公开(公告)号：US09767820B2

公开(公告)日：2017-09-19

申请号：US15175105

申请日：2016-06-07

申请人： Comcast Cable Communications, LLC

发明人： George Thomas Des Jardins

IPC分类号： H04R29/00 ,  G10L21/0308 ,  G10L25/84 ,  G10L19/018

CPC分类号： G10L21/0308 ,  G10L19/018 ,  G10L21/0208 ,  G10L21/0224 ,  G10L25/84

摘要： A system for removing noise from an audio signal is described. For example, noise caused by content playing in the background during a voice command or phone call may be removed from the audio signal representing the voice command or phone call. By removing noise, the signal to noise ratio of the audio signal may be improved.

公开(公告)号：US09711165B2

公开(公告)日：2017-07-18

申请号：US14984089

申请日：2015-12-30

申请人： AUDIONAMIX

发明人： Romain Hennequin

IPC分类号： G10L21/00 ,  H04B3/20 ,  G10L21/0308 ,  G10L19/02 ,  G10L21/02 ,  G10L21/0208

CPC分类号： G10L21/0308 ,  G10L19/02 ,  G10L21/0205 ,  G10L21/0208 ,  G10L2021/02082

摘要： Processes are described herein for transforming an audio mixture for which a specific component is affected by reverberation, into a specific dry component (i.e. unaffected by the reverberation) and a background component. In the process described herein, the long-term effects of reverberation are explicitly taken into account by modelling the spectrogram of the specific component as the result of a matrix convolution along time between the spectrogram of the specific dry component and a reverberation matrix. Parameters of the model are estimated iteratively by minimizing a cost-function measuring the divergence between the spectrogram of the mixture signal and the model of the spectrogram of the mixture signal.

公开(公告)号：US09679579B1

公开(公告)日：2017-06-13

申请号：US15186993

申请日：2016-06-20

申请人： Google Inc.

发明人： Richard Francis Lyon ,  Ron Weiss ,  Thomas Chadwick Walters

IPC分类号： G10L21/028 ,  G10L21/0356 ,  G10L25/51 ,  G10L21/0308 ,  G10L21/0388

CPC分类号： G10L21/0356 ,  G10L21/02 ,  G10L21/0272 ,  G10L21/028 ,  G10L21/0308 ,  G10L21/0388 ,  G10L25/51

摘要： Systems and methods facilitating removal of content from audio files are described. A method includes identifying a sound recording in a first audio file, identifying a reference file having at least a defined level of similarity to the sound recording, and processing the first audio file to remove the sound recording and generate a second audio file. In some embodiments, winner-take-all coding and Hough transforms are employed for determining alignment and rate adjustment of the reference file in the first audio file. After alignment, the reference file is filtered in the frequency domain to increase similarity between the reference file and the sound recording. The frequency domain representation (FR) of the filtered version is subtracted from the FR first audio and the result converted to a time representation of the second audio file. In some embodiments, spectral subtraction is also performed to generate a further improved second audio file.

公开(公告)号：US09437199B2

公开(公告)日：2016-09-06

申请号：US14431309

申请日：2013-09-25

申请人： UNIVERSITE BORDEAUX 1 ,  CENTRE NATIONAL DE LA RECHERCHE SCIENTIFIQUE (CNRS)

发明人： Sylvain Marchand ,  Stanislaw Gorlow

IPC分类号： G10L19/008 ,  G10L19/018 ,  G10L21/028

CPC分类号： G10L19/008 ,  G10L19/018 ,  G10L21/028 ,  G10L21/0308

摘要： The invention relates to a method and the associated device 1 for separating one or more particular digital audio source signals (si) contained in a mixed multichannel digital audio signal (smix) obtained by mixing a plurality of digital audio source signals (s1, . . . , sp). According to the invention: the modulus of the amplitude or the normalized power of the particular source signal(s) (si) is determined from representative values of said particular source signal(s) contained in the mixed signal; and then linearly constrained minimum variance spatial filtering is performed on the mixed signal in order to obtain each particular source signal (s′i), said filtering being based on the distribution of said particular source signal between at least two channels of the mixed signal, and the modulus of the amplitude or the normalized power of said particular source signal is used as a linear constraint of the filter.

公开(公告)号：US09373320B1

公开(公告)日：2016-06-21

申请号：US13972262

申请日：2013-08-21

申请人： Google Inc.

