公开(公告)号：US09947336B2

公开(公告)日：2018-04-17

申请号：US14775038

申请日：2014-03-10

Applicant: DOLBY LABORATORIES LICENSING CORPORATION

Inventor： Dong Shi ,  Xuejing Sun ,  Glenn N. Dickins

IPC: G10L21/00 ,  G10L21/0232 ,  H04M9/08 ,  G10L21/0208 ,  G10L25/21 ,  G10L25/60 ,  G10L21/0216

CPC classification number: G10L21/0232 ,  G10L21/0208 ,  G10L25/21 ,  G10L25/60 ,  G10L2021/02082 ,  G10L2021/02163 ,  H04M9/082

Abstract: The present application provides an acoustic echo mitigation apparatus and method, an audio processing apparatus and a voice communication terminal. According to an embodiment, an acoustic echo mitigation apparatus is provided, including: an acoustic echo canceller for cancelling estimated acoustic echo from a microphone signal and outputting an error signal; a residual echo estimator for estimating residual echo power; and an acoustic echo suppressor for further suppressing residual echo and noise in the error signal based on the residual echo power and noise power. Here, the residual echo estimator is configured to be continuously adaptive to power change in the error signal. According to the embodiments of the present application, the acoustic echo mitigation apparatus and method can, at least, be well adaptive to the change of power of the error signal after the AEC processing, such as that caused by change of double-talk status, echo path properties, noise level and etc.

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：US09571425B2

公开(公告)日：2017-02-14

申请号：US14384327

申请日：2013-03-21

Applicant: DOLBY LABORATORIES LICENSING CORPORATION

Inventor： Glenn N. Dickins ,  Xuejing Sun ,  Brendon Costa

IPC: H04L12/66 ,  H04L12/861 ,  H04M3/56 ,  G10L25/78 ,  G10L19/16 ,  H04L29/06

CPC classification number: H04L49/90 ,  G10L19/167 ,  G10L25/78 ,  H04L65/1066 ,  H04M3/569

Abstract: Voice communication method and apparatus and method and apparatus for operating jitter buffer are described. Audio blocks are acquired in sequence. Each of the audio blocks includes one or more audio frames. Voice activity detection is performed on the audio blocks. In response to deciding voice onset for a present one of the audio blocks, a subsequence of the sequence of the acquired audio blocks is retrieved. The subsequence precedes the present audio block immediately. The subsequence has a predetermined length and non-voice is decided for each audio block in the subsequence. The present audio block and the audio blocks in the subsequence are transmitted to a receiving party. The audio blocks in the subsequence are identified as reprocessed audio blocks. In response to deciding non-voice for the present audio block, the present audio block is cached.

Abstract translation: 描述了用于操作抖动缓冲器的语音通信方法和装置及方法和装置。 按顺序获取音频块。 每个音频块包括一个或多个音频帧。 对音频块执行语音活动检测。 响应于为音频块中的当前音频块确定语音开始，检索获取的音频块的序列的子序列。 该子序列立即在当前音频块之前。 子序列具有预定长度，并且在子序列中为每个音频块确定非声音。 当前音频块和子序列中的音频块被发送到接收方。 子序列中的音频块被识别为再处理的音频块。 响应于为当前音频块确定非语音，缓存当前音频块。

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：US09373343B2

公开(公告)日：2016-06-21

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: G10L25/78 ,  G10L25/48 ,  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。

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。

公开(公告)号：US20150036679A1

公开(公告)日：2015-02-05

申请号：US14384644

申请日：2013-03-21

Applicant: DOLBY LABORATORIES LICENSING CORPORATION

Inventor： Huiqun Deng ,  Xuejing Sun

IPC: G10L19/16 ,  H04L29/06

CPC classification number: G10L19/167 ,  G10L19/005 ,  G10L19/02 ,  H04L65/602 ,  H04L65/607

Abstract: Methods and corresponding apparatuses for transmitting and receiving audio signals are described. A transformation is performed on the audio signals in units of frame in order to obtain transformed audio data of each frame, said transformed audio data consisting of multiple signal components in the frequency domain. These signal components of each frame are distributed into multiple adjacent packets in order to generate packets in which signal components distributed from multiple frames are interleaved. Subsequently, the generated packets are transmitted. Accordingly, in case that packet loss occurs during transmission, the audio signals can be recovered based on the received signal components without consuming additional bandwidth. Therefore, robustness against packet loss can be achieved with little overhead.

Abstract translation: 描述用于发送和接收音频信号的方法和相应的装置。 以帧为单位对音频信号进行变换，以获得每帧的变换音频数据，所述变换的音频数据由频域中的多个信号分量组成。 每个帧的这些信号分量被分配到多个相邻的分组中，以便产生其中从多个帧分配的信号分量交错的分组。 随后，发送生成的分组。 因此，在发送期间发生分组丢失的情况下，可以基于接收到的信号分量来恢复音频信号，而不消耗额外的带宽。 因此，可以以很少的开销实现抗丢包的鲁棒性。

公开(公告)号：US20150023514A1

公开(公告)日：2015-01-22

申请号：US14382864

申请日：2013-03-21

Applicant: DOLBY LABORATORIES LICENSING CORPORATION

Inventor： Dong Shi ,  JiaQuan Huo ,  Xuejing Sun ,  Glenn N. Dickins

IPC: G10L21/0208

CPC classification number: G10L21/0208 ,  G10L21/02 ,  G10L25/12 ,  G10L2021/02082

Abstract: Embodiments of method and apparatus for acoustic echo control are described. According to the method, an echo energy-based doubletalk detection is performed to determine whether there is a doubletalk in a microphone signal with reference to a loudspeaker signal. A spectral similarity between spectra of the microphone signal and the loudspeaker signal is calculated. It is determined that there is no doubletalk in the microphone signal if the spectral similarity is higher than a threshold level. Adaption of an adaptive filter for applying acoustic echo cancellation or acoustic echo suppression on the microphone signal is enabled if it is determined that there is no doubletalk in the microphone signal through the echo energy-based doubletalk detection, or there is no doubletalk through the spectral similarity-based doubletalk detection.

Abstract translation: 描述了用于声学回声控制的方法和装置的实施例。 根据该方法，执行基于回波能量的双重检测，以确定麦克风信号中是否存在参考扬声器信号的双音节。 计算麦克风信号和扬声器信号的频谱之间的频谱相似度。 如果频谱相似度高于阈值电平，则确定麦克风信号中没有双音节。 如果通过基于回波能量的双重检测确定了麦克风信号中没有双重位，或者通过光谱没有双重反馈，则可以使用适用于对麦克风信号进行声学回声消除或声学回声抑制的自适应滤波器 基于相似度的双标检测。