公开(公告)号：US06735567B2

公开(公告)日：2004-05-11

申请号：US10409430

申请日：2003-04-08

申请人： Yang Gao ,  Adil Benyassine ,  Jes Thyssen ,  Eyal Shlomot ,  Huan-yu Su

发明人： Yang Gao ,  Adil Benyassine ,  Jes Thyssen ,  Eyal Shlomot ,  Huan-yu Su

IPC分类号： G10L1304

CPC分类号： G10L19/00 ,  G10L19/167 ,  G10L19/24 ,  H03G3/00

摘要： A speech compression system capable of encoding a speech signal into a bitstream for subsequent decoding to generate synthesized speech is disclosed. The speech compression system optimizes the bandwidth consumed by the bitstream by balancing the desired average bit rate with the perceptual quality of the reconstructed speech. The speech compression system comprises a full-rate codec, a half-rate codec, a quarter-rate codec and an eighth-rate codec. The codecs are selectively activated based on a rate selection. In addition, the full and half-rate codecs are selectively activated based on a type classification. Each codec is selectively activated to encode and decode the speech signals at different bit rates emphasizing different aspects of the speech signal to enhance overall quality of the synthesized speech.

摘要翻译： 公开了能够将语音信号编码为比特流以进行后续解码以产生合成语音的语音压缩系统。 语音压缩系统通过将期望的平均比特率与重构语音的感知质量进行平衡来优化比特流消耗的带宽。 语音压缩系统包括全速率编解码器，半速率编解码器，四分之一速率编解码器和八速率编解码器。 基于速率选择来选择性地激活编解码器。 此外，基于类型分类，全速率和半速率编解码器被选择性地激活。 选择性地激活每个编解码器以以强调语音信号的不同方面的不同比特率对语音信号进行编码和解码，以增强合成语音的整体质量。

公开(公告)号：US06581032B1

公开(公告)日：2003-06-17

申请号：US09662828

申请日：2000-09-15

申请人： Yang Gao ,  Adil Benyassine ,  Jes Thyssen ,  Eyal Shlomot ,  Huan-yu Su

发明人： Yang Gao ,  Adil Benyassine ,  Jes Thyssen ,  Eyal Shlomot ,  Huan-yu Su

IPC分类号： G10L1912

CPC分类号： G10L19/00 ,  G10L19/167 ,  G10L19/24 ,  H03G3/00

摘要： A speech compression system capable of encoding a speech signal into a bitstream for subsequent decoding to generate synthesized speech is disclosed. The speech compression system optimizes the bandwidth consumed by the bitstream by balancing the desired average bit rate with the perceptual quality of the reconstructed speech. The speech compression system comprises a full-rate codec, a half-rate codec, a quarter-rate codec and an eighth-rate codec. The codecs are selectively activated based on a rate selection. In addition, the full and half-rate codecs are selectively activated based on a type classification. Each codec is selectively activated to encode and decode the speech signals at different bit rates emphasizing different aspects of the speech signal to enhance overall quality of the synthesized speech.

摘要翻译： 公开了能够将语音信号编码为比特流以进行后续解码以产生合成语音的语音压缩系统。 语音压缩系统通过将期望的平均比特率与重构语音的感知质量进行平衡来优化比特流消耗的带宽。 语音压缩系统包括全速率编解码器，半速率编解码器，四分之一速率编解码器和八速率编解码器。 基于速率选择来选择性地激活编解码器。 此外，基于类型分类，全速率和半速率编解码器被选择性地激活。 选择性地激活每个编解码器以以强调语音信号的不同方面的不同比特率对语音信号进行编码和解码，以增强合成语音的整体质量。

公开(公告)号：US20090043574A1

公开(公告)日：2009-02-12

申请号：US12284623

申请日：2008-09-23

申请人： Yang Gao ,  Adil Benyassine ,  Jes Thyssen ,  Eyal Shlomot ,  Huan-yu Su

发明人： Yang Gao ,  Adil Benyassine ,  Jes Thyssen ,  Eyal Shlomot ,  Huan-yu Su

IPC分类号： G10L19/12 ,  G10L19/00

CPC分类号： G10L19/00 ,  G10L19/167 ,  G10L19/20 ,  G10L19/22 ,  G10L19/24 ,  G10L2019/0001 ,  H03G3/00

摘要： There is provided a method of decoding speech data generated from a speech signal. The method comprises receiving the speech data having at least one main pulse in a subframe of the speech data; generating a first predicted pulse, based on the at least one main pulse, on one side of the main pulse in the subframe of the speech data, wherein the first predicted pulse has a lower gain than the main pulse; generating a second predicted pulse, as a mirror image of the first predicted pulse on a reverse time scale, on the other side of the main pulse in the subframe of the speech data; reconstructing the speech signal using the at least one main pulse, the first predicted pulse and the second predicted pulse.

