公开(公告)号：US07426466B2

公开(公告)日：2008-09-16

申请号：US10897746

申请日：2004-07-22

Applicant: Arasanipalai K. Ananthapadmanabhan ,  Sharath Manjunath ,  Pengjun Huang ,  Eddie-Lun Tik Choy ,  Andrew P. DeJaco

Inventor： Arasanipalai K. Ananthapadmanabhan ,  Sharath Manjunath ,  Pengjun Huang ,  Eddie-Lun Tik Choy ,  Andrew P. DeJaco

IPC: G10L19/14

CPC classification number: G10L19/04 ,  G10L19/0204 ,  G10L19/032 ,  G10L19/08 ,  G10L19/097 ,  G10L19/26 ,  G10L25/12

Abstract: A method and apparatus for predictively quantizing voiced speech includes a parameter generator and a quantizer. The parameter generator is configured to extract parameters from frames of predictive speech such as voiced speech, and to transform the extracted information to a frequency-domain representation. The quantizer is configured to subtract a weighted sum of the parameters for previous frames from the parameter for the current frame. The quantizer is configured to quantize the difference value. A prototype extractor may be added to first extract a pitch period prototype to be processed by the parameter generator.

Abstract translation: 用于预测量化浊音的方法和装置包括参数发生器和量化器。 参数发生器被配置为从诸如有声语音的预测语音的帧中提取参数，并且将提取的信息变换为频域表示。 量化器被配置为从当前帧的参数中减去先前帧的参数的加权和。 量化器被配置为量化差值。 可以添加原型提取器以首先提取要由参数发生器处理的音调周期原型。

公开(公告)号：US06947888B1

公开(公告)日：2005-09-20

申请号：US09690915

申请日：2000-10-17

Applicant: Pengjun Huang

Inventor： Pengjun Huang

IPC: G10L19/04 ,  G10L19/18 ,  H03M7/30 ,  G10G19/04

CPC classification number: G10L19/12 ,  G10L19/083 ,  G10L19/18 ,  G10L25/93

Abstract: A low-bit-rate coding technique for unvoiced segments of speech, without loss of quality compared to the conventional Code Excited Linear Prediction (CELP) method operating at a much higher bit rate. A set of gains are derived from a residual signal after whitening the speech signal by a linear prediction filter. These gains are then quantized and applied to a randomly generated sparse excitation. The excitation is filtered, and its spectral characteristics are analyzed and compared to the spectral characteristics of the original residual signal. Based on this analysis, a filter is chosen to shape the spectral characteristics of the excitation to achieve optimal performance.

Abstract translation: 一种用于无声段语音的低比特率编码技术，与以更高比特率运行的常规码激励线性预测（CELP）方法相比，没有质量损失。 通过线性预测滤波器对语音信号进行白化后的残差信号得到一组增益。 然后将这些增益量化并应用于随机产生的稀疏激发。 对激发进行滤波，并对其光谱特征进行分析，并将其与原始残留信号的光谱特性进行比较。 基于该分析，选择滤波器来形成激发的光谱特征以实现最佳性能。

公开(公告)号：US06434519B1

公开(公告)日：2002-08-13

申请号：US09356861

申请日：1999-07-19

Applicant: Sharath Manjunath ,  Andrew P. Dejaco ,  Arasanipalai K. Ananthapadmanabhan ,  Pengjun Huang ,  Eddie Lun Tik Choy

Inventor： Sharath Manjunath ,  Andrew P. Dejaco ,  Arasanipalai K. Ananthapadmanabhan ,  Pengjun Huang ,  Eddie Lun Tik Choy

IPC: G10L1914

CPC classification number: G10L19/0208 ,  G10L19/10

Abstract: A method and apparatus for identifying frequency bands to compute linear phase shifts between frame prototypes in a speech coder includes partitioning the frequency spectrum of a prototype of a frame by dividing the frequency spectrum into segments, assigning one or more bands to each segment, and establishing, for each segment, a set of bandwidths for the bands. The bandwidths may be fixed and uniformly distributed in any given segment. The bandwidths may be fixed and non-uniformly distributed in any segment. The bandwidths may be variable and non-uniformly distributed in any given segment.

