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1. 发明授权

US06973127B1 Apparatus and method for memory saving wavelet based video coding 失效
标题翻译：用于存储器保存小波的视频编码的装置和方法
公开(公告)号：US06973127B1
公开(公告)日：2005-12-06
申请号：US09648988
申请日：2000-08-25
申请人： Victor D. Kolesnik , Irina E. Bocharova , Boris D. Kudryashov , Andrey Marsavin , Victor Simileysky , Felix A. Taubin
发明人： Victor D. Kolesnik , Irina E. Bocharova , Boris D. Kudryashov , Andrey Marsavin , Victor Simileysky , Felix A. Taubin
IPC分类号： H04N7/12 , H04N7/26
CPC分类号： H04N19/61 , H04N19/13 , H04N19/42 , H04N19/63
摘要： A method and apparatus for compressing video signals. The video compression may include row by row horizontal subband filtering of video data representing part of an image, and row by row concurrent vertical high and vertical low pass subband filtering of the results of the horizontal subband filtering. Less than the vertical filter length of rows of results of the horizontal subband filtering may be buffered at a time. The vertical high and vertical low pass subband filtering may include calculating a first result row by adding a first row of a recursion buffer to the results of multiplying a row of results from the horizontal subband filtering and a first vertical filter coefficient, calculating a second result row by adding the first row of the recursion buffer to the result of multiplying the row of results of the horizontal subband filtering and a second vertical filter coefficient, overwriting the first row of the recursion buffer with the second result row, and transmitting the first result row to be encoded.
摘要翻译：一种用于压缩视频信号的方法和装置。视频压缩可以包括对表示图像的一部分的视频数据进行逐行水平子带滤波，并且对水平子带滤波的结果逐行并行垂直高和低通滤波。可以一次缓冲小于水平子带滤波结果的行的垂直滤波器长度。垂直高低通滤波可以包括通过将递归缓冲器的第一行与从水平子带滤波结果行乘以与第一垂直滤波器系数相乘的结果来计算第一结果行，计算第二结果通过将递归缓冲器的第一行添加到将水平子带滤波的结果行与第二垂直滤波器系数相乘的结果，用第二结果行重写递归缓冲器的第一行，并发送第一结果行要编码。

2. 发明授权

US06263312B1 Audio compression and decompression employing subband decomposition of residual signal and distortion reduction 失效
标题翻译：音频压缩和解压缩采用子带分解残差信号和失真减少
公开(公告)号：US06263312B1
公开(公告)日：2001-07-17
申请号：US09033431
申请日：1998-03-02
申请人： Victor D. Kolesnik , Irina E. Bocharova , Boris D. Kudryashov , Eugene Ovsyannikov , Andrei N. Trofimov , Boris Troyanovsky
发明人： Victor D. Kolesnik , Irina E. Bocharova , Boris D. Kudryashov , Eugene Ovsyannikov , Andrei N. Trofimov , Boris Troyanovsky
IPC分类号： G10L2104
CPC分类号： G10L19/0208 , G10L19/0212 , G10L19/04
摘要： A method and apparatus to achieve relatively high quality audio data compression/decompression, while achieving relatively low bit rates (e.g., high compression ratios). According to one aspect of the invention, a residual signal is subband decomposed and adaptively quantized and encoded to capture frequency information that may provide higher quality compression and decompression relative to transform encoding techniques. According to a second aspect of the invention, an input audio signal is compared to an encoded signal based on the input audio signal to detect and reduce, as necessary, distortion in the encoded signal or portions thereof.
摘要翻译：一种实现相对高质量音频数据压缩/解压缩的方法和装置，同时实现相对较低的比特率（例如，高压缩比）。根据本发明的一个方面，残留信号被子带分解并被自适应地量化和编码以捕获相对于变换编码技术可以提供更高质量的压缩和解压缩的频率信息。根据本发明的第二方面，将输入音频信号与基于输入音频信号的编码信号进行比较，以根据需要检测和减少编码信号或其部分中的失真。

