Patents by Inventor David Giesbrecht

David Giesbrecht has filed for patents to protect the following inventions. This listing includes patent applications that are pending as well as patents that have already been granted by the United States Patent and Trademark Office (USPTO).

  • Patent number: 9443532
    Abstract: Systems and methods of improved noise reduction using direction of arrival information include: receiving audio signals from two or more acoustic sensors; applying a beamformer module to the audio signals to employ a first noise cancellation algorithm to the audio signals and combine the audio signals into an audio signal; applying a noise reduction post-filter module to the audio signal, the application of which includes: estimating a current noise spectrum of the audio signals after the application of the first noise cancellation algorithm; using spatial information derived from the audio signals received from the two or more acoustic sensors to determine a measured direction-of-arrival by estimating the current time-delay between the acoustic sensor inputs; comparing the measured direction-of-arrival to a target direction-of-arrival; applying a second noise reduction algorithm to the audio signal; and outputting a single audio stream with reduced background noise.
    Type: Grant
    Filed: July 23, 2013
    Date of Patent: September 13, 2016
    Assignee: QSOUND LABS, INC.
    Inventor: David Giesbrecht
  • Publication number: 20140037100
    Abstract: Systems and methods of improved noise reduction include the steps of: receiving an audio signal from two or more acoustic sensors; applying a beamformer to employ a first noise cancellation algorithm; applying a noise reduction post-filter module to the audio signal including: estimating a current noise spectrum of the received audio signal after the application of the first noise cancellation algorithm, wherein the current noise spectrum is estimated using the audio signal received by the second acoustic sensor; determining a punished noise spectrum using the time-average level difference between the audio signal received by the first acoustic sensor and the current noise spectrum; determining a final noise estimate by subtracting the punished noise spectrum from the current noise spectrum; and applying a second noise reduction algorithm to the audio signal received by the first acoustic sensor using the final noise estimate; and outputting an audio stream with reduced background noise.
    Type: Application
    Filed: August 5, 2013
    Publication date: February 6, 2014
    Applicant: QSound Labs, Inc.
    Inventor: David Giesbrecht
  • Publication number: 20140023199
    Abstract: Systems and methods of improved noise reduction using direction of arrival information include: receiving an audio signal from two or more acoustic sensors; applying a beamformer module to employ a first noise cancellation algorithm to the audio signal; applying a noise reduction post-filter module to the audio signal, the application of which includes: estimating a current noise spectrum of the received audio signal after the application of the first noise cancellation algorithm; using spatial information derived from the audio signal received from the two or more acoustic sensors to determine a measured direction-of-arrival by estimating the current time-delay between the acoustic sensor inputs; comparing the measured direction-of-arrival to a target direction-of-arrival; applying a second noise reduction algorithm to the audio signal in proportion to the difference between the measured direction-of-arrival and the target direction-of-arrival; and outputting a single audio stream with reduced background noi
    Type: Application
    Filed: July 23, 2013
    Publication date: January 23, 2014
    Applicant: QSound Labs, Inc.
    Inventor: David Giesbrecht
  • Patent number: 8566086
    Abstract: A method and system for enhancing the frequency response of speech signals are provided. An average speech spectral shape estimate is calculated over time based on the input speech signal. The average speech spectral shape estimate may be calculated in the frequency domain using a first order IIR filtering or “leaky integrators.” Thus, the average speech spectral shape estimate adapts over time to changes in the acoustic characteristics of the voice path or any changes in the electrical audio path that may affect the frequency response of the system. A spectral correction factor may be determined by comparing the average speech spectral shape estimate to a desired target spectral shape. The spectral correction factor may be added (in units of dB) to the spectrum of the input speech signal in order to enhance or adjust the spectrum of the input speech signal toward the desired spectral shape, and an enhanced speech signal re-synthesized from the corrected spectrum.
    Type: Grant
    Filed: June 28, 2005
    Date of Patent: October 22, 2013
    Assignee: QNX Software Systems Limited
    Inventors: David Giesbrecht, Phillip Hetherington
  • Publication number: 20130121498
    Abstract: A handheld device includes: an orientation sensor; an audio processor connected to the orientation sensor and adapted to receive orientation information from the orientation sensor; and a plurality of microphones through which audio content is captured, wherein the audio processor modifies the noise reduction algorithm applied to the audio content captured based, at least in part, on the orientation information.
    Type: Application
    Filed: November 11, 2011
    Publication date: May 16, 2013
    Inventor: David Giesbrecht
  • Patent number: 8311840
    Abstract: A system and methods are provided for extending the frequency bandwidth of a harmonic signal. Harmonic content of a band-limited signal is extended to frequencies outside the signal's passband by performing a non-linear transformation on the complex spectrum of the band-limited signal in the frequency domain. The non-linear transformation may be accomplishes by a linear convolution of the complex spectrum with itself. A system for extending the frequency bandwidth of a harmonic signal includes a signal processor with a forward transform module for transforming a time domain signal into the frequency domain, a non-linear transform module for performing the non-linear transformation on the complex spectrum of the harmonic signal, and a reverse transform module for transforming the extended spectrum of the harmonic signal back into the time domain.