Abstract translation: 用于识别用于计算语音编码器中的帧原型之间的线性相移的频带的方法和装置包括：通过将频谱划分成段，将一个或多个频带分配给每个分段来建立帧的原型的频谱，并建立 ，对于每个段，一组带宽的带宽。 带宽可以是固定的，并且均匀分布在任何给定的段中。 带宽可以是固定的，并且不均匀地分布在任何段中。 带宽可以是可变的，并且不均匀地分布在任何给定的段中。

公开(公告)号：US09083521B2

公开(公告)日：2015-07-14

申请号：US12477590

申请日：2009-06-03

Applicant: Christian Sgraja ,  Marc W. Werner ,  Christian Pietsch ,  Wolfgang Granzow ,  Nikolai K N Leung ,  Christoph A. Joetten ,  Pengjun Huang

Inventor： Christian Sgraja ,  Marc W. Werner ,  Christian Pietsch ,  Wolfgang Granzow ,  Nikolai K N Leung ,  Christoph A. Joetten ,  Pengjun Huang

IPC: G10L19/00 ,  H04L12/58 ,  H04L1/00 ,  H04L12/855 ,  H04J3/06 ,  H04L1/18 ,  G10L19/16 ,  H04L1/16

CPC classification number: H04L1/1858 ,  G10L19/167 ,  H04J3/06 ,  H04L1/0003 ,  H04L1/003 ,  H04L1/1607 ,  H04L1/189

Abstract: A system is provided for transmitting information through a speech codec (in-band) such as found in a wireless communication network. A modulator transforms the data into a spectrally noise-like signal based on the mapping of a shaped pulse to predetermined positions within a modulation frame, and the signal is efficiently encoded by a speech codec. A synchronization sequence provides modulation frame timing at the receiver and is detected based on analysis of a correlation peak pattern. A request/response protocol provides reliable transfer of data using message redundancy, retransmission, and/or robust modulation modes dependent on the communication channel conditions.

Abstract translation: 提供了一种用于通过诸如在无线通信网络中找到的语音编解码器（带内）来发送信息的系统。 调制器基于成形脉冲到调制帧内的预定位置的映射，将数据转换成频谱类似噪声的信号，并且该信号被语音编解码器有效地编码。 同步序列在接收机处提供调制帧定时，并且基于相关峰值模式的分析来检测。 请求/响应协议使用消息冗余，重传和/或依赖于通信信道条件的鲁棒调制模式提供可靠的数据传输。

公开(公告)号：US08825480B2

公开(公告)日：2014-09-02

申请号：US12477574

申请日：2009-06-03

Applicant: Christoph A. Joetten ,  Christian Sgraja ,  Georg Frank ,  Pengjun Huang ,  Christian Pietsch ,  Marc W. Werner ,  Ethan R. Duni ,  Eugene J. Baik

Inventor： Christoph A. Joetten ,  Christian Sgraja ,  Georg Frank ,  Pengjun Huang ,  Christian Pietsch ,  Marc W. Werner ,  Ethan R. Duni ,  Eugene J. Baik

IPC: G10L19/008 ,  H04L25/49 ,  H04L5/14 ,  H04L7/04 ,  G10L19/00 ,  H04L25/03 ,  H04L7/027 ,  H04J3/06

CPC classification number: H04L7/042 ,  G10L19/00 ,  G10L19/0019 ,  H04J3/0611 ,  H04L5/1415 ,  H04L5/1423 ,  H04L5/1453 ,  H04L7/027 ,  H04L25/03834 ,  H04L25/4902

Abstract: A system is provided for transmitting information through a speech codec (in-band) such as found in a wireless communication network. A modulator transforms the data into a spectrally noise-like signal based on the mapping of a shaped pulse to predetermined positions within a modulation frame, and the signal is efficiently encoded by a speech codec. A synchronization sequence provides modulation frame timing at the receiver and is detected based on analysis of a correlation peak pattern. A request/response protocol provides reliable transfer of data using message redundancy, retransmission, and/or robust modulation modes dependent on the communication channel conditions.

