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1. 发明授权

US06418408B1 Frequency domain interpolative speech codec system 有权
公开(公告)号：US06418408B1
公开(公告)日：2002-07-09
申请号：US09542792
申请日：2000-04-04
申请人： Bangalore R. Udaya Bhaskar , Srinivas Nandkumar , Kumar Swaminathan , Gaguk Zakaria
发明人： Bangalore R. Udaya Bhaskar , Srinivas Nandkumar , Kumar Swaminathan , Gaguk Zakaria
IPC分类号： G10L1904
CPC分类号： G10L19/18 , G10L19/005 , G10L19/02 , G10L19/0204 , G10L19/04 , G10L19/083 , G10L19/09 , G10L25/27 , G10L25/30 , G10L25/78 , G10L25/90 , G10L2019/0012 , G10L2025/783
摘要： Encoding of prototype waveform components applicable to GeoMobile and Telephony Earth Station (TES) providing improved voice quality enabling a dual-channel mode of operation which permits more users to communicate over the same physical channel. A prototype word (PW) gain is vector quantized using a vector quantizer (VQ) that explicitly populates the codebook by representative steady state and transient vectors of PW gain for tracking the abrupt variations in speech levels during onsets and other non-stationary events, while maintaining the accuracy of the speech level during stationary conditions. The rapidly evolving waveform (REW) and slowly evolving waveform (SEW) component vectors are converted to magnitude-phase. The variable dimension SEW magnitude vector is quantized using a hierarchical approach, i.e., a fixed dimension SEW mean vector computed by a sub-band averaging of SEW magnitude spectrum, and only the REW magnitude is explicitly encoded. The REW magnitude vector sequence is normalized to unity RMS value, resulting in a REW magnitude shape vector and a REW gain vector. The normalized REW magnitude vectors are modeled by a multi-band sub-band model which converts the variable dimension REW magnitude shape vectors, e.g., six dimensional REW sub-band vectors. The sub-band vectors are averaged over time, resulting in a single average REW sub-band vector for each frame. At the decoder, the full-dimension REW magnitude shape vector is obtained from the REW sub-band vector by a piecewise-constant construction. The REW phase vector is regenerated at the decoder based on the received REW gain vector and the voicing measure, which determines a weighted mixture of SEW component and a random noise that is passed through a high pass filter to generate the REW component. The high pass filter poles are adjusted based on the voicing measure to control the REW component characteristics. At the output the filter, the magnitude of the REW component is scaled to match the received REW magnitude vector.

2. 发明授权

US06691092B1 Voicing measure as an estimate of signal periodicity for a frequency domain interpolative speech codec system 有权
标题翻译：音频测量作为频域内插语音编解码系统的信号周期的估计
公开(公告)号：US06691092B1
公开(公告)日：2004-02-10
申请号：US09542390
申请日：2000-04-04
申请人： Bangalore R. Udaya Bhaskar , Srinivas Nandkumar , Kumar Swaminathan , Gaguk Zakaria
发明人： Bangalore R. Udaya Bhaskar , Srinivas Nandkumar , Kumar Swaminathan , Gaguk Zakaria
IPC分类号： G10L1304
CPC分类号： G10L19/097 , G10L2025/783
摘要： A system determines a voicing measure as a measure of the degree of signal periodicity and uses the determined voicing measure to quantize the spectral magnitude of the slowly evolving waveform (SEW) and the modeling of the SEW and rapidly evolving waveform (REW) phase spectra.
摘要翻译：系统将发声测量值确定为信号周期程度的度量，并使用确定的发声测量来量化慢速演化波形（SEW）的频谱幅度以及SEW和快速演变波形（REW）相位谱的建模。

