EP1903833A1 - Rückkopplungsunterdrückung in einem Tonsystem - Google Patents

Rückkopplungsunterdrückung in einem Tonsystem Download PDF

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Publication number
EP1903833A1
EP1903833A1 EP06121006A EP06121006A EP1903833A1 EP 1903833 A1 EP1903833 A1 EP 1903833A1 EP 06121006 A EP06121006 A EP 06121006A EP 06121006 A EP06121006 A EP 06121006A EP 1903833 A1 EP1903833 A1 EP 1903833A1
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EP
European Patent Office
Prior art keywords
sound
signal
sound signal
peak
filter
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Withdrawn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
EP06121006A
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English (en)
French (fr)
Inventor
Deepak Somasundaram
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.)
Phonic Ear Inc
Original Assignee
Phonic Ear Inc
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 Phonic Ear Inc filed Critical Phonic Ear Inc
Priority to EP06121006A priority Critical patent/EP1903833A1/de
Publication of EP1903833A1 publication Critical patent/EP1903833A1/de
Withdrawn legal-status Critical Current

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    • 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
    • H04R3/02—Circuits for transducers, loudspeakers or microphones for preventing acoustic reaction, i.e. acoustic oscillatory feedback
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R27/00—Public address systems
    • 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
    • H04R29/001—Monitoring arrangements; Testing arrangements for loudspeakers