    Type: Grant
    Filed: June 28, 2005
    Date of Patent: November 13, 2012
    Assignee: QNX Software Systems Limited
    Inventors: David Giesbrecht, Phillip Hetherington, Xueman Li
  • Patent number: 8284947
    Abstract: A signal processing system detects reverberation. The system may suppress the reverberation and improve signal quality. The system analyzes frequency bands of an input signal to determine whether reverberation characteristics are present. When reverberation is detected, the system may attenuate the reverberant frequency band to reduce or eliminate the reverberation.
    Type: Grant
    Filed: December 1, 2004
    Date of Patent: October 9, 2012
    Assignee: QNX Software Systems Limited
    Inventors: David Giesbrecht, Phillip Hetherington
  • Patent number: 8170879
    Abstract: A signal enhancement system improves the understandability of speech or other audio signals. The system reinforces selected parts of the signal, may attenuate selected parts of the signal, and may increase SNR. The system includes delay logic, a partitioned adaptive filter, and signal reinforcement logic. The partitioned adaptive filter may track and enhance the fundamental frequency and harmonics in the input signal. The partitioned filter output signals may approximately reproduce the input signal, delayed by an integer multiple of the period of the fundamental frequency of the input signal. The reinforcement logic combines the input signal and the filtered signals to produce an enhanced output signal.
    Type: Grant
    Filed: April 8, 2005
    Date of Patent: May 1, 2012
    Assignee: QNX Software Systems Limited
    Inventors: Rajeev Nongpiur, David Giesbrecht, Phillip Hetherington
  • Patent number: 8063764
    Abstract: A system for detecting and responding to emergency events includes a plurality of local emergency detection and response units positioned in a local area. Each unit includes one or more local sensing agents and a local detection manager. Each local sensing agent is operable to detect emergency events by a change in a given emergency factor in the local area and to convey data representative of the change in the emergency factor to a detection manager which is operable to receive the data and to assign a value to the emergency factor according to the data. A central location controller unit and/or the local emergency detection and response unit are operable for classifying the assigned value of the emergency function to form an assigned value classification and for initiating the local emergency event response agent to implement a response protocol according to the assigned value classification.
    Type: Grant
    Filed: May 22, 2009
    Date of Patent: November 22, 2011
    Assignee: Toronto Rehabilitation Institute
    Inventors: Alex Mihailidis, David Giesbrecht, Jesse Hoey, Tracy Lee, Vicky Young, Melinda Hamill, Jennifer Boger, John Paul Lobos, Babak Taati
  • Patent number: 7894598
    Abstract: A system for limiting a received audio signal in a communication system is provided. The receive audio signal is limited prior to being played over a loudspeaker to insure that the loudspeaker output will not be clipped when picked up by a nearby microphone associated with the communication system. By preventing clipping of the loudspeaker output at the microphone, the transfer function of the loudspeaker-enclosure-microphone system remains linear, facilitating accurate echo cancellation in the communication system.
    Type: Grant
    Filed: December 14, 2004
    Date of Patent: February 22, 2011
    Assignee: Nuance Communications, Inc.
    Inventors: Gerhard Uwe Schmidt, Tim Haulick, Clarence Chu, David Giesbrecht
  • Patent number: 7680652
    Abstract: A signal enhancement system improves the understandability of speech or other audio signals. The system reinforces selected parts of the signal, may attenuate selected parts of the signal, and may increase SNR. The system includes delay logic, an adaptive filter, and signal reinforcement logic. The adaptive filter may track one or more fundamental frequencies in the input signal and outputs a filtered signal. The filtered signal may approximately reproduce the input signal approximately delayed by an integer multiple of the signal's fundamental frequencies. The reinforcement logic combines the input signal and the filtered signal output to produce an enhanced signal output.
    Type: Grant
    Filed: October 26, 2004
    Date of Patent: March 16, 2010
    Assignee: QNX Software Systems (Wavemakers), Inc.
    Inventors: David Giesbrecht, Phillip Hetherington
  • Patent number: 7610196
    Abstract: A signal enhancement system improves the understandability of speech or other audio signals. The system reinforces selected parts of the signal, may attenuate selected parts of the signal, and may increase SNR. The system includes delay logic, a partitioned adaptive filter, and signal reinforcement logic. The partitioned adaptive filter may track and enhance the fundamental frequency and harmonics in the input signal. The partitioned filter output signals may approximately reproduce the input signal, delayed by an integer multiple of the period of the fundamental frequency of the input signal. The reinforcement logic combines the input signal and the filtered signals to produce an enhanced output signal.
    Type: Grant
    Filed: April 8, 2005
    Date of Patent: October 27, 2009
    Assignee: QNX Software Systems (Wavemakers), Inc.
