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：US5195137A

公开(公告)日：1993-03-16

申请号：US646122

申请日：1991-01-28

Applicant: Kumar Swaminathan

Inventor： Kumar Swaminathan

IPC: G06T9/00 ,  H03M7/30

CPC classification number: G06T9/008 ,  H03M7/3082

Abstract: In many applications involving the coding and processing of speech signals the relevant applicable codebook is one which may be termed a sparse codebook. That is, the majority of elements in the codebook are zero valued. The searching of such a sparse codebook is accelerated in accord with the present invention by generating auxiliary information defining the sparse nature of the codebok and using this information to assist and speed up searches of the codebook.In a particular method of searching the calculation of the distance between a target vector and a stored codebook vector is enhanced by use of a distortion metric derived from energy terms and correlation terms of the codebook entries. Calculation of these energy and correlation terms is speeded up by exploiting the sparseness of the codebook entries. The non-zero elements (NZE) of the space codebook are each identified and are defined by their offset from a reference point.

公开(公告)号：US06961320B1

公开(公告)日：2005-11-01

申请号：US09669283

申请日：2000-09-26

Applicant: Kumar Swaminathan ,  Udaya Bhaskar

Inventor： Kumar Swaminathan ,  Udaya Bhaskar

IPC: H04L5/22 ,  H04M3/42 ,  H04M11/00 ,  H04M11/06

CPC classification number: H04M3/42391 ,  H04M11/066 ,  H04M2207/18

Abstract: A method, system, and software product for transmitting TTY/TDD signals in a system employing low bit-rate voice compression are disclosed. The method includes receiving an input signal and generating a teletypewriter (TTY) indicator signal from the input signal. Whether or not the input signal is a TTY signal including a TTY character, is determined based on the TTY indicator signal. A TTY packet including the TTY character of the TTY signal is constructed and transmitted if the input signal is determined to be a TTY signal. A method, system, and software product for receiving and decoding TTY/TDD signal is also disclosed.

Abstract translation: 公开了一种在采用低比特率语音压缩的系统中传输TTY / TDD信号的方法，系统和软件产品。 该方法包括接收输入信号并从输入信号产生电传打字机（TTY）指示符信号。 是否基于TTY指示符信号确定输入信号是否包括TTY字符的TTY信号。 如果输入信号被确定为TTY信号，则构造并发送包括TTY信号的TTY字符的TTY分组。 还公开了用于接收和解码TTY / TDD信号的方法，系统和软件产品。

公开(公告)号：US06314393B1

公开(公告)日：2001-11-06

申请号：US09270918

申请日：1999-03-16

Applicant: Yue-Peng Zheng ,  Shvetal K. Patel ,  Kumar Swaminathan

Inventor： Yue-Peng Zheng ,  Shvetal K. Patel ,  Kumar Swaminathan

IPC: G10L1912

CPC classification number: G10L19/16 ,  G10L19/12

Abstract: An integrated circuit for processing a speech signal in accordance with a CELP standard includes a plurality of processing elements coupled to a data bus in parallel. Each processing element includes a multiplier and an accumulator. The integrated circuit further includes an auxiliary processing element, which is also coupled to the data bus and has a division unit and a comparator. The plurality of processing elements and the auxiliary processing element are also coupled in a pipeline formation.

Abstract translation: 用于根据CELP标准处理语音信号的集成电路包括并行耦合到数据总线的多个处理单元。 每个处理元件包括一个乘法器和一个累加器。 集成电路还包括辅助处理元件，其还耦合到数据总线，并具有除法单元和比较器。 多个处理元件和辅助处理元件也以管道结构耦合。

公开(公告)号：US5758256A

公开(公告)日：1998-05-26

申请号：US483426

申请日：1995-06-07

Applicant: Kirk H. Berry ,  Debabrata Chowdhury ,  Shrirang Jangi ,  Kumar Swaminathan

Inventor： Kirk H. Berry ,  Debabrata Chowdhury ,  Shrirang Jangi ,  Kumar Swaminathan

IPC: G10L19/00 ,  H04W88/18 ,  H04M11/00 ,  H04Q7/00

CPC classification number: H04W88/181 ,  G10L19/012

Abstract: A method of transporting speech information over a wireless cellular communications system is provided. By determining the existence and compatibility of the destination port with the origination port in a given telephone call, the present invention is capable of using only one compression step and one decompression step. Accordingly, voice signal degradation and delay associated with multiple compression/decompression steps may be reduced.