摘要翻译： 提供了一种对从语音信号产生的语音数据进行解码的方法。 该方法包括：接收语音数据的子帧中具有至少一个主脉冲的语音数据; 基于所述至少一个主脉冲在所述语音数据的子帧中的所述主脉冲的一侧产生第一预测脉冲，其中所述第一预测脉冲具有比所述主脉冲更低的增益; 在语音数据的子帧中的主脉冲的另一侧上产生第二预测脉冲作为反时限上的第一预测脉冲的镜像; 使用所述至少一个主脉冲，所述第一预测脉冲和所述第二预测脉冲来重构所述语音信号。

公开(公告)号：US07146309B1

公开(公告)日：2006-12-05

申请号：US10653874

申请日：2003-09-02

申请人： Adil Benyassine ,  Eyal Shlomot ,  Huan-Yu Su

发明人： Adil Benyassine ,  Eyal Shlomot ,  Huan-Yu Su

IPC分类号： G10L19/00

CPC分类号： G10L19/08

摘要： There are provided methods and devices for generating excitation values for a speech signal. In one aspect, an example method comprises obtaining one or more characteristics of a first speech frame of the speech signal, deriving a first seed value based on the one or more characteristics of the first speech frame, providing the first seed value to a Gaussian time series generator; and using the Gaussian time series generator to generate an excitation values for the first frame. The one or more characteristics may include a spectrum information of the first frame, an energy information of the first frame, or a gain information of the first frame.

摘要翻译： 提供了用于产生语音信号的激励值的方法和装置。 在一个方面，示例性方法包括获得语音信号的第一语音帧的一个或多个特征，基于第一语音帧的一个或多个特征导出第一种子值，将第一种子值提供给高斯时间 串联发电机; 并使用高斯时间序列发生器来产生第一帧的激励值。 一个或多个特征可以包括第一帧的频谱信息，第一帧的能量信息或第一帧的增益信息。

公开(公告)号：US07120578B2

公开(公告)日：2006-10-10

申请号：US09841764

申请日：2001-04-24

申请人： Jes Thyssen ,  Huan-yu Su ,  Adil Benyassine ,  Eyal Shlomot

发明人： Jes Thyssen ,  Huan-yu Su ,  Adil Benyassine ,  Eyal Shlomot

IPC分类号： G10L11/06 ,  G10L19/12

CPC分类号： G10L19/012

摘要： Speech coding systems include multi-rate speech codecs having an encoder and a decoder. Silence description coding for multi-rate speech coding systems that employ discontinued transmission is performed in either the encoder or the decoder of the multi-rate speech codec. It may also be performed in a distributed manner wherein it is performed partially in the encoder and partially in the decoder. The silence description coding is performed on a speech signal having a substantially non-speech-like characteristic. Voice activity detection classifies the speech signal as being either substantially speech-like or substantially non-speech-like. The silence description coding is selected from a plurality of coding modes. In certain embodiments of the invention, the silence description coding is a source coding mode that operates at a bit rate that fits within a bit rate budget as determined by all of the available source coding modes within the plurality of coding modes. The silence description coding is also accompanied with signaling coding and channel coding of the speech signal. Error checking is performed using an unused portion of a bandwidth of the multi-rate speech codec's bit rate. This error checking involves majority voting in certain embodiments of the invention.

摘要翻译： 语音编码系统包括具有编码器和解码器的多速率语音编解码器。 在多速率语音编解码器的编码器或解码器中执行采用中断传输的多速率语音编码系统的静音描述编码。 它也可以以分布式方式执行，其中部分地在编码器中执行，部分地在解码器中执行。 对具有基本上非语音的特征的语音信号执行静音描述编码。 语音活动检测将语音信号分类为基本上是语音的或基本上非语音的。 从多种编码模式中选择静音描述编码。 在本发明的某些实施例中，静默描述编码是以适合在多个编码模式内的所有可用源编码模式所确定的比特率预算中的比特率操作的源编码模式。 静音描述编码也伴随着语音信号的信令编码和信道编码。 使用多速率语音编解码器的比特率的带宽的未使用部分来执行错误检查。 在本发明的某些实施例中，该错误检查涉及多数投票。

