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71. 发明申请

US20060235680A1 Apparatus, method and computer program product for processing acoustical-signal 有权
标题翻译：用于处理声信号的装置，方法和计算机程序产品
公开(公告)号：US20060235680A1
公开(公告)日：2006-10-19
申请号：US11376130
申请日：2006-03-16
申请人： Koichi Yamamoto , Akinori Kawamura
发明人： Koichi Yamamoto , Akinori Kawamura
IPC分类号： G10L11/04
CPC分类号： G10L19/008 , G10L21/04
摘要： An acoustical-signal processing apparatus includes a feature extracting unit that extracts feature data common to each channel signal which forms a multichannel acoustical signal, based on a composite similarity obtained by combining similarities calculated from each channel signal; and a time-base companding unit that executes time compression and time expansion of the multichannel acoustical signal based on the extracted feature data.
摘要翻译：声信号处理装置包括特征提取单元，其基于通过组合从每个信道信号计算的相似度而获得的复合相似度，提取形成多声道声信号的每个信道信号共同的特征数据; 以及时基压缩单元，其基于所提取的特征数据执行多声道声信号的时间压缩和时间扩展。

72. 发明授权

US07092774B1 Multi-channel audio processing system with real-time program duration alteration 有权
标题翻译：具有实时程序持续时间变化的多声道音频处理系统
公开(公告)号：US07092774B1
公开(公告)日：2006-08-15
申请号：US09516303
申请日：2000-02-29
申请人： Christopher Scott Gifford , Leonard Keith Moeller
发明人： Christopher Scott Gifford , Leonard Keith Moeller
IPC分类号： G06F17/00 , G10L21/00
CPC分类号： G10L21/04 , G10L19/008 , G11B20/00007 , G11B20/10527 , G11B27/034 , G11B27/28 , G11B2020/10601 , H04H60/04 , H04H60/07
摘要： The total running time of an original multi-channel program signal is altered to generate a time-shortened (or time-lengthened) program signal. The original program signal may be thought of subdividable into a sequence of program signal portions, each program signal portion being further subdividable into a sequence of signal windows. Differencing circuitry determines, for each program siganl portion, a difference value indicative of a best differnce match between an initial signal window in that signal portion and subsequent signal windows in that signal portion in accordance with a predefined criterion. Removal circuitry then deletes from the original multi-channel program signal a multi-window segment of that signal portion, the deleted segment beginning with the initial signal window and ending with the subsequent signal window that generated the best difference match.
摘要翻译：改变原始多通道节目信号的总运行时间以产生缩短时间（或延长时间）的节目信号。原始程序信号可被认为可细分为一系列程序信号部分，每个程序信号部分进一步细分为一系列信号窗口。差分电路针对每个节目进口部分确定指示该信号部分中的初始信号窗口与该信号部分中的后续信号窗口之间根据预定准则的最佳差异匹配的差值。去除电路然后从原始多通道程序信号中删除该信号部分的多窗口段，该删除段从初始信号窗口开始，并且随后产生最佳差异匹配的信号窗口结束。

73. 发明授权

US07082220B2 Data processing apparatus 失效
公开(公告)号：US07082220B2
公开(公告)日：2006-07-25
申请号：US10239805
申请日：2002-01-24
申请人： Tetsujiro Kondo , Tsutomu Watanabe
发明人： Tetsujiro Kondo , Tsutomu Watanabe
IPC分类号： G06K9/36 , G06K9/48
CPC分类号： G10L21/04 , G10L19/26 , G10L25/12 , H04N19/105 , H04N19/117 , H04N19/122 , H04N19/124 , H04N19/13 , H04N19/137 , H04N19/14 , H04N19/147 , H04N19/176 , H04N19/18 , H04N19/192 , H04N19/40 , H04N19/44 , H04N19/46 , H04N19/60 , H04N19/61 , H04N19/80 , H04N19/85 , H04N19/98
摘要： The present invention relates to data processing apparatuses which improve the quality of decoded data, such as an image and sound, more. A decoding section 21 decodes, for example, encoded data obtained by JPEG-encoding an image, and an encoding section 22 encodes first decoded data obtained as a result to output re-encoded data. Further, a decoding section 23 decodes the re-encoded data to obtain second decoded data. A class-classification adaptive processing circuit 24 generates a prediction tap used for a predetermined prediction calculation with tap coefficients obtained by learning, from the first decoded data and the second decoded data, and performs the predetermined prediction calculation by using the prediction tap and the tap coefficients to obtain a prediction value corresponding to master data used as a master in the learning. The present invention can be applied to transfer systems for encoding images and sound to transfer them.