Abstract translation: 提供了一种通过无线蜂窝通信系统传送语音信息的方法。 通过在给定的电话呼叫中确定目的地端口与起始端口的存在和兼容性，本发明能够仅使用一个压缩步骤和一个解压缩步骤。 因此，可以减少与多个压缩/解压缩步骤相关联的语音信号衰减和延迟。

公开(公告)号：US5596676A

公开(公告)日：1997-01-21

申请号：US540637

申请日：1995-10-11

Applicant: Kumar Swaminathan ,  Kalyan Ganesan ,  Prabhat K. Gupta

Inventor： Kumar Swaminathan ,  Kalyan Ganesan ,  Prabhat K. Gupta

IPC: G10L11/04 ,  G10L11/06 ,  G10L19/00 ,  G10L19/12 ,  G10L19/14 ,  G10L9/12 ,  G10L9/14

CPC classification number: G10L19/012 ,  G10L19/12 ,  G10L19/26 ,  G10L25/90 ,  G10L25/09 ,  G10L25/18 ,  G10L25/24 ,  G10L25/93

Abstract: A method for encoding a signal that includes a speech component is described. First and second linear prediction windows of a frame are analyzed to generate sets of filter coefficients. First and second pitch analysis windows of the frame are analyzed to generate pitch estimates. The frame is classified in one of at least two modes, e.g. voiced, unvoiced and noise modes, based, for example, on pitch stationarity, short-term level gradient or zero crossing rate. Then the frame is encoded using the filter coefficients and pitch estimates in a particular manner depending upon the mode determination for the frame, preferably employing CELP based encoding algorithms.

Abstract translation: 描述了一种用于编码包括语音分量的信号的方法。 分析帧的第一和第二线性预测窗口以生成滤波器系数集合。 分析帧的第一和第二音调分析窗口以产生音调估计。 该帧被分类为至少两种模式之一，例如， 例如，基于音调稳定性，短期电平梯度或零交叉率的有声，无声和噪声模式。 然后，根据帧的模式确定，优选使用基于CELP的编码算法，以特定方式使用滤波器系数和音调估计来对帧进行编码。

公开(公告)号：US5495555A

公开(公告)日：1996-02-27

申请号：US905992

申请日：1992-06-25

Applicant: Kumar Swaminathan

Inventor： Kumar Swaminathan

IPC: G10L19/08 ,  G10L11/04 ,  G10L11/06 ,  G10L19/00 ,  G10L19/04 ,  G10L19/12 ,  G10L19/14 ,  G10L3/02

CPC classification number: G10L19/26 ,  G10L19/12 ,  G10L25/90 ,  G10L25/93

Abstract: Code excited linear prediction (CELP) is performed using two voiced and unvoiced sets of windows, each set is used both for linear prediction and pitch determination. The accompanying degradation in voice quality is comparable to the IS54 standard 8.0 Kbps voice coder employed in U.S. digital cellular systems. This is accomplished by using the same parametric model used in traditional CELP coders but determining, quantizing, encoding, and updating these parameters differently. The low bit rate speech decoder is like most CELP decoders except that it operates in two modes depending on the received mode bit. Both pitch prefiltering and global postfiltering are employed for enhancement of the synthesized speech. In addition, built-in error detection and error recovery schemes are used that help mitigate the effects of any uncorrectable transmission errors.

