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1. 发明授权

US06807528B1 Adding data to a compressed data frame 失效
标题翻译：将数据添加到压缩数据帧
公开(公告)号：US06807528B1
公开(公告)日：2004-10-19
申请号：US09851589
申请日：2001-05-08
申请人： Michael M. Truman , Matthew A. Watson
发明人： Michael M. Truman , Matthew A. Watson
IPC分类号： G10L2104
CPC分类号： G10L19/002
摘要： Many low bit rate digital audio encoding systems, including Dolby Digital and MPEG-2 AAC generate data streams in which unused dummy, fill, stuffing, or null bits exist whenever the bit allocation function in the encoder does not utilize all available bits from a bit pool. Instead, all or some of such wasted bits are used to carry information. This can be accomplished after an encoder generates a bitstream. The resulting bitstream is analyzed to identify the locations of some or all of the unused bits. Some or all of the identified unused bits are then replaced with information-carrying bits to embed information-carrying bits in locations formerly occupied by unused bits. Alternatively, instead of replacing some or all unused bits in the bitstream with information-carrying bits after encoding, a modified encoder may insert information-carrying bits in some or all of the unused bit positions during the encoding process.
摘要翻译：包括杜比数字和MPEG-2 AAC在内的许多低比特率数字音频编码系统产生数据流，其中当编码器中的位分配功能不利用来自一位的所有可用位时，不存在未使用的虚拟，填充，填充或空位游泳池相反，所有或一些这种浪费的比特用于携带信息。这可以在编码器生成比特流之后实现。分析所得比特流以识别部分或全部未使用位的位置。所识别的未使用的位中的一些或全部随后被信息携带位替换，以将信息携带位嵌入以前被未使用位占据的位置。或者，代替在编码之后用信息携带位替换比特流中的一些或所有未使用的比特，修改的编码器可以在编码处理期间将部分或全部未使用的比特位置插入信息携带比特。

2. 发明授权

US06785657B2 Digital signal processor 有权
标题翻译：数字信号处理器
公开(公告)号：US06785657B2
公开(公告)日：2004-08-31
申请号：US09725000
申请日：2000-11-29
申请人： Norio Hatanaka
发明人： Norio Hatanaka
IPC分类号： G10L2104
CPC分类号： H04N5/9265 , G11B27/3027 , G11B2220/2562 , H04N5/4401 , H04N5/602 , H04N5/607 , H04N5/85 , H04N21/42646 , H04N21/4396 , H04N21/44016 , H04N21/482 , H04N21/8106
摘要： A digital signal processor includes a coded audio data generation unit for generating coded audio data; an audio data generation unit for generating audio data; a signal switching unit for outputting one of the coded audio data and the audio data respectively supplied from the coded audio data generation unit and the audio data generation unit, and switching the output between these data; and a signal switching control unit for detecting the periodicity of the coded audio data outputted from the coded audio data generation unit, and controlling the signal switching unit so as to switch the output at the boundary of periods of the coded audio data.
摘要翻译：数字信号处理器包括用于产生编码音频数据的编码音频数据产生单元; 用于产生音频数据的音频数据生成单元; 信号切换单元，用于输出从编码音频数据生成单元和音频数据生成单元分别提供的编码音频数据和音频数据之一，并切换这些数据之间的输出; 以及信号切换控制单元，用于检测从编码音频数据生成单元输出的编码音频数据的周期性，并且控制信号切换单元，以便在编码音频数据的周期的边界处切换输出。

3. 发明授权

US06490553B2 Apparatus and method for controlling rate of playback of audio data 有权
标题翻译：控制音频数据播放速率的装置和方法
公开(公告)号：US06490553B2
公开(公告)日：2002-12-03
申请号：US09781828
申请日：2001-02-12
申请人： Jean-Manuel Van Thong , Davis Pan
发明人： Jean-Manuel Van Thong , Davis Pan
IPC分类号： G10L2104
CPC分类号： H04N5/278 , G10L15/26
摘要： The disclosed method and apparatus controls the rate of playback of audio data corresponding to a stream of speech. Using speech recognition, the rate of speech of the audio data is determined. The determined rate of speech is compared to a target rate. Based on the comparison, the playback rate is adjusted, i.e. increased or decreased, to match the target rate.
摘要翻译：所公开的方法和装置控制对应于语音流的音频数据的重放速率。使用语音识别，确定音频数据的语音速率。将确定的语速与目标速率进行比较。基于比较，调整播放速率，即增加或减少，以匹配目标速率。

