公开(公告)号：US09525845B2

公开(公告)日：2016-12-20

申请号：US14426134

申请日：2013-09-27

Applicant: DOLBY LABORATORIES LICENSING CORPORATION ,  DOLBY INTERNATIONAL AB

Inventor： Dong Shi ,  Xuejing Sun ,  Kai Li ,  Shen Huang ,  Harald Mundt ,  Heiko Purnhagen ,  Glenn Dickins

IPC: H04N7/14 ,  H04N7/15

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。

公开(公告)号：US20150081283A1

公开(公告)日：2015-03-19

申请号：US14384356

申请日：2013-03-21

Applicant: DOLBY LABORATORIES LICENSING CORPORATION

Inventor： Xuejing Sun ,  Zhiwei Shuang ,  Shen Huang

IPC: G10L25/78

CPC classification number: G10L25/78 ,  G10L25/18 ,  G10L25/81 ,  G10L25/84

Abstract: Embodiments are described for harmonicity estimation, audio classification, pitch determination and noise estimation. Measuring harmonicity of an audio signal includes calculation a log amplitude spectrum of audio signal. A first spectrum is derived by calculating each component of the first spectrum as a sum of components of the log amplitude spectrum on frequencies. In linear frequency scale, the frequencies are odd multiples of the component's frequency of the first spectrum. A second spectrum is derived by calculating each component of the second spectrum as a sum of components of the log amplitude spectrum on frequencies. In linear frequency scale, the frequencies are even multiples of the component's frequency of the second spectrum. A difference spectrum is derived subtracting the first spectrum from the second spectrum. A measure of harmonicity is generated as a monotonically increasing function of the maximum component of the difference spectrum within predetermined frequency range.

Abstract translation: 描述了用于谐波估计，音频分类，音调确定和噪声估计的实施例。 测量音频信号的谐波包括计算音频信号的对数幅度谱。 通过将第一频谱的每个分量计算为频率上的对数幅度谱的分量的和来导出第一频谱。 在线性频率标度中，频率是第一个频谱的分量频率的奇数倍。 通过将第二频谱的每个分量计算为频率上的对数幅度谱的分量的和来导出第二频谱。 在线性频率标度中，频率是第二个频谱的分量频率的偶数倍。 导出从第二个频谱减去第一个频谱的差分谱。 产生谐波度的度量作为预定频率范围内的差分频谱的最大分量的单调递增函数。

公开(公告)号：US11289103B2

公开(公告)日：2022-03-29

申请号：US16928918

申请日：2020-07-14

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.

公开(公告)号：US10455080B2

公开(公告)日：2019-10-22

申请号：US15539101

申请日：2015-12-23

Applicant: Dolby Laboratories Licensing Corporation

Inventor： Doh-Suk Kim ,  Shen Huang

IPC: H04L12/26 ,  H04M3/22 ,  G10L25/60

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 5 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.

公开(公告)号：US10224040B2

公开(公告)日：2019-03-05

申请号：US14899238

申请日：2014-07-02

Applicant: Dolby International AB ,  Dolby Laboratories Licensing Corporation

Inventor： Shen Huang ,  Xuejing Sun ,  Heiko Purnhagen

IPC: G10L19/005 ,  G10L19/02 ,  G10L19/008 ,  G10L19/16

Abstract: The present application relates to packet loss concealment apparatus and method, and audio processing system. According to an embodiment, the packet loss concealment apparatus is provided for concealing packet losses in a stream of audio packets, each audio packet comprising at least one audio frame in transmission format comprising at least one monaural component and at least one spatial component. The packet loss concealment apparatus may comprises a first concealment unit for creating the at least one monaural component for a lost frame in a lost packet and a second concealment unit for creating the at least one spatial component for the lost frame. According to the embodiment, spatial artifacts such as incorrect angle and diffuseness may be avoided as far as possible in PLC for multi-channel spatial or sound field encoded audio signals.

