公开(公告)号：US08346544B2

公开(公告)日：2013-01-01

申请号：US11625802

申请日：2007-01-22

申请人： Sharath Manjunath ,  Ananthapadmanabhan Aasanipalai Kandhadai ,  Eddie L. T. Choy

发明人： Sharath Manjunath ,  Ananthapadmanabhan Aasanipalai Kandhadai ,  Eddie L. T. Choy

IPC分类号： G10L19/00

CPC分类号： G10L19/24

摘要： In a device configurable to encode speech performing an closed loop re-decision may comprise representing a speech signal by amplitude components and phase components for a current frame and a past frame. In a first closed loop stage, a first set of compressed components and a first set of uncompressed components for a current frame may be generated. A first set of features may be generated by comparing current and past frame amplitude and/or phase components. In a second closed loop stage, a second set of compressed components for the current frame may be generated by compressing the first set of compressed components and compressing the first set of uncompressed components. Generation of a second set of features may be based on the second set of compressed components from the current frame and a combination of amplitude and/or phase components from the past frame.

摘要翻译： 在可配置为对执行闭环重新判定的语音进行编码的设备中的设备可以包括通过当前帧和过去帧的幅度分量和相位分量表示语音信号。 在第一闭环阶段中，可以产生用于当前帧的第一组压缩组件和第一组未压缩组件。 可以通过比较当前和过去的帧幅度和/或相位分量来生成第一组特征。 在第二闭环阶段中，可以通过压缩第一组压缩分量并压缩第一组未压缩分量来生成用于当前帧的第二组压缩分量。 第二组特征的生成可以基于来自当前帧的第二组压缩分量和来自过去帧的幅度和/或相位分量的组合。

公开(公告)号：US08223988B2

公开(公告)日：2012-07-17

申请号：US12022037

申请日：2008-01-29

申请人： Song Wang ,  Dinesh Ramakrishnan ,  Samir Gupta ,  Eddie L. T. Choy

发明人： Song Wang ,  Dinesh Ramakrishnan ,  Samir Gupta ,  Eddie L. T. Choy

IPC分类号： H04R3/00 ,  H04R1/02 ,  H04R9/06

CPC分类号： G10L21/028 ,  G10L2021/02166 ,  H04R3/005 ,  H04R25/40

摘要： An enhanced blind source separation technique is provided to improve separation of highly correlated signal mixtures. A beamforming algorithm is used to precondition correlated first and second input signals in order to avoid indeterminacy problems typically associated with blind source separation. The beamforming algorithm may apply spatial filters to the first signal and second signal in order to amplify signals from a first direction while attenuating signals from other directions. Such directionality may serve to amplify a desired speech signal in the first signal and attenuate the desired speech signal from the second signal. Blind source separation is then performed on the beamformer output signals to separate the desired speech signal and the ambient noise and reconstruct an estimate of the desired speech signal. To enhance the operation of the beamformer and/or blind source separation, calibration may be performed at one or more stages.

摘要翻译： 提供增强的盲源分离技术来改善高度相关的信号混合物的分离。 波束形成算法用于预处理相关的第一和第二输入信号，以避免通常与盲源分离相关联的不确定性问题。 波束成形算法可以对第一信号和第二信号应用空间滤波器，以便在衰减来自其它方向的信号的同时放大来自第一方向的信号。 这种方向性可以用于在第一信号中放大期望的语音信号，并从第二信号中衰减所需的语音信号。 然后对波束形成器输出信号执行盲源分离，以分离所需的语音信号和环境噪声，并重建所需语音信号的估计。 为了增强波束形成器和/或盲源分离的操作，可以在一个或多个阶段执行校准。

