathematische(n) Transformation(en) geeignet sind, und
    - eines zweiten rechteckigen Zwischenfensters.
  5. Verfahren nach einem der Ansprüche 1 bis 4,
    dadurch gekennzeichnet, dass der erste Verarbeitungsschritt, der auf jede erste Sequenz angewandt wird, unter anderem umfasst:
    - einen vorgegebenen Verarbeitungsteilschritt (303), der auf die erste Sequenz angewandt wird;
    - einen Teilschritt der inversen mathematischen Transformation (304), die auf die verarbeiteten Muster der ersten Sequenz angewandt wird, und
    - einen Schritt zur Addition (305) der Sprachmuster, die aus dem Teilschritt der auf die erste Sequenz angewandten inversen mathematischen Transformation hervorgegangen sind, und der entsprechenden Sprachmuster, die aus dem Teilschritt der auf die vorangegangene erste Sequenz angewandten inversen mathematischen Transformation hervorgegangen sind.
  6. Verfahren nach Anspruch 5,
    dadurch gekennzeichnet, dass der Teilschritt der vorgegebenen Verarbeitung eine Verringerung oder Unterdrückung des Rauschens in diesem Audiosignal beinhaltet.
  7. Verfahren nach einem der Ansprüche 5 und 6,
    dadurch gekennzeichnet, dass der Teilschritt der vorgegebenen Verarbeitung mindestens eine Verarbeitung aus der Gruppe umfasst, welche beinhaltet:
    - eine Verringerung oder Unterdrückung des Echos in diesem Audiosignal;
    - eine Spracherkennung in diesem Audiosignal.
  8. Verfahren nach einem der Ansprüche 1 bis 7,
    dadurch gekennzeichnet, dass die mathematische(n) Transformation(en) der Gruppe angehören, die umfasst:
    - die Fast Fourier Transformationen (FFT) und ihre Varianten,
    - die Fast Hadamard Transformationen (FHT) und ihre Varianten,
    - die Diskreten Cosinus Transformationen (DCT) und ihre Varianten.
  9. Verfahren nach einem der Ansprüche 1 bis 8,
    dadurch gekennzeichnet, dass das Quellenaudiosignal ein Sprachsignal ist.
  10. Vorrichtung zur Verarbeitung eines Audiosignals, umfassend:
    - erste Mittel zur Verarbeitung eines Quellenaudiosignals, die mindestens eine mathematische Transformation einsetzen, die auf erste Mustersequenzen angewandt wird, welche durch Anwendung von ersten Segmentierungsfenstern auf das Quellenaudiosignal erhalten werden, und
    - zweite Mittel zur Audioverarbeitung, die auf zweite Mustersequenzen angewandt werden, die durch Anwendung von zweiten Segmentierungsfenstern auf das vom ersten Schritt ausgegebene Signal erhalten werden, wobei die Länge der zweiten Segmentierungsfenster von der Länge der ersten Segmentierungsfenster verschieden ist,
    dadurch gekennzeichnet, dass zwei aufeinander folgende erste Fenster und/oder zwei aufeinander folgende zweite Fenster sich überschneiden, wobei diese Überschneidungen dergestalt sind, dass die Segmentierungen synchron sind und dass die Segmentierungen auf das Ende des ersten und des zweiten Fensters synchronisiert sind.
  11. Computerprogramm, das über ein Kommunikationsnetzwerk heruntergeladen werden kann und/oder auf einem von einem Computer lesbaren Medium gespeichert ist und/oder von einem Prozessor ausgeführt werden kann,
    dadurch gekennzeichnet, dass es Programmcodeanweisungen zur Durchführung des Verfahrens zur Verarbeitung eines Audiosignals nach mindestens einem der Ansprüche 1 bis 9 umfasst.
EP02743323A 2001-05-15 2002-05-15 Einrichtung und verfahren zur verarbeitung eines audiosignals Expired - Lifetime EP1395981B1 (de)

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FR0106412 2001-05-15
FR0106412A FR2824978B1 (fr) 2001-05-15 2001-05-15 Dispositif et procede de traitement d'un signal audio
PCT/FR2002/001640 WO2002093558A1 (fr) 2001-05-15 2002-05-15 Dispositif et procede de traitement d'un signal audio.

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US (1) US7295968B2 (de)
EP (1) EP1395981B1 (de)
JP (1) JP2004527797A (de)
KR (1) KR20040005965A (de)
CN (1) CN1223991C (de)
AT (1) ATE377244T1 (de)
DE (1) DE60223246D1 (de)
FR (1) FR2824978B1 (de)
IL (2) IL158797A0 (de)
WO (1) WO2002093558A1 (de)

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CN1520589A (zh) 2004-08-11
IL158797A0 (en) 2004-05-12
EP1395981A1 (de) 2004-03-10
DE60223246D1 (de) 2007-12-13
KR20040005965A (ko) 2004-01-16
US20040236572A1 (en) 2004-11-25
US7295968B2 (en) 2007-11-13
WO2002093558A1 (fr) 2002-11-21
CN1223991C (zh) 2005-10-19
FR2824978A1 (fr) 2002-11-22
IL158797A (en) 2009-02-11
FR2824978B1 (fr) 2003-09-19
ATE377244T1 (de) 2007-11-15
JP2004527797A (ja) 2004-09-09

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