Abstract translation: 提供了一种用于通过诸如在无线通信网络中找到的语音编解码器（带内）来发送信息的系统。 调制器基于成形脉冲到调制帧内的预定位置的映射，将数据转换成频谱类似噪声的信号，并且该信号被语音编解码器有效地编码。 同步序列提供接收机处的调制帧定时，并根据相关峰值模式的分析来检测。 请求/响应协议使用消息冗余，重传和/或依赖于通信信道条件的鲁棒调制模式提供可靠的数据传输。

公开(公告)号：US08503517B2

公开(公告)日：2013-08-06

申请号：US12477544

申请日：2009-06-03

Applicant: Pengjun Huang ,  Christian Pietsch ,  Christian Sgraja ,  Georg Frank ,  Christoph A. Joetten ,  Marc W. Werner ,  Wolfgang Granzow

Inventor： Pengjun Huang ,  Christian Pietsch ,  Christian Sgraja ,  Georg Frank ,  Christoph A. Joetten ,  Marc W. Werner ,  Wolfgang Granzow

IPC: H04B3/36

CPC classification number: H04L7/042 ,  H04J3/0611 ,  H04L5/1415 ,  H04L5/1423 ,  H04L5/1453 ,  H04L7/027 ,  H04L25/03834 ,  H04L25/4902

Abstract: A system is provided for transmitting information through a speech codec (in-band) such as found in a wireless communication network. A modulator transforms the data into a spectrally noise-like signal based on the mapping of a shaped pulse to predetermined positions within a modulation frame, and the signal is efficiently encoded by a speech codec. A synchronization sequence provides modulation frame timing at the receiver and is detected based on analysis of a correlation peak pattern. A request/response protocol provides reliable transfer of data using message redundancy, retransmission, and/or robust modulation modes dependent on the communication channel conditions.

公开(公告)号：US20100036668A1

公开(公告)日：2010-02-11

申请号：US12537906

申请日：2009-08-07

Applicant: Khaled H. El-Maleh ,  Eddie-Lun Tik Choy ,  Arasanipalai K. Ananthapadmanabhan ,  Andrew P. DaJaco ,  Pengjun Huang

Inventor： Khaled H. El-Maleh ,  Eddie-Lun Tik Choy ,  Arasanipalai K. Ananthapadmanabhan ,  Andrew P. DaJaco ,  Pengjun Huang

IPC: G10L11/00

CPC classification number: H04L1/08 ,  H04L1/0046 ,  H04L1/201

Abstract: A system and method for detection of rate determination algorithm errors in variable rate communications system receivers. The disclosed embodiments prevent rate determination algorithm errors from causing audible artifacts such as screeches or beeps. The disclosed system and method detects frames with incorrectly determined data rates and performs frame erasure processing and/or memory state clean up to prevent propagation of distortion across multiple frames. Frames with incorrectly determined data rates are detected by checking illegal rate transitions, reserved bits, validating unused filter type bit combinations and analyzing relationships between fixed code-book gains and linear prediction coefficient gains.

Abstract translation: 一种用于在可变速率通信系统接收机中检测速率确定算法错误的系统和方法。 所公开的实施例防止速率确定算法错误引起可听见的伪影，例如吱吱声或嘟嘟声。 所公开的系统和方法检测具有错误确定的数据速率的帧，并执行帧擦除处理和/或存储器状态清理，以防止跨多个帧的失真传播。 通过检查非法速率转换，保留位，验证未使用的过滤器类型位组合以及分析固定代码簿增益和线性预测系数增益之间的关系来检测具有不正确确定的数据速率的帧。