发明人： Richard Francis Lyon ,  Ron Weiss ,  Thomas Chadwick Walters

IPC分类号： G10L21/028 ,  G10L13/033 ,  G10L21/02 ,  G10L21/0272 ,  G10L21/0308

CPC分类号： G10L21/0356 ,  G10L21/02 ,  G10L21/0272 ,  G10L21/028 ,  G10L21/0308 ,  G10L21/0388 ,  G10L25/51

摘要： Systems and methods facilitating removal of content from audio files are described. A method includes identifying a sound recording in a first audio file, identifying a reference file having at least a defined level of similarity to the sound recording, and processing the first audio file to remove the sound recording and generate a second audio file. In some embodiments, winner-take-all coding and Hough transforms are employed for determining alignment and rate adjustment of the reference file in the first audio file. After alignment, the reference file is filtered in the frequency domain to increase similarity between the reference file and the sound recording. The frequency domain representation (FR) of the filtered version is subtracted from the FR first audio and the result converted to a time representation of the second audio file. In some embodiments, spectral subtraction is also performed to generate a further improved second audio file.

摘要翻译： 描述了有助于从音频文件中移除内容的系统和方法。 一种方法包括：识别在第一音频文件中的声音记录，识别具有至少与所述声音记录的确定的相似度水平的参考文件，以及处理所述第一音频文件以去除所述声音记录并生成第二音频文件。 在一些实施例中，采用获胜者全部编码和霍夫变换来确定第一音频文件中的参考文件的对准和速率调整。 对齐后，参考文件在频域中进行过滤，以增加参考文件和录音之间的相似性。 从FR第一音频中减去滤波版本的频域表示（FR），并将结果转换为第二音频文件的时间表示。 在一些实施例中，还执行频谱减法以产生进一步改进的第二音频文件。

公开(公告)号：US09368112B2

公开(公告)日：2016-06-14

申请号：US13891198

申请日：2013-05-10

申请人： Huawei Technologies Co., Ltd.

发明人： Zhe Wang

IPC分类号： G10L15/20 ,  G10L25/78

CPC分类号： G10L21/0308 ,  G10L15/20 ,  G10L19/0208 ,  G10L19/22 ,  G10L25/78 ,  G10L25/84 ,  G10L2025/786

摘要： The disclosure provides a method and an apparatus for detecting a voice activity in an input audio signal composed of frames. A noise characteristic of the input signal is determined based on a received frame of the input audio signal. A voice activity detection (VAD) parameter is derived based on the noise characteristic of the input audio signal using an adaptive function. The derived VAD parameter is compared with a threshold value to provide a voice activity detection decision. The input audio signal is processed according to the voice activity detection decision.

摘要翻译： 本公开提供了一种用于检测由帧组成的输入音频信号中的语音活动的方法和装置。 基于输入音频信号的接收帧来确定输入信号的噪声特性。 基于使用自适应功能的输入音频信号的噪声特性导出语音活动检测（VAD）参数。 将导出的VAD参数与阈值进行比较以提供语音活动检测决策。 输入音频信号根据语音活动检测决定进行处理。

公开(公告)号：US20160155457A1

公开(公告)日：2016-06-02

申请号：US15019242

申请日：2016-02-09

申请人： TELEFONAKTIEBOLAGET L M ERICSSON (PUBL)

发明人： Stefan Bruhn

IPC分类号： G10L25/84 ,  G10L21/0308 ,  G10L19/06

CPC分类号： G10L19/087 ,  G10L19/012 ,  G10L19/06 ,  G10L19/26 ,  G10L21/02 ,  G10L21/0308 ,  G10L25/84 ,  G10L2021/02168

摘要： In a method for coding of information for enhancing a background noise representation, voice activity of an input speech signal is determined. A noisiness parameter is determined for an inactive speech signal, wherein the noisiness parameter is based on a ratio of prediction gains of two Linear Predictive Coder (LPC) prediction filters with different orders. The noisiness parameter is quantized, and the quantized noisiness parameter is encoded for transmission.

