INTEGRATED SPEECH INTELLIGIBILITY ENHANCEMENT SYSTEM AND ACOUSTIC ECHO CANCELLER
    51.
    发明申请
    INTEGRATED SPEECH INTELLIGIBILITY ENHANCEMENT SYSTEM AND ACOUSTIC ECHO CANCELLER 有权
    集成语音智能增强系统和耳塞式收音机

    公开(公告)号:US20090281805A1

    公开(公告)日:2009-11-12

    申请号:US12464624

    申请日:2009-05-12

    IPC分类号: G10L15/20

    摘要: A system and method is described that improves the intelligibility of a far-end telephone speech signal to a user of a telephony device in the presence of near-end background noise. As described herein, the system and method improves the intelligibility of the far-end telephone speech signal in a manner that does not require user input and that minimizes the distortion of the far-end telephone speech signal. The system is integrated with an acoustic echo canceller and shares information therewith.

    摘要翻译: 描述了一种在存在近端背景噪声的情况下提高远端电话语音信号给​​电话设备的用户的可懂度的系统和方法。 如这里所述,该系统和方法以不需要用户输入并使远端电话语音信号的失真最小化的方式提高了远端电话语音信号的可懂度。 该系统与声学回声消除器集成并与其共享信息。

    PACKET LOSS CONCEALMENT FOR SUB-BAND PREDICTIVE CODING BASED ON EXTRAPOLATION OF SUB-BAND AUDIO WAVEFORMS
    52.
    发明申请
    PACKET LOSS CONCEALMENT FOR SUB-BAND PREDICTIVE CODING BASED ON EXTRAPOLATION OF SUB-BAND AUDIO WAVEFORMS 有权
    基于子带音频波形扩展的子带预测编码的分组丢失隐藏

    公开(公告)号:US20090240492A1

    公开(公告)日:2009-09-24

    申请号:US12474855

    申请日:2009-05-29

    IPC分类号: G10L19/00

    摘要: A technique is described for concealing the effect of a lost frame in a series of frames representing an encoded audio signal in a sub-band predictive coding system. In accordance with the technique, a first synthesized sub-band audio signal is synthesized, wherein synthesizing the first synthesized sub-band audio signal comprises performing waveform extrapolation based on a stored first sub-band decoded audio signal. A second synthesized sub-band audio signal is also synthesized, wherein synthesizing the second synthesized sub-band audio signal comprises performing waveform extrapolation based on the stored second sub-band decoded audio signal. The first synthesized sub-band audio signal and the second synthesized sub-band audio signal are combined to generate a synthesized full-band output audio signal corresponding to a lost frame.

    摘要翻译: 描述了一种用于在子带预测编码系统中隐藏表示编码音频信号的一系列帧中的丢失帧的影响的技术。 根据该技术,合成第一合成子带音频信号,其中合成第一合成子带音频信号包括基于存储的第一子带解码音频信号执行波形外推。 还合成了第二合成子带音频信号,其中合成第二合成子带音频信号包括基于所存储的第二子带解码音频信号执行波形外推。 第一合成子带音频信号和第二合成子带音频信号被组合以产生对应于丢失帧的合成全频带输出音频信号。

    Method for adaptive filtering
    53.
    发明授权
    Method for adaptive filtering 有权
    自适应滤波方法

    公开(公告)号:US07478040B2

    公开(公告)日:2009-01-13

    申请号:US10968333

    申请日:2004-10-20

    IPC分类号: G10L21/02 G10L21/00 G10L11/06

    CPC分类号: G10L19/26 G10L21/0364

    摘要: A method for adaptive long-term filtering of an audio signal, such as a decoded speech signal. The method includes measuring a smoothed periodicity of an audio signal segment, such as an audio frame, wherein the smoothed periodicity is measured by low-pass filtering an instantaneous periodicity of the audio signal segment. The periodicity of the audio signal segment is then increased in a manner that depends upon whether the smoothed periodicity is less than a predetermined threshold. By utilizing a smoothed periodicity measurement in this fashion, more accurate control of the post-filter is provided as compared to conventional solutions. Additionally, the method includes deriving filters by interpolating between filter responses of adjacent audio signal segments to minimize distortion at segment boundaries.

