公开(公告)号：US20090287496A1

公开(公告)日：2009-11-19

申请号：US12510445

申请日：2009-07-28

申请人： Jes Thyssen ,  Wilfrid LeBlanc ,  Juin-Hwey Chen

发明人： Jes Thyssen ,  Wilfrid LeBlanc ,  Juin-Hwey Chen

IPC分类号： G10L19/00 ,  H03G3/00

CPC分类号： H03G9/025 ,  H03G7/007 ,  H03G9/005

摘要： A loudness enhancement system and method is described that increases the loudness of an audio signal being played back by an audio device that places limits on the dynamic range of the audio signal. In an embodiment, the loudness enhancement system and method compresses the audio signal to an adaptively-determined compression limit that is greater than or equal to a maximum desired output level and then applies an adaptively-determined degree of soft clipping to the compressed audio signal. The compression limit and degree of soft clipping may be determined based on an overload measure that is calculated for successive portions of the audio signal. The loudness enhancement system and method advantageously operates in a manner that generates less distortion than the method of simply over-driving the audio signal such that hard-clipping occurs.

摘要翻译： 描述了一种响度增强系统和方法，其增加了对限制音频信号的动态范围的音频设备正在播放的音频信号的响度。 在一个实施例中，响度增强系统和方法将音频信号压缩到大于或等于最大期望输出电平的自适应确定的压缩极限，然后对压缩音频信号应用自适应确定的软限幅度。 可以基于为音频信号的连续部分计算的过载测量来确定软限幅的压缩极限和程度。 响度增强系统和方法有利地以比简单地过度驱动音频信号的方法产生更少的失真的方式进行操作，从而发生硬削波。

公开(公告)号：US20090281803A1

公开(公告)日：2009-11-12

申请号：US12464590

申请日：2009-05-12

申请人： Juin-Hwey Chen ,  Jes Thyssen

发明人： Juin-Hwey Chen ,  Jes Thyssen

IPC分类号： G10L19/14

CPC分类号： G10L21/0208 ,  G10L19/012 ,  G10L21/0232

摘要： A speech intelligibility enhancement (SIE) system and method is described that improves the intelligibility of a speech signal to be played back by an audio device when the audio device is located in an environment with loud acoustic background noise. In an embodiment, the audio device comprises a near-end telephony terminal and the speech signal comprises a speech signal received over a communication network from a far-end telephony terminal for playback at the near-end telephony terminal.

摘要翻译： 描述了一种语音清晰度增强（SIE）系统和方法，当音频设备位于具有大声背景噪声的环境中时，提高了由音频设备重放的语音信号的清晰度。 在一个实施例中，音频设备包括近端电话终端，并且语音信号包括通过通信网络从远端电话终端接收的用于在近端电话终端回放的语音信号。

公开(公告)号：US07324937B2

公开(公告)日：2008-01-29

申请号：US10968300

申请日：2004-10-20

申请人： Jes Thyssen ,  Juin-Hwey Chen

发明人： Jes Thyssen ,  Juin-Hwey Chen

IPC分类号： G10L19/08

CPC分类号： G10L19/26 ,  G10L19/005 ,  G10L19/12

摘要： A method for performing packet loss concealment (PLC) and/or frame erasure concealment (FEC) in a speech decoder of a voice communication system. In accordance with the method, if a segment of an encoded speech signal is determined to be bad, an excitation signal is derived by scaling a random sequence of samples, and long-term and short-term predictive parameters are derived based on parameters associated with a previously-decoded segment. The excitation signal is then filtered by a long-term synthesis filter and a short-term synthesis filter under the control of the respective long-term and short-term predictive parameters. If the number of consecutively-received bad segments exceeds a predetermined threshold, the decoded speech signal is gradually reduced.

