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41. 发明公开

EP2154680A3 Method and apparatus for speech coding 有权
标题翻译：用于语音编码的方法和设备
公开(公告)号：EP2154680A3
公开(公告)日：2011-12-21
申请号：EP09014423.9
申请日：1998-12-07
申请人： Mitsubishi Electric Corporation
发明人： Yamaura, Tadashi
IPC分类号： G10L19/10 , G10L19/12
CPC分类号： G10L19/135 , G10L13/02 , G10L19/012 , G10L19/06 , G10L19/083 , G10L19/09 , G10L19/107 , G10L19/12 , G10L19/125 , G10L19/18 , G10L21/0264 , G10L25/93 , G10L2019/0002 , G10L2019/0005 , G10L2019/0007 , G10L2019/0011 , G10L2019/0012 , G10L2019/0016
摘要： A speech encoding method and apparatus for encoding a speech according to code-excited linear prediction CELP, comprising:
analyzing the speech to obtain a linear prediction parameter;
obtaining a linear prediction parameter code by encoding the linear prediction parameter;
obtaining an adaptive code vector concerning an adaptive code from an adaptive codebook;
obtaining pitch information corresponding to the adaptive code; evaluating a noise level of the speech based on the pitch information; obtaining a weight based on the evaluated noise level;
obtaining an excitation code by comparing a coded speech and the speech, wherein the coded speech is obtained by using the adaptive code vector, an excitation codebook and the weight; and outputting a speech code including the adaptive code, the linear prediction parameter code, and the excitation code.
摘要翻译：一种语音编码方法和装置，用于根据码激励线性预测CELP对语音进行编码，包括：分析语音以获得线性预测参数; 通过对线性预测参数进行编码来获得线性预测参数代码; 从自适应码本获得关于自适应码的自适应码矢量; 获得对应于自适应码的音调信息; 基于音高信息评估语音的噪声水平; 基于评估的噪音水平获得重量; 通过比较编码语音和语音获得激励码，其中通过使用自适应码矢量，激励码本和加权获得编码语音; 并输出包括自适应码，线性预测参数码和激励码的语音码。

42. 发明公开

EP2187390A1 Speech signal decoding 有权
标题翻译：语音信号解码
公开(公告)号：EP2187390A1
公开(公告)日：2010-05-19
申请号：EP10152526.9
申请日：2000-10-31
申请人： NEC CORPORATION
发明人： MURASHIMA, Atsushi
IPC分类号： G10L21/02 , G10L19/14
CPC分类号： G10L19/083 , G10L21/0364 , G10L2019/0012
摘要： The quality of reconstructed speech on which background noise is superimposed is improved in a speech signal decoding apparatus for generating a speech signal by driving a filter, which is constituted by linear prediction coefficients, by an excitation signal. A smoothing circuit smoothes sound source gain in a noise segment using sound source gain that was obtained in the past. A smoothing-quantity limiting circuit calculates an amount of fluctuation represented by dividing, by the sound source gain, the absolute value of the difference between the sound source gain and the sound source gain that has been smoothed, and limits the value of the smoothed gain in such a manner that the amount of fluctuation will not exceed a certain threshold value.
摘要翻译：在用于通过激励信号驱动由线性预测系数构成的滤波器来产生语音信号的语音信号解码装置中改善了叠加有背景噪声的重构语音的质量。平滑电路使用过去获得的声源增益来平滑噪声段中的声源增益。平滑量限制电路计算通过将声源增益除以声源增益和已平滑的声源增益之间的差的绝对值表示的波动量，并且限制平滑增益的值以这种方式波动量不会超过某个阈值。

