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公开(公告)号：US09877108B2
公开(公告)日：2018-01-23
申请号：US15519047
申请日：2015-10-13
Applicant: Dolby Laboratories Licensing Corporation
Inventor： Guilin Ma , Xiguang Zheng , Chen Zhang , Xuejing Sun
IPC: H04R29/00 , H04R3/04 , H04R1/40 , H04R3/00 , G10L21/0232 , G10L21/0216
CPC classification number: H04R3/04 , G10L21/0232 , G10L2021/02166 , H04M1/6016 , H04M1/605 , H04R1/326 , H04R1/406 , H04R3/005 , H04R29/001 , H04R29/004 , H04R29/005 , H04R2201/40 , H04R2201/401 , H04R2201/403 , H04R2430/20 , H04R2499/11 , H04S7/302
Abstract: Example embodiments disclosed herein relate to user experience oriented audio signal processing. There is provided a method for user experience oriented audio signal processing. The method includes obtaining a first audio signal from an audio sensor of an electronic device; computing, based on the first audio signal, a compensation factor for an acoustic path from the electronic device to a listener and applying the compensation factor to a second audio signal outputted from the electronic device. Corresponding system and computer program products are disclosed.
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22.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：US20150104022A1
公开(公告)日：2015-04-16
申请号：US14384439
申请日：2013-03-21
Applicant: Dolby Laboratories Licensing Corporation
Inventor： Huiqun Deng , Xuejing Sun
IPC: H04S1/00
CPC classification number: H04S1/007 , G10L21/0364 , G10L25/87 , H04S2400/01 , H04S2400/05
Abstract: An audio processing method and apparatus are described. In one embodiment, at least one first sub-band of a first audio signal is suppressed to obtain a reduced first audio signal with reserved sub-bands; suppressing at least one second sub-band of the at least one second audio signal to obtain at least one reduced second audio signal with reserved sub-bands; and mixing the reduced first audio signal and at least one reduced second audio signal. Alternatively, a first spatial auditory property is assigned to a first audio signal so that the first audio signal may be perceived as originating from a first position. Alternatively, rhythmic similarity between at least two audio signals is detected, and time scaling is applied to an audio signal in response to relatively high rhythmic similarity between the audio signal and the other audio signal(s); and then at least two audio signals are mixed.
Abstract translation: 描述音频处理方法和装置。 在一个实施例中，抑制第一音频信号的至少一个第一子带以获得具有保留子带的减小的第一音频信号; 抑制所述至少一个第二音频信号的至少一个第二子带以获得具有保留的子带的至少一个缩减的第二音频信号; 以及混合减小的第一音频信号和至少一个缩小的第二音频信号。 或者，第一空间听觉属性被分配给第一音频信号，使得第一音频信号可以被感知为源自第一位置。 或者，检测至少两个音频信号之间的节律相似度，并且响应于音频信号和其它音频信号之间相对较高的节律相似度，将时间缩放应用于音频信号; 然后混合至少两个音频信号。
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公开(公告)号：US20150071446A1
公开(公告)日：2015-03-12
申请号：US14365072
申请日：2012-12-12
Applicant: Dolby Laboratories Licensing Corporation
Inventor： Xuejing Sun , Glenn Dickins , Huiqun Deng , Zhiwei Shuang , Bin Cheng
CPC classification number: H04S5/00 , G10L19/26 , G10L21/0364 , H04S7/302
Abstract: An audio processing method and an audio processing apparatus are described. A mono-channel audio signal is transformed into a plurality of first subband signals. Proportions of a desired component and a noise component are estimated in each of the subband signals. Second subband signals corresponding respectively to a plurality of channels are generated from each of the first subband signals. Each of the second subband signals comprises a first component and a second component obtained by assigning a spatial hearing property and a perceptual hearing property different from the spatial hearing property to the desired component and the noise component in the corresponding first subband signal respectively, based on a multi-dimensional auditory presentation method. The second subband signals are transformed into signals for rendering with the multi-dimensional auditory presentation method. By assigning different hearing properties to desired sound and noise, the intelligibility of the audio signal can be improved.