Definitions

  • This invention relates to a method and system for cancellation of acoustical feedback in a sound system, such as a public address sound system or a classroom sound system i.e. assistive learning system.
  • the computer determines a maximum magnitude frequency which is then compared with the magnitude of one or more harmonics and/or subharmonics of the maximum magnitude frequency so as to establish whether the maximum magnitude frequency is greater by a predetermined factor, which would indicate a candidate resonating feedback frequency.
  • the presence of a candidate resonating feedback frequency in a plurality of a predetermined number of successive frequency spectrums indicate the candidate resonant frequency is a resonating feedback frequency to be attenuated.
  • the computer subsequently establishes programmable notch-filters to suppress the resonating feedback frequency.
  • An object of the present invention is to provide a sound system having a simple and effective means for eliminating acoustical feedback, and which requires only a few processing steps.
  • a particular advantage of the present invention is the provision of one or more notch-filters engaging in the signal path softly thus avoiding unnecessary processing artefacts.
  • a particular feature of the present invention is utilisation of an understanding of the statistical distribution of a speech signal in the frequency domain.
  • a sound system for processing acoustical sound comprising a microphone adapted to convert an acoustical sound to an electrical sound signal, a processor adapted to process said sound signal and to generate a processed sound signal, and a speaker adapted to convert said processed sound signal to a processed acoustical sound
  • said processor comprising a calculating unit adapted to calculate a threshold value based on mean magnitude and standard deviation of said sound signal, a FFT unit adapted to transform said sound signal into frequency domain, a peak identification unit adapted to identify a peak in said sound signal in frequency domain and to generate a peak signal, a comparator adapted to compare said threshold value with said peak signal and to generate a control signal identifying frequency of said peak, and a programmable notch-filter unit adapted to receive said control signal and operable to filter out a bandwidth of said sound signal
  • the sound system according to the first aspect of the present invention thus may advantageously utilise the fact that vocal sound has a Gaussian distribution in the time domain and the fact that most energy is of the vocal sound is within one standard deviation from the centre frequency. Hence the sound system may be particularly useful in situations where vocal sound is to be amplified.
  • a is in this context to be construed as one or more, a plurality, or a multiplicity of elements.
  • processor is in this context to be construed as a unit capable of performing a wide range of mathematical processes such as achieved by a microprocessor, a microcontroller, a central processing unit, and/or a digital signal processor.
  • the processor is capable of implementing a transfer function for a sound signal, i.e. providing a required gain in accordance with frequency.
  • the programmable notch-filter according to the first aspect may comprise a leaky integrator operable to control attack time of said programmable notch-filter.
  • the leaky integrator ensures that the notch-filter gradually reduces the sound signal in the frequency domain in a bandwidth of the notch-filter so that artefacts introduced by steep edged notch-filters are avoided.
  • the leaky integrator is computationally efficient for the sound system since it requires only three mathematical operations.
  • the leaky integrator may be operable to control the attack times of the programmable notch-filter in accordance with frequency. That is, the leaky integrator may be adapted to be operable having a first attack time for a first frequency bandwidth and having a second attack time for a second frequency bandwidth. Thus the leaky integrator may be operable having a long attack time in the high frequency part of said sound signal in said frequency domain and having a short attack time in the low frequency part of said sound signal in said frequency domain.
  • attack time is in this context to be construed as the time it takes for the programmable notch-filter from receiving a control signal to fully engaging the filter. Further, “attack time” is in this context to be construed as similar to “release time” being the opposite, namely the time it takes for the programmable notch-filter from receiving a control signal to fully disengaging the filter.
  • the processor according to the first aspect of the present invention may further comprise a counter unit adapted to count a number of frequencies of said sound signal in the frequency domain having magnitudes above said threshold value.
  • the counter unit may be adapted to provide a gain control signal to said processor when the count of said frequencies is above a predetermined number.
  • the processor when receiving the gain control signal may reduce gain throughout the frequency spectrum. This is, particularly, advantageous since by identifying a plurality of frequencies in the sound signal in the frequency domain may demonstrate an acoustical feedback is present.
  • the predetermined number may be in the range between 2 to 10 such as 3.
  • the programmable notch-filter according to the first aspect of the present invention may be operable to establish a number of parallel notch-filters each having a selected operating bandwidth. Obviously, any number of parallel notch-filters may be established each having a selected operating bandwidth and centre frequency; however, the number may be limited to the predetermined number defined above.
  • the programmable notch-filter according to the first aspect of the present invention may be operable to receive the sound signal in the time domain or to receive the sound signal in the frequency domain.
  • the configuration of the programmable notch-filter is thus not limiting to the sound system.
  • the programmable notch-filter according to the first aspect of the present invention may comprise amplifying means adapted to amplify the sound signal in accordance with a predetermined transfer function.
  • the programmable notch-filter may be implemented as an active filter such as an infinite impulse response filter.
  • the method according to the second aspect of the present invention may comprise any elements of the sound system according to the first aspect of the present invention.
  • FIG. 1 shows a block diagram of a sound system according to the first embodiment of the present invention and designated in entirety by reference numeral 100.
  • the sound system 100 comprises a microphone unit 102 converting a sound to an analogue electrical sound signal.
  • the analogue electrical sound signal is communicated through a first communication path 104 to an analogue-to-digital (A/D) converter 106, which converts the analogue electrical sound signal into a digital sound signal.
  • the digital sound signal is communicated through a second communication path 108 to a sound processor 110, which processes the digital signal in accordance with a predetermined transfer function.
  • the second communication path 108 may be a multi-channel bus.
  • the sound processor 110 generates a processed digital signal and communicates this through a third communication path 112 to a digital-to-analogue (D/A) converter 114.
  • the third communication path 112 may be identical to the second communication path 108 i.e. a controlled multi-channel bus.
  • the D/A converter 114 converts the processed digital signal into a processed analogue signal and communicates this through a fourth communication path 116 to a driver 118.
  • the driver 118 is connected to a loud speaker 120 through a fifth communication path 122 and is adapted to drive the loud speaker 112 to present a processed sound.
  • a large part of the sound system 100 may in fact be implemented as integrated elements so that the sound system 100 comprises the microphone unit 102, the speaker unit 120 and a digital signal processor 124.
  • the sound processor 110 as shown in figure 2 comprises an input buffer unit 202 adapted to buffer the digital signals into a number (N) of frames, which are communicated to a FFT unit 204 transforming the frames into frequency domain signals and to a threshold calculation unit 206 adapted to calculate a threshold value from the frame based on mean magnitude (m) and standard deviation ( ⁇ ) of the frames.
  • the calculation of the threshold value may further be adjusted by a bias.
  • the multiplication factor " ⁇ " may have any real number; however the presently preferred number is 2, since this provides for most of the energy of the frame if the frame contain vocal information.
  • the transformed frame is forwarded from the FFT unit 204 to a peak identification unit 208 adapted to identify peaks in the transformed frame and to generate a peak signal for each peak identified in the transformed frame.
  • the peak signal provides information of magnitude and frequency of the peak.
  • the peak identification unit 208 may be configured to identify any number of peaks such as in the range one to ten, for example identifying the three largest peaks in each transformed frame.
  • the peak identification unit 208 may comprise a counter for counting number of peaks and may be adapted to generate a flag signal when the number of peaks identified equals a preselected number.
  • the threshold calculation unit 206 generates a threshold signal for each frame and forwards the threshold signal to a comparator unit 210, which compares the threshold signal to the peak signals received from the peak identification unit 208.
  • the calculation of the mean magnitude of the frequency spectrum in a frame may advantageously be established by a squared addition of the real and imaginary parts of the digital signals. Further, the calculation of the mean magnitude of the digital signals may advantageously be established by a vector magnitude computation such as suggested by Richard G. Lyons in "Understanding Digital Signal Processing” 2nd edition (the ⁇ Max + ⁇ Min method ). It should be understood that any calculation or estimation know to a person skilled in the art may be employed.
  • the comparator unit 210 generates a filter control signal in case the peak signal is greater than the threshold value, which filter control signal is forwarded to a filter/amplifier unit 212.
  • the filter/amplifier unit 212 comprises a programmable notch-filter 214 and an amplifier 216, and is adapted to receive the digital sound signal and filter the digital sound signal according to the filter control signal by means of the programmable notch-filter 214, and to amplify the potentially filtered digital sound signal according to a predetermined transfer function by means of the amplifier 216.
  • amplify is to be construed as increasing or decreasing any particular frequency regions.
  • the filter/amplifier unit 212 may be implemented as an active filter such as an infinite impulse response (IIR) filter.
  • IIR infinite impulse response
  • the programmable notch-filter 214 may comprise a leaky integrator adapted to provide a gradual engagement of the notch-filter 214 so as to avoid artefacts caused by the notch-filter's 214 sharp edges to be generated.
  • the leaky integrator may be operable so that the effect of the notch-filter is engaged and disengaged slowly.
  • the leaky integrator may be implemented by any means know to a person skilled in the art.
  • the comparator 210 In case the peak identification unit 208 identifies a maximum number of peaks within a frame the comparator 210 generates an alert signal, which causes the filter/amplifier unit 212 to reduce gain of the amplifier 216. The effect of the reduction of the gain is monitored on the following frames. That is, if the peak identification unit 208 fails to identify new peaks in the next frames then the gain is gradually increased.
  • FIG 3 shows a block diagram of a sound processor 110' according to a second embodiment of the present invention, which comprises the same elements of the sound processor 110 and these are referenced by the same numerals.
  • the sound processor 110' differs from the sound processor 110 by having the FFT unit 204 transforming the frames into frequency domain signals, which are then communicated to the threshold calculation unit 206 in this case being adapted to calculate a threshold value from the frame based on mean magnitude and standard deviation of the frequency spectrum of the frame.
  • FIG 3 shows a block diagram of a sound processor 110" according to a second embodiment of the present invention.
  • the sound processor 110' comprises the same elements of the sound processor 110 and 110' and these are referenced by the same numerals.
  • the sound processor 110 differs from the sound processor 110' by having the filter/amplifier unit 212 receive frames from the buffer unit 202 and thus perform filtering and amplifying operations on the frames rather than directly on the digital sound signal.
  • FIG 4 shows a further block diagram of a sound processor 110"' according to a third embodiment of the present invention.
  • the sound processor 110"' comprises the same elements of the sound processors 110, 110' and 110" and these are referenced by the same numerals.
  • the sound processor 110 differs from the sound processors 110 and 110' by having a filter/amplification unit 300 receiving the sound signal in the frequency domain from the FFT unit 204 and thus performing the filtering and amplifying operations on the sound signal in the frequency domain rather than on the digital sound signal or on the frames.
  • the filter/amplification unit 300 further comprises an inverse FFT unit 302 for inverting the processed sound signal in the frequency domain back into a processed sound signal in the time domain.

