公开(公告)号：US08706479B2

公开(公告)日：2014-04-22

申请号：US12614153

申请日：2009-11-06

申请人： Robert W. Zopf ,  Laurent Pilati

发明人： Robert W. Zopf ,  Laurent Pilati

IPC分类号： G10L19/00

CPC分类号： G10L19/005 ,  G10L19/0204

摘要： Packet loss concealment systems and methods are described that may be used in conjunction with a Bluetooth® Low-Complexity Sub-band Coding (LC-SBC) codec or other sub-band codecs, including but not limited to an MPEG-1 Audio Layer 3 (MP3) codec, an Advanced Audio Coding (AAC) codec, and a Dolby AC-3 codec.

摘要翻译： 描述了可以与蓝牙低复杂度子带编码（LC-SBC）编解码器或其他子带编解码器结合使用的分组丢失隐藏系统和方法，包括但不限于MPEG-1音频层3 （MP3）编解码器，高级音频编码（AAC）编解码器和杜比AC-3编解码器。

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：US08578247B2

公开(公告)日：2013-11-05

申请号：US12431184

申请日：2009-04-28

申请人： Robert W. Zopf ,  Vivek Kumar ,  Michael S. Gustafson

发明人： Robert W. Zopf ,  Vivek Kumar ,  Michael S. Gustafson

IPC分类号： H03M13/00

CPC分类号： G10L19/005 ,  H04L1/0045

摘要： Systems and methods are described for managing bit errors present in an encoded bit stream representative of a portion of an audio signal, wherein the encoded bit stream is received via a channel in a wireless communications system. The channel may comprise, for example, a Synchronous Connection-Oriented (SCO) channel or an Extended SCO (eSCO) channel in a Bluetooth® wireless communications system.

摘要翻译： 描述了用于管理存在于表示音频信号的一部分的编码比特流中的比特错误的系统和方法，其中经由无线通信系统中的信道接收编码比特流。 该信道可以包括例如Bluetooth®无线通信系统中的同步面向连接（SCO）信道或扩展SCO（eSCO）信道。

公开(公告)号：US08340977B2

公开(公告)日：2012-12-25

申请号：US12436472

申请日：2009-05-06

申请人： Robert W. Zopf

发明人： Robert W. Zopf

IPC分类号： G10L19/00 ,  G10L21/02

CPC分类号： H03M3/024 ,  G10L19/005

摘要： A system and method is described for compensating for the effects of a corrupted Continuously Variable Delta Slope Modulation (CVSD) decoder memory state on a decoded audio signal. In accordance with the system and method, a first estimated step size associated with a first frame of the decoded audio signal is calculated and a second estimated step size associated with a replacement frame generated to conceal bit errors in the first frame of the decoded audio signal is calculated. At least a second frame of the decoded audio signal is then modified based on the first estimated step size and the second estimated step size.

摘要翻译： 描述了一种系统和方法，用于补偿经解码的音频信号的损坏的连续可变增量斜率调制（CVSD）解码器存储器状态的影响。 根据系统和方法，计算与解码音频信号的第一帧相关联的第一估计步长，并且产生与产生的替换帧相关联的第二估计步长，以隐藏解码音频信号的第一帧中的位错误 被计算。 然后，基于第一估计步长和第二估计步长修改解码音频信号的至少第二帧。

公开(公告)号：US20110209029A1

公开(公告)日：2011-08-25

申请号：US13007020

申请日：2011-01-14

申请人： Robert W. Zopf

发明人： Robert W. Zopf

IPC分类号： H03M13/29 ,  G06F11/10

CPC分类号： H03M13/15 ,  H03M13/09 ,  H03M13/37 ,  H03M13/6306 ,  H04L1/0045 ,  H04L1/0057

摘要： Low complexity error correction using cyclic redundancy check (CRC). Communications between at communication devices, sometimes including at least one redundant transmission from a transmitter to a receiver, undergo low complexity error correction. CRC may be employed in conjunction with using any desired type of ECC or using uncoded modulation. Based on CRC determined bit-errors, as few as a singular syndrome associated with a singular bit-error or a linear combination of syndromes associated with two or more singular bit-errors within two or more received signal sequences are employed to perform error correction of the received signal. Real time combinations of multiple syndromes associated with respective single bit-errors (that may themselves be calculated off-line) are employed in accordance with error correction. In addition to CRC, any ECC may be employed including convolutional code, RS code, turbo code, TCM code, TTCM code, LDPC code, or BCH code.