3. 发明授权

US5729655A Method and apparatus for speech compression using multi-mode code excited linear predictive coding 失效
标题翻译：使用多模式码激励线性预测编码的语音压缩方法和装置
公开(公告)号：US5729655A
公开(公告)日：1998-03-17
申请号：US716771
申请日：1996-09-24
申请人： Victor D. Kolesnik , Andrey N. Trofimov , Irina E. Bocharova , Victor Yu Krachkovsky , Boris D. Kudryashov , Eugeny P. Ovsjannikov , Boris K. Trojanovsky , Sergei I. Kovalov
发明人： Victor D. Kolesnik , Andrey N. Trofimov , Irina E. Bocharova , Victor Yu Krachkovsky , Boris D. Kudryashov , Eugeny P. Ovsjannikov , Boris K. Trojanovsky , Sergei I. Kovalov
IPC分类号： G10L11/06 , G10L19/00 , G10L19/12 , G10L19/14 , G10L3/02
CPC分类号： G10L19/18 , G10L19/12 , G10L2025/935 , G10L25/24
摘要： An apparatus and method of coding speech. The apparatus includes a first circuit being coupled to receive a first signal, the first signal corresponds to the speech signal. The first circuit is for generating a first set of parameters corresponding to the first frame. The apparatus includes a second circuit, being coupled to receive a second signal and the first set of parameters, the second signal corresponding to the speech signal, and the second circuit is for generating a third signal. The apparatus further includes a pulse train analyzer, being coupled to the second circuit, for generating a third match value, a third set of parameters, and a third excitation value. The apparatus further including a fourth circuit, being coupled to the second circuit, for generating a fourth match value, a fourth set of parameters, and a fourth excitation value. The apparatus further including a fifth circuit, being coupled to the third circuit and the fourth circuit, for selecting a mode corresponding to a match value. The apparatus further including a sixth circuit, being coupled to the fifth circuit, for selecting a selected set of parameters and a selected excitation corresponding to the mode. The apparatus further including a seventh circuit, being coupled to the first circuit and the sixth circuit, for generating an encoded signal responsive to the selected set of parameters and the mode.
摘要翻译：编码语音的装置和方法。该装置包括被耦合以接收第一信号的第一电路，第一信号对应于语音信号。第一电路用于产生对应于第一帧的第一组参数。该装置包括第二电路，被耦合以接收第二信号和第一组参数，第二信号对应于语音信号，第二电路用于产生第三信号。该装置还包括脉冲串分析器，耦合到第二电路，用于产生第三匹配值，第三组参数和第三激励值。该装置还包括耦合到第二电路的第四电路，用于产生第四匹配值，第四组参数和第四激励值。该装置还包括耦合到第三电路和第四电路的第五电路，用于选择对应于匹配值的模式。该装置还包括耦合到第五电路的第六电路，用于选择所选择的一组参数和对应于该模式的选定激励。该装置还包括耦合到第一电路和第六电路的第七电路，用于响应所选择的一组参数和该模式产生编码信号。

4. 发明授权

US06249614B1 Video compression and decompression using dynamic quantization and/or encoding 失效
标题翻译：使用动态量化和/或编码的视频压缩和解压缩
公开(公告)号：US06249614B1
公开(公告)日：2001-06-19
申请号：US09036883
申请日：1998-03-06
申请人： Victor D. Kolesnik , Irina E. Bocharova , Boris D. Kudryashov , Andrey Malkov , Andrey Marsavin , Victor Simileysky , Felix A. Taubin
发明人： Victor D. Kolesnik , Irina E. Bocharova , Boris D. Kudryashov , Andrey Malkov , Andrey Marsavin , Victor Simileysky , Felix A. Taubin
IPC分类号： G06K936
CPC分类号： H04N19/14 , H04N19/10 , H04N19/124 , H04N19/61 , H04N19/63 , H04N19/96
摘要： A method and apparatus for performing video compression and decompression using dynamic quantization and/or encoding. According to one aspect of the compression method, an image is recursively filtered into its constituent components each represented by a matrix of coefficients. Next, the level of correlation of a first of the matrices of coefficients is determined. Based on the level of correlation of this first coefficient matrix, one of a first quantization technique and a second quantization technique is selected. The first coefficient matrix is then quantized using the selected quantization technique. According to another aspect of the compression method, an image is digitally filtered into its constituent components, each represented by a matrix of coefficients. At least certain matrices are quantized to generated a first quantized matrix. The first quantized matrix is then recursively divided until the first quantized matrix or each of the resulting submatrices is sufficiently dense or sufficiently sparse. Each of the sufficiently dense matrices are encoded using a first technique, while each of the sufficiently sparse matrices are encoded a second technique.
摘要翻译：一种使用动态量化和/或编码来执行视频压缩和解压缩的方法和装置。根据压缩方法的一个方面，图像被递归地滤波成其各自由系数矩阵表示的组成部分。接下来，确定第一系数矩阵的相关性水平。基于该第一系数矩阵的相关度，选择第一量化技术和第二量化技术之一。然后使用所选择的量化技术量化第一系数矩阵。根据压缩方法的另一方面，图像被数字滤波成其组成分量，每个由系数矩阵表示。至少某些矩阵被量化以产生第一量化矩阵。然后将第一量化矩阵递归地划分直到第一量化矩阵或所得到的子矩阵中的每一个足够密集或足够稀疏。使用第一技术对足够密集的矩阵中的每一个进行编码，而每一个足够稀疏的矩阵都被编码为第二种技术。