43. 发明公开

EP2154681A2 Method and apparatus for speech decoding 审中-公开
标题翻译： Verfahren und Vorrichtung zur Sprachdekodierung
公开(公告)号：EP2154681A2
公开(公告)日：2010-02-17
申请号：EP09014424.7
申请日：1998-12-07
申请人： Mitsubishi Denki Kabushiki Kaisha
发明人： Yamaura, Tadashi
IPC分类号： G10L19/10
CPC分类号： G10L19/135 , G10L13/02 , G10L19/012 , G10L19/06 , G10L19/083 , G10L19/09 , G10L19/107 , G10L19/12 , G10L19/125 , G10L19/18 , G10L21/0264 , G10L25/93 , G10L2019/0002 , G10L2019/0005 , G10L2019/0007 , G10L2019/0011 , G10L2019/0012 , G10L2019/0016
摘要： A high quality speech is reproduced with a small data amount in speech coding and decoding for performing compression coding and decoding of a speech signal to a digital signal.
In speech decoding method according to a code-excited linear prediction (CELP) speech, a noise level of a speech in a concerning decoding period is evaluated by using a decoded gain of the speech based on a gain code included in the coded speech.
摘要翻译：在用于对语音信号进行数字信号的压缩编码和解码的语音编码和解码中以小的数据量再现高质量语音。在根据码激励线性预测（CELP）语音的语音解码方法中，通过使用基于包括在编码语音中的增益码的语音的解码增益来评估相关解码周期中的语音的噪声电平。

44. 发明授权

EP1480201B1 REDUCTION OF QUANTIZATION-INDUCED BLOCK-DISCONTINUITIES IN AN AUDIO CODER 有权
标题翻译：异径QUANTIZE由数据块中断，音频编码器
公开(公告)号：EP1480201B1
公开(公告)日：2009-03-11
申请号：EP04076676.8
申请日：2000-05-25
申请人： America Online, Inc.
发明人： Wu, Shuwu , Mantegna, John , Perlmutter, Keren
IPC分类号： G10L19/02 , G06F17/14
CPC分类号： G10L19/00 , G10L19/0212 , G10L19/022 , G10L19/028 , G10L19/038 , G10L2019/0012
摘要： A method and system for reduction of quantization-induced block-discontinuities arising from lossy compression and decompression of continuous signals, especially audio signals. One embodiment encompasses a general purpose, ultra-low latency, efficient audio codec algorithm. More particularly, the invention includes a method and apparatus for compression and decompression of audio signals using a novel boundary analysis and synthesis framework to substantially reduce quantization-induced frame or block-discontinuity; a novel adaptive cosine packet transform (ACPT) as the transform of choice to effectively capture the input audio characteristics; a signal-residue classifier to separate the strong signal clusters form the noise and weak signal components (collectively called residue); and adaptive sparse vector quantization (ASVQ) algorithm for signal components; a stochastic noise model for the residue; and an associated rate control algorithm. The invention further includes corresponding computer program implementations of these and other algorithms.

45. 发明授权

EP1596368B1 Method and apparatus for speech decoding 有权
标题翻译：用于解码语音信号的方法和装置
公开(公告)号：EP1596368B1
公开(公告)日：2007-05-23
申请号：EP05015793.2
申请日：1998-12-07
申请人： MITSUBISHI DENKI KABUSHIKI KAISHA
发明人： Yamaura, Tadashi, Mitsubishi Denki K.K.
IPC分类号： G10L19/10
CPC分类号： G10L19/135 , G10L13/02 , G10L19/012 , G10L19/06 , G10L19/083 , G10L19/09 , G10L19/107 , G10L19/12 , G10L19/125 , G10L19/18 , G10L21/0264 , G10L25/93 , G10L2019/0002 , G10L2019/0005 , G10L2019/0007 , G10L2019/0011 , G10L2019/0012 , G10L2019/0016
摘要： A high quality speech is reproduced with a small data amount in speech coding and decoding for performing compression coding and decoding of a speech signal to a digital signal. In speech decoding method according to a code-excited linear prediction (CELP) speech, a noise level of a speech in a concerning decoding period is evaluated by using a decoded gain of the speech based on a gain code included in the coded speech.