3. 发明授权

US06493664B1 Spectral magnitude modeling and quantization in a frequency domain interpolative speech codec system 有权
标题翻译：频域内插语音编解码系统中的频谱幅度建模和量化
公开(公告)号：US06493664B1
公开(公告)日：2002-12-10
申请号：US09542793
申请日：2000-04-04
申请人： Bangalore R. Udaya Bhaskar , Srinivas Nandkumar , Kumar Swaminathan , Gaguk Zakaria
发明人： Bangalore R. Udaya Bhaskar , Srinivas Nandkumar , Kumar Swaminathan , Gaguk Zakaria
IPC分类号： G10L1912
CPC分类号： G10L19/18 , G10L19/005 , G10L19/02 , G10L19/0204 , G10L19/04 , G10L19/083 , G10L19/09 , G10L25/27 , G10L25/30 , G10L25/78 , G10L25/90 , G10L2019/0012 , G10L2025/783
摘要： Encoding of prototype waveform components applicable to telecommunication systems provides improved voice quality enabling a dual-channel mode of operation which permits more users to communicate over the same physical channel. A prototype word (PW) gain is vector quantized using a vector quantizer (VQ) that explicitly populates a codebook by representative steady state and transient vectors of PW gain for tracking the abrupt variations in speech levels during onsets and other non-stationary events, while maintaining the accuracy of the speech level during stationary conditions.
摘要翻译：适用于电信系统的原型波形组件的编码提供改进的语音质量，从而实现双通道操作模式，允许更多的用户通过相同的物理信道进行通信。原型字（PW）增益使用矢量量化器（VQ）进行矢量量化，矢量量化器（VQ）通过代表性的稳定状态和PW增益的瞬态矢量明确地填充码本，用于跟踪在开始和其他非平稳事件期间语音电平的突然变化，而在静止状态下保持语音水平的准确性。

4. 发明授权

US06453289B1 Method of noise reduction for speech codecs 有权
标题翻译：语音编解码器降噪方法
公开(公告)号：US06453289B1
公开(公告)日：2002-09-17
申请号：US09361015
申请日：1999-07-23
申请人： Filiz Basbug Ertem , Srinivas Nandkumar , Kumar Swaminathan
发明人： Filiz Basbug Ertem , Srinivas Nandkumar , Kumar Swaminathan
IPC分类号： G10L2102
CPC分类号： G10L25/78 , G10L21/0208
摘要： An improved noise reduction algorithm is provided, as well as a voice activity detector, for use in a voice communication system. The voice activity detector allows for a reliable estimate of noise and enhancement of noise reduction. The noise reduction algorithm and voice activity detector can be implemented integrally in an encoder or applied independently to speech coding application. The voice activity detector employs line spectral frequencies and enhanced input speech which has undergone noise reduction to generate a voice activity flag. The noise reduction algorithm employs a smooth gain function determined from a smoothed noise spectral estimate and smoothed input noisy speech spectra. The gain function is smoothed both across frequency and time in an adaptive manner based on the estimate of the signal-to-noise ratio. The gain function is used for spectral amplitude enhancement to obtain a reduced noise speech signal. Smoothing employs critical frequency bands corresponding to the human auditory system. Swirl reduction is performed to improve overall human perception of decoded speech.
摘要翻译：提供了一种改进的降噪算法，以及用于语音通信系统中的语音活动检测器。语音活动检测器允许噪声的可靠估计和噪声降低的增强。噪声降低算法和语音活动检测器可以在编码器中一体地实现或独立地应用于语音编码应用。语音活动检测器采用经过降噪的线谱频率和增强输入语音以产生语音活动标志。噪声降低算法采用从平滑噪声谱估计和平滑输入噪声语音谱确定的平滑增益函数。基于信噪比的估计，以自适应方式在频率和时间上平滑增益功能。增益函数用于频谱振幅增强，以获得降噪噪声语音信号。平滑采用对应于人类听觉系统的临界频带。进行旋转减少以改善对解码语音的整体人感知。