摘要翻译： 使用循环冗余校验（CRC）的低复杂度纠错。 在通信设备之间的通信，有时包括从发射机到接收机的至少一个冗余传输，经历低复杂度的纠错。 可以结合使用任何期望类型的ECC或使用未编码调制来采用CRC。 基于CRC确定的位错误，使用与在两个或更多个接收信号序列内与两个或多个奇异位错误相关联的奇异位错误或综合征的线性组合相关联的奇异综合来执行错误校正 接收信号。 根据纠错采用与相应的单个位错误相关联的多个综合征的实时组合（其本身可以离线计算）。 除了CRC之外，可以使用任何ECC，包括卷积码，RS码，turbo码，TCM码，TTCM码，LDPC码或BCH码。

公开(公告)号：US07971108B2

公开(公告)日：2011-06-28

申请号：US12506727

申请日：2009-07-21

申请人： Robert W. Zopf ,  Siukai Mak

发明人： Robert W. Zopf ,  Siukai Mak

IPC分类号： G06F11/00

CPC分类号： G10L19/005

摘要： Systems and methods are described for managing bit errors present in a series of encoded bits representative of a portion of an audio signal, wherein the series of encoded bits is received over a communication link in an audio communications system. At least one characteristic of a portion of a received modulated carrier signal that is demodulated to produce the series of encoded bits is determined. A number of bit errors present in the series of encoded bits is then determined based on the at least one characteristic. Based on the estimated number of bit errors, one of a plurality of methods for producing a series of digital audio samples representative of the portion of the audio signal is selectively performed. The series of digital audio samples produced by the selected method is then converted into a form suitable for playback to a user.

摘要翻译： 描述了用于管理存在于表示音频信号的一部分的一系列编码比特中的比特错误的系统和方法，其中通过音频通信系统中的通信链路接收一系列编码比特。 确定被解调以产生一系列编码比特的接收调制载波信号的一部分的至少一个特性。 然后基于该至少一个特性确定存在于一系列编码比特中的多个比特错误。 基于估计的位错误数量，选择性地执行用于产生表示音频信号部分的一系列数字音频样本的多种方法之一。 然后，通过所选择的方法产生的一系列数字音频样本被转换成适合于播放给用户的形式。

公开(公告)号：US20110099014A1

公开(公告)日：2011-04-28

申请号：US12887353

申请日：2010-09-21

申请人： Robert W. Zopf

发明人： Robert W. Zopf

IPC分类号： G10L13/00

CPC分类号： G10L21/00 ,  G10L17/00 ,  G10L19/0018 ,  G10L19/16

摘要： Systems and methods are described for performing packet loss concealment (PLC) to mitigate the effect of one or more lost frames within a series of frames that represent a speech signal. In accordance with the exemplary systems and methods, PLC is performed by searching a codebook of speech-related parameter profiles to identify content that is being spoken and by selecting a profile associated with the identified content for use in predicting or estimating speech-related parameter information associated with one or more lost frames of a speech signal. The predicted/estimated speech-related parameter information is then used to synthesize one or more frames to replace the lost frame(s) of the speech signal.

摘要翻译： 描述了用于执行分组丢失隐藏（PLC）以减轻表示语音信号的一系列帧内的一个或多个丢失帧的影响的系统和方法。 根据示例性系统和方法，通过搜索语音相关参数简档的码本来识别正在被说出的内容并且通过选择与所标识的内容相关联的简档来用于预测或估计语音相关参数信息来执行PLC 与语音信号的一个或多个丢失帧相关联。 然后，使用预测/估计的语音相关参数信息来合成一个或多个帧来代替语音信号的丢失帧。

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：US07272556B1

公开(公告)日：2007-09-18

申请号：US09159481

申请日：1998-09-23

申请人： Joseph Gerard Aguilar ,  David A. Campana ,  Raymond Chen ,  Robert B. Dunn ,  Robert J. McAulay ,  Xiaoquin Sun ,  Wei Wang ,  Craig Watkins ,  Robert W. Zopf

发明人： Joseph Gerard Aguilar ,  David A. Campana ,  Raymond Chen ,  Robert B. Dunn ,  Robert J. McAulay ,  Xiaoquin Sun ,  Wei Wang ,  Craig Watkins ,  Robert W. Zopf