公开(公告)号：US10014005B2

公开(公告)日：2018-07-03

申请号：US14384356

申请日：2013-03-21

Applicant: DOLBY LABORATORIES LICENSING CORPORATION

Inventor： Xuejing Sun ,  Zhiwei Shuang ,  Shen Huang

IPC: G10L25/00 ,  G10L25/78 ,  G10L25/18 ,  G10L25/84 ,  G10L25/81

CPC classification number: G10L25/78 ,  G10L25/18 ,  G10L25/81 ,  G10L25/84

Abstract: Embodiments are described for harmonicity estimation, audio classification, pitch determination and noise estimation. Measuring harmonicity of an audio signal includes calculation a log amplitude spectrum of audio signal. A first spectrum is derived by calculating each component of the first spectrum as a sum of components of the log amplitude spectrum on frequencies. In linear frequency scale, the frequencies are odd multiples of the component's frequency of the first spectrum. A second spectrum is derived by calculating each component of the second spectrum as a sum of components of the log amplitude spectrum on frequencies. In linear frequency scale, the frequencies are even multiples of the component's frequency of the second spectrum. A difference spectrum is derived subtracting the first spectrum from the second spectrum. A measure of harmonicity is generated as a monotonically increasing function of the maximum component of the difference spectrum within predetermined frequency range.

公开(公告)号：US09881621B2

公开(公告)日：2018-01-30

申请号：US15369768

申请日：2016-12-05

Applicant: Dolby Laboratories Licensing Corporation

Inventor： Shen Huang ,  Xuejing Sun

IPC: G10L19/005

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 results 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 (412, 413) is described. A lost packet (412) is a packet which is deemed to be lost by a transform-based audio decoder. Each of the one or more lost packets (412, 413) comprises a set of transform coefficients (313). A set of transform coefficients (313) is used by the transform-based audio decoder to generate a corresponding frame (412, 413) of a time domain audio signal. The method (200) comprises determining (205) for a current lost packet (412) of the one or more lost packets (412, 413) a number of preceding lost packets from the one or more lost packets (313); 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 (422) of the audio signal using the determined PLC scheme (204, 207, 208); wherein the current frame (422) corresponds to the current lost packet (412).

公开(公告)号：US09558744B2

公开(公告)日：2017-01-31

申请号：US14650214

申请日：2013-11-27

Applicant: DOLBY LABORATORIES LICENSING CORPORATION

Inventor： Xuejing Sun ,  Shen Huang ,  Poppy Crum ,  Hannes Muesch ,  Glenn N. Dickins ,  Michael Eckert

IPC: G06F17/00 ,  G06F17/27 ,  G10L19/00 ,  G10L21/00 ,  G10L15/20 ,  G10L21/02 ,  H04M3/56

CPC classification number: G10L15/20 ,  G10L21/02 ,  H04M3/568

Abstract: An audio processing apparatus and an audio processing method are described. In one embodiment, the audio processing apparatus include an audio masker separator for separating from a first audio signal an audio material comprising a sound other than stationary noise and utterance meaningful in semantics, as an audio masker candidate. The apparatus also includes a first context analyzer for obtaining statistics regarding contextual information of detected audio masker candidates, and a masker library builder for building a masker library or updating an existing masker library by adding, based on the statistics, at least one audio masker candidate as an audio masker into the masker library, wherein audio maskers in the maker library are used to be inserted into a target position in a second audio signal to conceal defects in the second audio signal.

Abstract translation: 描述音频处理装置和音频处理方法。 在一个实施例中，音频处理设备包括一个音频掩蔽器分离器，用于将音频材料与第一音频信号分离，该音频材料包括除了固定噪声之外的声音以及在语义上有意义的话语作为音频掩蔽者候选者。 该装置还包括用于获得关于检测到的音频掩蔽者候选者的上下文信息的统计信息的第一上下文分析器，以及用于构建掩蔽程序库或通过基于统计信息添加至少一个音频掩码选择器来构建掩蔽程序库或更新现有掩蔽程序库的掩码程序库构建器 作为音频掩蔽器进入掩蔽器库，其中制造商库中的音频掩蔽器被用于插入第二音频信号中的目标位置以隐藏第二音频信号中的缺陷。

公开(公告)号：US20150255079A1

公开(公告)日：2015-09-10

申请号：US14431256

申请日：2013-09-27

Applicant: DOLBY LABORATORIES LICENSING CORPORATION

Inventor： Shen Huang ,  Xuejing Sun

IPC: 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。