5. 发明授权

US5966688A Speech mode based multi-stage vector quantizer 失效
标题翻译：基于语音模式的多级矢量量化器
公开(公告)号：US5966688A
公开(公告)日：1999-10-12
申请号：US958143
申请日：1997-10-28
申请人： Srinivas Nandkumar , Kumar Swaminathan
发明人： Srinivas Nandkumar , Kumar Swaminathan
IPC分类号： G10L11/06 , G10L19/00 , G10L19/06 , G10L3/02
CPC分类号： G10L19/07 , G10L25/93
摘要： A speech mode based multi-stage vector quantizer is disclosed which quantizes and encodes line spectral frequency (LSF) vectors that were obtained by transforming the short-term predictor filter coefficients in a speech codec that utilizes linear predictive techniques. The quantizer includes a mode classifier that classifies each speech frame of a speech signal as being associated with one of a voiced, spectrally stationary (Mode A) speech frame, a voiced, spectrally non-stationary (Mode B) speech frame and an unvoiced (Mode C) speech frame. A converter converts each speech frame of the speech signal into an LSF vector and an LSF vector quantizer includes a 12-bit, two-stage, backward predictive vector encoder that encodes the Mode A speech frames and a 22 bit, four-stage backward predictive vector encoder that encodes the Mode 13 and the Mode C speech frames.
摘要翻译：公开了一种基于语音模式的多级矢量量化器，其对通过使用线性预测技术的语音编解码器中的短期预测器滤波器系数进行变换而获得的线谱频率（LSF）矢量进行量化和编码。量化器包括模式分类器，其将语音信号的每个语音帧分类为与有声，频谱平稳（模式A）语音帧，有声，频谱非平稳（模式B）语音帧和无声（模式C）语音帧。 A转换器将语音信号的每个语音帧转换为LSF向量，并且LSF向量量化器包括对模A语音帧进行编码的12位两级反向预测向量编码器和22位四级后向预测编码模式13和模式C语音帧的矢量编码器。

6. 发明授权

US5781540A Device and method for communicating in a mobile satellite system 失效
标题翻译：用于在移动卫星系统中进行通信的装置和方法
公开(公告)号：US5781540A
公开(公告)日：1998-07-14
申请号：US497582
申请日：1995-06-30
申请人： James Eryx Malcolm , Daniel Fraley , Adrian Morris , David Roos , Kumar Swaminathan , Seok Ho Kim , Robert Carroll Marquart
发明人： James Eryx Malcolm , Daniel Fraley , Adrian Morris , David Roos , Kumar Swaminathan , Seok Ho Kim , Robert Carroll Marquart
IPC分类号： H04B7/185 , H04J3/06 , H04L7/04 , H04L7/10 , H04B7/212
CPC分类号： H04J3/0605 , H04B7/18532 , H04L7/046 , H04L7/10
摘要： The present invention relates to a device and a method for communicating in a mobile communication system. The method provides a carrier signal having a plurality of frames. Each frame has a plurality of time slots, and each time slot comprises a plurality of transmission bits. A group of time slots are assigned to a communication channel. A traffic burst signal having a plurality of traffic symbols is transmitted over the communication channel by transmitting a first preamble over one of the assigned time slots, and transmitting a second preamble and at least one of the traffic symbols over at least one of the other assigned time slots. The second preamble occupies fewer transmission bits than the first preamble. The apparatus for transmitting a telephony signal over an RF channel includes a modem receiving a digitized PCM telephony signal and producing a traffic burst signal, and a transmitting unit in communication with the modem for transmitting a FDMA/TDMA signal carrying a plurality of traffic burst signals. At least one of the traffic burst signals carries a limited preamble message including a header field and a unique word field and at least one digitized voice message associated with a telephone call. Another traffic burst signal carries at least one signal acquisition message including a unique word field.
摘要翻译：本发明涉及用于在移动通信系统中进行通信的装置和方法。该方法提供具有多个帧的载波信号。每个帧具有多个时隙，并且每个时隙包括多个传输比特。一组时隙被分配给通信信道。通过在所分配的时隙中的一个发送第一前同步码，通过通信信道发送具有多个业务符号的业务突发信号，并且通过至少一个其他分配的业务符号发送第二前导码和至少一个业务符号时隙。第二前导码占用比第一前同步码少的传输比特。用于通过RF信道发送电话信号的装置包括调制解调器，接收数字化的PCM电话信号并产生业务脉冲串信号，以及与调制解调器通信的发送单元，用于发送携带多个话务脉冲串信号的FDMA / TDMA信号。业务突发信号中的至少一个携带有限的前导消息，其包括报头字段和唯一字字段以及与电话呼叫相关联的至少一个数字化语音消息。另一业务脉冲串信号携带至少一个信号获取消息，其包括唯一的字字段。