IPC分类号： G10L21/00

CPC分类号： G10L19/002 ,  G10L19/093 ,  G10L19/24

摘要： A system and method for processing of audio and speech signals is disclosed, which provide compatibility over a range of communication devices operating at different sampling frequencies and/or bit rates. The analyzer of the system divides the input signal in different portions, at least one of which carries information sufficient to provide intelligible reconstruction of the input signal. The analyzer also encodes separate information about other portions of the signal in an embedded manner, so that a smooth transition can be achieved from low bit-rate to high bit-rate applications. Accordingly, communication devices operating at different sampling rates and/or bit-rates can extract corresponding information from the output bit stream of the analyzer. In the present invention embedded information generally relates to separate parameters of the input signal, or to additional resolution in the transmission of original signal parameters. Non-linear techniques for enhancing the overall performance of the system are also disclosed. Also disclosed is a novel method of improving the quantization of signal parameters. In a specific embodiment the input signal is processed in two or more modes dependent on the state of the signal in a frame. When the signal is determined to be in a transition state, the encoder provides phase information about N sinusoids, which the decoder end uses to improve the quality of the output signal at low bit rates.

摘要翻译： 公开了一种用于处理音频和语音信号的系统和方法，其提供了在不同采样频率和/或比特率下操作的通信设备的范围上的兼容性。 系统的分析器将输入信号分成不同的部分，其中至少一个传送足以提供输入信号的可理解的重建的信息。 分析器还以嵌入的方式编码关于信号的其他部分的单独信息，从而可以从低比特率到高比特率应用实现平滑的转换。 因此，以不同的采样率和/或比特率工作的通信设备可以从分析器的输出比特流中提取相应的信息。 在本发明中，嵌入信息通常涉及输入信号的单独参数，或者涉及原始信号参数传输中的附加分辨率。 还公开了用于增强系统的整体性能的非线性技术。 还公开了一种改进信号参数量化的新方法。 在具体实施例中，输入信号以两个或更多个模式被处理，这取决于帧中的信号的状态。 当信号被确定为处于转换状态时，编码器提供关于N个正弦曲线的相位信息，解码器端用于以低比特率提高输出信号的质量。

公开(公告)号：US09047865B2

公开(公告)日：2015-06-02

申请号：US11889332

申请日：2007-08-10

申请人： Joseph Gerard Aguilar ,  David A. Campana ,  Juin-Hwey Chen ,  Robert B. Dunn ,  Robert J. McAulay ,  Xiaoquin Sun ,  Wei Wang ,  Craig Watkins ,  Robert W. Zopf

发明人： Joseph Gerard Aguilar ,  David A. Campana ,  Juin-Hwey Chen ,  Robert B. Dunn ,  Robert J. McAulay ,  Xiaoquin Sun ,  Wei Wang ,  Craig Watkins ,  Robert W. Zopf

IPC分类号： G10L19/18 ,  G10L19/093 ,  G10L19/24

CPC分类号： G10L19/002 ,  G10L19/093 ,  G10L19/24

摘要： A system and method for processing of audio and speech signals is disclosed, which provide compatibility over a range of communication devices operating at different sampling frequencies and/or bit rates. The analyzer of the system divides the input signal in different portions, at least one of which carries information sufficient to provide intelligible reconstruction of the input signal. The analyzer also encodes separate information about other portions of the signal in an embedded manner, so that a smooth transition can be achieved from low bit-rate to high bit-rate applications. Accordingly, communication devices operating at different sampling rates and/or bit-rates can extract corresponding information from the output bit stream of the analyzer. In the present invention embedded information generally relates to separate parameters of the input signal, or to additional resolution in the transmission of original signal parameters. Non-linear techniques for enhancing the overall performance of the system are also disclosed. Also disclosed is a novel method of improving the quantization of signal parameters. In a specific embodiment the input signal is processed in two or more modes dependent on the state of the signal in a frame. When the signal is determined to be in a transition state, the encoder provides phase information about N sinusoids, which the decoder end uses to improve the quality of the output signal at low bit rates.

摘要翻译： 公开了一种用于处理音频和语音信号的系统和方法，其提供了在不同采样频率和/或比特率下操作的通信设备的范围上的兼容性。 系统的分析器将输入信号分成不同的部分，其中至少一个传送足以提供输入信号的可理解的重建的信息。 分析器还以嵌入的方式编码关于信号的其他部分的单独信息，从而可以从低比特率到高比特率应用实现平滑的转换。 因此，以不同的采样率和/或比特率工作的通信设备可以从分析器的输出比特流中提取相应的信息。 在本发明中，嵌入信息通常涉及输入信号的单独参数，或者涉及原始信号参数传输中的附加分辨率。 还公开了用于增强系统的整体性能的非线性技术。 还公开了一种改进信号参数量化的新方法。 在具体实施例中，输入信号以两个或更多个模式被处理，这取决于帧中的信号的状态。 当信号被确定为处于转换状态时，编码器提供关于N个正弦曲线的相位信息，解码器端用于以低比特率提高输出信号的质量。