# US 10,455,326 B2 · Audio feedback reduction utilizing adaptive filters and nonlinear processing > Agents: this data is available as JSON from the patentgrep API and MCP server — see https://patentgrep.com/llms.txt - Number: US 10,455,326 B2 - Document ID: US10455326B2 - Status: Granted - Kind: B2 - Application number: 16/144,202 - Filed: 2018-09-27 - Published: 2019-08-15 - Granted: 2019-10-22 - Earliest claimed: 2017-03-31 - Assignees: Polycom, Inc. - Applicants: Polycom, Inc. - Inventors: Kwan Truong; Peter L. Chu - Web page: https://patentgrep.com/patent/US10455326B2 ## CPC - H04R 3/02: for preventing acoustic reaction {, i.e. acoustic oscillatory feedback (specially adapted for hearing aids H04R25/453)} - G11B 27/031: Electronic editing of digitised analogue information signals, e.g. audio or video signals - H03G 5/025: {Equalizers; Volume or gain control in limited frequency bands} - H03G 5/165: {Equalizers; Volume or gain control in limited frequency bands} - H04R 27/00: Public address systems (circuits for preventing acoustic reaction H04R3/02; circuits for distributing signals to loudspeakers H04R3/12; {monitoring or testing arrangements for public address systems H04R29/007}; amplifiers H03F) - H04R 2227/007: Electronic adaptation of audio signals to reverberation of the listening space for PA - H04R 2430/03: Synergistic effects of band splitting and sub-band processing - H04R: LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; ELECTRIC HEARING AIDS; PUBLIC ADDRESS SYSTEMS (producing sounds with frequency not determined by supply frequency G10K) - G11B: INFORMATION STORAGE BASED ON RELATIVE MOVEMENT BETWEEN RECORD CARRIER AND TRANSDUCER (recording measured values in a way that does not require playback through a transducer G01D9/00; recording or playback apparatus using mechanically marked tape, e.g. punched paper tape, or using unit records, e.g. punched or magnetically marked cards G06K; transferring data from one type of record carrier to another G06K1/18; circuits for coupling output of reproducer to radio receiver H04B1/20; gramophone pick-ups or like acoustic electromechanical transducers or circuits therefor H04R) - H03G: CONTROL OF AMPLIFICATION ## Abstract Traditional audio feedback elimination systems may attempt to reduce the effect of the audio feedback by simply scaling down the audio volume of the signal frequencies that are prone to howling. Other traditional feedback elimination systems may also employ adaptive notch filtering to detect and “notch” the so-called “singing” or “howling” frequencies as they occur in real-time. Such devices may typically have several knobs and buttons needing tuning, for example: the number of adaptive parametric equalizers (PEQs) versus fixed PEQs; attack and decay timers; and/or PEQ bandwidth. Rather than removing the singing frequencies with PEQs, the devices described herein attempt to holistically model the feedback audio and then remove the entire feedback signal. Two advantages of the devices described herein are: 1.) the system can operate at a much larger loop-gain (and hence with a much higher loudspeaker volume); and 2) setup is greatly simplified (i.e., no tuning knobs or buttons). ## Claims ### Claim 1 A system for eliminating audio acoustic feedback, comprising: one or more speakers configured to produce an output audio signal; one or more microphones configured to capture an input audio signal, wherein the input audio signal may comprise at least a portion of the output audio signal; an audio mixer operatively coupled to the one or more microphones; an audio amplifier operatively coupled to the one or more speakers; and a feedback elimination system operatively coupled between the audio mixer and the audio amplifier, the feedback elimination system configured to: analyze, using a subband data structure having a crossover frequency, the input audio signal, wherein the subband data structure uses a first subband decomposition operation and a second subband decomposition operation, and wherein each subband comprises two or more adaptive filter taps; estimate an echo return loss in the input audio signal; detect a double talk condition in the input audio signal; suppress echo in the input audio signal based, at least in part, on the analysis of the input audio signal, the estimated echo return loss, and the detected double talk condition; and synthesize the output audio signal based, at least in part, on the input audio signal with suppressed echo. ### Claim 2 (depends on 1) The system of claim 1, wherein each adaptive filter tap is smoothed over time. ### Claim 3 (depends on 1) The system of claim 1, wherein the crossover frequency is 1000 Hz. ### Claim 4 (depends on 1) The system of claim 1, wherein the first subband decomposition operation is a 128-subband decomposition operation and the second subband decomposition operation is a 256-subband decomposition operation. ### Claim 5 (depends on 1) The system of claim 1, wherein synthesizing the output audio signal based, at least in part, on the input audio signal