5. 发明授权

US5832443A Method and apparatus for adaptive audio compression and decompression 失效
标题翻译：用于自适应音频压缩和解压缩的方法和装置
公开(公告)号：US5832443A
公开(公告)日：1998-11-03
申请号：US806075
申请日：1997-02-25
申请人： Victor D. Kolesnik , Irina Bocharova , Boris Kudryashov , Eugene Ovsyannikov , Andrei Trofimov , Boris Troyanovsky
发明人： Victor D. Kolesnik , Irina Bocharova , Boris Kudryashov , Eugene Ovsyannikov , Andrei Trofimov , Boris Troyanovsky
IPC分类号： G10L19/02 , G10L3/02 , G10L9/00
CPC分类号： G10L19/0212
摘要： A method and apparatus for compression and decompression of an audio signal. In encoding an input audio signal, at least a portion of the audio signal is transformed into a set of coefficients. A set of binary vectors associated with the set of coefficients are generated for digitizing the transformed audio signal using a fixed rate adaptive quantization. Information based on the set of binary vectors is combinatorially encoded and output as a bit stream of encoded audio data. The encoded audio data may be stored, transmitted, and/or decoded.
摘要翻译：一种用于压缩和解压缩音频信号的方法和装置。在对输入音频信号进行编码时，音频信号的至少一部分被变换成一组系数。产生与该组系数相关联的一组二进制向量，用于使用固定速率自适应量化来对经变换的音频信号进行数字化。基于二进制矢量集的信息被组合编码并作为编码音频数据的比特流输出。可以存储，发送和/或解码编码的音频数据。

6. 发明授权

US5659659A Speech compressor using trellis encoding and linear prediction 失效
标题翻译：语音压缩器采用网格编码和线性预测
公开(公告)号：US5659659A
公开(公告)日：1997-08-19
申请号：US665642
申请日：1996-06-18
申请人： Victor D. Kolesnik , Victor Yu Krachkovsky , Boris D. Kudrjashov , Eugene P. Ovsjannikov , Boris K. Trojanovsky , Vladimir V. Egorov
发明人： Victor D. Kolesnik , Victor Yu Krachkovsky , Boris D. Kudrjashov , Eugene P. Ovsjannikov , Boris K. Trojanovsky , Vladimir V. Egorov
IPC分类号： G10L19/12 , G10L3/02 , G10L9/00
CPC分类号： G10L19/12
摘要： A speech compressor utilizing Trellis Encoding and Linear Prediction (TELP). A TELP speech compressor provides improved signal generation and search technique for a code-excited linear prediction (CELP) speech encoder. TELP is a frame oriented coding that breaks the quantized speech signals into frames of prescribed length N and each frame into subframes of prescribed length L, which are processed as dependent units utilizing an analysis-by-synthesis approach. The approach is based on constructing the best mean square linear predicting filter and searching the best exciting sequence for the filter in order to produce synthesized speech. A trellis encoder is used instead of a stochastic code book. The Q-ary analysis of a given subframe and previous excitations is proposed for a fast vector search in an adaptive code book. It simplifies the implementation of digital speech compression.
摘要翻译：利用网格编码和线性预测（TELP）的语音压缩器。 TELP语音压缩器为代码激励线性预测（CELP）语音编码器提供改进的信号生成和搜索技术。 TELP是一种面向帧的编码，其将量化的语音信号分解成规定长度N和每帧的帧，成为规定长度L的子帧，其被用作依赖于分析的方法处理。该方法基于构建最佳均方线性预测滤波器并搜索滤波器的最佳激励序列以产生合成语音。使用网格编码器代替随机码本。提出了一种给定子帧和先前激励的Q元分析，用于自适应码本中的快速向量搜索。它简化了数字语音压缩的实现。

