Patents by Inventor David Gunawan
David Gunawan 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: 9373341Abstract: Method for measuring level of speech determined by an audio signal in a manner which corrects for and reduces the effect of modification of the signal by the addition of noise thereto and/or amplitude compression thereof, and a system configured to perform any embodiment of the method. In some embodiments, the method includes steps of generating frequency banded, frequency-domain data indicative of an input speech signal, determining from the data a Gaussian parametric spectral model of the speech signal, and determining from the parametric spectral model an estimated mean speech level and a standard deviation value for each frequency band of the data; and generating speech level data indicative of a bias corrected mean speech level for each frequency band, including using at least one correction value to correct the estimated mean speech level for the frequency band, where each correction value has been predetermined using a reference speech model.Type: GrantFiled: March 21, 2013Date of Patent: June 21, 2016Assignee: Dolby Laboratories Licensing CorporationInventors: David Gunawan, Glenn Dickins
-
Patent number: 9373343Abstract: An audio signal with a temporal sequence of blocks or frames is received or accessed. Features are determined as characterizing aggregately the sequential audio blocks/frames that have been processed recently, relative to current time. The feature determination exceeds a specificity criterion and is delayed, relative to the recently processed audio blocks/frames. Voice activity indication is detected in the audio signal. VAD is based on a decision that exceeds a preset sensitivity threshold and is computed over a brief time period, relative to blocks/frames duration, and relates to current block/frame features. The VAD and the recent feature determination are combined with state related information, which is based on a history of previous feature determinations that are compiled from multiple features, determined over a time prior to the recent feature determination time period. Decisions to commence or terminate the audio signal, or related gains, are outputted based on the combination.Type: GrantFiled: March 21, 2013Date of Patent: June 21, 2016Assignee: Dolby Laboratories Licensing CorporationInventors: Glenn N. Dickins, Zhiwei Shuang, David Gunawan, Xuejing Sun
-
Patent number: 9349384Abstract: In some embodiments, a method for adaptive control of gain applied to an audio signal, including steps of analyzing segments of the signal to identify audio objects (e.g., voices of participants in a voice conference); storing information regarding each distinct identified object; using at least some of the information to determine at least one of a target gain, or a gain change rate for reaching a target gain, for each identified object; and applying gain to segments of the signal indicative of an identified object such that the gain changes (typically, at the gain change rate for the object) from an initial gain to the target gain for the object. The information stored may include a scene description. Aspects of the invention include a system configured (e.g., programmed) to perform any embodiment of the inventive method.Type: GrantFiled: September 11, 2013Date of Patent: May 24, 2016Assignee: Dolby Laboratories Licensing CorporationInventors: David Gunawan, Glenn N. Dickins
-
Publication number: 20160035367Abstract: Improved audio data processing method and systems are provided. Some implementations involve dividing frequency domain audio data into a plurality of subbands and determining amplitude modulation signal values for each of the plurality of subbands. A band-pass filter may be applied to the amplitude modulation signal values in each subband, to produce band-pass filtered amplitude modulation signal values for each subband. The band-pass filter may have a central frequency that exceeds an average cadence of human speech. A gain may be determined for each subband based, at least in part, on a function of the amplitude modulation signal values and the band-pass filtered amplitude modulation signal values. The determined gain may be applied to each subband.Type: ApplicationFiled: March 31, 2014Publication date: February 4, 2016Applicant: DOLBY LABORATORIES LICENSING CORPORATIONInventors: Erwin GOESNAR, Glenn N. DICKINS, David GUNAWAN
-
Publication number: 20160006879Abstract: A method in a soundfield-capturing endpoint and the capturing endpoint that comprises a microphone array capturing soundfield, and an input processor pre-processing and performing auditory scene analysis to detect local sound objects and positions, de-clutter the sound objects, and integrate with auxiliary audio signals to form a de-cluttered local auditory scene that has a measure of plausibility and perceptual continuity. The input processor also codes the resulting de-cluttered auditory scene to form coded scene data comprising mono audio and additional scene data to send to others. The endpoint includes an output processor generating signals for a display unit that displays a summary of the de-cluttered local auditory scene and/or a summary of activity in the communication system from received data, the display including a shaped ribbon display element that has an extent with locations on the extent representing locations and other properties of different sound objects.Type: ApplicationFiled: July 1, 2015Publication date: January 7, 2016Applicant: Dolby Laboratories Licensing CorporationInventors: Glenn N. Dickins, Gary Spittle, David Gunawan, Anthony Tucker
-
Publication number: 20150264314Abstract: A system and method for initiating conference calls with external devices are disclosed. Call participants are sent conference invitation and conference information regarding the designated conference call. This conference information is stored on the participant's external device. When the participants arrive at a conference call location having a conferencing device, the conferencing device is capable of communicating with the external device, initiating communications, exchanging conference information. If the participant is verified and/or authorized, the conference system may send the IP address of the conference device to the conference system to initiate the conference call. In one embodiment, the conference device uses an ultrasound acoustic communication band to initiate the call with the external device on a semi-automated basis. An acoustic signature comprising a pilot sequence for communications synchronization may be generated to facilitate the call.Type: ApplicationFiled: October 11, 2013Publication date: September 17, 2015Applicant: Dolby Laboratories Licensing CorporationInventors: Erwin Goesnar, David Gunawan, Glenn N. Dickins
-
