-
公开(公告)号：US20160155447A1
公开(公告)日：2016-06-02
申请号：US14392287
申请日：2014-06-26
Inventor： Janusz KLEJSA , Leif Jonas SAMUELSSON , Heiko PURNHAGEN , Glenn N. DICKINS
IPC: G10L19/008 , G10L19/02 , G10L19/002 , G10L19/035
CPC classification number: G10L19/008 , G10L19/002 , G10L19/0204 , G10L19/0212 , G10L19/032 , G10L19/035
Abstract: An encoding system (100) encodes a first (E1) and further (E2, E3) audio signals as a layered bitstream (B), wherein a quantizer for each frequency band of each signal is selected using a rate allocation rule based on signal-specific rate allocation data, a spectral envelope of the signal and a reference level (EnvE1Max), which is determined based on the spectral envelope of the first signal and is not necessarily included in the bitstream. Further disclosed is a decoding system for reconstructing the audio signals based on the bitstream. In embodiments, the bitstream has a basic layer (BE1), which contains data that enable decoding of the first audio signal, and a spatial layer (Bspatial) facilitating decoding of the further audio signal(s). In embodiments, the encoding system prepares the bitstream subject to a basic-layer bitrate constraint and a total bitrate constraint.
Abstract translation: 编码系统（100）将第一（E1）和另外（E2，E3）音频信号编码为分层比特流（B），其中使用基于信号的比特率的速率分配规则来选择每个信号的每个频带的量化器， 特定速率分配数据，信号的频谱包络和基于第一信号的频谱包络确定的参考电平（EnvE1Max），并且不一定包括在比特流中。 还公开了一种用于基于比特流重建音频信号的解码系统。 在实施例中，比特流具有包含能够对第一音频信号进行解码的数据的基本层（BE1）以及便于对其它音频信号进行解码的空间层（B空间）。 在实施例中，编码系统根据基本层比特率约束和总比特率约束准备比特流。
-
公开(公告)号：US20230319190A1
公开(公告)日：2023-10-05
申请号：US17628732
申请日：2020-07-29
Inventor： Glenn N. DICKINS , Christopher Graham HINES , David GUNAWAN , Richard J. CARTWRIGHT , Alan J. SEEFELDT , Daniel Arteaga , Mark R.P. THOMAS , Joshua B. LANDO
CPC classification number: H04M9/082 , G10L2015/223 , G10L15/22
Abstract: An audio processing method may involve receiving output signals from each microphone of a plurality of microphones in an audio environment, the output signals corresponding to a current utterance of a person and determining, based on the output signals, one or more aspects of context information relating to the person, including an estimated current proximity of the person to one or more microphone locations. The method may involve selecting two or more loudspeaker-equipped audio devices based, at least in part, on the one or more aspects of the context information, determining one or more types of audio processing changes to apply to audio data being rendered to loudspeaker feed signals for the audio devices and causing one or more types of audio processing changes to be applied. In some examples, the audio processing changes have the effect of increasing a speech to echo ratio at one or more microphones.
-
公开(公告)号：US20220345820A1
公开(公告)日：2022-10-27
申请号：US17631024
申请日：2020-07-27
Inventor： Glenn N. DICKINS , Richard J. CARTWRIGHT , David GUNAWAN , Christopher Graham HINES , Mark R. P. THOMAS , Alan J. SEEFELDT , Joshua B. LANDO , Carlos Eduardo Medaglia DYONISIO , Daniel ARTEAGA
Abstract: An audio session management method for an audio environment having multiple audio devices may involve receiving, from a first device implementing a first application and by a device implementing an audio session manager, a first route initiation request to initiate a first route for a first audio session. The first route initiation request may indicate a first audio source and a first audio environment destination. The first audio environment destination may correspond with at least a first person in the audio environment, but in some instances will not indicate an audio device. The method may involve establishing a first route corresponding to the first route initiation request. Establishing the first route may involve determining a first location of at least the first person in the audio environment, determining at least one audio device for a first stage of the first audio session and initiating or scheduling the first audio session.
-
公开(公告)号：US20220197592A1
公开(公告)日：2022-06-23
申请号：US17601199
申请日：2020-04-03
Applicant: Dolby Laboratories Licensing Corporation
Inventor： Glenn N. DICKINS , Feng DENG , Michael ECKERT , Craig JOHNSTON , Paul HOLMBERG
IPC: G06F3/16 , H04M3/56 , H04L65/403
Abstract: A communication system, method, and computer-readable medium therefor comprise a media server configured to receive a plurality of audio streams from a corresponding plurality of client devices, the media server including circuitry configured to rank the plurality of audio streams based on a predetermined metric, group a first portion of the plurality of audio streams into a first set, the first portion of the plurality of audio streams being the N highest-ranked audio streams, group a second portion of the plurality of audio streams into a second set, the second portion of the plurality of audio streams being the M lowest-ranked audio streams, forward respective audio streams of the first set to a receiver device, and discard respective audio streams of the second set, wherein N and M are independent integers.