公开(公告)号：US06961698B1

公开(公告)日：2005-11-01

申请号：US10420654

申请日：2003-04-21

申请人： Yang Gao ,  Adil Benyassine ,  Jes Thyssen ,  Eyal Shlomot ,  Huan-yu Su

发明人： Yang Gao ,  Adil Benyassine ,  Jes Thyssen ,  Eyal Shlomot ,  Huan-yu Su

IPC分类号： G10L19/00 ,  G10L13/00 ,  G10L13/04 ,  G10L19/02 ,  G10L19/04 ,  G10L19/08 ,  G10L19/10 ,  G10L19/12 ,  G10L19/14 ,  H03M7/30 ,  H03M7/36

CPC分类号： G10L19/00 ,  G10L19/167 ,  G10L19/24 ,  H03G3/00

摘要： A speech compression system capable of encoding a speech signal into a bitstream for subsequent decoding to generate synthesized speech is disclosed. The bitstream comprises a type component and a gain component. The type component is representative of a type classification of a frame of speech signal that is transmitted. The type component comprises a first type and second type. The gain component represents an adaptive codebook gain and a fixed codebook gain component comprises a fixed codebook gain component and an adaptive codebook gain component exclusively encoded as separate components of the bitstream as a function of the bit rate when the type classification is the second type.

摘要翻译： 公开了能够将语音信号编码为比特流以进行后续解码以产生合成语音的语音压缩系统。 比特流包括类型分量和增益分量。 类型分量代表传输的语音信号帧的类型分类。 类型组件包括第一类型和第二类型。 增益分量表示自适应码本增益，并且固定码本增益分量包括固定码本增益分量和自适应码本增益分量，该类型分类作为第二类型时，作为比特率的单独分量专门编码。

公开(公告)号：US06463414B1

公开(公告)日：2002-10-08

申请号：US09547832

申请日：2000-04-12

申请人： Huan-Yu Su ,  Eyal Shlomot ,  Jes Thyssen ,  Adil Benyassine ,  Yang Gao

发明人： Huan-Yu Su ,  Eyal Shlomot ,  Jes Thyssen ,  Adil Benyassine ,  Yang Gao

IPC分类号： G10L1102

CPC分类号： G10L19/173

摘要： There is provided a conference bridge or transcoder configured to intelligently handle multiple speech channels in the contest of a packet network, wherein various speech channels may adhere to variety of speech encoding standards. For example, the conference bridge establishes framing and alignment of multiple incoming speech channels associated with multiple participants, extracts parameters from the speech samples, mixes the parameters, and re-encodes the resulting speech samples for transmission to the participants. In one aspect, a speech processing method comprises decoding a first bitstream according to a first coding scheme to generate first speech samples and a first side information; generating second speech samples and a second side information using the first speech samples and the first side information, for use according to a second coding scheme; and creating a second bitstream, encoded based on the second coding scheme, using the second speech samples and the second side information.

摘要翻译： 提供了一种配置成在分组网络的比赛中智能地处理多个语音信道的会议桥或代码转换器，其中各种语音信道可以遵循各种语音编码标准。 例如，会议桥建立与多个参与者相关联的多个输入语音信道的成帧和对准，从语音样本中提取参数，混合参数，并对所得到的语音样本进行重新编码以传输给参与者。 一方面，语音处理方法包括根据第一编码方案对第一比特流进行解码，以产生第一语音样本和第一侧信息; 使用第一语音样本和第一侧信息生成第二语音样本和第二侧信息，以便根据第二编码方案使用; 以及使用所述第二语音样本和所述第二侧信息来创建基于所述第二编码方案编码的第二比特流。

公开(公告)号：US06757649B1

公开(公告)日：2004-06-29

申请号：US10409404

申请日：2003-04-08

申请人： Yang Gao ,  Adil Benyassine ,  Jes Thyssen ,  Eyal Shlomot ,  Huan-yu Su

发明人： Yang Gao ,  Adil Benyassine ,  Jes Thyssen ,  Eyal Shlomot ,  Huan-yu Su

IPC分类号： G10L1912

CPC分类号： G10L19/00 ,  G10L19/167 ,  G10L19/24 ,  H03G3/00

摘要： A speech compression system capable of encoding a speech signal into a bitstream for subsequent decoding to generate synthesized speech is disclosed. The speech compression system optimizes the bandwidth consumed by the bitstream by balancing the desired average bit rate with the perceptual quality of the reconstructed speech. The speech compression system comprises a full-rate codec, a half-rate codec, a quarter-rate codec and an eighth-rate codec. The codecs are selectively activated based on a rate selection. In addition, the full and half-rate codecs are selectively activated based on a type classification. Each codec is selectively activated to encode and decode the speech signals at different bit rates emphasizing different aspects of the speech signal to enhance overall quality of the synthesized speech.