74. 发明授权

US06999922B2 Synchronization and overlap method and system for single buffer speech compression and expansion 失效
标题翻译：用于单缓冲语音压缩和扩展的同步和重叠方法和系统
公开(公告)号：US06999922B2
公开(公告)日：2006-02-14
申请号：US10607639
申请日：2003-06-27
申请人： Marc Andre Boillot , John Gregory Harris , Thomas Lawrence Reinke
发明人： Marc Andre Boillot , John Gregory Harris , Thomas Lawrence Reinke
IPC分类号： G10L19/00 , G10L13/06
CPC分类号： G10L21/04
摘要： The present invention (110) permits a user to speed up and slow down speech without changing the speakers pitch (102, 110, 112, 128, 402–416). It is a user adjustable feature to change the spoken rate to the listeners' preferred listening rate or comfort. It can be included on the phone as a customer convenience feature without changing any characteristics of the speakers voice besides the speaking rate with soft key button (202) combinations (in interconnect or normal). From the users perspective, it would seem only that the talker changed his speaking rate, and not that the speech was digitally altered in any way. The pitch and general prosody of the speaker are preserved. The following uses of the time expansion/compression feature are listed to compliment already existing technologies or applications in progress including messaging services, messaging applications and games, real-time feature to slow down the listening rate.
摘要翻译：本发明（110）允许用户在不改变扬声器音高（102,110,112,128,402-416）的情况下加速和减慢语音。这是一个用户可调整的功能，可以将听众的倾听率或舒适度改变为口语速率。除了使用软按键（202）组合（互连或正常）组合的说话率外，它可以作为客户便利功能包括在手机中，而不会改变扬声器声音的任何特性。从用户的角度来看，谈话者似乎只是改变了他的发言率，而不是以任何方式对演讲进行数字化改变。扬声器的音调和广泛韵律得以保留。列出了以下使用时间扩展/压缩功能来补充已经存在的技术或正在进行的应用程序，包括消息传递服务，消息传递应用程序和游戏，实时功能可以降低收听速度。

75. 发明申请

US20060004578A1 Method for controlling duration in speech synthesis 有权
标题翻译：用于控制语音合成中的持续时间的方法
公开(公告)号：US20060004578A1
公开(公告)日：2006-01-05
申请号：US10527779
申请日：2003-08-05
申请人： Ercan Gigi
发明人： Ercan Gigi
IPC分类号： G10L13/06
CPC分类号： G10L13/07 , G10L21/04
摘要： The present invention relates to a method of synthesizing of a speech signal, comprising: -assigning of a first identifier to a first class of intervals of an original speech signal and assigning of a second identifier to a second class of intervals of the original speech signal, -windowing the original speech signal to provide a number of pitch bells, -processing the pitch bells having the first identifier assigned thereto for modifying a duration of the speech signal, -performing an overlap and add operation on the processed pitch bells.
摘要翻译：本发明涉及一种合成语音信号的方法，包括：将第一标识符分配给原始语音信号的第一类间隔，并将第二标识符分配给原始语音信号的第二类间隔 - 将原始语音信号窗口化以提供多个音调钟， - 处理具有分配给其的第一标识符的音调铃，以修改语音信号的持续时间， - 执行重叠并对处理的音调铃添加操作。