公开(公告)号：US08175871B2

公开(公告)日：2012-05-08

申请号：US11864906

申请日：2007-09-28

申请人： Song Wang ,  Samir Kumar Gupta ,  Eddie L. T. Choy

发明人： Song Wang ,  Samir Kumar Gupta ,  Eddie L. T. Choy

IPC分类号： G10L21/02 ,  G10L19/00 ,  G10L17/00 ,  H04B15/00 ,  H04M9/08

CPC分类号： H04M9/082 ,  G10L2021/02082 ,  G10L2021/02166

摘要： Multiple microphone noise suppression apparatus and methods are described herein. The apparatus and methods implement a variety of noise suppression techniques and apparatus that can be selectively applied to signals received using multiple microphones. The microphone signals received at each of the multiple microphones can be independently processed to cancel echo signal components that can be generated from a local audio source. The echo cancelled signals may be processed by some or all modules within a signal separator that operates to separate or otherwise isolate a speech signal from noise signals. The signal separator can include a pre-processing de-correlator followed by a blind source separator. The output of the blind source separator can be post filtered to provide post separation de-correlation. The separated speech and noise signals can be non-linearly processed for further noise reduction, and additional post processing can be implemented following the non-linear processing.

摘要翻译： 本文描述了多麦克风噪声抑制装置和方法。 该装置和方法实现了可以选择性地应用于使用多个麦克风接收的信号的各种噪声抑制技术和装置。 可以独立地处理在多个麦克风中的每一个处接收的麦克风信号以消除可从本地音频源产生的回波信号分量。 回波消除信号可以由信号分离器内的一些或所有模块进行处理，信号分离器用于分离或以其他方式将语音信号与噪声信号隔离开来。 信号分离器可以包括预处理去相关器，其后是盲源分离器。 盲源分离器的输出可以后过滤以提供后分离去相关。 分离的语音和噪声信号可以被非线性处理以进一步降低噪声，并且可以在非线性处理之后实现附加的后处理。

公开(公告)号：US08041057B2

公开(公告)日：2011-10-18

申请号：US11449454

申请日：2006-06-07

申请人： Pei Xiang ,  Eddie L. T. Choy ,  Prajakt V. Kulkarni ,  Samir Kumar Gupta

发明人： Pei Xiang ,  Eddie L. T. Choy ,  Prajakt V. Kulkarni ,  Samir Kumar Gupta

IPC分类号： H04B1/00

CPC分类号： H04S7/30 ,  H04M1/6016 ,  H04M1/72558 ,  H04R2499/11

摘要： This disclosure describes audio mixing techniques that intelligently combine two or more audio signals into an output signal. The techniques allow audio to be combined, yet create perceptual differentiation between the different audio signals. The result is that a user is able to hear both audio signals in a combined output, but the different audio signals do not perceptually interfere with one another. The techniques are relatively simple to implement and are well suited for radio telephones.

摘要翻译： 本公开描述了将两个或多个音频信号智能地组合成输出信号的音频混合技术。 这些技术允许音频被组合，但是在不同音频信号之间产生感知差异。 结果是用户能够听到组合输出中的两个音频信号，但是不同的音频信号不会感知到彼此干扰。 这些技术实现起来相对简单，并且非常适合于无线电话。

公开(公告)号：US20090135976A1

公开(公告)日：2009-05-28

申请号：US11946253

申请日：2007-11-28

申请人： Dinesh Ramakrishnan ,  Song Wang ,  Eddie L. T. Choy ,  Samir Kumar Gupta

发明人： Dinesh Ramakrishnan ,  Song Wang ,  Eddie L. T. Choy ,  Samir Kumar Gupta

IPC分类号： H04L7/027

CPC分类号： H04J3/0632 ,  G10L19/005

摘要： In a digital system with more than one clock source, lack of synchronization between the clock sources may cause overflow or underflow in sample buffers, also called sample slipping. Sample slipping may lead to undesirable artifacts in the processed signal due to discontinuities introduced by the addition or removal of extra samples. To smooth out discontinuities caused by sample slipping, samples are filtered to when a buffer overflow condition occurs, and the samples are interpolated to produce additional samples when a buffer underflow condition occurs. The interpolated samples may also be filtered. The filtering and interpolation operations can be readily implemented without adding significant burden to the computational complexity of a real-time digital system.