    摘要翻译: 一种用于对诸如解码语音信号的音频信号进行自适应长期滤波的方法。 该方法包括测量诸如音频帧的音频信号段的平滑的周期性,其中通过对音频信号段的瞬时周期进行低通滤波来测量平滑周期。 然后以取决于平滑周期是否小于预定阈值的方式增加音频信号段的周期性。 通过以这种方式利用平滑的周期性测量,与常规解决方案相比,提供了对后置滤波器的更精确的控制。 此外,该方法包括通过在相邻音频信号段的滤波器响应之间进行插值来导出滤波器,以最小化分段边界处的失真。

    Re-phasing of Decoder States After Packet Loss
    54.
    发明申请
    Re-phasing of Decoder States After Packet Loss 有权
    数据包丢失后重新分解解码器状态

    公开(公告)号:US20080046237A1

    公开(公告)日:2008-02-21

    申请号:US11838905

    申请日:2007-08-15

    IPC分类号: G10L21/00

    摘要: A technique is described herein for updating a state of a decoder configured to decode a series of frames representing an encoded audio signal. In accordance with the technique, an output audio signal associated with a lost frame in the series of frames is synthesized. The decoder state is set to align with the synthesized output audio signal at a frame boundary. An extrapolated signal is generated based on the synthesized output audio signal. A time lag is calculated between the extrapolated signal and a decoded audio signal associated with a first received frame after the lost frame in the series of frames, wherein the time lag represents a phase difference between the extrapolated signal and the decoded audio signal. The decoder state is then reset based on the time lag.

    摘要翻译: 本文描述了一种用于更新被配置为对表示编码音频信号的一系列帧进行解码的解码器的状态的技术。 根据该技术,合成与一系列帧中的丢失帧相关联的输出音频信号。 解码器状态被设置为与帧边界处的合成输出音频信号对准。 基于合成的输出音频信号产生外插信号。 在所述一系列帧中的丢失帧之后,在外推信号和与第一接收帧相关联的解码音频信号之间计算时间滞后,其中所述时间延迟表示外推信号和解码音频信号之间的相位差。 然后根据时间滞后重新设置解码器状态。

    Constrained and Controlled Decoding After Packet Loss
    55.
    发明申请
    Constrained and Controlled Decoding After Packet Loss 审中-公开
    丢包后约束和受控解码

    公开(公告)号:US20080046236A1

    公开(公告)日:2008-02-21

    申请号:US11838899

    申请日:2007-08-15

    IPC分类号: G10L21/02

    摘要: A technique is described herein for reducing audible artifacts in an audio output signal generated by decoding a received frame in a series of frames representing an encoded audio signal in a predictive coding system. In accordance with the technique, it is determined if the received frame is one of a predefined number of received frames that follow a lost frame in the series of the frames. Responsive to determining that the received frame is one of the predefined number of received frames, at least one parameter or signal associated with the decoding of the received frame is altered from a state associated with normal decoding. The received frame is then decoded in accordance with the at least one parameter or signal to generate a decoded audio signal. The audio output signal is then generated based on the decoded audio signal.

    摘要翻译: 本文描述了一种技术,用于通过在代表预测编码系统中的编码音频信号的一系列帧中对接收到的帧进行解码而产生的音频输出信号中减少可听见的伪影。 根据该技术,确定接收到的帧是否是在一系列帧中的丢失帧之后的预定数量的接收帧中的一个。 响应于确定接收到的帧是预定数量的接收帧之一,与所接收的帧的解码相关联的至少一个参数或信号从与正常解码相关联的状态改变。 接收的帧然后根据至少一个参数或信号被解码以产生解码的音频信号。 然后基于解码的音频信号产生音频输出信号。