4. 发明授权

US06477502B1 Method and apparatus for using non-symmetric speech coders to produce non-symmetric links in a wireless communication system 有权
标题翻译：在无线通信系统中使用非对称语音编码器来产生非对称链路的方法和装置
公开(公告)号：US06477502B1
公开(公告)日：2002-11-05
申请号：US09643360
申请日：2000-08-22
申请人： Ananth Ananthpadmanabhan , Andrew P. DeJaco
发明人： Ananth Ananthpadmanabhan , Andrew P. DeJaco
IPC分类号： G10L2104
CPC分类号： H04W88/181 , G10L19/18
摘要： A method and apparatus for balancing the forward link capacity of a wireless communication system with the reverse link capacity of the system is presented. Speech coders with selectable modes are implemented in both links so that a forward link speech coder will not be operating with a mode set that is identical to the mode set used by a reverse link speech coder. Since the reverse link has a higher user capacity than a forward link, the reverse link speech coder can operate with a higher average data rate. Hence, the mode set used by the reverse link speech coder can be implemented without low average data rate modes. Elimination of modes from the mode set reduces complexity of the speech coder.
摘要翻译：提出了一种用于平衡无线通信系统的前向链路容量与系统的反向链路容量的方法和装置。具有可选模式的语音编码器在两个链路中实现，使得前向链路语音编码器将不以与由反向链路语音编码器使用的模式集相同的模式集来操作。由于反向链路具有比前向链路更高的用户容量，反向链路语音编码器可以以较高的平均数据速率进行操作。因此，可以在没有低平均数据速率模式的情况下实现由反向链路语音编码器使用的模式集合。从模式集中消除模式降低了语音编码器的复杂性。

5. 发明授权

US06230133B1 Home office communication system and method 失效
标题翻译：家庭办公通讯系统及方法
公开(公告)号：US06230133B1
公开(公告)日：2001-05-08
申请号：US09103825
申请日：1998-06-24
申请人： Raymond Walden Bennett, III , Robert Wesley Bossemeyer, Jr. , Wayne Robert Heinmiller , Jordan Howard Light
发明人： Raymond Walden Bennett, III , Robert Wesley Bossemeyer, Jr. , Wayne Robert Heinmiller , Jordan Howard Light
IPC分类号： G10L2104
CPC分类号： H04L12/2878 , H04L12/2856 , H04M1/247 , H04M1/2478 , H04M1/2535 , H04M1/271 , H04M1/723 , H04M3/42314 , H04M3/4234 , H04M3/53333 , H04M7/0033 , H04M7/0054 , H04M7/1205 , H04M11/06 , H04M2250/74
摘要： A system for utilizing multiple simultaneous communication channels has a control and display system (16). A modem (12) is controlled by the control and display system (16) and connected to a telephone line (14). A multiplexer (18) is connected to the modem (12) and multiplexes a plurality of outgoing data streams in a predefined format and demultiplexes an incoming data stream from the modem (12). A speech encoding system (20) generates a plurality of speech packets to form one of the plurality of outgoing data streams. A telephone line control system (22) generates a plurality of control packets forming one of the plurality of outgoing data streams and an information channel (24) forms one of the plurality of outgoing data streams.
摘要翻译：一种用于利用多个同时通信信道的系统具有控制和显示系统（16）。调制解调器（12）由控制和显示系统（16）控制并连接到电话线（14）。多路复用器（18）连接到调制解调器（12），并以预定格式复用多个输出数据流，并从调制解调器（12）解复用输入数据流。语音编码系统（20）生成多个语音分组以形成多个输出数据流中的一个。电话线控制系统（22）产生形成多个输出数据流中的一个的多个控制分组，以及形成多个输出数据流之一的信息信道（24）。