7. 发明授权

US5751903A Low rate multi-mode CELP codec that encodes line SPECTRAL frequencies utilizing an offset 失效
标题翻译：低速多模CELP编解码器，利用偏移编码线路频谱频率
公开(公告)号：US5751903A
公开(公告)日：1998-05-12
申请号：US359116
申请日：1994-12-19
申请人： Kumar Swaminathan , Murthy Vemuganti
发明人： Kumar Swaminathan , Murthy Vemuganti
IPC分类号： G10L11/06 , G10L19/00 , G10L19/14 , G10L9/18
CPC分类号： G10L19/06 , G10L19/18 , G10L25/24 , G10L25/93
摘要： The present invention provides a multi-mode CELP encoding and decoding method and device for digitized speech signals providing improvements over prior art codecs and coding methods by selectively utilizes backward prediction for the short-term predictor parameters and fixed codebook gain of a speech signal. In order to achieve these improvements, the present invention provides a coding method comprising the steps of classifying a segment of the digitized speech signal as one of a plurality of predetermined modes, determining a set of unquantized line spectral frequencies to represent the short term predictor parameters for that segment, and quantizing the determined set of unquantized line spectral frequencies using a mode-specific combination of scalar quantization and vector quantization, which utilizes backward prediction for modes with voiced speech signals. Furthermore, backward prediction is selectively applied to the fixed codebook gain in the modes that are free of transients so that it may be used in the fixed codebook search and fixed codebook gain quantization in those modes.
摘要翻译：本发明提供了一种用于数字化语音信号的多模式CELP编码和解码方法和装置，其通过选择性地利用对短期预测参数的反向预测和语音信号的固定码本增益来提供超过现有技术编解码器和编码方法的改进。为了实现这些改进，本发明提供了一种编码方法，包括以下步骤：将数字化语音信号的片段分类为多个预定模式之一，确定一组未量化的线谱频率以表示短期预测参数并且使用标量量化和矢量量化的模式特定组合量化所确定的未量化线谱频率的集合，其利用具有有声语音信号的模式的反向预测。此外，在没有瞬变的模式中，有选择地将后向预测应用于固定码本增益，使得其可以用于这些模式中的固定码本搜索和固定码本增益量化。