摘要翻译： 在具有多个时钟源的数字系统中，时钟源之间的同步缺乏可能导致采样缓冲器中的溢出或下溢，也称为样品打滑。 由于添加或除去额外的样品引起的不连续性，样品打滑可能导致处理过的信号中的不期望的伪影。 为了平滑由样品滑动引起的不连续性，将样品过滤到发生缓冲液溢出状态时，当发生缓冲液下溢条件时，样品被内插以产生附加样品。 内插样本也可以被过滤。 可以容易地实现滤波和插值操作，而不会对实时数字系统的计算复杂度造成重大负担。

公开(公告)号：US08954324B2

公开(公告)日：2015-02-10

申请号：US11864897

申请日：2007-09-28

申请人： Song Wang ,  Samir Kumar Gupta ,  Eddie L. T. Choy

发明人： Song Wang ,  Samir Kumar Gupta ,  Eddie L. T. Choy

IPC分类号： G10L15/20 ,  G10L11/06 ,  G10L19/00 ,  G10L25/78 ,  G10L21/0216

CPC分类号： G10L25/78 ,  G10L2021/02165

摘要： Voice activity detection using multiple microphones can be based on a relationship between an energy at each of a speech reference microphone and a noise reference microphone. The energy output from each of the speech reference microphone and the noise reference microphone can be determined. A speech to noise energy ratio can be determined and compared to a predetermined voice activity threshold. In another embodiment, the absolute value of the autocorrelation of the speech and noise reference signals are determined and a ratio based on autocorrelation values is determined. Ratios that exceed the predetermined threshold can indicate the presence of a voice signal. The speech and noise energies or autocorrelations can be determined using a weighted average or over a discrete frame size.

摘要翻译： 使用多个麦克风的语音活动检测可以基于语音基准麦克风和噪声参考麦克风各自的能量之间的关系。 可以确定来自每个语音参考麦克风和噪声参考麦克风的能量输出。 可以确定语音能量比，并将其与预定的语音活动阈值进行比较。 在另一个实施例中，确定语音和噪声参考信号的自相关的绝对值，并且确定基于自相关值的比率。 超过预定阈值的比率可以指示语音信号的存在。 可以使用加权平均值或离散的帧大小来确定语音和噪声能量或自相关性。

公开(公告)号：US08532714B2

公开(公告)日：2013-09-10

申请号：US12362098

申请日：2009-01-29

申请人： Kuntal Dilipsinh Sampat ,  Samir Kumar Gupta ,  Eddie L. T. Choy ,  Joel Linsky

发明人： Kuntal Dilipsinh Sampat ,  Samir Kumar Gupta ,  Eddie L. T. Choy ,  Joel Linsky

IPC分类号： H04B1/00

CPC分类号： H04W8/245 ,  G06F9/44526

摘要： An executable is downloaded to an audio output device over a communications link. The executable may configure the audio output device to decode audio encoded in a specified format. The executable may also or alternatively include other audio processing software. The audio may include voice and/or audio playback, e.g., music playback. The ability to download an audio executable allows dynamic provisioning of various decoding and/or audio process capabilities to an audio output device. This may eliminate the need to transcode digitized audio for playback at the audio output device, and may also allow the audio output device to decode multiple audio formats without having multiple audio decoders permanently residing within the audio output device.

摘要翻译： 通过通信链路将可执行文件下载到音频输出设备。 可执行程序可以配置音频输出设备来解码以指定格式编码的音频。 可执行程序还可以或者可选地包括其他音频处理软件。 音频可以包括语音和/或音频播放，例如音乐播放。 下载音频可执行文件的能力允许向音频输出设备动态地提供各种解码和/或音频处理能力。 这可以消除将数字化音频转码为在音频输出设备处回放的需要，并且还可以允许音频输出设备解码多种音频格式，而不会使多个音频解码器永久驻留在音频输出设备内。

公开(公告)号：US08090573B2

公开(公告)日：2012-01-03

申请号：US11625797

申请日：2007-01-22

申请人： Sharath Manjunath ,  Ananthapadmanabhan Arasanipalai Kandhadai ,  Eddie L. T. Choy

发明人： Sharath Manjunath ,  Ananthapadmanabhan Arasanipalai Kandhadai ,  Eddie L. T. Choy

IPC分类号： G10L19/00

CPC分类号： G10L19/22

摘要： In a device configurable to encode speech performing an open loop re-decision may comprise representing a speech signal by amplitude components and phase components for a current frame and a past frame. During the current frame, there may be an extraction of uncompressed amplitude components and uncompressed phase components. The amplitude components and the phase components from the past frame may then be retrieved. A set of features may be generated based on the uncompressed amplitude components from the current frame, the uncompressed phase components from the current frame, the amplitude components from the past frame, and the phase components from the past frame. The set of features may be checked as part of the open loop re-decision, and determining a final encoding decision based on the checking may be performed. The final encoding decision may be an encoding mode and/or encoding rate.