    Packet Loss Concealment for Sub-band Predictive Coding Based on Extrapolation of Full-band Audio Waveform
    56.
    发明申请
    Packet Loss Concealment for Sub-band Predictive Coding Based on Extrapolation of Full-band Audio Waveform 审中-公开
    基于全波段音频波形外推的子带预测编码的丢包隐藏

    公开(公告)号:US20080046233A1

    公开(公告)日:2008-02-21

    申请号:US11838885

    申请日:2007-08-15

    IPC分类号: G10L11/00

    摘要: A technique for concealing the effect of a lost frame in a series of frames representing an encoded audio signal in a sub-band predictive coding system is provided. In accordance with the technique, one or more received frames in the series of frames are decoded to generate a full-band output audio signal, wherein the full-band output audio signal comprises a combination of at least a first sub-band decoded audio signal and a second sub-band decoded audio signal. The full-band output audio signal corresponding to the one or more received frames is stored. Then, a full-band output audio signal corresponding to the lost frame is synthesized, wherein synthesizing the full-band output audio signal corresponding to the lost frame comprises performing waveform extrapolation based on the stored full-band output audio signal corresponding to the one or more received frames.

    摘要翻译: 提供了一种用于在子带预测编码系统中隐藏表示编码音频信号的一系列帧中的丢失帧的影响的技术。 根据该技术,对一系列帧中的一个或多个接收帧进行解码以产生全频带输出音频信号,其中全频带输出音频信号包括至少第一子带解码音频信号 和第二子带解码音频信号。 存储对应于一个或多个接收帧的全频带输出音频信号。 然后,合成对应于丢失帧的全频带输出音频信号,其中合成对应于丢失帧的全频带输出音频信号包括基于所存储的全频带输出音频信号对应于一个或者 更多收到的帧。

    Packet Loss Concealment for a Sub-band Predictive Coder Based on Extrapolation of Excitation Waveform
    57.
    发明申请
    Packet Loss Concealment for a Sub-band Predictive Coder Based on Extrapolation of Excitation Waveform 有权
    基于激发波形外推的子带预测编码器的丢包隐藏

    公开(公告)号:US20080040122A1

    公开(公告)日:2008-02-14

    申请号:US11835716

    申请日:2007-08-08

    IPC分类号: G10L19/00

    CPC分类号: G10L19/0208 G10L19/005

    摘要: Systems and methods are described for performing packet loss concealment using an extrapolation of an excitation waveform in a sub-band predictive speech coder, such as an ITU-T Recommendation G.722 wideband speech coder. The systems and methods are useful for concealing the quality-degrading effects of packet loss in a sub-band predictive coder and address some sub-band architectural issues when applying excitation extrapolation techniques to such sub-band predictive coders.

    摘要翻译: 描述了使用在诸如ITU-T G.722建议书G.722宽带语音编码器的子带预测语音编码器中外推激励波形来执行分组丢失隐藏的系统和方法。 这些系统和方法对于隐藏子带预测编码器中的分组丢失的质量降级效应是有用的,并且当向这种子带预测编码器应用激励外推技术时,解决某些子带架构问题。

    Efficient excitation quantization in noise feedback coding with general noise shaping
    58.
    发明授权
    Efficient excitation quantization in noise feedback coding with general noise shaping 有权
    噪声反馈编码中的高效激励量化与通用噪声整形

    公开(公告)号:US07206740B2

    公开(公告)日:2007-04-17

    申请号:US10216276

    申请日:2002-08-12

    IPC分类号: G10L19/04 G10L19/12 G10L19/00

    CPC分类号: G10L19/26

    摘要: In a Noise Feedback Coding (NFC) system operable in a ZERO-STATE condition and a ZERO-INPUT condition, the NFC system including at least one filter having a filter memory, a method of updating the filter memory. The method comprises: (a) producing a ZERO-STATE contribution to the filter memory when the NFC system is in the ZERO-STATE condition; (b) producing a ZERO-INPUT contribution to the filter memory when the NFC system is in the ZERO-INPUT condition; and (c) updating the filter memory as a function of both the ZERO-STATE contribution and the ZERO-INPUT contribution.

