公开(公告)号：US06772115B2

公开(公告)日：2004-08-03

申请号：US09843938

申请日：2001-04-30

申请人： Kazutoshi Yasunaga ,  Toshiyuki Morii ,  Hiroyuki Ehara

发明人： Kazutoshi Yasunaga ,  Toshiyuki Morii ,  Hiroyuki Ehara

IPC分类号： G10L1912

CPC分类号： G10L19/135 ,  G10L19/12 ,  G10L2019/0007 ,  G10L2019/0013

摘要： A random code vector reading section and a random codebook of a conventional CELP type speech coder/decoder are respectively replaced with an oscillator for outputting different vector streams in accordance with values of input seeds, and a seed storage section for storing a plurality of seeds. This makes it unnecessary to store fixed vectors as they are in a fixed codebook (ROM), thereby considerably reducing the memory capacity.

摘要翻译： 传统CELP型语音编码器/解码器的随机码矢量读取部分和随机码本分别被替换为根据输入种子的值输出不同矢量流的振荡器和用于存储多个种子的种子存储部分。 这使得不必像固定码本（ROM）一样存储固定向量，从而显着地降低了存储器容量。

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：US06594627B1

公开(公告)日：2003-07-15

申请号：US09533232

申请日：2000-03-23

申请人： Vivek K. Goyal ,  Jonathan Adam Kelner ,  Jelena Kovacevic

发明人： Vivek K. Goyal ,  Jonathan Adam Kelner ,  Jelena Kovacevic

IPC分类号： G10L1912

CPC分类号： G10L19/12 ,  G10L2019/0004

摘要： A lattice-structured multiple description vector quantization (LSMDVQ) encoder generates M descriptions of a signal to be encoded, each of the descriptions being transmittable over a corresponding one of M channels. The encoder is configured based at least in part on a distortion measure which is a function of a central distortion and at least one side distortion. For example, if M=2, the distortion measure may be an average mean-squared error (AMSE) function of the form ƒ(D0, D1, D2), where D0 is a central distortion resulting from reconstruction based on receipt of both a first and a second description, and D1 and D2 are side distortions resulting from reconstruction using only a first description and a second description, respectively. Further performance improvements may be obtained through perturbation of the lattice points. The LSMDVQ techniques of the invention can also be extended to cases of M greater than two, for which the encoder may utilize an ordered set of M codebooks &Lgr;1, &Lgr;2, . . . , &Lgr;M of increasing size, with the coarsest codebook corresponding to a lattice. In such cases, for each number k of descriptions received, there may be a single decoding function that maps the received vector to a corresponding one of the codebooks &Lgr;k, such that reconstruction of the signal requires no more than M such decoding functions.

摘要翻译： 格子结构的多描述矢量量化（LSMDVQ）编码器生成要编码的信号的M个描述，每个描述可以通过M个通道中的相应一个传送。 该编码器至少部分地基于作为中心失真和至少一个侧面失真的函数的失真度量来配置。 例如，如果M = 2，失真测量可以是形式ƒ（D0，D1，D2）的平均均方误差（AMSE）函数，其中D0是基于接收到a 第一和第二描述，D1和D2分别是仅使用第一描述和第二描述的重建产生的侧面失真。 通过扰动晶格点可以获得进一步的性能改进。 本发明的LSMDVQ技术也可以扩展到大于2的M的情况，编码器可以使用M个码本LAMBD1，LAMBD2的有序集合。 。 。 ，LAMBDM的尺寸越来越大，其中最粗的码本对应于格子。 在这种情况下，对于接收到的每个k个描述，可以存在将接收到的矢量映射到码本LAMBDk中的相应一个的单个解码功能，使得该信号的重建需要不超过M个这样的解码功能。

公开(公告)号：US06345248B1

公开(公告)日：2002-02-05

申请号：US09433002

申请日：1999-11-02

申请人： Huan-Yu Su ,  Tom Hong Li

发明人： Huan-Yu Su ,  Tom Hong Li

IPC分类号： G10L1912

CPC分类号： G10L19/08 ,  G10L19/06 ,  G10L2019/0011

摘要： A pitch lag coding device and method using interframe correlation inherent in pitch lag values to reduce coding bit requirements. A pitch lag value is extracted for a given speech frame, and then refined for each subframe. For every speech frame having N samples of speech, LPC analysis and vector quantization are performed for the whole coding frame. The LPC residual obtained for each frame is then processed such that pitch lag values for all subframes within the coding frame are analyzed concurrently. The remaining coding parameters, i.e., the codebook search, gain parameters, and excitation signal, are then analyzed sequentially according to their respective subframes.