6. 发明授权

US06185525B2 Method and apparatus for digital signal compression without decoding 有权
标题翻译：无需解码的数字信号压缩方法和装置
公开(公告)号：US06185525B2
公开(公告)日：2001-02-06
申请号：US09170744
申请日：1998-10-13
申请人： David B. Taubenheim , Miriam R. Boudreaux , Sunil Satyamurti
发明人： David B. Taubenheim , Miriam R. Boudreaux , Sunil Satyamurti
IPC分类号： G10L2104
CPC分类号： G10L19/087 , G10L19/07
摘要： A method (100) of compressing a digital signal that is parametrically modeled and encoded includes the steps of storing (102) the digital signal in a memory in a plurality of frames having a plurality of parameters in each frame of the plurality of frames, wherein the digital signal was encoded at a higher rate and converting the digital signal to a lower rate by selecting (106) from each frame of the plurality of frames a subset of the plurality of parameters and discarding (108) the subset of the plurality of parameters within each frame of the plurality of frames.
摘要翻译：一种对参数建模和编码的数字信号进行压缩的方法（100）包括以下步骤：将数字信号存储在具有多个帧中的每个帧中的多个参数的多个帧中的存储器中，其中以更高的速率对数字信号进行编码，并通过从多个帧的每个帧中选择（106）多个参数的子集，并将（108）多个参数的子集丢弃（106）来将数字信号转换为较低的速率在多个帧的每个帧内。

7. 发明授权

US06178405B1 Concatenation compression method 失效
标题翻译：串联压缩法
公开(公告)号：US06178405B1
公开(公告)日：2001-01-23
申请号：US08751876
申请日：1996-11-18
申请人： Jing-Zheng Ouyang , Nan-Sheng Lin
发明人： Jing-Zheng Ouyang , Nan-Sheng Lin
IPC分类号： G10L2104
CPC分类号： H03M7/48
摘要： A data signal compression technique for real-time voice and data processing where the digitized signal is first compressed to obtain a first compressed signal, and the first compressed signal is then compressed again to obtain a second compressed signal. Within a digital signal processor, digital signals first undergo time scale compression after which the compressed signals undergo audio compression to achieve multiple-compressed signals. Upon reception in a second digital signal processor, the multiple-compressed signals are correspondingly decompressed to achieve a high-quality estimation of the original digital signals.
摘要翻译：一种用于实时语音和数据处理的数据信号压缩技术，其中首先压缩数字化信号以获得第一压缩信号，然后再次压缩第一压缩信号以获得第二压缩信号。在数字信号处理器中，数字信号首先进行时标压缩，之后压缩信号进行音频压缩以实现多压缩信号。在第二数字信号处理器中接收时，相应地对多个压缩信号进行解压缩，以实现原始数字信号的高质量估计。

8. 发明授权

US06763329B2 Method of converting the speech rate of a speech signal, use of the method, and a device adapted therefor 有权
标题翻译：转换语音信号的语音速率，使用该方法的方法以及适用于其的设备
公开(公告)号：US06763329B2
公开(公告)日：2004-07-13
申请号：US09827195
申请日：2001-04-05
申请人： Cecilia Brandel , Henrik Johannisson
发明人： Cecilia Brandel , Henrik Johannisson
IPC分类号： G10L2104
CPC分类号： G10L21/04
摘要： A method of converting the speech rate of a speech signal with a pitch period below a maximum expected pitch period comprises the steps of dividing the speech signal into segments, estimating the pitch period in a segment, copying a fraction of the speech signal in the segment, said fraction having a duration equal to said estimated pitch period, providing from the fraction an intermediate signal having the same duration, and expanding the segment by inserting the intermediate signal pitch synchronously into the speech signal of the segment. A segment size longer than the maximum expected pitch period but shorter than twice the maximum expected pitch period is used. A considerably smaller amount of data has to be processed for each segment, so that the method can be implemented with the limited computational resources of e.g. a mobile telephone. A similar device is also provided.
摘要翻译：将音频周期的语音信号的语音速率转换为最大预期音调周期的方法包括以下步骤：将语音信号划分成段，估计段中的音调周期，复制段中的语音信号的一部分所述分数具有等于所述估计的音调周期的持续时间，从该分数提供具有相同持续时间的中间信号，并且通过将中间信号间距同步地插入该分段的语音信号来扩展分段。使用比最大预期音调周期长但小于最大预期音调周期的两倍的音调段大小。必须对每个段处理相当少量的数据，使得该方法可以用例如有限的计算资源来实现。一个移动电话。还提供了类似的装置。

