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公开(公告)号：US11223669B2
公开(公告)日：2022-01-11
申请号：US16927785
申请日：2020-07-13
Applicant: Dolby Laboratories Licensing Corporation
Inventor： Shen Huang , Doh-Suk Kim , Xuejing Sun
Abstract: 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.
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公开(公告)号：US10714098B2
公开(公告)日：2020-07-14
申请号：US16228690
申请日：2018-12-20
Applicant: DOLBY LABORATORIES LICENSING CORPORATION
Inventor： Shen Huang , Michael Eckert , Glenn N. Dickins
IPC: G10L19/005 , G10L19/008 , H04S3/00 , H04L1/00 , G10L19/02
Abstract: Systems and methods for providing forward error correction for a multi-channel audio signal are described. Blocks of an audio stream are buffered into a frame. A transformation can be applied that compacts the energy of each block into a plurality of transformed channels. The energy compaction transform may compact the most energy of a block into the first transformed channel and to compact decreasing amounts of energy into each subsequent transformed channel. The transformed frame may be encoded using any suitable codec and transmitted in a packet over a network. Improved forward error correction may be provided by attaching a low bit rate encoding of the first transformed channel to a subsequent packet. To reconstruct a lost packet, the low bit rate encoding of the first channel for the lost packet may be combined with a packet loss concealment version of the other channels, constructed from a previously-received packet.
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3.发明申请NEAR-END INDICATION THAT THE END OF SPEECH IS RECEIVED BY THE FAR END IN AN AUDIO OR VIDEO CONFERENCE 有权在音频或视频会议末尾接收到的语音结束的最终指示公开(公告)号：US20150237301A1
公开(公告)日：2015-08-20
申请号：US14426134
申请日：2013-09-27
Inventor： Dong Shi , Xuejing Sun , Kai Li , Shen Huang , Harald Mundt , Heiko Purnhagen , Glenn N. Dickins
CPC classification number: H04N7/147 , H04M3/569 , H04M9/082 , H04M2201/12 , H04M2201/14 , H04M2201/38 , H04M2203/258 , H04M2203/352 , H04N7/15
Abstract: Embodiments of client device and method for audio or video conferencing are described. An embodiment includes an offset detecting unit, a configuring unit, an estimator and an output unit. The offset detecting unit detects an offset of speech input to the client device. The configuring unit determines a voice latency from the client device to every far end. The estimator estimates a time when a user at the far end perceives the offset based on the voice latency. The output unit outputs a perceivable signal indicating that a user at the far end perceives the offset based on the time estimated for the far end. The perceivable signal is helpful to avoid collision between parties.
Abstract translation: 描述用于音频或视频会议的客户端设备和方法的实施例。 实施例包括偏移检测单元，配置单元，估计器和输出单元。 偏移检测单元检测输入到客户端设备的语音偏移。 配置单元确定从客户端设备到每个远端的语音延迟。 估计器估计在远端的用户基于语音延迟感知到偏移的时间。 输出单元输出可感知的信号，指示远端的用户基于为远端估计的时间感知偏移。 可感知的信号有助于避免各方之间的冲突。
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公开(公告)号：US12046247B2
公开(公告)日：2024-07-23
申请号：US17702698
申请日：2022-03-23
Applicant: Dolby Laboratories Licensing Corporation
Inventor： Shen Huang , Michael Eckert , Glenn N. Dickins
IPC: G10L19/005 , G10L19/008 , G10L19/02 , H04L1/00 , H04S3/00
CPC classification number: G10L19/005 , G10L19/008 , G10L19/0212 , H04L1/0011 , H04L1/0041 , H04S3/008 , H04S2400/01
Abstract: Systems and methods for providing forward error correction for a multi-channel audio signal are described. Blocks of an audio stream are buffered into a frame. A transformation can be applied that compacts the energy of each block into a plurality of transformed channels. The energy compaction transform may compact the most energy of a block into the first transformed channel and to compact decreasing amounts of energy into each subsequent transformed channel. The transformed frame may be encoded using any suitable codec and transmitted in a packet over a network. Improved forward error correction may be provided by attaching a low bit rate encoding of the first transformed channel to a subsequent packet. To reconstruct a lost packet, the low bit rate encoding of the first channel for the lost packet may be combined with a packet loss concealment version of the other channels, constructed from a previously-received packet.
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公开(公告)号：US11070666B2
公开(公告)日：2021-07-20
申请号：US16659485
申请日：2019-10-21
Applicant: Dolby Laboratories Licensing Corporation
Inventor： Doh-Suk Kim , Shen Huang
Abstract: This disclosure falls into the field of voice communication systems, more specifically it is related to the field of voice quality estimation in a packet based voice communication system. In particular the disclosure provides a method and device for reducing a prediction error of the voice quality estimation by considering the content of lost packets. Furthermore, this disclosure provides a method and device which uses a voice quality estimating algorithm to calculate the voice quality estimate based on an input which is switchable between a first and a second input mode.
