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61. 发明授权

US08688248B2 Method and system for content sampling and identification 有权
标题翻译：内容抽样和识别的方法和系统
公开(公告)号：US08688248B2
公开(公告)日：2014-04-01
申请号：US11547996
申请日：2005-04-19
申请人： Avery Li-Chun Wang
发明人： Avery Li-Chun Wang
IPC分类号： G06F17/00 , G06F13/00 , G10L21/00 , G10L25/00 , G10L19/04 , G11B27/11 , H04H60/58 , H04H20/14 , G10L15/22 , G10L17/26 , G10L15/26
CPC分类号： G11B27/11 , G10L15/22 , G10L15/265 , G10L17/26 , G10L19/04 , H04H20/14 , H04H60/58
摘要： A method and system for content sampling (106) and identification is presented. A data stream is recorded, and samples of the stream are identified. Samples (106) can be initially taken at random for identification. Once a sample (106) is identified and segmented within the data stream, the next time to sample (106) may be calculated to be outside the time frame of the identified sample (106). Thus, the sampling period can be adaptively adjusted to be at times after identified tracks.
摘要翻译：提出了一种内容抽样（106）和识别的方法和系统。记录数据流，并识别流的样本。样品（106）可以最初随机进行鉴定。一旦样本（106）在数据流内被识别和分段，则下一次采样（106）可被计算为在识别的样本（106）的时间范围之外。因此，采样周期可以被自适应地调整为在所识别的轨迹之后的时间。

62. 发明申请

US20140029752A1 AUDIO DECODING DEVICE AND AUDIO DECODING METHOD 有权
标题翻译：音频解码设备和音频解码方法
公开(公告)号：US20140029752A1
公开(公告)日：2014-01-30
申请号：US13904165
申请日：2013-05-29
申请人： Fujitsu Limited
发明人： Yohei KISHI , Akira Kamano , Shunsuke Takeuchi , Miyuki Shirakawa , Masanao Suzuki
IPC分类号： G10L19/008 , G10L19/04
CPC分类号： G10L19/008 , G10L19/0204 , G10L19/04 , G10L25/12
摘要： An audio decoding device includes a processor; and a memory which stores a plurality of instructions, which when executed by the processor, cause the processor to execute, decoding, using a first channel signal and a second channel signal included in a plurality of channels of an audio signal having a first frequency range and a second frequency range, a first prediction coefficient of the first frequency range and a second prediction coefficient of the second frequency range, both selected from a code book when prediction-encoding a third channel signal that is not subjected to prediction encoding and that is included in the plurality of channels; decoding a residual signal included in the first frequency range, the residual signal representing an error occurring in prediction encoding; and prediction-decoding the third channel signal subjected to prediction-encoding in the second frequency range from the first channel signal, the second channel signal.
摘要翻译：音频解码装置包括处理器; 以及存储器，其存储多个指令，所述指令在由所述处理器执行时使所述处理器执行，解码使用包括在具有第一频率范围的音频信号的多个通道中的第一通道信号和第二通道信号以及第二频率范围，第一频率范围的第一预测系数和第二频率范围的第二预测系数，两者都是从对未进行预测编码的第三频道信号进行预测编码时的码本中选择的，包括在多个通道中; 对包含在第一频率范围内的残差信号进行解码，残差信号表示在预测编码中发生的误差; 并从第一信道信号，第二信道信号预测解码在第二频率范围内进行预测编码的第三信道信号。

63. 发明申请

US20140019143A1 METHOD AND AN APPARATUS FOR PROCESSING AN AUDIO SIGNAL 有权
公开(公告)号：US20140019143A1
公开(公告)日：2014-01-16
申请号：US13960467
申请日：2013-08-06
申请人： Industry-Academic Cooperation Foundation, Yonsei University
发明人： Hyen-O Oh , Chang Heon Lee , Hong Goo Kang , Jung Wook Song
IPC分类号： G10L19/04
CPC分类号： G10L19/00 , G10L19/022 , G10L19/04 , G10L19/167 , G10L19/18 , G11B20/00007 , G11B2020/00028 , G11B2020/10546
摘要： An apparatus for processing an audio signal and method thereof are disclosed. The present invention includes receiving, by an audio processing apparatus, an audio signal including a first data of a first block encoded with rectangular coding scheme and a second data of a second block encoded with non-rectangular coding scheme; receiving a compensation signal corresponding to the second block; estimating a prediction of an aliasing part using the first data; and, obtaining a reconstructed signal for the second block based on the second data, the compensation signal and the prediction of aliasing part.