46. 发明公开

EP1688918A1 Speech decoding 审中-公开
标题翻译：语音解码
公开(公告)号：EP1688918A1
公开(公告)日：2006-08-09
申请号：EP06112720.5
申请日：2000-09-08
申请人： NEC CORPORATION
发明人： Murashima, Atsushi
IPC分类号： G10L19/14 , G10L21/02
CPC分类号： G10L19/083 , G10L21/0364 , G10L2019/0012
摘要： A method of decoding a speech signal based on a CELP (Code Excited Linear Prediction) with improvement in degradation of decoded sound quality in a noise period. The method includes the steps of: calculating a norm of an excitation vector for each fixed period in a noise period; smoothing the calculated norm using a norm obtained in a previous period; changing the amplitude of the excitation vector in the period using the calculated norm and the smoothed norm; and driving a synthesizing filter by the excitation vector with the changed amplitude.
摘要翻译：一种基于CELP（码激励线性预测）对语音信号进行解码的方法，其中噪声周期中解码音质的劣化得到改善。该方法包括以下步骤：在噪声周期中计算每个固定周期的激励向量的范数; 使用在前一时期获得的范数平滑计算的范数; 使用所计算的范数和平滑范数来改变该周期中的激励向量的幅度; 并且通过具有改变的幅度的激励矢量来驱动合成滤波器。

47. 发明公开

EP1577881A2 A speech communication system and method for handling lost frames 审中-公开
标题翻译： Sprachübertragungssystemund Verfahren zur Behandlung verlorener Datenrahmen
公开(公告)号：EP1577881A2
公开(公告)日：2005-09-21
申请号：EP05012550.9
申请日：2001-07-09
申请人： Mindspeed Technologies, Inc.
发明人： Benyassine, Adil c/oConexant Systems, Inc. , Shlomot, Eyal , Su, Huan-Yu
IPC分类号： G10L19/06 , G10L19/00 , G10L19/14
CPC分类号： G10L19/08 , G10L19/005 , G10L19/07 , G10L19/083 , G10L25/90 , G10L2019/0012
摘要： A speech communication system and method that has an improved way of handling information lost during transmission from the encoder to the decoder. If the speech communication system does not transmit fixed codebook excitation values to the decoder, the improved encoder/decoder generates the same random excitation values for a given frame by using a seed whose value is determined by information in that frame.
摘要翻译：一种语音通信系统和方法，其具有处理从编码器到解码器的传输期间丢失的信息的改进方式。如果语音通信系统不向解码器发送固定码本激励值，则改进的编码器/解码器通过使用其值由该帧中的信息确定的种子为给定帧生成相同的随机激励值。

48. 发明公开

EP1199710A1 DEVICE FOR ENCODING/DECODING VOICE AND FOR VOICELESS ENCODING, DECODING METHOD, AND RECORDED MEDIUM ON WHICH PROGRAM IS RECORDED 有权
标题翻译：设备，方法和存储介质具有程序DECODE清音股份
公开(公告)号：EP1199710A1
公开(公告)日：2002-04-24
申请号：EP00931614.2
申请日：2000-05-31
申请人： NEC Corporation
发明人： SERIZAWA, Masahiro , ITO, Hironori
IPC分类号： G10L19/00 , H04B14/04
CPC分类号： G10L19/012 , G10L2019/0012
摘要： A speech decoding device of the invention smoothes, in decoding speech signal in a voice-less period, RMS and filter coefficients which is discontinuously transmitted, and provides them to a synthesis filter. Thereby, it is capable of preventing discontinuous changing of the filter coefficient caused by the intermittent transmission of the filter coefficient. As a result, a quality of decoding can be improved. Also, to remove an effect, caused by the smoothing process, from the filter coefficients or the RMS which are transmitted in the past frames, a smoothing factor is adjusted not to perform smoothing while a certain time period (or a certain number of frames) from when a transition is made from a voice period from a voice-less period, or when a decoded feature parameter satisfies a predetermined condition.
摘要翻译：在语音期间较少发明平滑的一种语音解码装置，在解码语音信号，RMS和滤波器系数的所有这是不连续的反式mitted，并将它们提供给一个合成滤波器。由此，能够防止由过滤器系数的间歇发送引起的滤波器系数的不连续变化的。其结果是，解码的质量可以被改善。因此，要除去到效果，通过平滑化处理引起的，从滤波器系数或哪些是在过去的帧反式mitted的RMS，平滑因子被调节为不进行平滑化，同时在一定的时间段（或帧的特定数量的）当从由语音时段由从语音较少期间，或当一个解码的特征参数SATIS FIES预定条件的转变。

