公开(公告)号：US5630016A

公开(公告)日：1997-05-13

申请号：US614777

申请日：1996-03-07

Applicant: Kumar Swaminathan ,  Brian M. McCarthy

Inventor： Kumar Swaminathan ,  Brian M. McCarthy

IPC: G10L19/00 ,  G10L9/00

CPC classification number: G10L19/012

Abstract: A digital discontinuous cellular communication system has a transmitter that transmits two frames of data following detection of voice inactivity. A receiver includes a comfort noise generator that uses the two frames of data to output noise to the speaker during period of voice inactivity. The comfort noise generator includes synthesis codebook with samples scaled by actual background noise and excitation codebook with samples filtered and scaled by the background noise that are combined to produce comfort noise having attributes and loudness level of the received background noise prior to interruption of transmission. The scaled signals are weighted to vary the loudness level and spectral attributes.

Abstract translation: 数字不连续的蜂窝通信系统具有在检测到语音不活动之后发送两帧数据的发射机。 接收机包括舒适噪声发生器，其在语音不活动期间使用两帧数据来向扬声器输出噪声。 舒适噪声发生器包括具有通过实际背景噪声缩放的样本的合成码本和激励码本，其中滤波和被背景噪声缩放的样本被组合以产生在传输中断之前具有接收到的背景噪声的属性和响度水平的舒适噪声。 加权比例的信号以改变响度级别和频谱属性。

公开(公告)号：US5537509A

公开(公告)日：1996-07-16

申请号：US890747

申请日：1992-05-28

Applicant: Kumar Swaminathan ,  Brian M. McCarthy

Inventor： Kumar Swaminathan ,  Brian M. McCarthy

IPC: G10L19/00 ,  H04B7/26 ,  H04J3/17 ,  H04L12/28 ,  H04L12/56 ,  H04W28/06 ,  H04W72/04 ,  G10L9/00

CPC classification number: H04W72/044 ,  G10L19/012 ,  H04B7/26 ,  H04B7/2643 ,  H04J3/172 ,  H04W28/06 ,  H04W72/0406

Abstract: A digital discontinuous cellular communication system has a transmitter that transmits two frames of data following detection of voice inactivity. A receiver includes a comfort noise generator that uses the two frames of data to output noise to the speaker during period of voice inactivity. The comfort noise generator includes synthesis codebook with samples scaled by actual background noise and excitation codebook with samples filtered and scaled by the background noise that are combined to produce comfort noise having attributes and loudness level of the received background noise prior to interruption of transmission. The scaled signals are weighted to vary the loudness level and spectral attributes.

Abstract translation: 数字不连续的蜂窝通信系统具有在检测到语音不活动之后发送两帧数据的发射机。 接收机包括舒适噪声发生器，其在语音不活动期间使用两帧数据来向扬声器输出噪声。 舒适噪声发生器包括具有通过实际背景噪声缩放的样本的合成码本和激励码本，其中滤波和被背景噪声缩放的样本被组合以产生在传输中断之前具有接收到的背景噪声的属性和响度水平的舒适噪声。 加权比例的信号以改变响度级别和频谱属性。

公开(公告)号：US5781540A

公开(公告)日：1998-07-14

申请号：US497582

申请日：1995-06-30

Applicant: James Eryx Malcolm ,  Daniel Fraley ,  Adrian Morris ,  David Roos ,  Kumar Swaminathan ,  Seok Ho Kim ,  Robert Carroll Marquart

Inventor： James Eryx Malcolm ,  Daniel Fraley ,  Adrian Morris ,  David Roos ,  Kumar Swaminathan ,  Seok Ho Kim ,  Robert Carroll Marquart

IPC: H04B7/185 ,  H04J3/06 ,  H04L7/04 ,  H04L7/10 ,  H04B7/212

CPC classification number: H04J3/0605 ,  H04B7/18532 ,  H04L7/046 ,  H04L7/10

Abstract: 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.

Abstract translation: 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。

公开(公告)号：US5751903A

公开(公告)日：1998-05-12

申请号：US359116

申请日：1994-12-19

Applicant: Kumar Swaminathan ,  Murthy Vemuganti

Inventor： Kumar Swaminathan ,  Murthy Vemuganti

IPC: G10L11/06 ,  G10L19/00 ,  G10L19/14 ,  G10L9/18

CPC classification number: G10L19/06 ,  G10L19/18 ,  G10L25/24 ,  G10L25/93

Abstract: 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.

Abstract translation: 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。

公开(公告)号：US06996523B1

公开(公告)日：2006-02-07

申请号：US10073128

申请日：2002-02-13

Applicant: Udaya Bhaskar ,  Kumar Swaminathan

Inventor： Udaya Bhaskar ,  Kumar Swaminathan

IPC: G10L19/10

CPC classification number: G10L19/097 ,  G10L19/032

Abstract: 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).