摘要翻译： 在可配置为对执行开环重新判定的语音进行编码的装置中的装置可以包括通过当前帧和过去帧的幅度分量和相位分量表示语音信号。 在当前帧中，可以提取未压缩幅度分量和未压缩相位分量。 然后可以检索来自过去帧的幅度分量和相位分量。 可以基于来自当前帧的未压缩幅度分量，来自当前帧的未压缩相位分量，来自过去帧的幅度分量和来自过去帧的相位分量来生成一组特征。 可以将这组特征作为开环重新判定的一部分进行检查，并且可以执行基于检查来确定最终编码决定。 最终编码决定可以是编码模式和/或编码速率。

公开(公告)号：US07970564B2

公开(公告)日：2011-06-28

申请号：US11551509

申请日：2006-10-20

申请人： Song Wang ,  Eddie L. T. Choy ,  Samir Kumar Gupta

发明人： Song Wang ,  Eddie L. T. Choy ,  Samir Kumar Gupta

IPC分类号： G01R13/00

CPC分类号： G06K9/6243 ,  G10L21/0272

摘要： This disclosure describes signal processing techniques that can improve the performance of blind source separation (BSS) techniques. In particular, the described techniques propose pre-processing steps that can help to de-correlate the different signals from one another prior to execution of the BSS techniques. In addition, the described techniques also propose optional post-processing steps that can further de-correlate the different signals following execution of the BSS techniques. The techniques may be particularly useful for improving BSS performance with highly correlated audio signals, e.g., from two microphones that are in close spatial proximity to one another.

摘要翻译： 本公开描述了可以提高盲源分离（BSS）技术的性能的信号处理技术。 特别地，所描述的技术提出了预处理步骤，其可以有助于在执行BSS技术之前将不同信号彼此相关联。 此外，所描述的技术还提出了可选的后处理步骤，其可以在执行BSS技术之后进一步使不同信号去相关。 这些技术对于通过高度相关的音频信号（例如来自彼此紧密地空间接近的两个麦克风）来改善BSS性能可能特别有用。

公开(公告)号：US07663046B2

公开(公告)日：2010-02-16

申请号：US12042170

申请日：2008-03-04

申请人： Prajakt Kulkarni ,  Eddie L. T. Choy ,  Nidish Ramachandra Kamath ,  Samir K Gupta ,  Stephen Molloy ,  Suresh Devalapalli

发明人： Prajakt Kulkarni ,  Eddie L. T. Choy ,  Nidish Ramachandra Kamath ,  Samir K Gupta ,  Stephen Molloy ,  Suresh Devalapalli

IPC分类号： G10H1/00

CPC分类号： G10H1/0066 ,  G10H7/004

摘要： This disclosure describes techniques for processing audio files that comply with the musical instrument digital interface (MIDI) format. In particular, various tasks associated with MIDI file processing are delegated between software operating on a general purpose processor, firmware associated with a digital signal processor (DSP), and dedicated hardware that is specifically designed for MIDI file processing. Alternatively, a multi-threaded DSP may be used instead of a general purpose processor and the DSP. In one aspect, this disclosure provides a method comprising parsing MIDI files and scheduling MIDI events associated with the MIDI files using a first process, processing the MIDI events using a second process to generate MIDI synthesis parameters, and generating audio samples using a hardware unit based on the synthesis parameters.

摘要翻译： 本公开描述了用于处理符合乐器数字接口（MIDI）格式的音频文件的技术。 具体而言，与在通用处理器上运行的软件，与数字信号处理器（DSP）相关联的固件以及专门为MIDI文件处理而专门设计的专用硬件之间的任何与MIDI文件处理相关的各种任务被委派。 或者，可以使用多线程DSP来代替通用处理器和DSP。 在一个方面，本公开提供了一种方法，包括使用第一处理解析MIDI文件和调度与MIDI文件相关联的MIDI事件，使用第二处理来处理MIDI事件以产生MIDI合成参数，以及使用基于硬件单元生成音频样本 对合成参数。