7. 发明授权

US06965859B2 Method and apparatus for audio compression 失效
标题翻译：用于音频压缩的方法和装置
公开(公告)号：US06965859B2
公开(公告)日：2005-11-15
申请号：US10378455
申请日：2003-03-03
申请人： Victor D. Kolesnik , Boris D. Kudryashov , Sergey Petrov , Evgeny Ovsyannikov , Boris Trojanovsky , Andrey Trofimov
发明人： Victor D. Kolesnik , Boris D. Kudryashov , Sergey Petrov , Evgeny Ovsyannikov , Boris Trojanovsky , Andrey Trofimov
IPC分类号： G10L19/02 , G10L19/14 , G10L21/04 , H04B20060101 , H04B1/66
CPC分类号： G10L19/02
摘要： A method and apparatus for audio compression receives an audio signal. Transform coding is applied to the audio signal to generate a sequence of transform frequency coefficients. The sequence of transform frequency coefficients is partitioned into a plurality of non-uniform width frequency ranges and then zero value frequency coefficients are inserted at the boundaries of the non-uniform width frequency ranges. As a result, certain of the transform frequency coefficients that represent high frequencies are dropped.
摘要翻译：用于音频压缩的方法和装置接收音频信号。将变换编码应用于音频信号以产生变换频率系数序列。将变换频率系数的序列划分为多个不均匀的宽度频率范围，然后在非均匀宽度频率范围的边界插入零值频率系数。结果，表示高频的某些变换频率系数下降。

8. 发明授权

US5818535A Method and apparatus for adaptive hybrid motion video compression and decompression 失效
标题翻译：用于自适应混合运动视频压缩和解压缩的方法和装置
公开(公告)号：US5818535A
公开(公告)日：1998-10-06
申请号：US719834
申请日：1996-09-30
申请人： Ilya L. Asnis , Victor D. Kolesnik , Sergey I. Kovalev , Kirill B. Shabunov
发明人： Ilya L. Asnis , Victor D. Kolesnik , Sergey I. Kovalev , Kirill B. Shabunov
IPC分类号： H04N7/26 , H04N7/34 , H04N7/36 , H04N7/50 , H04N7/32
CPC分类号： H04N19/593 , H04N19/503 , H04N19/51 , H04N19/61 , H04N19/107
摘要： A method and apparatus for compression and decompression of motion video data. According to one embodiment, a method for compression of a first frame of motion video is provided, wherein the first frame is compared to a base frame. Based on the comparison, an altered base frame is created that contains a set of one or more regions that are based on a set of one or more regions in the first frame and a set of one or more regions from the base frame. To encode the first frame, the altered base frame is compared to the first frame to derive a motion compensation vector based on a region in the altered base frame.
摘要翻译：一种用于压缩和解压缩运动视频数据的方法和装置。根据一个实施例，提供了一种用于压缩第一运动视频的方法，其中将第一帧与基本帧进行比较。基于比较，创建改变的基本帧，其包含基于第一帧中的一个或多个区域的集合的一组或多个区域，以及来自基本帧的一个或多个区域的集合。为了对第一帧进行编码，将改变的基本帧与第一帧进行比较，以基于改变的基本帧中的区域导出运动补偿向量。