摘要翻译： 公开了能够将语音信号编码为比特流以进行后续解码以产生合成语音的语音压缩系统。 语音压缩系统通过将期望的平均比特率与重构语音的感知质量进行平衡来优化比特流消耗的带宽。 语音压缩系统包括全速率编解码器，半速率编解码器，四分之一速率编解码器和八速率编解码器。 基于速率选择来选择性地激活编解码器。 此外，基于类型分类，全速率和半速率编解码器被选择性地激活。 选择性地激活每个编解码器以以强调语音信号的不同方面的不同比特率对语音信号进行编码和解码，以增强合成语音的整体质量。

公开(公告)号：US06636829B1

公开(公告)日：2003-10-21

申请号：US09617191

申请日：2000-07-14

申请人： Adil Benyassine ,  Eyal Shlomot ,  Huan-Yu Su

发明人： Adil Benyassine ,  Eyal Shlomot ,  Huan-Yu Su

IPC分类号： G10L1900

CPC分类号： G10L19/08 ,  G10L19/005 ,  G10L19/07 ,  G10L19/083 ,  G10L25/90 ,  G10L2019/0012

摘要： An exemplary decoder comprises a receiver that receives parameters of a speech signal on a frame-by-frame basis, a control logic for decoding parameters and for resynthesizing the speech signal, the control logic including a minimum spacing indicative of a minimum difference required between LSFs of consecutive frames, a frame recovery logic that, when a lost frame detector detects a lost frame, sets the minimum spacing for the lost frame to a first value which is greater than the minimum spacing for the previously received frame, and/or uses pitch lag parameters of a plurality of previously received frames to extrapolate a pitch lag parameter for the lost frame, and/or sets gain parameter of a subframe of the lost frame in a first manner if the lost gain parameter is an adaptive codebook gain parameter and in a second manner if the lost gain parameter is a fixed codebook gain parameter.

摘要翻译： 示例性解码器包括接收器，其逐帧地接收语音信号的参数，用于解码参数并用于再合成语音信号的控制逻辑，所述控制逻辑包括指示LSF之间所需的最小差异的最小间隔 连续帧的帧恢复逻辑，当丢失帧检测器检测到丢失帧时，将丢失帧的最小间隔设置为大于先前接收帧的最小间隔的第一值，和/或使用间距 多个先前接收的帧的滞后参数，以推断丢失帧的音调滞后参数，和/或以丢失的增益参数为自适应码本增益参数，以第一种方式设置丢失帧的子帧的增益参数，并且 丢失增益参数是固定码本增益参数的第二种方式。

公开(公告)号：US5920853A

公开(公告)日：1999-07-06

申请号：US702780

申请日：1996-08-23

申请人： Adil Benyassine ,  Huan-Yu Su ,  Eyal Shlomot

发明人： Adil Benyassine ,  Huan-Yu Su ,  Eyal Shlomot

IPC分类号： H04N7/26 ,  G06T9/00 ,  H03M7/30 ,  G06F17/30 ,  G06F5/00

CPC分类号： G06T9/008 ,  H03M7/3082 ,  Y10S707/99931

摘要： A signal compression system includes a coder and a decoder. The coder includes an extract unit for extracting an input feature vector from an input signal, a coder memory unit for storing a predesigned vector quantization (VQ) table for the coder such that the coder memory unit uses a set of primary indices to address entries within the pre-designed VQ table, a coder mapping unit for mapping indices from a set of secondary indices to the first set of indices, and a search unit for searching for one index out of the set of secondary indices, wherein the index from the set of secondary indices corresponds to an entry in the coder memory unit, and the entry best represents the input feature vector according to some predetermined criteria. On the decoder side, the decoder includes a decoder memory unit for storing the same pre-designed VQ table and set of primary indices as the coder memory unit, a decoder mapping unit, and a retrieval unit, wherein the entry indicated by the index best represents the input feature vector.

摘要翻译： 信号压缩系统包括编码器和解码器。 编码器包括用于从输入信号提取输入特征向量的提取单元，编码器存储单元，用于存储用于编码器的预先设计的矢量量化（VQ）表，使得编码器存储单元使用一组主要索引来寻址 预先设计的VQ表，用于映射从一组二次索引到第一组索引的索引的编码器映射单元，以及用于搜索该次要索引集合中的一个索引的搜索单元，其中来自该集合的索引 次要索引对应于编码器存储单元中的条目，并且条目最好地表示根据某些预定标准的输入特征向量。 在解码器侧，解码器包括解码器存储器单元，用于存储与编码器存储单元相同的预先设计的VQ表和一组主要索引，解码器映射单元和检索单元，其中由索引最佳指示的条目 代表输入特征向量。