CA2518684A1 - Multi-channel adaptive speech signal processing with noise reduction - Google Patents
Multi-channel adaptive speech signal processing with noise reduction Download PDFInfo
- Publication number
- CA2518684A1 CA2518684A1 CA002518684A CA2518684A CA2518684A1 CA 2518684 A1 CA2518684 A1 CA 2518684A1 CA 002518684 A CA002518684 A CA 002518684A CA 2518684 A CA2518684 A CA 2518684A CA 2518684 A1 CA2518684 A1 CA 2518684A1
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- signals
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- microphone
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Links
- 230000003044 adaptive effect Effects 0.000 title claims abstract 31
- 238000012545 processing Methods 0.000 title claims abstract 25
- 238000007781 pre-processing Methods 0.000 claims abstract 12
- 230000006978 adaptation Effects 0.000 claims 10
- 238000000034 method Methods 0.000 claims 8
- 230000000903 blocking effect Effects 0.000 claims 4
- 239000011159 matrix material Substances 0.000 claims 4
- 238000003491 array Methods 0.000 claims 1
- 238000004590 computer program Methods 0.000 claims 1
Classifications
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- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; SPEECH OR AUDIO CODING OR DECODING
- G10L21/00—Processing of the speech or voice signal 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
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- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; SPEECH OR AUDIO CODING OR DECODING
- G10L15/00—Speech recognition
- G10L15/20—Speech recognition techniques specially adapted for robustness in adverse environments, e.g. in noise, of stress induced speech
-
- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; 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
- G10L19/008—Multichannel audio signal coding or decoding using interchannel correlation to reduce redundancy, e.g. joint-stereo, intensity-coding or matrixing
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
- H04R3/00—Circuits for transducers, loudspeakers or microphones
- H04R3/005—Circuits for transducers, loudspeakers or microphones for combining the signals of two or more microphones
-
- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; SPEECH OR AUDIO CODING OR DECODING
- G10L21/00—Processing of the speech or voice signal 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
- G10L2021/02161—Number of inputs available containing the signal or the noise to be suppressed
- G10L2021/02166—Microphone arrays; Beamforming
-
- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; SPEECH OR AUDIO CODING OR DECODING
- G10L21/00—Processing of the speech or voice signal 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
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
- H04R2430/00—Signal processing covered by H04R, not provided for in its groups
- H04R2430/20—Processing of the output signals of the acoustic transducers of an array for obtaining a desired directivity characteristic
- H04R2430/25—Array processing for suppression of unwanted side-lobes in directivity characteristics, e.g. a blocking matrix
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
- H04R2499/00—Aspects covered by H04R or H04S not otherwise provided for in their subgroups
- H04R2499/10—General applications
- H04R2499/13—Acoustic transducers and sound field adaptation in vehicles
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
- H04R29/00—Monitoring arrangements; Testing arrangements
- H04R29/004—Monitoring arrangements; Testing arrangements for microphones
- H04R29/005—Microphone arrays
- H04R29/006—Microphone matching
Abstract
The present invention relates to a system for speech signal processing with noise reduction, comprising a microphone array comprising at least two microphones to detect microphone signals, a pre-processing means connected to the microphone array to receive microphone signals comprising means configured to perform a time delay compensation of the microphone signals to obtain pre-processed signals, a first signal processing means connected to the pre-processing means to receive the pre-processed signals comprising a means configured to generate at least one noise reference signal, in particular, one noise reference signal for each microphone signal, second signal processing means connected to the pre-processing means to receive the pre-processed signals comprising an adaptive beam former to obtain a beamformed signal, and an adaptive noise cancelling means configured to reduce noise of the beamformed signal on the basis of the at least one noise reference signal.
Claims (19)
1. System for speech signal processing with noise reduction, comprising a microphone array comprising at least two microphones to detect microphone signals, a pre-processing means connected to the microphone array to receive micro-phone signals comprising means configured to perform a time delay compen-sation of the microphone signals to obtain pre-processed signals, a first signal processing means connected to the pre-processing means to re-ceive the pre-processed signals comprising a means configured to generate at least one noise reference signal, in particular, one noise reference signal for each microphone signal, a second signal processing means connected to the pre-processing means to receive the pre-processed signals comprising an adaptive beam former to ob-tain a beamformed signal, and an adaptive noise cancelling means configured to reduce noise of the beam-formed signal on the basis of the at least one noise reference signal.
2. System according to Claim 1, wherein the means configured to generate the at least one noise reference signal comprises a non-adaptive or an adaptive blocking matrix.
3. System according to any one of Claims 1 or 2, wherein the pre-processing means further comprises an adaptive self-calibration means configured to match the phases and amplitudes of the time delay compensated microphone signals.
4. System according to any one of Claims 1 - 3, further comprising an adaptation control configured to control the adaptation of the adaptive processing means, in particular, the adaptive beamformer and the adaptive noise cancelling means.