64. 发明申请

US20130317833A1 Methods and Systems for Generating Filter Coefficients and Configuring Filters 有权
标题翻译：用于生成滤波器系数和配置滤波器的方法和系统
公开(公告)号：US20130317833A1
公开(公告)日：2013-11-28
申请号：US13983892
申请日：2012-02-08
申请人： Mark F. Davis
发明人： Mark F. Davis
IPC分类号： G10L19/04
CPC分类号： G10L19/04 , G10L19/0017
摘要： Methods for generating a palette of feedback (IIR) filter coefficient sets and using the palette to configure (e.g., adaptively update) a prediction filter which includes a feedback filter, and a system for performing any of the methods. Examples of the system include an encoder, including a prediction filter and configured to encode data indicative of a waveform signal (e.g., samples of an audio signal), and a decoder. In some embodiments, the prediction filter is included in an encoder operable to generate (and assert to a decoder) encoded data including filter coefficient data indicative of the selected IIR coefficient set with which the prediction filter was configured during generation of the encoded data. In some embodiments, the timing with which adaptive updating of prediction filter configuration occurs or is allowed to occur is constrained (e.g., to optimize efficiency of prediction encoding).
摘要翻译：用于生成反馈调色板（IIR）滤波器系数组并使用调色板来配置（例如，自适应地更新）包括反馈滤波器的预测滤波器的方法和用于执行任何方法的系统的方法。系统的示例包括编码器，包括预测滤波器并且被配置为对表示波形信号（例如，音频信号的样本）的数据和解码器进行编码。在一些实施例中，预测滤波器被包括在可操作以生成（并且向解码器断言）编码数据的编码数据中，所述编码数据包括指示在生成编码数据期间配置了预测滤波器的所选择的IIR系数组的滤波器系数数据。在一些实施例中，发生或允许发生预测过滤器配置的自适应更新的定时被限制（例如，以优化预测编码的效率）。

65. 发明申请

US20130246073A1 CODING DEVICE, CODING METHOD, DECODING DEVICE, DECODING METHOD, AND STORAGE MEDIUM 有权
标题翻译：编码设备，编码方法，解码设备，解码方法和存储介质
公开(公告)号：US20130246073A1
公开(公告)日：2013-09-19
申请号：US13727370
申请日：2012-12-26
申请人： CASIO COMPUTER CO., LTD.
发明人： Goro SAKATA
IPC分类号： G10L19/04
CPC分类号： G10L19/04 , G10H1/0041 , G10H7/002 , G10H2230/041 , G10H2250/601 , G10L19/08 , H03M7/30
摘要： For respective sampling data of waveform data of sounds to be coded, a prediction residual value is calculated as sampling residual data, and an effective bit length is calculated from this residual waveform data. Then, for the effective bit length data, a maximum effective bit length among processing targets is generated as common effective actual data, and coded data in which this common effective actual data and information indicating the common effective bit length are arranged in a predetermined configuration format are generated. The information included in the coded data is analyzed and each of the plurality of the common effective bit information is extracted. Then, waveform data of the sounds are decoded by performing inverse linear prediction processing from an analysis result on the residual waveform data decompressed by performing bit extension which adds a portion other than the common effective bit length.
摘要翻译：对于要编码的声音的波形数据的各个采样数据，计算预测残差值作为采样残差数据，并根据该残差波形数据计算有效位长度。然后，对于有效比特长度数据，生成作为公共有效实际数据的处理对象之间的最大有效比特长度，以及其中表示公共有效比特长度的公共有效实际数据和信息以预定配置格式排列的编码数据被生成。分析包含在编码数据中的信息，并且提取多个公共有效位信息中的每一个。然后，通过对通过添加除了公共有效位长度以外的部分的比特扩展进行解压缩的残差波形数据的分析结果进行逆线性预测处理来解码声音的波形数据。