8. 发明授权

US06996523B1 Prototype waveform magnitude quantization for a frequency domain interpolative speech codec system 有权
标题翻译：用于频域内插语音编解码系统的原型波形幅度量化
公开(公告)号：US06996523B1
公开(公告)日：2006-02-07
申请号：US10073128
申请日：2002-02-13
申请人： Udaya Bhaskar , Kumar Swaminathan
发明人： Udaya Bhaskar , Kumar Swaminathan
IPC分类号： G10L19/10
CPC分类号： G10L19/097 , G10L19/032
摘要： A system and method is provided that employs a frequency domain interpolative CODEC system for low bit rate coding of speech which comprises a linear prediction (LP) front end adapted to process an input signal that provides LP parameters which are quantized and encoded over predetermined intervals and used to compute a LP residual signal. An open loop pitch estimator adapted to process the LP residual signal, a pitch quantizer, and a pitch interpolator and provide a pitch contour within the predetermined intervals is also provided. Also provided is a signal processor responsive to the LP residual signal and the pitch contour and adapted to perform the following: provide a voicing measure, where the voicing measure characterizes a degree of voicing of the input speech signal and is derived from several input parameters that are correlated to degrees of periodicity of the signal over the predetermined intervals; extract a prototype waveform (PW) from the LP residual and the open loop pitch contour for a number of equal sub-intervals within the predetermined intervals; normalize the PW by a gain value of the PW; encode a magnitude of the PW; and directly quantize the PW in a magnitude domain without further decomposition of the PW into complex components, where the direct quantization is performed by a hierarchical quantization method based on a voicing classification using fixed dimension vector quantizers (VQ's).
摘要翻译：提供了一种系统和方法，其采用用于语音的低比特率编码的频域内插CODEC系统，其包括适于处理提供经过预定间隔量化和编码的LP参数的输入信号的线性预测（LP）前端，以及用于计算LP残差信号。还提供了适于处理LP残差信号的开环音调估计器，音调量化器和音调内插器，并且在预定间隔内提供音调轮廓。还提供了响应于LP残留信号和音调轮廓的信号处理器，并且适于执行以下操作：提供语音测量，其中语音测量表征输入语音信号的发音程度，并且从几个输入参数导出，与预定间隔的信号的周期度相关; 从所述LP残差和所述开环节距轮廓中提取所述预定间隔内的多个相等子间隔的原型波形（PW）; 通过PW的增益值对PW进行归一化; 编码PW的大小; 并且在幅度域中直接量化PW，而不会将PW进一步分解成复分量，其中通过使用固定维度矢量量化器（VQ's）的基于语音分类的分层量化方法来执行直接量化。

9. 发明授权

US5666370A High performance error control coding in channel encoders and decoders 失效
标题翻译：通道编码器和解码器中的高性能错误控制编码
公开(公告)号：US5666370A
公开(公告)日：1997-09-09
申请号：US591127
申请日：1996-01-25
申请人： Kalyan Ganesan , Kumar Swaminathan , Prabhat Gupta , P. Vijay Kumar
发明人： Kalyan Ganesan , Kumar Swaminathan , Prabhat Gupta , P. Vijay Kumar
IPC分类号： H03M13/29 , H03M13/35 , H04L1/00 , H04L1/20 , G06F11/08
CPC分类号： H04L1/0065 , H03M13/09 , H03M13/21 , H03M13/27 , H03M13/29 , H03M13/2903 , H03M13/2927 , H03M13/35 , H04L1/0054 , H04L1/007 , H04L1/201 , H04L1/208
摘要： An improved error control coding scheme is implemented in low bit rate coders in order to improve their performance in the presence of transmission errors typical of the digital cellular channel. The error control coding scheme exploits the nonlinear block codes (NBCs) for purposes of tailoring those codes to a fading channel in order to provide superior error protection to the compressed half rate speech data. For a half rate speech codec assumed to have a frame size of 40 ms, the speech encoder puts out a fixed number of bits per 40 ms. These bits are divided into three distinct classes, referred to as Class 1, Class 2 and Class 3 bits. A subset of the Class 1 bits are further protected by a CRC for error detection purposes. The Class 1 bits and the CRC bits are encoded by a rate 1/2 Nordstrom Robinson code with codeword length of 16. The Class 2 bits are encoded by a punctured version of the Nordstrom Robinson code. It has an effective rate of 8/14 with a codeword length 14. The Class 3 bits are left unprotected. The coded Class 1 plus CRC bits, coded Class 2 bits, and the Class 3 bits are mixed in an interleaving array of size 16.times.17 and interleaved over two slots in a manner that optimally divides each codeword between the two slots. At the receiver the coded Class 1 plus CRC bits, coded Class 2 bits, and Class 3 bits are extracted after de-interleaving.
摘要翻译：在低比特率编码器中实现改进的误差控制编码方案，以便在存在数字蜂窝信道典型的传输错误的情况下提高它们的性能。错误控制编码方案利用非线性块码（NBC），以便将这些码定制到衰落信道，以便为压缩的半速率语音数据提供优良的错误保护。对于假定帧大小为40ms的半速率语音编解码器，语音编码器每40ms提出固定数量的位。这些位被分为三个不同的类，称为1类，2类和3类。 Class 1位的一个子集进一步受到CRC的保护，以便进行错误检测。 1比特和CRC比特由码字长度为16的速率+ E，fra 1/2 + EE Nordstrom Robinson码编码。2类比特由Nordstrom Robinson码的穿孔版本编码。它的有效速率为+ E，带有码字长度为14的8/14 + EE。3类比特未被保护。经编码的1类加CRC比特，2类编码和3类比特混合在大小为16×17的交织阵列中，并且以两个时隙之间的每个码字最佳分割的方式在两个时隙上进行交织。在接收器处，解码后提取编码的1类加CRC比特，2类编码和3类比特。

