Dolby Laboratories Licensing Corporation

Dolby International Ab

Dobly Laboratories Licensing Corporation

Dobly International Ab

Nanjing Twirling Technology Co., Ltd.

Beijing Twirling in Time Co., Ltd.

Xuejing Sun

Glenn N Dickins

Dong Shi

Shen Huang

Kai Lin Li

Zhiwei Shuang

Xiguang Zheng

Guilin Ma

David Gunawan

Heiko Purnhagen

Richard James Cartwright

Chen Zhang

Michael Eckert

Gary Spittle

Huiqun Deng

JiaQuan Huo

Doh-Suk Kim

Bin Cheng

Brendon Costa

Peng Yin

Ming Wang

Taoran Lu

Lie Lu

Mark Stuart Vinton

Harald Mundt

Hannes Muesch

Xingtao Zhang

Lianwu Chen

Paul Francis Holmberg

Sen Xu

Andreas Ehret

Michael P Hollier

Craig Johnston

Poppy Crum

Yen-Liang Shue

Hariharan Ganapathy-Kathirvelu

David P Doney

A method, an apparatus, and logic to post-process raw gains determined by input processing to generate post-processed gains, comprising using one or both of delta gain smoothing and decision-directed gain smoothing. The delta gain smoothing comprises applying a smoothing filter to the raw gain with a smoothing factor that depends on the gain delta: the absolute value of the difference between the raw gain for the current frame and the post-processed gain for a previous frame. The decision-directed gain smoothing comprises converting the raw gain to a signal-to-noise ratio, applying a smoothing filter with a smoothing factor to the signal-to-noise ratio to calculate a smoothed signal-to-noise ratio, and converting the smoothed signal-to-noise ratio to determine the second smoothed gain, with smoothing factor possibly dependent on the gain delta.

Post-processing gains for signal enhancement

Apparatus and methods for controlling a jitter buffer are described. In one embodiment, the apparatus for controlling a jitter buffer includes an inter-talkspurt delay jitter estimator for estimating an offset value of the delay of a first frame in the current talkspurt with respect to the delay of a latest anchor frame in a previous talkspurt, and a jitter buffer controller for adjusting a length of the jitter buffer based on a long term length of the jitter buffer for each frame and the offset value.

Controlling a jitter buffer

Example embodiments disclosed herein relate to orientation-aware surround sound playback. A method for processing audio on an electronic device that includes a plurality of loudspeakers is disclosed, the loudspeakers arranged in more than one dimension of the electronic device. The method includes, responsive to receipt of a plurality of received audio streams, generating a rendering component associated with the plurality of received audio streams, determining an orientation dependent component of the rendering component, processing the rendering component by updating the orientation dependent component according to an orientation of the loudspeakers and dispatching the received audio streams to the plurality of loudspeakers for playback based on the processed rendering component. Corresponding system and computer program products are also disclosed.

Orientation-aware surround sound playback

Some implementations involve analyzing audio packets received during a time interval that corresponds with a conversation analysis segment to determine network jitter dynamics data and conversational interactivity data. The network jitter dynamics data may provide an indication of jitter in a network that relays the audio data packets. The conversational interactivity data may provide an indication of interactivity between participants of a conversation represented by the audio data. A jitter buffer size may be controlled according to the network jitter dynamics data and the conversational interactivity data. The time interval may include a plurality of talkspurts.

Jitter buffer control based on monitoring of delay jitter and conversational dynamics

Example embodiments disclosed herein relate to filter coefficient updating in time domain filtering. A method of processing an audio signal is disclosed. The method includes obtaining a predetermined number of target gains for a first portion of the audio signal by analyzing the first portion of the audio signal. Each of the target gains is corresponding to a subband of the audio signal. The method also includes determining filter coefficients for time domain filtering the first portion of the audio signal so as to approximate a frequency response given by the target gains. The filter coefficients are determined by iteratively selecting at least one target gain from the target gains and updating the filter coefficient based on the selected at least one target gain. Corresponding system and computer program product for processing an audio signal are also disclosed.

Filter coefficient updating in time domain filtering

In an audio processing system (), a filtering section (): receives subband signals () corresponding to audio content of a reference signal () in respective frequency subbands; receives subband signals () corresponding to audio content of a response signal () in the respective subbands; and forms filtered inband references () by applying respective filters () to the subband signals of the reference signal. For a frequency subband: filtered crossband references () are formed by multiplying, by scalar factors (), filtered inband references of other subbands; a composite filtered reference () is formed by summing the filtered inband reference of the subband () and the filtered crossband references; a residual signal () is computed as a difference between the composite filtered reference and the subband signal of the response signal corresponding to the subband; and the scalar factors and the filter applied to the subband signal of the reference signal corresponding to the subband are adjusted based on the residual signal.

Adaptive audio filtering

A service request for communication services for communication clients is received. In response, a communication service network is set up to support the communication services. Routing metadata is generated for each of the communication clients. The routing metadata is to be used by each of the communication clients for sharing service quality information with a respective peer communication client over a light-weight peer-to-peer (P2P) network. The routing metadata is downloaded to each of the communication clients. A communication client may exchange service signaling packets or service data packets over the communication service network. When the communication client determines that there is a problematic region in a bitstream received from the communication server, the communication client can request a peer communication client for a service quality information portion related to the problematic region.

In-service quality monitoring system with intelligent retransmission and interpolation

Various disclosed implementations involve processing and/or playback of a recording of a conference involving a plurality of conference participants. Some implementations disclosed herein involve receiving audio data corresponding to a recording of at least one conference involving a plurality of conference participants. In some examples, only a portion of the received audio data will be selected as playback audio data. The selection process may involve a topic selection process, a talkspurt filtering process and/or an acoustic feature selection process. Some examples involve receiving an indication of a target playback time duration. Selecting the portion of audio data may involve making a time duration of the playback audio data within a threshold time difference of the target playback time duration.

Selective conference digest

A method and an apparatus for locating a sound source are provided. The method includes: obtaining M channels of audio signals of a preset format by microphone arrays located in different planes (S); preprocessing the M channels of audio signals of the preset format, and projecting them onto the same plane, so as to obtain N channels of audio signals, where M&#x2265;N (S); performing a time-frequency transformation on each of the N channels of audio signals, so as to obtain frequency domain signals of the N channels of audio signals (S); further calculating a covariance matrix of the frequency domain signals and performing a smoothing process (S); performing an eigenvalue decomposition of the smoothed covariance matrix (S); estimating the sound source direction according to an eigenvector corresponding to the maximum eigenvalue, so as to obtain a sound source orientation parameter (S).

Method and device for sound source localization

Growing community of inventors

Beijing, China

Xuejing Sun