摘要翻译： 一种在语音通信系统的语音解码器中执行分组丢失隐藏（PLC）和/或帧擦除隐藏（FEC）的方法。 根据该方法，如果确定编码语音信号的段是坏的，则通过缩放样本的随机序列来导出激励信号，并且基于与...相关的参数导出长期和短期预测参数 先前解码的段。 然后在长期和短期预测参数的控制下，通过长期合成滤波器和短期合成滤波器对激发信号进行滤波。 如果连续接收的坏段的数量超过预定阈值，则解码的语音信号逐渐减少。

公开(公告)号：US20120123772A1

公开(公告)日：2012-05-17

申请号：US13295818

申请日：2011-11-14

申请人： Jes Thyssen ,  Huaiyu Zeng ,  Juin-Hwey Chen ,  Nelson Sollenberger ,  Xianxian Zhang

发明人： Jes Thyssen ,  Huaiyu Zeng ,  Juin-Hwey Chen ,  Nelson Sollenberger ,  Xianxian Zhang

IPC分类号： G10L21/02

CPC分类号： G10L21/0208 ,  G10L21/0272 ,  G10L2021/02165 ,  H04R1/245 ,  H04R2410/07

摘要： Multi-channel noise suppression systems and methods are described that omit the traditional delay-and-sum fixed beamformer in devices that include a primary speech microphone and at least one noise reference microphone with the desired speech being in the near-field of the device. The multi-channel noise suppression systems and methods use a blocking matrix (BM) to remove desired speech in the input speech signal received by the noise reference microphone to get a “cleaner” background noise component. Then, an adaptive noise canceler (ANC) is used to remove the background noise in the input speech signal received by the primary speech microphone based on the “cleaner” background noise component to achieve noise suppression. The filters implemented by the BM and ANC are derived using closed-form solutions that require calculation of time-varying statistics of complex frequency domain signals in the noise suppression system.

摘要翻译： 描述了多通道噪声抑制系统和方法，其省略了传统的延迟和总和固定波束形成器，其包括主话音麦克风和至少一个噪声参考麦克风，其中所需语音位于设备的近场。 多声道噪声抑制系统和方法使用阻塞矩阵（BM）去除由噪声参考麦克风接收的输入语音信号中的期望语音以获得“更干净”的背景噪声分量。 然后，使用自适应噪声消除器（ANC）来基于“更干净的”背景噪声分量来消除由主话音麦克风接收的输入语音信号中的背景噪声，以实现噪声抑制。 由BM和ANC实现的滤波器是使用需要在噪声抑制系统中计算复杂频域信号的时变统计的闭式解决方案导出的。

公开(公告)号：US08000960B2

公开(公告)日：2011-08-16

申请号：US11838891

申请日：2007-08-15

申请人： Juin-Hwey Chen ,  Robert W. Zopf ,  Jes Thyssen

发明人： Juin-Hwey Chen ,  Robert W. Zopf ,  Jes Thyssen

IPC分类号： G10L19/10

CPC分类号： G10L19/0204 ,  G10L19/005 ,  G10L19/04

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。

公开(公告)号：US07684521B2

公开(公告)日：2010-03-23

申请号：US11048916

申请日：2005-02-03

申请人： Jes Thyssen ,  Juin-Hwey Chen ,  Nambi Seshadri

发明人： Jes Thyssen ,  Juin-Hwey Chen ,  Nambi Seshadri

IPC分类号： H04L27/06 ,  H03M13/00

CPC分类号： H04L1/0045

摘要： Typical communication systems operate with a single channel decoder, and hence would have to settle for the performance from the single channel decoder regardless of the conditions of the communications channel. The present invention uses a hybrid channel decoder comprising multiple channel decoders, each configured to optimize the quality of the re-constructed signal for different channel conditions. Therefore, the desired decoder can be selected as conditions of the communications channel, or the data signal, change over time, so as to optimize the re-constructed data signal. In embodiments, the data signal is a speech signal.

摘要翻译： 典型的通信系统使用单个信道解码器进行操作，因此无论通信信道的条件如何，都必须从单信道解码器处理性能。 本发明使用包括多个信道解码器的混合信道解码器，每个信道解码器被配置为优化用于不同信道条件的重构信号的质量。 因此，可以选择期望的解码器作为通信信道的条件或数据信号随时间变化，以便优化重构的数据信号。 在实施例中，数据信号是语音信号。

公开(公告)号：US20090281800A1

公开(公告)日：2009-11-12

申请号：US12464517

申请日：2009-05-12

申请人： Wilfrid LeBlanc ,  Juin-Hwey Chen ,  Jes Thyssen

发明人： Wilfrid LeBlanc ,  Juin-Hwey Chen ,  Jes Thyssen

IPC分类号： G10L19/14

CPC分类号： G10L21/0208 ,  G10L19/012 ,  G10L21/0232

摘要： A speech intelligibility enhancement (SIE) system and method is described that improves the intelligibility of a speech signal to be played back by an audio device when the audio device is located in an environment with loud acoustic background noise. In an embodiment, the audio device comprises a near-end telephony terminal and the speech signal comprises a speech signal received over a communication network from a far-end telephony terminal for playback at the near-end telephony terminal.