Abstract translation: 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。

公开(公告)号：US06453289B1

公开(公告)日：2002-09-17

申请号：US09361015

申请日：1999-07-23

Applicant: Filiz Basbug Ertem ,  Srinivas Nandkumar ,  Kumar Swaminathan

Inventor： Filiz Basbug Ertem ,  Srinivas Nandkumar ,  Kumar Swaminathan

IPC: G10L2102

CPC classification number: G10L25/78 ,  G10L21/0208

Abstract: 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.

Abstract translation: 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。

公开(公告)号：US5666370A

公开(公告)日：1997-09-09

申请号：US591127

申请日：1996-01-25

Applicant: Kalyan Ganesan ,  Kumar Swaminathan ,  Prabhat Gupta ,  P. Vijay Kumar

Inventor： Kalyan Ganesan ,  Kumar Swaminathan ,  Prabhat Gupta ,  P. Vijay Kumar

IPC: H03M13/29 ,  H03M13/35 ,  H04L1/00 ,  H04L1/20 ,  G06F11/08

CPC classification number: H04L1/0065 ,  H03M13/09 ,  H03M13/21 ,  H03M13/27 ,  H03M13/29 ,  H03M13/2903 ,  H03M13/2927 ,  H03M13/35 ,  H04L1/0054 ,  H04L1/007 ,  H04L1/201 ,  H04L1/208

Abstract: 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.

Abstract translation: 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。

公开(公告)号：US07013269B1

公开(公告)日：2006-03-14

申请号：US10073406

申请日：2002-02-13

Applicant: Udaya Bhaskar ,  Kumar Swaminathan

Inventor： Udaya Bhaskar ,  Kumar Swaminathan

IPC: G10L19/02

CPC classification number: G10L19/097 ,  G10L25/93

Abstract: 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.

Abstract translation: 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。

公开(公告)号：US20050008240A1

公开(公告)日：2005-01-13

申请号：US10836672

申请日：2004-04-30

Applicant: Ashish Banerji ,  Kannan Panchapakesan ,  Kumar Swaminathan

Inventor： Ashish Banerji ,  Kannan Panchapakesan ,  Kumar Swaminathan

IPC: G06K9/46 ,  H04N5/235 ,  H04N5/262 ,  H04N13/02 ,  H04N19/89 ,  H04N19/895

CPC classification number: H04N7/15 ,  H04N5/2624 ,  H04N19/40 ,  H04N19/46 ,  H04N19/467 ,  H04N19/573 ,  H04N19/65 ,  H04N19/70 ,  H04N19/89 ,  H04N19/895

Abstract: A drift-free hybrid method of performing video stitching is provided. The method includes decoding a plurality of video bitstreams and storing prediction information. The decoded bitstreams form video images, spatially composed into a combined image. The image comprises frames of ideal stitched video sequence. The method uses prediction information in conjunction with previously generated frames to predict pixel blocks in the next frame. A stitched predicted block in the next frame is subtracted from a corresponding block in a corresponding frame to create a stitched raw residual block. The raw residual block is forward transformed, quantized, entropy encoded and added to the stitched video bitstream along with the prediction information. Also, the stitched raw residual block is inverse transformed and dequantized to create a stitched decoded residual block. The residual block is added to the predicted block to generate the stitched reconstructed block in the next frame of the sequence.

Abstract translation: 提供了一种执行视频拼接的无漂移混合方法。 该方法包括解码多个视频位流并存储预测信息。 解码的比特流形成视频图像，空间地组成组合图像。 该图像包括理想的拼接视频序列的帧。 该方法结合先前生成的帧使用预测信息来预测下一帧中的像素块。 从相应帧中的对应块中减去下一帧中的拼接预测块，以创建缝合的原始残留块。 将原始残留块与预测信息一起进行前向变换，量化，熵编码并添加到拼接视频比特流中。 此外，缝合的原始残留块被逆变换和去量化以产生缝合解码的残余块。 将残余块添加到预测块以在序列的下一帧中产生缝合的重构块。

公开(公告)号：US06418408B1

公开(公告)日：2002-07-09

申请号：US09542792

申请日：2000-04-04

Applicant: Bangalore R. Udaya Bhaskar ,  Srinivas Nandkumar ,  Kumar Swaminathan ,  Gaguk Zakaria

Inventor： Bangalore R. Udaya Bhaskar ,  Srinivas Nandkumar ,  Kumar Swaminathan ,  Gaguk Zakaria

IPC: G10L1904

CPC classification number: 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

Abstract: 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.