10. 发明授权

US07013269B1 Voicing measure for a speech CODEC system 有权
标题翻译：语音CODEC系统的语音测量
公开(公告)号：US07013269B1
公开(公告)日：2006-03-14
申请号：US10073406
申请日：2002-02-13
申请人： Udaya Bhaskar , Kumar Swaminathan
发明人： Udaya Bhaskar , Kumar Swaminathan
IPC分类号： G10L19/02
CPC分类号： G10L19/097 , G10L25/93
摘要： A system and method is provided that employs a frequency domain interpolative CODEC system for low bit rate coding of speech which comprises a linear prediction (LP) front end adapted to process an input signal providing LP parameters which are quantized and encoded over predetermined intervals and used to compute a LP residual signal. An open loop pitch estimator adapted to process the LP residual signal, a pitch quantizer, and a pitch interpolator also provides a pitch contour within the predetermined intervals. A voice activity detector adapted to process the LP parameters and the open loop pitch contour over the predetermined intervals is also provided as well as a signal processor responsive to the LP residual signal and the pitch contour and adapted to perform the following functions: extract a prototype waveform (PW) from the LP residual and the open loop pitch contour for a number of equal sub-intervals within the predetermined invervals; normalize the PW by a gain value of the PW; encode a magnitude of the PW; and provide a voicing measure where the voicing measure characterizes a degree of vocing of the input speech signal and is derived from several input parameters that are correlated to degrees of periodicity of the signal over the predetermined intervals. The voicing measure is provided for the purpose of regenerating a PW phase at a decoder; and providing improved quantization of the PW magnitude at an encoder. The voicing measure is encoded jointly with a PW nonstationarity measure vector using a spectrally weighted vector quantizer having a codebook partioned based on a voiced and unvoiced mode.
摘要翻译：提供了一种系统和方法，其采用用于语音的低比特率编码的频域内插编解码器系统，其包括线性预测（LP）前端，其适于处理提供经过预定间隔量化和编码的LP参数的输入信号，并使用以计算LP残差信号。适于处理LP残差信号的开环音调估计器，音调量化器和音调内插器也在预定间隔内提供音调轮廓。还提供了适于在预定间隔上处理LP参数和开环音调轮廓的语音活动检测器以及响应于LP残差信号和音调轮廓的信号处理器，并且适于执行以下功能：提取原型来自LP残差的波形（PW）和开环节距轮廓线，用于在预定的反相中的多个相等子间隔; 通过PW的增益值对PW进行归一化; 编码PW的大小; 并且提供发声测量，其中所述发声测量表征所述输入语音信号的声音程度，并且从与所述预定间隔上的所述信号的周期度相关的若干输入参数导出。提供发声措施是为了在解码器处再生PW相; 并且在编码器处提供对PW幅度的改进的量化。发声测量与PW非平稳测量向量一起编码，其使用具有基于有声和无声模式分组的码本的频谱加权矢量量化器。

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