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Patent Agency Ranking
公开(公告)号：US10715575B2
公开(公告)日：2020-07-14
申请号：US15170271
申请日：2016-06-01
Applicant: Dolby Laboratories Licensing Corporation
Inventor： Shen Huang , Doh-Suk Kim , Xuejing Sun
Abstract: 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.
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公开(公告)号：US09514755B2
公开(公告)日：2016-12-06
申请号：US14431256
申请日：2013-09-27
Applicant: Dolby Laboratories Licensing Corporation
Inventor： Shen Huang , Xuejing Sun
IPC: G10L19/005 , G10L19/00
CPC classification number: G10L19/005 , G10L19/0017
Abstract: The present document relates to audio signal processing in general, and to the concealment of artifacts that result from loss of audio packets during audio transmission over a packet-switched network, in particular. A method (200) for concealing one or more consecutive lost packets is described. A lost packet is a packet which is deemed to be lost transform-based audio decoder. Each of the one or more lost packets comprises a set of transform coefficients. A set of transform coefficients is used by the transform-based audio decoder to generate a corresponding frame of a time domain audio signal. The method (200) comprises determining (205) for a current lost packet of the one or more lost packets a number of preceding lost packets from the one or more lost packets; wherein the determined number is referred to as a loss position. Furthermore, the method comprises determining a packet loss concealment, referred to as PLC, scheme based on the loss position of the current packet; and determining (204, 207, 208) an estimate of a current frame of the audio signal using the determined PLC scheme (204, 207, 208); wherein the current frame corresponds to the current lost packet.
Abstract translation: 本文件一般涉及音频信号处理，特别涉及在通过分组交换网络的音频传输期间由于音频分组丢失而导致的伪影的隐藏。 描述用于隐藏一个或多个连续丢失分组的方法（200）。 丢失的分组是被认为是丢失的基于变换的音频解码器的分组。 一个或多个丢失分组中的每一个包括一组变换系数。 基于变换的音频解码器使用一组变换系数来生成时域音频信号的相应帧。 所述方法（200）包括：从所述一个或多个丢失分组确定（205）所述一个或多个丢失分组的当前丢失分组的若干先前丢失分组; 其中所确定的数量被称为损失位置。 此外，该方法包括基于当前分组的丢失位置确定称为PLC的分组丢失隐藏; 以及使用所确定的所述PLC方案（204,207,208）确定所述音频信号的当前帧的估计（204,207,208）; 其中当前帧对应于当前丢失分组。
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公开(公告)号：US10516782B2
公开(公告)日：2019-12-24
申请号：US15548245
申请日：2016-02-03
Applicant: DOLBY LABORATORIES LICENSING CORPORATION
Inventor： Richard J. Cartwright , Shen Huang
Abstract: 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. The audio data may include conference participant speech data from multiple endpoints, recorded separately and/or conference participant speech data from a single endpoint corresponding to multiple conference participants and including spatial information for each conference participant of the multiple conference participants. A search of the audio data may be based on one or more search parameters. The search may be a concurrent search for multiple features of the audio data. Instances of conference participant speech may be rendered to at least two different virtual conference participant positions of a virtual acoustic space.
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公开(公告)号：US09626970B2
公开(公告)日：2017-04-18
申请号：US14971401
申请日：2015-12-16
Applicant: Dolby Laboratories Licensing Corporation
Inventor： Shen Huang , Xuejing Sun
CPC classification number: G10L17/005 , G10L15/30 , G10L17/00 , G10L17/04 , G10L25/03 , G10L25/24 , G10L25/78 , H04M3/568 , H04M2201/41 , H04M2203/6045
Abstract: Embodiments of the present invention relate to speaker identification using spatial information. A method of speaker identification for audio content being of a format based on multiple channels is disclosed. The method comprises extracting, from a first audio clip in the format, a plurality of spatial acoustic features across the multiple channels and location information, the first audio clip containing voices from a speaker, and constructing a first model for the speaker based on the spatial acoustic features and the location information, the first model indicating a characteristic of the voices from the speaker. The method further comprises identifying whether the audio content contains voices from the speaker based on the first model. Corresponding system and computer program product are also disclosed.
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公开(公告)号：US20170103761A1
公开(公告)日：2017-04-13
申请号：US15287953
申请日：2016-10-07
Applicant: Dolby Laboratories Licensing Corporation
Inventor： Xuejing Sun , Kai Li , Mark S. Vinton , Shen Huang
IPC: G10L19/005 , G10L19/02 , G10L19/028
CPC classification number: G10L19/005 , G10L19/0204 , G10L19/028
Abstract: A method of encoding audio information for forward error correction reconstruction of a transmitted audio stream over a lossy packet switched network, the method including the steps of: (a) dividing the audio stream into audio frames; (b) determining a series of corresponding audio frequency bands for the audio frames; (c) determining a series of power envelopes for the frequency bands; (d) encoding the envelopes as a low bit rate version of the audio frame in a redundant transmission frame.
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