5. System according to Claim 4, wherein the adaptation control is configured to initiate adaptation of a first adaptive processing means depending on an adap-tation of a second adaptive processing means, wherein the first and the second adaptive processing means are one of the adaptive beamformer, the adaptive blocking matrix and the adaptive noise cancelling means.
6. System according to Claim 4 or 5, wherein the adaptation control is configured to initiate adaptation in dependence on instantaneous SNR and/or whether speech is detected and/or whether the energy of a detected speech signal ex-ceeds the energy of concurrently detected background noise by some prede-termined value and/or the direction where acoustic signals are coming from.
7. System according to any one of Claims 1 - 6, wherein the pre-processing means and/or the first and/or the second processing means are configured to perform processing in the time domain or in the frequency domain or in the subband frequency domain.
8. System according to any one of Claims 1 - 7, wherein the microphone array comprises at least one directional microphone, in particular, more than one di-rectional microphone pointing in different directions.
9. System according to any one of Claims 1 - 8, wherein the microphone array comprises at least two sub-arrays.
10. System according to any one of Claims 1 - 9, further comprising a frame wherein each microphone of the microphone array is arranged in a predeter-mined, in particular, fixed position in or on the frame.
11. Hands-free system comprising a system for speech signal processing with noise reduction according to one of the preceding claims.
12. Method for speech signal processing with noise reduction, comprising detecting microphone signals by a microphone array comprising at least two microphones, pre-processing the microphone signals, wherein the pre-processing comprises time delay compensating of the microphone signals to obtain pre-processed signals, using the pre-processed signals as input signals for a first signal processing and a second signal processing, wherein the first signal processing comprises generating at least one noise reference signal and the second signal process-ing comprises adaptive beamforming to obtain a beamformed signal, and reducing noise of the beamformed signal based on the noise reference signals using an adaptive noise cancelling means.
13. Method according to Claim 12, wherein the at least one noise reference signal is generated using a non-adaptive or an adaptive blocking matrix.
14. Method according to any one of Claims 12 or 13, wherein the pre-processing of the microphone signals further comprises adaptive self calibrating the time de-lay compensated signals to match the phases and the amplitudes of these sig-nals.
15. The method of any one of Claims 12 - 14, further comprising controlling the adaptation of the adaptive processing steps, in particular, the adaptive beam-forming and the adaptive noise cancelling.
16. The method of Claim 15, wherein controlling comprises adapting a first adap-tive processing means depending on an adaptation of a second adaptive proc-essing means, wherein the first and the second adaptive processing means are one of the adaptive beamformer, the adaptive blocking matrix and the adaptive noise cancelling means.
17. The method of any one of Claims 12 - 16, wherein the adaptation is controlled in dependence on the instantaneous SNR and/or whether speech is detected and/or whether the energy of a detected speech signal exceeds the energy of concurrently detected background noise by some predetermined value and/or the direction where acoustic signals are coming from.
18. The method of any one of Claims 12 - 17, wherein the pre-processing means and/or the first and/or the second processing means perform processing in the time domain or in the frequency domain or in the subband frequency domain.
19. Computer program product, comprising one or more computer readable media having computer-executable instructions for performing the steps of the method according to one of the Claims 12 - 18.
Applications Claiming Priority (2)
Application Number | Priority Date | Filing Date | Title |
---|---|---|---|
EP04022677.1 | 2004-09-23 | ||
EP04022677A EP1640971B1 (en) | 2004-09-23 | 2004-09-23 | Multi-channel adaptive speech signal processing with noise reduction |
Publications (2)
Publication Number | Publication Date |
---|---|
CA2518684A1 true CA2518684A1 (en) | 2006-03-23 |
CA2518684C CA2518684C (en) | 2015-07-21 |
Family
ID=34926680
Family Applications (1)
Application Number | Title | Priority Date | Filing Date |
---|---|---|---|
CA2518684A Expired - Fee Related CA2518684C (en) | 2004-09-23 | 2005-09-09 | Multi-channel adaptive speech signal processing with noise reduction |
Country Status (8)
Country | Link |
---|---|
US (1) | US8194872B2 (en) |
EP (1) | EP1640971B1 (en) |
JP (1) | JP4734070B2 (en) |
KR (1) | KR101239604B1 (en) |
CN (1) | CN1753084B (en) |
AT (1) | ATE405925T1 (en) |
CA (1) | CA2518684C (en) |
DE (1) | DE602004015987D1 (en) |
Cited By (1)
Publication number | Priority date | Publication date | Assignee | Title |
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CN111462743A (en) * | 2020-03-30 | 2020-07-28 | 北京声智科技有限公司 | Voice signal processing method and device |
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CN111462743B (en) * | 2020-03-30 | 2023-09-12 | 北京声智科技有限公司 | Voice signal processing method and device |
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