Abstract translation: 代码激励线性预测（CELP）是使用两个有声和无声的窗口组来执行的，每组都用于线性预测和音调确定。 伴随的语音质量下降与美国数字蜂窝系统中使用的IS54标准8.0Kbps语音编码器相当。 这通过使用与传统CELP编码器中使用的相同的参数模型来实现，但是以不同的方式确定，量化，编码和更新这些参数。 低比特率语音解码器像大多数CELP解码器一样，除了它根据接收模式位在两种模式下操作。 采用两种音调预滤波和全局后置滤波来增强合成语音。 此外，使用内置的错误检测和错误恢复方案，有助于减轻任何不可纠正的传输错误的影响。

公开(公告)号：US4956871A

公开(公告)日：1990-09-11

申请号：US252250

申请日：1988-09-30

Applicant: Kumar Swaminathan

Inventor： Kumar Swaminathan

IPC: H03M1/12 ,  G10L19/02 ,  G10L19/12 ,  H03M7/30 ,  H04B1/66 ,  H04B14/04

CPC classification number: H04B1/667

Abstract: A sub-band speech coding arrangement divides the speech spectrum into sub-bands and allocates bits to encode the time frame interval samples of each sub-band responsive to the speech energies of the sub-bands. The sub-band samples are quantized according to the sub-band energy bit allocation and the time frame quantized samples and speech energy signals are coded. A signal representative of the residual difference between the each time frame interval speech sample of the sub-band and the corresponding quantized speech sample of the sub-band is generated. The quality of the sub-band coded signal is improved by selecting the sub-bands with the largest residual differences, producing a vector signal from the sequence of residual difference signals of each selected sub-band, and matching the sub-band vector signal to one of a set of stored Gaussian codebook entries to generate a reduced bit code for the selected vector signal. The coded time frame interval quantized signals, speech energy signals and reduced bit codes for the selected residual differences are combined to form a multiplexed stream for the speech pattern of the time frame interval.

Abstract translation: 子带语音编码装置将语音频谱划分为子频带，并且分配比特以响应于子频带的语音能量对每个子频带的时间间隔采样进行编码。 根据子带能量比特分配对子带样本进行量化，并对时间帧量化样本和语音能量信号进行编码。 产生表示子带的每个时间间隔语音样本与子带的对应的量化语音样本之间的残差的信号。 通过选择具有最大残差的子带来改善子带编码信号的质量，从每个选择的子带的残差差信号的序列产生矢量信号，并将子带向量信号与 一组存储的高斯码本条目中的一个，以生成所选择的矢量信号的缩减比特码。 对所选择的残差进行编码的时间间隔量化信号，语音能量信号和降低的比特码被组合以形成用于时间间隔的语音模式的多路复用流。

公开(公告)号：US09894379B2

公开(公告)日：2018-02-13

申请号：US10074765

申请日：2002-02-12

Applicant: Ashish Banerji ,  Kumar Swaminathan

Inventor： Ashish Banerji ,  Kumar Swaminathan

IPC: H04N7/12 ,  H04N19/517 ,  H04N19/46 ,  H04N19/61

CPC classification number: H04N19/517 ,  H04N19/46 ,  H04N19/61

Abstract: A system and method for compressing video is disclosed, in which video frames that between consecutive I-frames are grouped into a video data set. The video data set is split into separate homogeneous files, and each of the homogeneous files are individually compressed. In one embodiment, the individually compressed files are multiplexed to form a bit stream.

公开(公告)号：US06931373B1

公开(公告)日：2005-08-16

申请号：US10073423

申请日：2002-02-13

Applicant: Udaya Bhaskar ,  Kumar Swaminathan

Inventor： Udaya Bhaskar ,  Kumar Swaminathan

IPC: G10L19/02 ,  G10L19/08

CPC classification number: G10L19/08 ,  G10L19/097

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 separate stationary and nonstationary components of the PW using a low complexity alignment process and a filtering process that introduce no delay. The ratio of the energy of the nonstationary component of the PW to that of the stationary component of the PW is averaged across 5 subbands to compute the nonstationarity measure as a frequency dependent vector entity. A measure of the degree of voicing of the residual is also computed using openloop pitchgain, pitch variance, relative signal power, PW correlation and PW nonstationarity in low frequency subbands. The nonstationarity measure and voicing measure are encoded using a 6-bit spectrally weighted vector quantization scheme using a codebook partitioned based on a voiced/unvoiced decision. At the decoder, a stationary component of PW is reconstructed as a weighted combination of the previous PW phase vector, a random phase perturbation and a fixed phase vector obtained from a voiced pitch pulse.