9. 发明授权

US5602961A Method and apparatus for speech compression using multi-mode code excited linear predictive coding 失效
标题翻译：使用多模式码激励线性预测编码的语音压缩方法和装置
公开(公告)号：US5602961A
公开(公告)日：1997-02-11
申请号：US251471
申请日：1994-05-31
申请人： Victor D. Kolesnik , Andrey N. Trofimov , Irina E. Bocharova , Victor Y. Krachkovsky , Boris D. Kudryashov , Eugeny P. Ovsjannikov , Boris K. Trojanovsky , Sergei I. Kovalov
发明人： Victor D. Kolesnik , Andrey N. Trofimov , Irina E. Bocharova , Victor Y. Krachkovsky , Boris D. Kudryashov , Eugeny P. Ovsjannikov , Boris K. Trojanovsky , Sergei I. Kovalov
IPC分类号： G10L11/06 , G10L19/00 , G10L19/12 , G10L19/14 , G10L3/02
CPC分类号： G10L19/18 , G10L19/12 , G10L2025/935 , G10L25/24
摘要： An apparatus and method of coding speech. The apparatus includes a first circuit being coupled to receive a first signal, the first signal corresponds to the speech signal. The first circuit is for generating a first set of parameters corresponding to the first frame. The apparatus includes a second circuit, being coupled to receive a second signal and the first set of parameters, the second signal corresponding to the speech signal, and the second circuit is for generating a third signal. The apparatus further includes a pulse train analyzer, being coupled to the second circuit, for generating a third match value, a third set of parameters, and a third excitation value. The apparatus further including a fourth circuit, being coupled to the second circuit, for generating a fourth match value, a fourth set of parameters, and a fourth excitation value. The apparatus further including a fifth circuit, being coupled to the third circuit and the fourth circuit, for selecting a mode corresponding to a match value. The apparatus further including a sixth circuit, being coupled to the fifth circuit, for selecting a selected set of parameters and a selected excitation corresponding to the mode. The apparatus further including a seventh circuit, being coupled to the first circuit and the sixth circuit, for generating an encoded signal responsive to the selected set of parameters and the mode.
摘要翻译：编码语音的装置和方法。该装置包括被耦合以接收第一信号的第一电路，第一信号对应于语音信号。第一电路用于产生对应于第一帧的第一组参数。该装置包括第二电路，被耦合以接收第二信号和第一组参数，第二信号对应于语音信号，第二电路用于产生第三信号。该装置还包括脉冲串分析器，耦合到第二电路，用于产生第三匹配值，第三组参数和第三激励值。该装置还包括耦合到第二电路的第四电路，用于产生第四匹配值，第四组参数和第四激励值。该装置还包括耦合到第三电路和第四电路的第五电路，用于选择对应于匹配值的模式。该装置还包括耦合到第五电路的第六电路，用于选择所选择的一组参数和对应于该模式的选定激励。该装置还包括耦合到第一电路和第六电路的第七电路，用于响应所选择的一组参数和该模式产生编码信号。

10. 发明授权

US07181404B2 Method and apparatus for audio compression 失效
标题翻译：用于音频压缩的方法和装置
公开(公告)号：US07181404B2
公开(公告)日：2007-02-20
申请号：US11078975
申请日：2005-03-11
申请人： Victor D. Kolesnik , Boris D. Kudryashov , Sergey Petrov , Evgeny Ovsyannikov , Boris Trojanovsky , Andrey Trofimov
发明人： Victor D. Kolesnik , Boris D. Kudryashov , Sergey Petrov , Evgeny Ovsyannikov , Boris Trojanovsky , Andrey Trofimov
IPC分类号： G10L19/02
CPC分类号： G10L19/02
摘要： A method and apparatus for audio compression receives an audio signal. Transform coding is applied to the audio signal to generate a sequence of transform frequency coefficients. The sequence of transform frequency coefficients is partitioned into a plurality of non-uniform width frequency ranges and then zero value frequency coefficients are inserted at the boundaries of the non-uniform width frequency ranges. As a result, certain of the transform frequency coefficients that represent high frequencies are dropped.
摘要翻译：用于音频压缩的方法和装置接收音频信号。将变换编码应用于音频信号以产生变换频率系数序列。将变换频率系数的序列划分为多个不均匀的宽度频率范围，然后在非均匀宽度频率范围的边界插入零值频率系数。结果，表示高频的某些变换频率系数下降。

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