    摘要翻译: 在可在零状态和零输入条件下操作的噪声反馈编码(NFC)系统中,NFC系统包括至少一个具有滤波器存储器的滤波器,更新滤波器存储器的方法。 该方法包括:(a)当NFC系统处于零状态时,向滤波器存储器产生零状态贡献; (b)当NFC系统处于零输入状态时,向滤波器存储器产生ZERO-INPUT贡献; 和(c)根据ZERO-STATE贡献和ZERO-INPUT贡献的函数来更新滤波器存储器。

    Efficient excitation quantization in noise feedback coding with general noise shaping
    59.
    发明授权
    Efficient excitation quantization in noise feedback coding with general noise shaping 有权
    噪声反馈编码中的高效激励量化与通用噪声整形

    公开(公告)号:US06751587B2

    公开(公告)日:2004-06-15

    申请号:US10216442

    申请日:2002-08-12

    IPC分类号: G01L2102

    CPC分类号: G10L19/06

    摘要: In a Noise Feedback Coding (NFC) system having a corresponding ZERO-STATE filter structure, the first ZERO-STATE filter structure including multiple filters, a method of producing a ZERO-STATE response error vector. The method includes: (a) transforming the first ZERO-STATE filter structure to a second ZERO-STATE filter structure including only an all-zero filter, the all-zero filter having a filter response substantially equivalent to a filter response of the ZERO-STATE filter structure including multiple filters; and (b) filtering a VQ codevector with the all-zero filter to produce the ZERO-STATE response error vector corresponding to the VQ codevector.

    摘要翻译: 在具有相应的零状态滤波器结构的噪声反馈编码(NFC)系统中,包括多个滤波器的第一零状态滤波器结构,产生零状态响应误差向量的方法。 该方法包括:(a)将第一ZERO-STATE滤波器结构转换为仅包括全零滤波器的第二ZERO-STATE滤波器结构,全零滤波器具有基本上等于ZERO-STATE滤波器响应的滤波器响应, STATE滤波器结构包括多个滤波器; 和(b)用全零滤波器对VQ码矢量进行滤波以产生对应于VQ码矢量的零状态响应误差向量。

    NOISE SUPPRESSION USING MULTIPLE SENSORS OF A COMMUNICATION DEVICE
    60.
    发明申请
    NOISE SUPPRESSION USING MULTIPLE SENSORS OF A COMMUNICATION DEVICE 有权
    使用通信设备的多个传感器的噪声抑制

    公开(公告)号:US20120185246A1

    公开(公告)日:2012-07-19

    申请号:US13174964

    申请日:2011-07-01

    IPC分类号: G10L21/02

    摘要: Techniques are described herein that suppress noise using multiple sensors (e.g., microphones) of a communication device. Noise modeling (e.g., estimation of noise basis vectors and noise weighting vectors) is performed with respect to a noise signal during operation of a communication device to provide a noise model. The noise model includes noise basis vectors and noise coefficients that represent noise provided by audio sources other than a user of the communication device. Speech modeling (e.g., estimation of speech basis vectors and speech weighting) is performed to provide a speech model. The speech model includes speech basis vectors and speech coefficients that represent speech of the user. A noisy speech signal is processed using the noise basis vectors, the noise coefficients, the speech basis vectors, and the speech coefficients to provide a clean speech signal.

    摘要翻译: 本文描述了使用通信设备的多个传感器(例如,麦克风)抑制噪声的技术。 在通信设备的操作期间,相对于噪声信号执行噪声建模(例如噪声基矢量和噪声加权矢量的估计)以提供噪声模型。 噪声模型包括噪声基矢量和表示由通信设备的用户以外的音频源提供的噪声的噪声系数。 执行语音建模(例如,语音基本向量的估计和语音加权)以提供语音模型。 语音模型包括表示用户语音的语音基向量和语音系数。 使用噪声基矢量,噪声系数,语音基矢量和语音系数来处理噪声语音信号以提供干净的语音信号。