    Inventors: Rajeev Nongpiur, David Giesbrecht, Phillip Hetherington
  • Publication number: 20060293882
    Abstract: A method and system for enhancing the frequency response of speech signals are provided. An average speech spectral shape estimate is calculated over time based on the input speech signal. The average speech spectral shape estimate may be calculated in the frequency domain using a first order IIR filtering or “leaky integrators.” Thus, the average speech spectral shape estimate adapts over time to changes in the acoustic characteristics of the voice path or any changes in the electrical audio path that may affect the frequency response of the system. A spectral correction factor may be determined by comparing the average speech spectral shape estimate to a desired target spectral shape. The spectral correction factor may be added (in units of dB) to the spectrum of the input speech signal in order to enhance or adjust the spectrum of the input speech signal toward the desired spectral shape, and an enhanced speech signal re-synthesized from the corrected spectrum.
    Type: Application
    Filed: June 28, 2005
    Publication date: December 28, 2006
    Inventors: David Giesbrecht, Phillip Hetherington
  • Publication number: 20060293016
    Abstract: A system and methods are provided for extending the frequency bandwidth of a harmonic signal. Harmonic content of a band-limited signal is extended to frequencies outside the signal's passband by performing a non-linear transformation on the complex spectrum of the band-limited signal in the frequency domain. The non-linear transformation may be accomplishes by a linear convolution of the complex spectrum with itself. A system for extending the frequency bandwidth of a harmonic signal includes a signal processor with a forward transform module for transforming a time domain signal into the frequency domain, a non-linear transform module for performing the non-linear transformation on the complex spectrum of the harmonic signal, and a reverse transform module for transforming the extended spectrum of the harmonic signal back into the time domain.
    Type: Application
    Filed: June 28, 2005
    Publication date: December 28, 2006
    Inventors: David Giesbrecht, Phillip Hetherington, Xueman Li
  • Publication number: 20060126822
    Abstract: A system for limiting a received audio signal in a communication system is provided. The receive audio signal is limited prior to being played over a loudspeaker to insure that the loudspeaker output will not be clipped when picked up by a nearby microphone associated with the communication system. By preventing clipping of the loudspeaker output at the microphone, the transfer function of the loudspeaker-enclosure-microphone system remains linear, facilitating accurate echo cancellation in the communication system.
    Type: Application
    Filed: December 14, 2004
    Publication date: June 15, 2006
    Inventors: Gerhard Schmidt, Tim Haulick, Clarence Chu, David Giesbrecht
  • Publication number: 20060115095
    Abstract: A signal processing system detects reverberation. The system may suppress the reverberation and improve signal quality. The system analyzes frequency bands of an input signal to determine whether reverberation characteristics are present. When reverberation is detected, the system may attenuate the reverberant frequency band to reduce or eliminate the reverberation.
    Type: Application
    Filed: December 1, 2004
    Publication date: June 1, 2006
    Inventors: David Giesbrecht, Phillip Hetherington
  • Publication number: 20060098809
    Abstract: A signal enhancement system improves the understandability of speech or other audio signals. The system reinforces selected parts of the signal, may attenuate selected parts of the signal, and may increase SNR. The system includes delay logic, a partitioned adaptive filter, and signal reinforcement logic. The partitioned adaptive filter may track and enhance the fundamental frequency and harmonics in the input signal. The partitioned filter output signals may approximately reproduce the input signal, delayed by an integer multiple of the period of the fundamental frequency of the input signal. The reinforcement logic combines the input signal and the filtered signals to produce an enhanced output signal.
    Type: Application
    Filed: April 8, 2005
    Publication date: May 11, 2006
    Inventors: Rajeev Nongpiur, David Giesbrecht, Phillip Hetherington
  • Publication number: 20060089958
    Abstract: A signal enhancement system improves the understandability of speech or other audio signals. The system reinforces selected parts of the signal, may attenuate selected parts of the signal, and may increase SNR. The system includes delay logic, an adaptive filter, and signal reinforcement logic. The adaptive filter may track one or more fundamental frequencies in the input signal and outputs a filtered signal. The filtered signal may approximately reproduce the input signal approximately delayed by an integer multiple of the signal's fundamental frequencies. The reinforcement logic combines the input signal and the filtered signal output to produce an enhanced signal output.
    Type: Application
    Filed: October 26, 2004
    Publication date: April 27, 2006
    Inventors: David Giesbrecht, Phillip Hetherington
  • Publication number: 20060089959
    Abstract: A signal enhancement system improves the understandability of speech or other audio signals. The system reinforces selected parts of the signal, may attenuate selected parts of the signal, and may increase SNR. The system includes delay logic, a partitioned adaptive filter, and signal reinforcement logic. The partitioned adaptive filter may track and enhance the fundamental frequency and harmonics in the input signal. The partitioned filter output signals may approximately reproduce the input signal, delayed by an integer multiple of the period of the fundamental frequency of the input signal. The reinforcement logic combines the input signal and the filtered signals to produce an enhanced output signal.
    Type: Application
    Filed: April 8, 2005
    Publication date: April 27, 2006
    Inventors: Rajeev Nongpiur, David Giesbrecht, Phillip Hetherington