公开(公告)号：US20090234644A1

公开(公告)日：2009-09-17

申请号：US12255604

申请日：2008-10-21

Applicant: Yuriy Reznik ,  Pengjun Huang

Inventor： Yuriy Reznik ,  Pengjun Huang

IPC: G10L19/02 ,  G10L19/00

CPC classification number: G10L19/24 ,  G10L19/038

Abstract: A scalable speech and audio codec is provided that implements combinatorial spectrum encoding. A residual signal is obtained from a Code Excited Linear Prediction (CELP)-based encoding layer, where the residual signal is a difference between an original audio signal and a reconstructed version of the original audio signal. The residual signal is transformed at a Discrete Cosine Transform (DCT)-type transform layer to obtain a corresponding transform spectrum having a plurality of spectral lines. The transform spectrum spectral lines are transformed using a combinatorial position coding technique.The combinatorial position coding technique includes generating a lexicographical index for a selected subset of spectral lines, where each lexicographic index represents one of a plurality of possible binary strings representing the positions of the selected subset of spectral lines. The lexicographical index represents non-zero spectral lines in a binary string in fewer bits than the length of the binary string.

Abstract translation: 提供可实现组合频谱编码的可扩展语音和音频编解码器。 从基于码激励线性预测（CELP）的编码层获得残留信号，其中残留信号是原始音频信号和原始音频信号的重建版本之间的差异。 残差信号在离散余弦变换（DCT）型变换层处变换，以获得具有多个谱线的对应变换频谱。 使用组合位置编码技术对变换频谱谱线进行变换。 组合位置编码技术包括为选定的谱线子集生成词典索引，其中每个词典索引表示表示所选择的谱线子集的位置的多个可能的二进制串中的一个。 字典索引表示二进制串中的非零谱线，比二进制串的长度少。

公开(公告)号：US20080312917A1

公开(公告)日：2008-12-18

申请号：US12190524

申请日：2008-08-12

Applicant: Arasanipalai K. Ananthapadmanabhan ,  Sharath Manjunath ,  Pengjun Huang ,  Eddie-Lun Tik Choy ,  Andrew P. Dejaco

Inventor： Arasanipalai K. Ananthapadmanabhan ,  Sharath Manjunath ,  Pengjun Huang ,  Eddie-Lun Tik Choy ,  Andrew P. Dejaco

IPC: G10L19/00

CPC classification number: G10L19/04 ,  G10L19/0204 ,  G10L19/032 ,  G10L19/08 ,  G10L19/097 ,  G10L19/26 ,  G10L25/12

Abstract: A method and apparatus for predictively quantizing voiced speech includes a parameter generator and a quantizer. The parameter generator is configured to extract parameters from frames of predictive speech such as voiced speech, and to transform the extracted information to a frequency-domain representation. The quantizer is configured to subtract a weighted sum of the parameters for previous frames from the parameter for the current frame. The quantizer is configured to quantize the difference value. A prototype extractor may be added to first extract a pitch period prototype to be processed by the parameter generator.

Abstract translation: 用于预测量化浊音的方法和装置包括参数发生器和量化器。 参数发生器被配置为从诸如有声语音的预测语音的帧中提取参数，并且将提取的信息变换为频域表示。 量化器被配置为从当前帧的参数中减去先前帧的参数的加权和。 量化器被配置为量化差值。 可以添加原型提取器以首先提取要由参数发生器处理的音调周期原型。

公开(公告)号：US20050143980A1

公开(公告)日：2005-06-30

申请号：US11066356

申请日：2005-02-24

Applicant: Pengjun Huang

Inventor： Pengjun Huang

IPC: G10L19/04 ,  G10L19/18 ,  H03M7/30 ,  G10L11/04

CPC classification number: G10L19/12 ,  G10L19/083 ,  G10L19/18 ,  G10L25/93

Abstract: A low-bit-rate coding technique for unvoiced segments of speech, without loss of quality compared to the conventional Code Excited Linear Prediction (CELP) method operating at a much higher bit rate. A set of gains are derived from a residual signal after whitening the speech signal by a linear prediction filter. These gains are then quantized and applied to a randomly generated sparse excitation. The excitation is filtered, and its spectral characteristics are analyzed and compared to the spectral characteristics of the original residual signal. Based on this analysis, a filter is chosen to shape the spectral characteristics of the excitation to achieve optimal performance.