Publication number: 20150228293Abstract: In some embodiments, a method for adaptive control of gain applied to an audio signal, including steps of analyzing segments of the signal to identify audio objects (e.g., voices of participants in a voice conference); storing information regarding each distinct identified object; using at least some of the information to determine at least one of a target gain, or a gain change rate for reaching a target gain, for each identified object; and applying gain to segments of the signal indicative of an identified object such that the gain changes (typically, at the gain change rate for the object) from an initial gain to the target gain for the object. The information stored may include a scene description. Aspects of the invention include a system configured (e.g., programmed) to per form any embodiment of the inventive method.Type: ApplicationFiled: September 11, 2013Publication date: August 13, 2015Applicant: DOLBY LABORATORIES LICENSING CORPORATIONInventors: David Gunawan, Glenn N. Dickins
-
Publication number: 20150215467Abstract: The present document relates to audio communication systems. In particular, the present document relates to the control of the level of audio signals within audio communication systems. A method for leveling a near-end audio signal (211) using a leveling gain (214) is described. The near-end audio signal (211) comprises a sequence of segments, wherein the sequence of segments comprises a current segment and one or more preceding segments.Type: ApplicationFiled: September 9, 2013Publication date: July 30, 2015Applicant: DOLBY LABORATORIES LICENSING CORPORATIONInventors: Yen-Liang Shue, David Gunawan, Glenn N. Dickins
-
Publication number: 20150058010Abstract: Method for measuring level of speech determined by an audio signal in a manner which corrects for and reduces the effect of modification of the signal by the addition of noise thereto and/or amplitude compression thereof, and a system configured to perform any embodiment of the method. In some embodiments, the method includes steps of generating frequency banded, frequency-domain data indicative of an input speech signal, determining from the data a Gaussian parametric spectral model of the speech signal, and determining from the parametric spectral model an estimated mean speech level and a standard deviation value for each frequency band of the data; and generating speech level data indicative of a bias corrected mean speech level for each frequency band, including using at least one correction value to correct the estimated mean speech level for the frequency band, where each correction value has been predetermined using a reference speech model.Type: ApplicationFiled: March 21, 2013Publication date: February 26, 2015Applicant: DOLBY LABORATORIES LICENSING CORPORATIONInventors: David Gunawan, Glenn Dickins
-
Publication number: 20150032446Abstract: An audio signal with a temporal sequence of blocks or frames is received or accessed. Features are determined as characterizing aggregately the sequential audio blocks/frames that have been processed recently, relative to current time. The feature determination exceeds a specificity criterion and is delayed, relative to the recently processed audio blocks/frames. Voice activity indication is detected in the audio signal. VAD is based on a decision that exceeds a preset sensitivity threshold and is computed over a brief time period, relative to blocks/frames duration, and relates to current block/frame features. The VAD and the recent feature determination are combined with state related information, which is based on a history of previous feature determinations that are compiled from multiple features, determined over a time prior to the recent feature determination time period. Decisions to commence or terminate the audio signal, or related gains, are outputted based on the combination.Type: ApplicationFiled: March 21, 2013Publication date: January 29, 2015Applicant: DOLBY LABORATORIES LICENSING CORPORATIONInventors: Glenn N. Dickins, Zhiwei Shuang, David Gunawan, Xuejing Sun
-
Publication number: 20150032447Abstract: A method, an apparatus, and a computer-readable medium configured with instructions that when executed carry out the method for determining a measure of harmonicity. In one embodiment the method includes selecting candidate fundamental frequencies within a range, and for candidate determining a mask or retrieving a pre-calculated mask that has positive value for each frequency that contributed to harmonicity, and negative value for each frequency that contributes to inharmonicity. A candidate harmonicity measure is calculated for each candidate fundamental by summing the product of the mask and the magnitude measure spectrum. The harmonicity measure is selected as the maximum of the candidate harmonicity measures.Type: ApplicationFiled: March 21, 2013Publication date: January 29, 2015Applicant: DOLBY LABORATORIES LICENSING CORPORATIONInventors: David Gunawan, Glenn N. Dickins
-
Publication number: 20140278380Abstract: In some embodiments, a method for modifying noise captured at endpoints of a teleconferencing system, including steps of capturing noise at each endpoint, and modifying the captured noise to generate modified noise having a frequency-amplitude spectrum which matches a target spectrum and a spatial property set which matches a target spatial property set. In other embodiments, a teleconferencing method including steps of: at endpoints of a teleconferencing system, determining audio frames indicative of audio captured at each endpoint, each of a subset of the frames indicative of noise but not a significant level of speech; and at each endpoint, generating modified frames indicative of modified noise having a frequency-amplitude spectrum which matches a target spectrum and a spatial property set which matches a target spatial property set, and generating encoded audio including by encoding the modified frames. Other aspects are systems configured to perform any embodiment of the method.Type: ApplicationFiled: February 27, 2014Publication date: September 18, 2014Applicant: Dolby Laboratories Licensing CorporationInventors: David Gunawan, Glenn N. Dickins, Paul Holmberg, Richard J. Cartwright
-
Publication number: 20140241528Abstract: In one embodiment, a sound field is mapped by extracting spatial angle information, diffusivity information, and optionally, sound level information. The extracted information is mapped for representation in the form of a Riemann sphere, wherein spatial angle varies longitudinally, diffusivity varies latitudinally, and level varies radially along the sphere. A more generalized mapping employs mapping the spatial angle and diffusivity information onto a representative region exhibiting variations in direction of arrival that correspond to the extracted spatial information and variations in distance that correspond to the extracted diffusivity information.Type: ApplicationFiled: February 24, 2014Publication date: August 28, 2014Applicant: DOLBY LABORATORIES LICENSING CORPORATIONInventors: David Gunawan, Dong Shi, Glenn N. Dickins