公开(公告)号：US08670990B2

公开(公告)日：2014-03-11

申请号：US12847120

申请日：2010-07-30

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

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

IPC分类号： G10L21/04 ,  G10L11/06

CPC分类号： G10L19/22 ,  G10L19/08

摘要： Systems and methods are described that utilize dynamic time scale modification (TSM) to achieve reduced bit rate audio coding. In accordance with embodiments, different levels of TSM compression are selectively applied to segments of an input speech signal prior to encoding thereof by an encoder. Encoded TSM-compressed segments are received at a decoder which decodes such segments and then applies an appropriate level of TSM decompression to each based on information received from the encoder. By selectively applying different levels of TSM compression to segments of an input speech signal prior to encoding, a coding bit rate associated with the encoder/decoder is reduced. Furthermore, by selecting a level of TSM compression for each segment of the input speech signal that takes into account certain local characteristics of that signal, such bit rate reduction is provided without introducing unacceptable levels of distortion into an output speech signal produced by the decoder.

摘要翻译： 描述了利用动态时间尺度修正（TSM）来实现降低的比特率音频编码的系统和方法。 根据实施例，在由编码器对TSM压缩进行编码之前，将不同级别的TSM压缩选择性地应用于输入语音信号的段。 编码的TSM压缩段在解码器处被接收，解码器对这些段进行解码，然后基于从编码器接收的信息向每个TSM解压缩应用适当级别的TSM解压缩。 通过在编码之前选择性地对输入语音信号的段应用不同级别的TSM压缩，减少与编码器/解码器相关联的编码比特率。 此外，通过选择考虑到该信号的某些局部特性的输入语音信号的每个段的TSM压缩级别，提供这样的比特率降低，而不会将不可接受的失真电平引入到由解码器产生的输出语音信号中。

公开(公告)号：US20110029317A1

公开(公告)日：2011-02-03

申请号：US12847120

申请日：2010-07-30

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

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

IPC分类号： G10L19/00

CPC分类号： G10L19/22 ,  G10L19/08

摘要： Systems and methods are described that utilize dynamic time scale modification (TSM) to achieve reduced bit rate audio coding. In accordance with embodiments, different levels of TSM compression are selectively applied to segments of an input speech signal prior to encoding thereof by an encoder. Encoded TSM-compressed segments are received at a decoder which decodes such segments and then applies an appropriate level of TSM decompression to each based on information received from the encoder. By selectively applying different levels of TSM compression to segments of an input speech signal prior to encoding, a coding bit rate associated with the encoder/decoder is reduced. Furthermore, by selecting a level of TSM compression for each segment of the input speech signal that takes into account certain local characteristics of that signal, such bit rate reduction is provided without introducing unacceptable levels of distortion into an output speech signal produced by the decoder.

摘要翻译： 描述了利用动态时间尺度修正（TSM）来实现降低的比特率音频编码的系统和方法。 根据实施例，在由编码器对TSM压缩进行编码之前，将不同级别的TSM压缩选择性地应用于输入语音信号的段。 编码的TSM压缩段在解码器处被接收，解码器对这些段进行解码，然后基于从编码器接收的信息向每个TSM解压缩应用适当级别的TSM解压缩。 通过在编码之前选择性地对输入语音信号的段应用不同级别的TSM压缩，减少与编码器/解码器相关联的编码比特率。 此外，通过选择考虑到该信号的某些局部特性的输入语音信号的每个段的TSM压缩级别，提供这样的比特率降低，而不会将不可接受的失真电平引入到由解码器产生的输出语音信号中。

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：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：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。

公开(公告)号：US20120010882A1

公开(公告)日：2012-01-12

申请号：US13240283

申请日：2011-09-22

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

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

IPC分类号： G10L21/02

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

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。

公开(公告)号：US20110320213A1

公开(公告)日：2011-12-29

申请号：US13227239

申请日：2011-09-07

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： G10L19/00

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

摘要： A technique is described for use in a decoder configured to decode a series of frames representing an encoded audio signal. The technique is for transitioning between a lost frame and one or more received frames following the lost frame in the series of frames. In accordance with the technique, an output audio signal associated with the lost frame is synthesized. 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 the received frame(s), wherein the time lag represents a phase difference between the extrapolated signal and the decoded audio signal. The decoded audio signal is time-warped based on the time lag, wherein time-warping the decoded audio signal comprises stretching or shrinking the decoded audio signal in the time domain.

摘要翻译： 描述了一种用于解码器的技术，其被配置为对表示编码音频信号的一系列帧进行解码。 该技术用于在丢失帧与一系列帧中的丢失帧之后的一个或多个接收帧之间进行转换。 根据该技术，合成与丢失帧相关联的输出音频信号。 基于合成的输出音频信号产生外插信号。 在外推信号和与接收到的帧相关联的解码音频信号之间计算时间滞后，其中时间延迟表示外推信号和解码音频信号之间的相位差。 经解码的音频信号基于时间滞后而变化，其中解码的音频信号的时间扭曲包括在时域中拉伸或收缩解码的音频信号。

公开(公告)号：US08024192B2

公开(公告)日：2011-09-20

申请号：US11838908

申请日：2007-08-15

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： G10L13/06

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

摘要： A technique is described for use in a decoder configured to decode a series of frames representing an encoded audio signal. The technique is for transitioning between a lost frame and one or more received frames following the lost frame in the series of frames. In accordance with the technique, an output audio signal associated with the lost frame is synthesized. 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 the received frame(s), wherein the time lag represents a phase difference between the extrapolated signal and the decoded audio signal. The decoded audio signal is time-warped based on the time lag, wherein time-warping the decoded audio signal comprises stretching or shrinking the decoded audio signal in the time domain.

摘要翻译： 描述了一种用于解码器的技术，其被配置为对表示编码音频信号的一系列帧进行解码。 该技术用于在丢失帧与一系列帧中的丢失帧之后的一个或多个接收帧之间进行转换。 根据该技术，合成与丢失帧相关联的输出音频信号。 基于合成的输出音频信号产生外插信号。 在外推信号和与接收到的帧相关联的解码音频信号之间计算时间滞后，其中时间延迟表示外推信号和解码音频信号之间的相位差。 经解码的音频信号基于时间滞后而变化，其中解码的音频信号的时间扭曲包括在时域中拉伸或收缩解码的音频信号。

公开(公告)号：US08005678B2

公开(公告)日：2011-08-23

申请号：US11838905

申请日：2007-08-15

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

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

IPC分类号： G10L13/06

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

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。

公开(公告)号：US08214206B2

公开(公告)日：2012-07-03

申请号：US13240283

申请日：2011-09-22

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

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

IPC分类号： G10L19/14

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

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。