76. 发明授权

US06967599B2 Method of reproducing audio signals without causing tone variation in fast or slow playback mode and reproducing apparatus for the same 失效
标题翻译：在快速或慢速播放模式中再现音频信号而不引起音调变化的方法和用于其的再现装置的方法
公开(公告)号：US06967599B2
公开(公告)日：2005-11-22
申请号：US09871293
申请日：2001-05-30
申请人： Won-Yong Choi , Byoung-Chul Lee , Sang-Hun Jeong , Won-Sik Choi
发明人： Won-Yong Choi , Byoung-Chul Lee , Sang-Hun Jeong , Won-Sik Choi
IPC分类号： G10L21/04 , G11B20/10 , G11B27/00 , H03M7/00
CPC分类号： G10L21/04 , G11B20/10527 , G11B27/005 , G11B2220/2562 , G11B2220/90
摘要： Audio data decoded in an MPEG system to be stored in a storage unit is supplied to an audio output via a filtering processing. For performing the filtering processing, presentation time interval of respective audio data is changed to conform to a user's designated playback speed, and the decoded audio data stored in the storage unit by being synchronized with the changed presentation time interval is written on an input queue in the set unit. A TSM algorithm is performed in the frame unit with respect to the audio data of the input queue to decrease the quantity of the audio data when the designated playback speed is faster than a normal playback speed or to increase it when the designated playback speed is slower than the normal playback speed, in accordance with a value of the designated playback speed. The TSM audio data is transferred to a middle queue. With respect to the audio data of the middle queue, up-sampling or down-sampling is performed in accordance with the value of the designated playback speed. The quantity of the audio data after the sampling becomes substantially the same as that of the decoded audio data, and thus the sampled audio data have a tone substantially identical to that of the normal playback speed and are transmitted to an output queue. The audio data stored in the output queue is synchronized with the changed presentation time interval to be transmitted to the storage in the set unit, and then is reproduced via an audio output.
摘要翻译：要存储在存储单元中的MPEG系统中解码的音频数据经由滤波处理被提供给音频输出。为了执行滤波处理，改变各个音频数据的呈现时间间隔以符合用户指定的播放速度，并且通过与改变的呈现时间间隔同步将存储在存储单元中的解码音频数据写入到输入队列中设定单位。相对于输入队列的音频数据，在帧单元中执行TSM算法，以便当指定的播放速度快于正常重放速度时减小音频数据的数量，或者当指定的播放速度较慢时增加音频数据的数量比正常的播放速度，根据指定的播放速度的值。 TSM音频数据被传输到中间队列。对于中间队列的音频数据，根据指定的播放速度的值执行上采样或下采样。采样后的音频数据量与解码的音频数据基本相同，因此采样的音频数据具有与正常重放速度基本相同的音调，并被发送到输出队列。存储在输出队列中的音频数据与改变的呈现时间间隔同步，以被发送到设置单元中的存储器，然后经由音频输出再现。

77. 发明申请

US20050246170A1 Audio signal processing apparatus and method 审中-公开
标题翻译：音频信号处理装置及方法
公开(公告)号：US20050246170A1
公开(公告)日：2005-11-03
申请号：US10517913
申请日：2003-05-27
申请人： Fabio Vignoli , Tatiana Lashina
发明人： Fabio Vignoli , Tatiana Lashina
IPC分类号： G10L21/04 , G10L21/0364 , G10L21/02
CPC分类号： G10L21/02 , G10L21/04 , G10L2021/03646
摘要： An audio signal processing apparatus (1) comprises an audio input (3) for an entered audio signal, an audio output (5) for outputting an outgoing audio signal, and a processor (9) for performing a transformation (2) to improve the intelligibility of speech present in the entered audio signal. The transformation (2) transforms the entered audio signal into the outgoing audio signal, by modeling at least one aspect of the Lombard effect, based upon a noise level value (7). The Lombard effect is a specific way in which people change their speech, when speaking in noisy environments. The audio signal processing apparatus can be applied in a television receiver and a radio program receiver.
摘要翻译：音频信号处理装置（1）包括用于输入的音频信号的音频输入（3），用于输出输出音频信号的音频输出（5）和用于执行变换（2）的处理器（9）输入音频信号中存在的语音的可懂度。变换（2）通过基于噪声电平值（7）对伦巴底效应的至少一个方面进行建模，将输入的音频信号转换为输出音频信号。当在嘈杂的环境中说话时，伦巴第效应是人们改变言论的一种特定方式。音频信号处理装置可以应用于电视接收机和无线电节目接收机。