摘要翻译： 音调滞后编码装置和方法，使用音调滞后值固有的帧间相关性来减少编码比特要求。 为给定的语音帧提取音调滞后值，然后针对每个子帧进行细化。 对于具有N个语音样本的每个语音帧，对于整个编码帧执行LPC分析和矢量量化。 然后对每个帧获得的LPC残差进行处理，使得同时分析编码帧内的所有子帧的音调滞后值。 然后根据其各自的子帧依次分析剩余的编码参数，即码本搜索，增益参数和激励信号。

公开(公告)号：US06311154B1

公开(公告)日：2001-10-30

申请号：US09223363

申请日：1998-12-30

申请人： Allen Gersho ,  Vladimir Cuperman ,  Ajit V Rao ,  Tung-Chiang Yang ,  Sassan Ahmadi ,  Fenghua Liu

发明人： Allen Gersho ,  Vladimir Cuperman ,  Ajit V Rao ,  Tung-Chiang Yang ,  Sassan Ahmadi ,  Fenghua Liu

IPC分类号： G10L1912

CPC分类号： G10L19/18

摘要： A speech coder and a method for speech coding wherein the speech signal is represented by an excitation signal applied to a synthesis filter. The speech is partitioned into frames and subframes. A classifier identifies which of several categories the speech frame belongs to, and a different coding method is applied to represent the excitation for each category. For some categories, one or more windows are identified for the frame where all or most of the excitation signal samples are assigned by a coding scheme. Performance is enhanced by coding the important segments of the excitation more accurately. The window locations are determined from a linear prediction residual by identifying peaks of the smoothed residual energy contour. The method adjusts the frame and subframe boundaries so that each window is located entirely within a modified subframe or frame. This eliminates the artificial restriction incurred when coding a frame or subframe in isolation, without regard for the local behavior of the speech signal across frame or subframe boundaries.

摘要翻译： 语音编码器和用于语音编码的方法，其中语音信号由施加到合成滤波器的激励信号表示。 语音被分成帧和子帧。 分类器识别语音帧所属的几个类别中的哪一个，并且应用不同的编码方法来表示每个类别的激励。 对于某些类别，对于所有或大多数激励信号样本由编码方案分配的帧，识别一个或多个窗口。 通过更准确地对激发的重要部分进行编码来增强性能。 通过识别平滑剩余能量轮廓的峰值，从线性预测残差确定窗口位置。 该方法调整帧和子帧边界，使得每个窗口完全位于修改的子帧或帧内。 这消除了在隔离地编码帧或子帧时引起的人为限制，而不考虑跨帧或子帧边界的语音信号的本地行为。

公开(公告)号：US06813602B2

公开(公告)日：2004-11-02

申请号：US10105120

申请日：2002-03-22

申请人： Jes Thyssen

发明人： Jes Thyssen

IPC分类号： G10L1912

CPC分类号： G10L19/005 ,  G10L19/002 ,  G10L19/012 ,  G10L19/08 ,  G10L19/083 ,  G10L19/09 ,  G10L19/10 ,  G10L19/12 ,  G10L19/125 ,  G10L19/18 ,  G10L19/265 ,  G10L21/0364 ,  G10L2019/0005 ,  G10L2019/0007 ,  G10L2019/0011

摘要： A multi-rate speech codec supports a plurality of encoding bit rate modes by adaptively selecting encoding bit rate modes to match communication channel restrictions. In higher bit rate encoding modes, an accurate representation of speech through CELP (code excited linear prediction) and other associated modeling parameters are generated for higher quality decoding and reproduction. To achieve high quality in lower bit rate encoding modes, the speech encoder departs from the strict waveform matching criteria of regular CELP coders and strives to identify significant perceptual features of the input signal. The encoder generates pluralities of codevectors from a single, normalized codevector by shifting or other rearrangement. As a result, searching speeds are enhanced, and the physical size of a codebook built from such codevectors is greatly reduced.

摘要翻译： 多速率语音编解码器通过自适应地选择编码比特率模式以匹配通信信道限制来支持多种编码比特率模式。 在较高的比特率编码模式中，通过CELP（码激励线性预测）和其他相关联的建模参数的语音的精确表示被生成用于更高质量的解码和再现。 为了在低比特率编码模式下实现高质量，语音编码器脱离了常规CELP编码器的严格波形匹配标准，并努力识别输入信号的重要感知特征。 编码器通过移位或其他重新排列从单个归一化码矢量生成多个码矢量。 结果，提高了搜索速度，并且大大地减少了由这些代码矢量构建的码本的物理大小。

公开(公告)号：US06804639B1

公开(公告)日：2004-10-12

申请号：US09582039

申请日：2000-06-21

申请人： Hiroyuki Ehara

发明人： Hiroyuki Ehara

IPC分类号： G10L1912

CPC分类号： G10L19/08

摘要： A CELP type speech coder performs quantization of pitch differential value on pitch information between subframes. The coder limits the number of preliminary selected candidates using threshold processing. The coder includes specialized pitches for a subframe on which quantization of pitch differential value is not applied. When pitch preliminary selection is performed on such a subframe, the coder limits the number of preliminarily selected candidates using threshold processing to avoid outputting, as a preliminarily selected candidate, the above-mentioned specialized pitches. The coder improves the accuracy of the pitch search (adaptive codebook search) while avoiding adverse effects on the quantization of pitch differential value.

