Dolby Laboratories Licensing Corporation

Dolby International Ab

Sound Id

Hannes Muesch

Glenn N Dickins

David Gunawan

Richard James Cartwright

Michael Eckert

Kai Lin Li

Jeroen Koppens

Dong Shi

Paul Francis Holmberg

Erwin Goesnar

Jeffrey Charles Riedmiller

Xuejing Sun

Regunathan Radhakrishnan

Shen Huang

Kai Li

Caslav V Pavlovic

Nicholas R Michael

Ephram Cohen

Poppy Crum

Chirag Shah

Paul Holmberg

Amad Shamsoddini

Methods for echo estimation or echo management (echo suppression or cancellation) on an input audio signal, with at least one of adaptation of a sparse prediction filter set, modification (for example, truncation) of adapted prediction filter impulse responses, generation of a composite impulse response from adapted prediction filter impulse responses, or use of echo estimation and/or echo management resources in a manner determined at least in part by classification of the input audio signal as being (or not being) echo free. Other aspects are systems configured to perform any embodiment of any of the methods.

Echo estimation and management with adaptation of sparse prediction filter set

Disclosed is an apparatus and method operative to receive packets of media from a network including a receiver unit operative to receive the packets from the network, a jitter buffer data structure for receiving the packets in an ordered queue, the jitter buffer data structure having a tail into which the packets are input; a plurality of heads defining points in the jitter buffer data structure from which the ordered queue of packets are to be played back, the heads comprise an adjustable actual playback head coupled to an actual playback unit and at least one prototype head, each prototype head having associated therewith a target latency a processor having decision logic operable to determine a cost of achieving the associated target latency for each prototype head, wherein the decision logic compares the costs determined for each prototype head to identify a particular target latency and head location for the actual playback head of the buffer and a playback unit coupled to the processor for actual playback of the playback head of the buffer, such that the particular target latency of the jitter buffer data structure is determined at playback of the buffer rather than upon input of the packets into the jitter buffer data structure.

Jitter buffer apparatus and method

Systems and methods are described for determining orientation of an external audio device in a video conference, which may be used to provide congruent multimodal representation for a video conference. A camera of a video conferencing system may be used to detect a potential location of an external audio device within a room in which the video conferencing system is providing a video conference. Within the detected potential location, a visual pattern associated with the external audio device may be identified. Using the identified visual pattern, the video conferencing system may estimate an orientation of the external audio device, the orientation being used by the video conferencing system to provide spatial audio video congruence to a far end audience.

Multimodal spatial registration of devices for congruent multimedia communications

A method for hybrid speech enhancement which employs parametric-coded enhancement (or blend of parametric-coded and waveform-coded enhancement) under some signal conditions and waveform-coded enhancement (or a different blend of parametric-coded and waveform-coded enhancement) under other signal conditions. Other aspects are methods for generating a bitstream indicative of an audio program including speech and other content, such that hybrid speech enhancement can be performed on the program, a decoder including a buffer which stores at least one segment of an encoded audio bitstream generated by any embodiment of the inventive method, and a system or device (e.g., an encoder or decoder) configured (e.g., programmed) to perform any embodiment of the inventive method. At least some of speech enhancement operations are performed by a recipient audio decoder with Mid/Side speech enhancement metadata generated by an upstream audio encoder.

Methods and apparatus for decoding based on speech enhancement metadata

According to one aspect, a method for determining voice activity is disclosed, the method including receiving a frame of an input audio signal, the input audio signal having a sample rate, and spitting the audio signal into a plurality of subbands, the plurality of subbands including at least a lowest subband and a highest subband. The method further comprises filtering the lowest subband to reduce an energy of the lowest subband, estimating a noise level for at least some of the plurality of subbands, and computing a signal-to-noise ratio for at least some of the plurality of subbands. The method also includes determining a speech activity level based at least in part on the computed signal-to-noise ratios and an average of an energy of at least some of the plurality of subbands.

Voice activity detector for audio signals

Disclosed is a method and apparatus operative to process packets of media received from a network including a receiver unit operative, a jitter buffer data structure and a playback head defining a point in the jitter buffer data structure from which the ordered queue of packets are to be played back, and at least one prototype head. Each prototype head having a predetermined latency assigned thereto and defining a point in the jitter buffer data structure from which the ordered queue of packets is being played back containing said latency a processor operable to determine a measure of conversational quality associated with the ordered queue of packets being played back by each prototype head. Also described is a head selector operable to compare the measures of conversational quality associated with the ordered queue of packets being played back by each prototype head to select the prototype head with the highest measure of conversational quality and a playback unit coupled to the playback head.

According to one aspect, a method for detecting voice activity is disclosed, the method including receiving a frame of an input audio signal, the input audio signal having an sample rate; dividing the frame into a plurality of subbands based on the sample rate, the plurality of subbands including at least a lowest subband and a highest subband; filtering the lowest subband with a moving average filter to reduce an energy of the lowest subband; estimating a noise level for each of the plurality of subbands; calculating a signal to noise ratio value for each of the plurality of subbands; and determining a speech activity level of the frame based on an average of the calculated signal to noise ratio values and a weighted average of an energy of each of the plurality of subbands. Other aspects include audio decoders that decode audio that was encoded using the methods described herein.

Hybrid waveform-coded and parametric-coded speech enhancement

Various disclosed implementations involve processing and/or playback of a recording of a conference involving a plurality of conference participants. Some implementations involve receiving or determining conversational dynamics data. One or more variables of a cost function may be based, at least in part, on the conversational dynamics data. The cost function may be a spatial optimization cost function of a vector describing a virtual conference participant position for each of the conference participants in a virtual acoustic space. The virtual acoustic space may be determined relative to a listener's head. The virtual conference participant positions may be assigned according to a solution of the cost function.

Optimized virtual scene layout for spatial meeting playback

Described are: a method, an apparatus, and a tangible computer-readable storage medium comprising instructions to instruct one or more processors to carry out a method. One set of methods is for the transmit side of a communication link and another set of methods is for the receive side. A transmit side method includes assigning one of a set of classifications to media, e.g., voice/audio packets transmitted in a sequence, different classifications impacting differently a measure of perceptual quality calculated at the receive side if packets of the respective classifications are lost. A present packet is sent to the receive side containing the classification of a previous packet.

Call quality estimation by lost packet classification

A method and system for filtering a multi-channel audio signal having a speech channel and at least one non-speech channel, to improve intelligibility of speech determined by the signal. In typical embodiments, the method includes steps of determining at least one attenuation control value indicative of a measure of similarity between speech-related content determined by the speech channel and speech-related content determined by the non-speech channel, and attenuating the non-speech channel in response to the at least one attenuation control value. Typically, the attenuating step includes scaling of a raw attenuation control signal (e.g., a ducking gain control signal) for the non-speech channel in response to the at least one attenuation control value. Some embodiments are a general or special purpose processor programmed with software or firmware and/or otherwise configured to perform filtering in accordance the invention.

Growing community of inventors

Oakland, CA, United States of America

Hannes Muesch