摘要翻译： 在用于增强背景噪声表示的信息的编码方法中，确定输入语音信号的语音活动。 对于无效语音信号确定噪声参数，其中噪声参数基于具有不同阶数的两个线性预测编码器（LPC）预测滤波器的预测增益的比率。 噪声参数被量化，量化噪声参数被编码用于传输。

公开(公告)号：US20160155434A1

公开(公告)日：2016-06-02

申请号：US14951476

申请日：2015-11-24

申请人： Gregory C. Burnett ,  Eric F. Breitfeller

发明人： Gregory C. Burnett ,  Eric F. Breitfeller

IPC分类号： G10K11/178 ,  G10L21/02 ,  H04R1/46 ,  G10L21/0308

CPC分类号： G10L21/0205 ,  G10K11/178 ,  G10K2210/30232 ,  G10K2210/3028 ,  G10K2210/3045 ,  G10L19/0204 ,  G10L21/02 ,  G10L21/0208 ,  G10L21/0308 ,  G10L25/78 ,  G10L2021/02082 ,  G10L2021/02161 ,  G10L2021/02165 ,  G10L2021/02168 ,  H04R1/46

摘要： Acoustic noise suppression is provided in multiple-microphone systems using Voice Activity Detectors (VAD). A host system receives acoustic signals via multiple microphones. The system also receives information on the vibration of human tissue associated with human voicing activity via the VAD. In response, the system generates a transfer function representative of the received acoustic signals upon determining that voicing information is absent from the received acoustic signals during at least one specified period of time. The system removes noise from the received acoustic signals using the transfer function, thereby producing a denoised acoustic data stream.

摘要翻译： 在使用语音活动检测器（VAD）的多麦克风系统中提供噪声抑制。 主机系统通过多个麦克风接收声信号。 该系统还通过VAD接收与人体发声活动相关的人体组织的振动信息。 作为响应，当在至少一个指定的时间段内确定在接收到的声信号中不存在发声信息时，系统产生代表所接收的声信号的传递函数。 该系统使用传递函数从接收的声信号中去除噪声，从而产生去噪的声学数据流。

公开(公告)号：US20150271599A1

公开(公告)日：2015-09-24

申请号：US14441631

申请日：2012-11-12

申请人： Juha Petteri OJANPERA ,  NOKIA CORPORATION

发明人： Juha Petteri Ojanpera

IPC分类号： H04R3/00 ,  G01S3/80

CPC分类号： H04R3/005 ,  G01S3/8006 ,  G10L21/0308 ,  G11B27/031 ,  G11B27/10 ,  G11B27/28

摘要： An apparatus comprising an input configured to receive an audio signal comprising at least two audio shots separated by an audio shot boundary, and a comparator configured to compare the audio signal against a reference audio signal and to determine a location of the audio shot boundary within the audio signal based on the comparison of the audio signal against the reference audio signal.

摘要翻译： 一种装置，包括被配置为接收包括由音频镜头边界分开的至少两个音频镜头的音频信号的输入，以及比较器，被配置为将音频信号与参考音频信号进行比较，并且确定音频镜头边界的位置 基于音频信号与参考音频信号的比较的音频信号。

公开(公告)号：US20150243290A1

公开(公告)日：2015-08-27

申请号：US14431309

申请日：2013-09-25

申请人： UNIVERSITÉ BORDEAUX 1 ,  CENTRE NATIONAL DE LA RECHERCHE SCIENTIFIQUE (CNRS)

发明人： Sylvain Marchand ,  Stanislaw Gorlow

IPC分类号： G10L19/008 ,  G10L19/018 ,  G10L21/028

CPC分类号： G10L19/008 ,  G10L19/018 ,  G10L21/028 ,  G10L21/0308

摘要： The invention relates to a method and the associated device 1 for separating one or more particular digital audio source signals (si) contained in a mixed multichannel digital audio signal (smix) obtained by mixing a plurality of digital audio source signals (s1, . . . , sp). According to the invention: the modulus of the amplitude or the normalized power of the particular source signal(s) (si) is determined from representative values of said particular source signal(s) contained in the mixed signal; and then linearly constrained minimum variance spatial filtering is performed on the mixed signal in order to obtain each particular source signal (s′i), said filtering being based on the distribution of said particular source signal between at least two channels of the mixed signal, and the modulus of the amplitude or the normalized power of said particular source signal is used as a linear constraint of the filter.

摘要翻译： 本发明涉及一种用于分离通过混合多个数字音频源信号（s1，...）而获得的混合多声道数字音频信号（smix）中包含的一个或多个特定数字音频源信号（si）的方法和相关设备1。 。，sp）。 根据本发明：由混合信号中包含的所述特定源信号的代表值确定特定源信号（s）（s）的振幅或归一化功率的模数; 然后对混合信号执行线性约束最小方差空间滤波，以便获得每个特定源信号（s'i），所述滤波是基于混合信号的至少两个信道之间的所述特定源信号的分布， 并且所述特定源信号的振幅或归一化功率的模数被用作滤波器的线性约束。