78. 发明申请

US20050228672A1 Method and system of dynamically adjusting a speech output rate to match a speech input rate 有权
标题翻译：动态调整语音输出速率以匹配语音输入速率的方法和系统
公开(公告)号：US20050228672A1
公开(公告)日：2005-10-13
申请号：US10815309
申请日：2004-04-01
申请人： James Lewis , Peeyush Jaiswal
发明人： James Lewis , Peeyush Jaiswal
IPC分类号： G10L13/08 , G10L21/04
CPC分类号： G10L21/04
摘要： A method (10) and system of adjusting a speech output rate to match a speech input rate can include the steps of receiving (12) speech input, computing (14) a speech input rate, and dynamically adjusting (18 or 26) a speech output rate to match the speech input rate. If the type of speech output is TTS, then a rate of TTS can be adjusted (18). If the type of speech output is recorded and alternate text is available, then steps (22 and 24) of counting alternate text available from a recorded output and determining an audio file length is used to compute a default output rate to adjust a recorded output rate. If the type is recorded and alternate text is unavailable, then steps (21 and 24) of obtaining an output word count from a transcription of a recorded speech output and determining an audio file length is used.
摘要翻译：一种方法（10）和调整语音输出速率以匹配语音输入速率的系统可以包括以下步骤：接收（12）语音输入，计算（14）语音输入速率，以及动态地调整（18或26）语音输出速率匹配语音输入速率。如果语音输出的类型是TTS，则可以调整TTS的速率（18）。如果语音输出的类型被记录并且替代文本可用，则使用计数从记录输出可用的替代文本并确定音频文件长度的步骤（22和24）来计算默认输出速率以调整记录的输出速率。如果记录类型并且替代文本不可用，则使用从记录的语音输出的转录获得输出字数并确定音频文件长度的步骤（21和24）。

79. 发明申请

US20050227657A1 Method and apparatus for increasing perceived interactivity in communications systems 审中-公开
标题翻译：增加通信系统中感知交互性的方法和装置
公开(公告)号：US20050227657A1
公开(公告)日：2005-10-13
申请号：US10819376
申请日：2004-04-07
申请人： Tomas Frankkila , Jonas Svedberg , Krister Svanbro , Bjorn Svensson , Tomas Jonsson
发明人： Tomas Frankkila , Jonas Svedberg , Krister Svanbro , Bjorn Svensson , Tomas Jonsson
IPC分类号： G10L13/06 , H04L12/56 , H04Q7/38
CPC分类号： G10L21/04
摘要： Perceived interactivity in user communications is achieved by reducing a perceived delay switching the active transmitter in the communication without having to reduce actual transmission and setup delays associated with a communication exchange. A sound signal is identified in the user communication. The sound signal is analyzed to identify or estimate a sound signal segment. The sound signal segment is preferably (though not necessarily) located at the beginning or the end of the sound signal. The sound signal segment may be selected directly from the sound signal itself, from a modified version of the sound signal, or from a signal associated with the sound signal. A determination is made that a length or duration of the sound signal segment should be or can be modified. One or more modifications for the sound signal segment are determined and are provided to one or more processing units to perform the modification(s).
摘要翻译：用户通信中的感知交互性通过减少在通信中切换有源发射机的感知延迟而不用减少与通信交换相关联的实际传输和建立延迟来实现。在用户通信中识别声音信号。分析声音信号以识别或估计声音信号段。声音信号段优选地（尽管不一定）位于声音信号的开始或结束处。可以从声音信号本身，声音信号的修改版本或与声音信号相关联的信号直接选择声音信号段。确定声音信号段的长度或持续时间应该是或可以被修改。确定声音信号段的一个或多个修改，并将其提供给一个或多个处理单元以执行修改。

80. 发明申请

US20050222847A1 System and method for time domain audio slow down, while maintaining pitch 审中-公开
标题翻译：时域音频系统和方法减速，同时保持音高
公开(公告)号：US20050222847A1
公开(公告)日：2005-10-06
申请号：US10803286
申请日：2004-03-18
申请人： Manoj Singhal , Sunoj Koshy , Arun Rao
发明人： Manoj Singhal , Sunoj Koshy , Arun Rao
IPC分类号： G10L21/04
CPC分类号： G10L21/04
摘要： A system and method for slowing down an audio signal while maintaining the same pitch as the original audio signal. The slowing down being done by a decoder. The method involves replicating frames of the decoded signal at a rate corresponding to the desired slow playback speed, and windowing the replicated frames to smooth out any artifacts that may result from the replication. The desired slow playback speed can be a default value predefined in the system or a value programmable by a user of the system.
摘要翻译：一种用于在保持与原始音频信号相同音调的同时减慢音频信号的系统和方法。由解码器完成的减速。该方法涉及以对应于期望的慢回放速度的速率复制解码信号的帧，并且对复制帧进行窗口化以平滑可能由复制产生的任何伪影。期望的慢播放速度可以是在系统中预定义的默认值或由系统的用户可编程的值。

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