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: 9443532Abstract: 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: GrantFiled: July 23, 2013Date of Patent: September 13, 2016Assignee: QSOUND LABS, INC.Inventor: David Giesbrecht
-
Publication number: 20140037100Abstract: 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: ApplicationFiled: August 5, 2013Publication date: February 6, 2014Applicant: QSound Labs, Inc.Inventor: David Giesbrecht
-
Publication number: 20140023199Abstract: 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 noiType: ApplicationFiled: July 23, 2013Publication date: January 23, 2014Applicant: QSound Labs, Inc.Inventor: David Giesbrecht
-
Patent number: 8566086Abstract: 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: GrantFiled: June 28, 2005Date of Patent: October 22, 2013Assignee: QNX Software Systems LimitedInventors: David Giesbrecht, Phillip Hetherington
-
Publication number: 20130121498Abstract: 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: ApplicationFiled: November 11, 2011Publication date: May 16, 2013Inventor: David Giesbrecht
-
Patent number: 8311840Abstract: 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: GrantFiled: June 28, 2005Date of Patent: November 13, 2012Assignee: QNX Software Systems LimitedInventors: David Giesbrecht, Phillip Hetherington, Xueman Li
-
Patent number: 8284947Abstract: 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: GrantFiled: December 1, 2004Date of Patent: October 9, 2012Assignee: QNX Software Systems LimitedInventors: David Giesbrecht, Phillip Hetherington
-
Patent number: 8170879Abstract: 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: GrantFiled: April 8, 2005Date of Patent: May 1, 2012Assignee: QNX Software Systems LimitedInventors: Rajeev Nongpiur, David Giesbrecht, Phillip Hetherington
-
Patent number: 8063764Abstract: 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: GrantFiled: May 22, 2009Date of Patent: November 22, 2011Assignee: Toronto Rehabilitation InstituteInventors: Alex Mihailidis, David Giesbrecht, Jesse Hoey, Tracy Lee, Vicky Young, Melinda Hamill, Jennifer Boger, John Paul Lobos, Babak Taati
-
Patent number: 7894598Abstract: 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: GrantFiled: December 14, 2004Date of Patent: February 22, 2011Assignee: Nuance Communications, Inc.Inventors: Gerhard Uwe Schmidt, Tim Haulick, Clarence Chu, David Giesbrecht
-
Patent number: 7680652Abstract: 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: GrantFiled: October 26, 2004Date of Patent: March 16, 2010Assignee: QNX Software Systems (Wavemakers), Inc.Inventors: David Giesbrecht, Phillip Hetherington
-
Patent number: 7610196Abstract: 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: GrantFiled: April 8, 2005Date of Patent: October 27, 2009Assignee: QNX Software Systems (Wavemakers), Inc.Inventors: Rajeev Nongpiur, David Giesbrecht, Phillip Hetherington
-
Publication number: 20060293882Abstract: 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: ApplicationFiled: June 28, 2005Publication date: December 28, 2006Inventors: David Giesbrecht, Phillip Hetherington
-
Publication number: 20060293016Abstract: 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: ApplicationFiled: June 28, 2005Publication date: December 28, 2006Inventors: David Giesbrecht, Phillip Hetherington, Xueman Li
-
Publication number: 20060126822Abstract: 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: ApplicationFiled: December 14, 2004Publication date: June 15, 2006Inventors: Gerhard Schmidt, Tim Haulick, Clarence Chu, David Giesbrecht
-
Publication number: 20060115095Abstract: 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: ApplicationFiled: December 1, 2004Publication date: June 1, 2006Inventors: David Giesbrecht, Phillip Hetherington
-
Publication number: 20060098809Abstract: 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: ApplicationFiled: April 8, 2005Publication date: May 11, 2006Inventors: Rajeev Nongpiur, David Giesbrecht, Phillip Hetherington
-
Publication number: 20060089958Abstract: 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: ApplicationFiled: October 26, 2004Publication date: April 27, 2006Inventors: David Giesbrecht, Phillip Hetherington
-
Publication number: 20060089959Abstract: 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: ApplicationFiled: April 8, 2005Publication date: April 27, 2006Inventors: Rajeev Nongpiur, David Giesbrecht, Phillip Hetherington