49. 发明公开

EP1181686A1 REDUCTION OF QUANTIZATION-INDUCED BLOCK-DISCONTINUITIES IN AN AUDIO CODER 有权
标题翻译： THE QUANTIZE的减少造成的数据块的不连续性音频编码器
公开(公告)号：EP1181686A1
公开(公告)日：2002-02-27
申请号：EP00936311.0
申请日：2000-05-25
申请人： America Online, Inc.
发明人： WU, Shuwu , MANTEGNA, John , PERLMUTTER, Keren
IPC分类号： G10L19/02
CPC分类号： G10L19/00 , G10L19/0212 , G10L19/022 , G10L19/028 , G10L19/038 , G10L2019/0012
摘要： A method and system for reduction of quantization-induced block-discontinuities arising from lossy compression and decompression of continuous signals, especially audio signals. One embodiment encompasses a general purpose, ultra-low latency, efficient audio codec algorithm. More particularly, the invention includes a method and apparatus for compression and decompression of audio signals using a novel boundary analysis and synthesis framework to substantially reduce quantization-induced frame or block-discontinuity; a novel adaptive cosine packet transform (ACPT) as the transform of choice to effectively capture the input audio characteristics; a signal-residue classifier to separate the strong signal clusters form the noise and weak signal components (collectively called residue); and adaptive sparse vector quantization (ASVQ) algorithm for signal components; a stochastic noise model for the residue; and an associated rate control algorithm. The invention further includes corresponding computer program implementations of these and other algorithms.

50. 发明公开

EP1052620A1 SOUND ENCODING METHOD AND SOUND DECODING METHOD, AND SOUND ENCODING DEVICE AND SOUND DECODING DEVICE 有权
标题翻译：音乐会 - 德克萨斯州 - 弗罗伦萨
公开(公告)号：EP1052620A1
公开(公告)日：2000-11-15
申请号：EP98957197.1
申请日：1998-12-07
申请人： MITSUBISHI DENKI KABUSHIKI KAISHA
发明人： YAMAURA, Tadashi
IPC分类号： G10L9/14 , H03M7/30 , H04B14/04
CPC分类号： G10L19/135 , G10L13/02 , G10L19/012 , G10L19/06 , G10L19/083 , G10L19/09 , G10L19/107 , G10L19/12 , G10L19/125 , G10L19/18 , G10L21/0264 , G10L25/93 , G10L2019/0002 , G10L2019/0005 , G10L2019/0007 , G10L2019/0011 , G10L2019/0012 , G10L2019/0016
摘要： A high quality speech is reproduced with a small data amount in speech coding and decoding for performing compression coding and decoding of a speech signal to a digital signal.
In speech coding method according to a code-excited linear prediction (CELP) speech coding, a noise level of a speech in a concerning coding period is evaluated by using a code or coding result of at least one of spectrum information, power information, and pitch information, and various excitation codebooks are used based on an evaluation result
摘要翻译：在用于对语音信号进行数字信号的压缩编码和解码的语音编码和解码中以小的数据量再现高质量语音。在根据码激励线性预测（CELP）语音编码的语音编码方法中，通过使用频谱信息，功率信息和频谱信息中的至少一个的代码或编码结果来评估相关编码周期中的语音的噪声电平基于评估结果使用音调信息和各种激励码本

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