-
公开(公告)号：US20210232360A1
公开(公告)日：2021-07-29
申请号：US17259543
申请日：2019-07-10
Applicant: Dolby Laboratories Licensing Corporation
Inventor： David GUNAWAN , Glenn N. DICKINS
IPC: G06F3/16 , G10L25/78 , G10L25/51 , G10L21/034 , H04R1/08
Abstract: An apparatus and method of transmission control for an audio device. The audio device uses sources other than the microphone to determine nuisance, and uses this to calculate a gain as well as to make the transmit decision. Using the gain results in a more nuanced nuisance mitigation than using the transmit decision on its own.
-
Patent Agency Ranking
公开(公告)号：US20200177837A1
公开(公告)日：2020-06-04
申请号：US16786799
申请日：2020-02-10
Applicant: Dolby Laboratories Licensing Corporation
Inventor： Glenn N. DICKINS , Ludovic Christophe MALFAIT , David GUNAWAN
Abstract: Systems and methods are described for detecting and remedying potential incongruence in a video conference. A camera of a video conferencing system may capture video images of a conference room. A processor of the video conferencing system may identify locations of a plurality of participants within an image plane of a video image. Using face and shape detection, a location of a center point of each identified participant's torso may be calculated. A region of congruence bounded by key parallax lines may be calculated, the key parallax lines being a subset of all parallax lines running through the center points of each identified participant. When the audio device location is not within the region of congruence, audio captured by an audio device may be adjusted to reduce effects of incongruence when the captured audio is replayed at a far end of the video conference.
-
公开(公告)号：US20200092422A1
公开(公告)日：2020-03-19
申请号：US16691487
申请日：2019-11-21
Applicant: DOLBY LABORATORIES LICENSING CORPORATION
Inventor： Glenn N. DICKINS , Richard J. CARTWRIGHT
IPC: H04M3/56 , H04M3/42 , H04M3/22 , G10L21/0232 , G10L21/0316
Abstract: Teleconference audio data including a plurality of individual uplink data packet streams, may be received during a teleconference. Each uplink data packet stream may corresponding to a telephone endpoint used by one or more teleconference participants. The teleconference audio data may be analyzed to determine a plurality of suppressive gain coefficients, which may be applied to first instances of the teleconference audio data during the teleconference, to produce first gain-suppressed audio data provided to the telephone endpoints during the teleconference. Second instances of the teleconference audio data, as well as gain coefficient data corresponding to the plurality of suppressive gain coefficients, may be sent to a memory system as individual uplink data packet streams. The second instances of the teleconference audio data may be less gain-suppressed than the first gain-suppressed audio data.
-
公开(公告)号：US20190392855A1
公开(公告)日：2019-12-26
申请号：US16564532
申请日：2019-09-09
Applicant: Dolby Laboratories Licensing Corporation
Inventor： Dong SHI , Glenn N. DICKINS , David GUNAWAN , Xuejing SUN
IPC: G10L21/0232 , G10L21/0208 , H04M9/08 , G10L21/028 , G10L25/21
Abstract: In an audio processing system (300), a filtering section (350, 400): receives subband signals (410, 420, 430) corresponding to audio content of a reference signal (301) in respective frequency subbands; receives subband signals (411, 421, 431) corresponding to audio content of a response signal (304) in the respective subbands; and forms filtered inband references (412, 422, 432) by applying respective filters (413, 423, 433) to the subband signals of the reference signal. For a frequency subband: filtered crossband references (424, 425) are formed by multiplying, by scalar factors (426, 427), filtered inband references of other subbands; a composite filtered reference (428) is formed by summing the filtered inband reference of the subband (422) and the filtered crossband references; a residual signal (429) 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.
-
公开(公告)号：US20190287548A1
公开(公告)日：2019-09-19
申请号：US16429552
申请日：2019-06-03
Applicant: Dolby Laboratories Licensing Corporation
Inventor： Xuejing SUN , Glenn N. DICKINS
IPC: G10L21/0364 , G10K11/16 , G10L21/0316 , H03G3/30 , H03G3/32 , G10L21/034 , G10L21/0224 , G10L25/78
Abstract: 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.
-
公开(公告)号：US20190045312A1
公开(公告)日：2019-02-07
申请号：US16079071
申请日：2017-02-16
Applicant: Dolby Laboratories Licensing Corporation
Inventor： David GUNAWAN , Glenn N. DICKINS
IPC: H04R29/00 , H04R1/40 , H04R3/00 , G10L21/0232 , H04M3/56
Abstract: Described herein are audio capture systems and methods. One embodiment provides an audio capture system (1) including: microphones (9-11) positioned to capture respective audio signals from different directions or locations within an audio environment; a mixing module (7) configured to mix the audio signals in accordance with a mixing control signal to produce an output audio mix, wherein, upon the detection of vibration activity, the mixing control signal controls the mixing module (7) to selectively temporarily modify one or more of the audio signals to reduce the presence of noise associated with vibration activity in the output audio mix.
-
-
-
-
-
-
-
-
-
Search Results
63
records
(took 0.036 seconds)