66. 发明授权

US08438030B2 Automated distortion classification 有权
标题翻译：自动失真分类
公开(公告)号：US08438030B2
公开(公告)日：2013-05-07
申请号：US12626101
申请日：2009-11-25
申请人： Gaurav Talwar , Rathinavelu Chengalvarayan
发明人： Gaurav Talwar , Rathinavelu Chengalvarayan
IPC分类号： G10L19/04
CPC分类号： G10L17/26 , G10L15/20 , G10L21/0208
摘要： A method of and system for automated distortion classification. The method includes steps of (a) receiving audio including a user speech signal and at least some distortion associated with the signal; (b) pre-processing the received audio to generate acoustic feature vectors; (c) decoding the generated acoustic feature vectors to produce a plurality of hypotheses for the distortion; and (d) post-processing the plurality of hypotheses to identify at least one distortion hypothesis of the plurality of hypotheses as the received distortion. The system can include one or more distortion models including distortion-related acoustic features representative of various types of distortion and used by a decoder to compare the acoustic feature vectors with the distortion-related acoustic features to produce the plurality of hypotheses for the distortion.
摘要翻译：一种自动失真分类的方法和系统。该方法包括以下步骤：（a）接收包括用户语音信号的音频和至少一些与该信号相关的失真; （b）预处理所接收的音频以产生声学特征向量; （c）对生成的声学特征向量进行解码以产生用于失真的多个假设; 以及（d）后处理所述多个假设以将所述多个假设中的至少一个失真假设识别为接收到的失真。该系统可以包括一个或多个失真模型，包括代表各种类型的失真的失真相关的声学特征，并被解码器用于将声学特征向量与失真相关的声学特征进行比较，以产生用于失真的多个假设。

67. 发明授权

US08433563B2 Predictive speech signal coding 有权
标题翻译：预测语音信号编码
公开(公告)号：US08433563B2
公开(公告)日：2013-04-30
申请号：US12455478
申请日：2009-06-02
申请人： Koen Bernard Vos , Soren Skak Jensen
发明人： Koen Bernard Vos , Soren Skak Jensen
IPC分类号： G10L19/04 , G10L19/08
CPC分类号： G10L19/12 , G10L19/09
摘要： A method, system and computer program for encoding speech according to a source-filter model. The method comprises deriving a spectral envelope signal representative of a modelled filter and a first remaining signal representative of a modelled source signal, and deriving a second remaining signal from the first remaining signal by, at intervals during the encoding: exploiting a correlation between approximately periodic portions in the first remaining signal to generate a predicted version of a later portion from a stored version of an earlier portion, and using the predicted-version of the later portion to remove an effect of said periodicity from the first remaining signal. The method further comprises, once every number of intervals, transforming the stored version of the earlier portion of the first remaining signal prior to generating the predicted version of the respective later portion.
摘要翻译：一种根据源滤波器模型对语音进行编码的方法，系统和计算机程序。该方法包括导出代表建模过滤器的频谱包络信号和表示建模源信号的第一剩余信号，以及在编码期间间隔从第一剩余信号导出第二剩余信号：利用近似周期性的相关性第一剩余信号中的部分，以从早期部分的存储版本生成稍后部分的预测版本，并且使用后面部分的预测版本来从第一剩余信号中去除所述周期性的影响。该方法还包括：每产生一次间隔之后，在生成相应较后部分的预测版本之前变换第一剩余信号的较早部分的存储版本。