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  • Health & Medical Sciences (AREA)
  • General Health & Medical Sciences (AREA)
  • Otolaryngology (AREA)
  • Physics & Mathematics (AREA)
  • Engineering & Computer Science (AREA)
  • Acoustics & Sound (AREA)
  • Signal Processing (AREA)
  • Circuit For Audible Band Transducer (AREA)
EP06121006A 2006-09-21 2006-09-21 Rückkopplungsunterdrückung in einem Tonsystem Withdrawn EP1903833A1 (de)

Priority Applications (1)

Application Number Priority Date Filing Date Title
EP06121006A EP1903833A1 (de) 2006-09-21 2006-09-21 Rückkopplungsunterdrückung in einem Tonsystem

Applications Claiming Priority (1)

Application Number Priority Date Filing Date Title
EP06121006A EP1903833A1 (de) 2006-09-21 2006-09-21 Rückkopplungsunterdrückung in einem Tonsystem

Publications (1)

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EP1903833A1 true EP1903833A1 (de) 2008-03-26

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EP06121006A Withdrawn EP1903833A1 (de) 2006-09-21 2006-09-21 Rückkopplungsunterdrückung in einem Tonsystem

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Citations (4)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
EP0599450A2 (de) * 1992-11-25 1994-06-01 Matsushita Electric Industrial Co., Ltd. Tonverstärkervorrichtung mit automatischer Unterdrückung akustischer Rückkopplung
US5677987A (en) * 1993-11-19 1997-10-14 Matsushita Electric Industrial Co., Ltd. Feedback detector and suppressor
EP0843502A1 (de) * 1996-11-13 1998-05-20 Yamaha Corporation Schaltung zur Erkennung und Verhinderung von akustischen Rückkopplungen und Verwendung einer derartigen Schaltung in einem Lautsprechersystem
US20030144840A1 (en) * 2002-01-30 2003-07-31 Changxue Ma Method and apparatus for speech detection using time-frequency variance

Patent Citations (4)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
EP0599450A2 (de) * 1992-11-25 1994-06-01 Matsushita Electric Industrial Co., Ltd. Tonverstärkervorrichtung mit automatischer Unterdrückung akustischer Rückkopplung
US5677987A (en) * 1993-11-19 1997-10-14 Matsushita Electric Industrial Co., Ltd. Feedback detector and suppressor
EP0843502A1 (de) * 1996-11-13 1998-05-20 Yamaha Corporation Schaltung zur Erkennung und Verhinderung von akustischen Rückkopplungen und Verwendung einer derartigen Schaltung in einem Lautsprechersystem
US20030144840A1 (en) * 2002-01-30 2003-07-31 Changxue Ma Method and apparatus for speech detection using time-frequency variance

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