with suppressed echo, comprises overlap-adding a first frequency range from zero Hz to the crossover frequency with a second frequency range greater than the crossover frequency. ### Claim 6 (depends on 1) The system of claim 1, wherein the two or more adaptive filter taps comprise a transient filter tap and a slow filter tap. ### Claim 7 (depends on 1) The system of claim 1, wherein the first subband decomposition operation is different from the second subband decomposition operation. ### Claim 8 A non-transitory computer readable storage medium storing instructions executable by at least one processor and comprising instructions to: receive an input audio signal from one or more microphones, the input audio signal comprising at least a portion of a first output audio signal produced by one or more speakers, the input audio signal having a frequency range; analyze, using a subband data structure with at least one crossover frequency, the input audio signal, wherein the subband data structure uses a first subband decomposition operation and a second subband decomposition operation, and wherein each subband comprises one or more adaptive filter taps; estimate an echo return loss in the input audio signal; detect a double talk condition in the input audio signal; suppress echo in the input audio signal based, at least in part, on the analysis of the input audio signal, the estimated echo return loss, and the detected double talk condition; and synthesize a second output audio signal based, at least in part, on the input audio signal with suppressed echo. ### Claim 9 (depends on 8) The non-transitory computer readable storage medium of claim 8, wherein each adaptive filter tap is smoothed over time. ### Claim 10 (depends on 8) The non-transitory computer readable storage medium of claim 8, wherein the at least one crossover frequency is 1000 Hz. ### Claim 11 (depends on 8) The non-transitory computer readable storage medium of claim 8, wherein the first subband decomposition operation is a 128-subband decomposition operation and the second subband decomposition operation is a 256-subband decomposition operation. ### Claim 12 (depends on 8) The non-transitory computer readable storage medium of claim 8, wherein the instructions to synthesize the output audio signal based, at least in part, on the input audio signal with suppressed echo, comprise instructions to overlap-add a first frequency range from zero Hz to the crossover frequency with a second frequency range greater than the crossover frequency. ### Claim 13 (depends on 8) The non-transitory computer readable storage medium of claim 8, wherein the one or more adaptive filter taps comprise a transient filter tap and a slow filter tap. ### Claim 14 (depends on 8) The non-transitory computer readable storage medium of claim 8, wherein the instructions further comprise instructions to transmit the second output audio signal to the one or more speakers. ### Claim 15 (depends on 8) The non-transitory computer readable storage medium of claim 8, wherein the first subband decomposition operation is different from the second subband decomposition operation. ### Claim 16 An audio feedback elimination system comprising one or processors configured to: receive an input audio signal from one or more microphones, the input audio signal comprising at least a portion of a first output audio signal produced by one or more speakers, the input audio signal having a frequency range; analyze, using a subband data structure with at least one crossover frequency, the input audio signal, wherein the subband data structure uses a first subband decomposition operation and a second subband decomposition operation, and wherein each subband comprises one or more adaptive filter taps; estimate an echo return loss in the input audio signal; detect a double talk condition in the input audio signal; suppress echo in the input audio signal based, at least in part, on the analysis of the input audio signal, the estimated echo return loss, and the detected double talk condition; and synthesize a second output audio signal based, at least in part, on the input audio signal with suppressed echo. ### Claim 17 (depends on 16) The audio feedback elimination system of claim 16, wherein each adaptive filter tap is smoothed over time. ### Claim 18 (depends on 16) The audio feedback elimination system of claim 16, wherein the at least one crossover frequency is 1000 Hz. ### Claim 19 (depends on 16) The audio feedback elimination system of claim 16, wherein the first subband decomposition operation is a 128-subband decomposition operation and the second subband decomposition operation is a 256-subband decomposition operation. ### Claim 20 (depends on 16) The audio feedback elimination system of claim 16, wherein synthesizing the output audio signal based, at least in part, on the input audio signal with suppressed echo, comprises overlap-adding a first frequency range from zero Hz to the crossover frequency with a second frequency range greater than the crossover frequency.