摘要翻译： 描述了一种语音清晰度增强（SIE）系统和方法，当音频设备位于具有大声背景噪声的环境中时，提高了由音频设备重放的语音信号的可懂度。 在一个实施例中，音频设备包括近端电话终端，并且语音信号包括通过通信网络从远端电话终端接收的用于在近端电话终端回放的语音信号。

公开(公告)号：US20090248405A1

公开(公告)日：2009-10-01

申请号：US12474809

申请日：2009-05-29

申请人： Juin-Hwey Chen ,  Jes Thyssen ,  Robert W. Zopf

发明人： Juin-Hwey Chen ,  Jes Thyssen ,  Robert W. Zopf

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。

公开(公告)号：US07512535B2

公开(公告)日：2009-03-31

申请号：US10183554

申请日：2002-06-28

申请人： Juin-Hwey Chen ,  Jes Thyssen

发明人： Juin-Hwey Chen ,  Jes Thyssen

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

CPC分类号： G10L19/26

摘要： A filter controller processes a decoded speech (DS) signal. The DS signal has a spectral envelope including a first plurality of formant peaks having different respective amplitudes. The controller produces, from the DS signal, a spectrally-flattened DS signal that is a time-domain signal. The spectrally-flattened time-domain DS signal has a spectral envelope including a second plurality of formant peaks. Each of the second plurality of formant peaks approximately coincides in frequency with a respective one of the first plurality of formant peaks. Also, the second plurality of formant peaks have approximately equal respective amplitudes. Next, the controller derives, from the spectrally-flattened time-domain DS signal, a set of filter coefficients representative of a filter response that is to be used to filter the DS signal.

摘要翻译： 滤波器控制器处理解码语音（DS）信号。 DS信号具有包括具有不同相应振幅的第一多个共振峰的频谱包络。 控制器从DS信号产生作为时域信号的频谱平坦化的DS信号。 频谱平坦化的时域DS信号具有包括第二多个共振峰的频谱包络。 所述第二多个共振峰中的每一个峰值与所述第一多个共振峰中的相应一个峰值大致重合。 而且，第二多个共振峰具有大致相等的相应振幅。 接下来，控制器从频谱平坦化的时域DS信号中导出代表要用于滤波DS信号的滤波器响应的一组滤波器系数。

公开(公告)号：US20080033718A1

公开(公告)日：2008-02-07

申请号：US11734800

申请日：2007-04-13

申请人： Robert W. Zopf ,  Juin-Hwey Chen ,  Jes Thyssen

发明人： Robert W. Zopf ,  Juin-Hwey Chen ,  Jes Thyssen

IPC分类号： G10L19/02

CPC分类号： G10L19/005 ,  G10L25/78

摘要： An audio decoding system performs frame loss concealment (FLC) when portions of a bit stream representing an audio signal are lost within the context of a digital communication system. The audio decoding system employs two different FLC methods: one designed to perform well for music, and the other designed to perform well for speech. When a frame is deemed lost, the audio decoding system analyzes a previously-decoded audio signal corresponding to previously-decoded frames of an audio bit-stream. Based on the results of the analysis, the lost frame is classified as either speech or music. Using this classification, other signal analysis, and knowledge of the employed FLC methods, the audio decoding system selects the appropriate FLC method which then performs FLC on the lost frame.

摘要翻译： 音频解码系统在数字通信系统的上下文中丢失表示音频信号的比特流的部分时执行帧丢失隐藏（FLC）。 音频解码系统采用两种不同的FLC方法：一种被设计为对音乐表现良好，另一种被设计为对演讲表现良好。 当帧被认为丢失时，音频解码系统分析对应于音频比特流的先前解码的帧的先前解码的音频信号。 基于分析结果，丢失的帧被分类为语音或音乐。 使用这种分类，其他信号分析和所采用的FLC方法的知识，音频解码系统选择适当的FLC方法，然后在丢帧上执行FLC。