68. 发明授权

US08428943B2 Quantization matrices for digital audio 有权
公开(公告)号：US08428943B2
公开(公告)日：2013-04-23
申请号：US13046530
申请日：2011-03-11
申请人： Wei-Ge Chen , Naveen Thumpudi , Ming-Chieh Lee
发明人： Wei-Ge Chen , Naveen Thumpudi , Ming-Chieh Lee
IPC分类号： G10L19/02 , G10L19/04
CPC分类号： G10L19/008 , G10L19/02 , G10L19/0204
摘要： Quantization matrices facilitate digital audio encoding and decoding. An audio encoder generates and compresses quantization matrices; an audio decoder decompresses and applies the quantization matrices. The invention includes several techniques and tools, which can be used in combination or separately. For example, the audio encoder can generate quantization matrices from critical band patterns for blocks of audio data. The encoder can compute the quantization matrices directly from the critical band patterns, which can be computed from the same audio data that is being compressed. The audio encoder/decoder can use different modes for generating/applying quantization matrices depending on the coding channel mode of multi-channel audio data. The audio encoder/decoder can use different compression/decompression modes for the quantization matrices, including a parametric compression/decompression mode.

69. 发明申请

US20130096928A1 METHOD AND APPARATUS FOR PROCESSING AN AUDIO SIGNAL 有权
标题翻译：用于处理音频信号的方法和装置
公开(公告)号：US20130096928A1
公开(公告)日：2013-04-18
申请号：US13636922
申请日：2011-03-23
申请人： Gyuhyeok Jeong , Daehwan Kim , Changheon Lee , Lagyoung Kim , Hyejeong Jeon , Byungsuk Lee , Ingyu Kang
发明人： Gyuhyeok Jeong , Daehwan Kim , Changheon Lee , Lagyoung Kim , Hyejeong Jeon , Byungsuk Lee , Ingyu Kang
IPC分类号： G10L19/04
CPC分类号： G10L19/04 , G10L19/06 , G10L19/22 , G10L19/24
摘要： The present invention relates to a method for processing an audio signal, comprising: determining bandwidth information indicating to which of a plurality of bands the current frame corresponds; determining information on the order corresponding to the present frame on the basis of the bandwidth information; performing a linear predictive analysis of the present frame to generate a first set linear predictive transform coefficient of a first order; performing a vector quantization on the first set linear predictive coefficient to generate a first index; performing a linear predictive analysis of the current frame to generate a second set linear predictive transform coefficient of a second order in accordance with the information on the order; and performing a vector quantization on a second set difference by using the first set index and the second set linear predictive transform coefficient, when the second set linear predictive coefficient is generated.
摘要翻译：本发明涉及一种用于处理音频信号的方法，包括：确定指示当前帧对应于多个频带中的哪个频带的带宽信息; 基于所述带宽信息确定与所述当前帧相对应的顺序的信息; 对当前帧执行线性预测分析以产生第一阶的第一组线性预测变换系数; 对所述第一组线性预测系数执行矢量量化以产生第一指标; 执行当前帧的线性预测分析，以根据关于订单的信息生成二阶的第二组线性预测变换系数; 以及当产生所述第二组线性预测系数时，通过使用所述第一设定索引和所述第二组线性预测变换系数，对第二设定差执行向量量化。

70. 发明授权

US08392187B2 Dynamic pruning for automatic speech recognition 有权
标题翻译：动态修剪自动语音识别
公开(公告)号：US08392187B2
公开(公告)日：2013-03-05
申请号：US12362668
申请日：2009-01-30
申请人： Qifeng Zhu
发明人： Qifeng Zhu
IPC分类号： G10L19/04
CPC分类号： G10L15/083
摘要： Methods, speech recognition systems, and computer readable media are provided that recognize speech using dynamic pruning techniques. A search network is expanded based on a frame from a speech signal, a best hypothesis is determined in the search network, a default beam threshold is modified, and the search network is pruned using the modified beam threshold. The search network may be further pruned based on the search depth of the best hypothesis and/or the average number of frames per state for a search path.
摘要翻译：提供了使用动态修剪技术来识别语音的方法，语音识别系统和计算机可读介质。基于来自语音信号的帧扩展搜索网络，在搜索网络中确定最佳假设，修改默认波束阈值，并且使用修改的波束阈值修剪搜索网络。可以基于搜索路径的最佳假设的搜索深度和/或每个状态的平均帧数来进一步修剪搜索网络。

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