Abstract translation: 描述了音频处理方法和音频处理装置。 单声道音频信号被变换成多个第一子带信号。 在每个子带信号中估计所需分量和噪声分量的比例。 从第一子带信号中的每一个产生分别对应于多个信道的第二子带信号。 每个第二子带信号包括第一分量和第二分量，该第一分量和第二分量通过分别将空间听觉特性和不同于空间听觉特性的感知听觉特性分配给相应的第一子带信号中的期望分量和噪声分量，基于 多维听觉呈现方法。 第二子带信号被转换成用于用多维听觉呈现方法渲染的信号。 通过将不同的听觉属性分配给期望的声音和噪声，可以提高音频信号的可懂度。
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公开(公告)号：US20150032446A1
公开(公告)日：2015-01-29
申请号：US14382667
申请日：2013-03-21
Applicant: DOLBY LABORATORIES LICENSING CORPORATION
Inventor： Glenn N. Dickins , Zhiwei Shuang , David Gunawan , Xuejing Sun
IPC: G10L25/84
CPC classification number: G10L25/84 , G10L25/78 , G10L2025/783
Abstract: 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.
Abstract translation: 具有块或帧的时间序列的音频信号被接收或访问。 确定特征是综合表征最近相对于当前时间最近处理的顺序音频块/帧。 相对于最近处理的音频块/帧，特征确定超过特定性标准并被延迟。 在音频信号中检测到语音活动指示。 VAD基于超过预设灵敏度阈值的决定，并且相对于块/帧持续时间在短时间段内计算，并且涉及当前块/帧特征。 VAD和最近的特征确定与状态相关信息相结合，状态相关信息基于在最近的特征确定时间段之前的时间确定的从多个特征编译的先前特征确定的历史。 基于该组合输出开始或终止音频信号或相关增益的决定。
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Patent Agency Ranking
公开(公告)号：US11632318B2
公开(公告)日：2023-04-18
申请号：US16988571
申请日：2020-08-07
Applicant: DOLBY LABORATORIES LICENSING CORPORATION
Inventor： Kai Li , Xuejing Sun , Gary Spittle
IPC: H04L43/0852 , H04L65/80 , H04J3/06 , H04L65/00 , G10L15/08 , G10L25/93 , H04L43/16 , H04L65/403 , G10L25/48 , G10L25/78
Abstract: 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.
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公开(公告)号：US11264045B2
公开(公告)日：2022-03-01
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.
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公开(公告)号：US11076052B2
公开(公告)日：2021-07-27
申请号：US15548265
申请日：2016-02-03
Applicant: DOLBY LABORATORIES LICENSING CORPORATION
Inventor： Richard J. Cartwright , Xuejing Sun
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. 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.
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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：US10667055B2
公开(公告)日：2020-05-26
申请号：US16554654
申请日：2019-08-29
Applicant: Dolby Laboratories Licensing Corporation
Inventor： Dong Shi , Xuejing Sun
IPC: G06F3/16 , H03G3/30 , H03G5/02 , H03G5/16 , H04R3/04 , G10L21/02 , G10L25/18 , G10L25/21 , G10L25/48 , G10L21/0232
Abstract: Example embodiments disclosed herein relate to separated audio analysis and processing. A system for processing an audio signal is disclosed. The system includes an audio analysis module configured to analyze an input audio signal to determine a processing parameter for the input audio signal, the input audio signal being represented in time domain. The system also includes an audio processing module configured to process the input audio signal in parallel with the audio analysis module. The audio processing module includes a time domain filter configured to filter the input audio signal to obtain an output audio signal in the time domain, and a filter controller configured to control a filter coefficient of the time domain filter based on the processing parameter determined by the audio analysis module. Corresponding method and computer program product of processing an audio signal are also disclosed.
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