摘要翻译： CELP型语音编码器对子帧之间的音调信息进行音调差分值的量化。 编码器使用阈值处理来限制初步选择的候选者的数量。 编码器包括用于不施加音调差分值的量化的子帧的专用音调。 当在这样的子帧上执行音调预选时，编码器使用阈值处理来限制预先选择的候选的数量，以避免作为预先选择的候选者输出上述专门音高。 编码器提高音调搜索的精度（自适应码本搜索），同时避免对音调差分值的量化的不利影响。

公开(公告)号：US06795805B1

公开(公告)日：2004-09-21

申请号：US09830331

申请日：2001-07-23

申请人： Bruno Bessette ,  Redwan Salami ,  Roch Lefebvre

发明人： Bruno Bessette ,  Redwan Salami ,  Roch Lefebvre

IPC分类号： G10L1912

CPC分类号： G10L19/26 ,  G10L2019/0011

摘要： An alternative approach by which periodicity enhancement of an excitation signal is achieved through filtering an innovative codevector by an innovation filter to reduce low frequency content of the innovative codevector and enhance the periodicity at low frequencies more than high frequencies.

摘要翻译： 激励信号的周期性增强的另一种方法是通过用创新滤波器过滤创新码矢来实现，以减少创新码矢量的低频内容，并提高低频率下的周期性高于高频。

公开(公告)号：US06785646B2

公开(公告)日：2004-08-31

申请号：US09855821

申请日：2001-05-14

申请人： Yunbiao Wang ,  John Simons

发明人： Yunbiao Wang ,  John Simons

IPC分类号： G10L1912

CPC分类号： G10L19/097 ,  G10L2019/0013

摘要： The present invention provides a method and system to improve the cookbook search algorithm used in a coding/decoding device or routine. The codebook search algorithm is performed by a processing system that allows for parallel execution of instructions, for example a DSP. An embodiment of the present invention provides a method for coding of a first waveform. First a plurality of vectors determined from a plurality of waveforms is stored in a memory. Next a minimum weighted error using a plurality of filter coefficients and the plurality of vectors is determined. The minimum weighted error gives a closest match between the first waveform and a second waveform synthesized from a selected vector of the plurality of vectors. Then an indication of said selected vector is provided as part of a code of the first waveform. The plurality of filter coefficients have added to them at least one duplicate filter coefficient such that the performance of determining the minimum weighted error is improved, by for example, at least one clock cycle.

摘要翻译： 本发明提供了一种改进在编码/解码装置或程序中使用的食谱搜索算法的方法和系统。 码本搜索算法由允许并行执行指令的处理系统执行，例如DSP。 本发明的实施例提供了一种用于编码第一波形的方法。 首先，从多个波形确定的多个向量存储在存储器中。 接下来，使用多个滤波器系数进行最小加权误差，并确定多个向量。 最小加权误差给出了第一波形与从多个向量的选定向量合成的第二波形之间的最接近的匹配。 然后，提供所述选择的向量的指示作为第一波形的代码的一部分。 多个滤波器系数已经向它们添加了至少一个重复滤波器系数，使得通过例如至少一个时钟周期来改善确定最小加权误差的性能。

公开(公告)号：US06633840B1

公开(公告)日：2003-10-14

申请号：US09351326

申请日：1999-07-12

申请人： Pierre Bonnard ,  Jean Varaldi

发明人： Pierre Bonnard ,  Jean Varaldi

IPC分类号： G10L1912

CPC分类号： H04W76/20 ,  H04L1/0059 ,  H04L1/0061 ,  H04L1/007 ,  H04W88/181

摘要： To transmit data over a mobile telephone speech channel a source encoder is replaced by a transcoder, a conversion table and/or a concatenation circuit in order to choose from the words produced by the source encoder the ones that are the most robust and which can without difficulty withstand speech synthesis followed by an inverse analysis to reconstitute streams of the bits of the data to be transmitted.

摘要翻译： 为了通过移动电话语音信道发送数据，源编码器由代码转换器，转换表和/或级联电路代替，以便从源编码器产生的字中选择最健壮的，可以没有 难以忍受语音合成，然后进行逆分析，以重构要发送的数据的比特流。