9. 发明授权

US06718309B1 Continuously variable time scale modification of digital audio signals 失效
标题翻译：数字音频信号的连续可变时标修改
公开(公告)号：US06718309B1
公开(公告)日：2004-04-06
申请号：US09626046
申请日：2000-07-26
申请人： Roger Selly
发明人： Roger Selly
IPC分类号： G10L2104
CPC分类号： G10L21/01
摘要： A method for time scale modification of a digital audio signal produces an output signal that is at a different playback rate, but at the same pitch, as the input signal. The method is an improved version of the synchronized overlap-and-add (SOLA) method, and overlaps sample blocks in the input signal with sample blocks in the output signal in order to compress the signal. Samples are overlapped at a location that produces the best possible output quality. A correlation function is calculated for each possible overlap lag, and the location producing the highest value of the function is chosen. The range of possible overlap lags is equal to the sum of the size of the two sample blocks. A computationally efficient method for calculating the correlation function computes a discrete frequency transform of the input and output sample blocks, calculates the correlation, and then performs an inverse frequency transform of the correlation function, which has a maximum at the optimal lag. Also provided is a method for time scale modification of a multi-channel digital audio signal, in which each channel is processed independently. The listener integrates the different channels, and perceives a high quality multi-channel signal.
摘要翻译：用于数字音频信号的时间尺度修改的方法产生与输入信号处于不同的重放速率但是以相同音高的输出信号。该方法是同步重叠和加法（SOLA）方法的改进版本，并且将输入信号中的采样块与输出信号中的采样块重叠以便压缩该信号。样品在产生最佳输出质量的位置重叠。对于每个可能的重叠滞后计算相关函数，并且选择产生该函数的最高值的位置。可能的重叠延迟的范围等于两个样本块的大小的总和。用于计算相关函数的计算有效的方法计算输入和输出采样块的离散频率变换，计算相关性，然后执行最佳滞后最大的相关函数的逆频变换。还提供了一种用于多通道数字音频信号的时间尺度修改的方法，其中每个信道被独立地处理。听众整合不同的频道，并感知到高质量的多声道信号。

10. 发明授权

US06567782B1 Portable information terminal, method of processing audio data, recording medium, and program 有权
标题翻译：便携式信息终端，处理音频数据的方法，记录介质和程序
公开(公告)号：US06567782B1
公开(公告)日：2003-05-20
申请号：US09614863
申请日：2000-07-12
申请人： Takayuki Wakimura
发明人： Takayuki Wakimura
IPC分类号： G10L2104
CPC分类号： G11B20/00007 , G10L19/0212 , G11B20/10527 , H04B1/665
摘要： An expansion processor has a buffer defining unit for defining one of two buffers as a present inverse quantization buffer and defining one of two buffers as a present restoration buffer, an inverse quantization processor for inversely quantizing a quantized value read for each sample from a DCT data buffer, an IDCT processor for effecting an IDCT process on the inversely quantized data to restore time-domain audio data from frequency-domain data, a low-pass filter processor for removing a high-frequency component from the restored audio data, and an audio data output unit for outputting successive restored samples of audio data to a DAC to output sound from a speaker.
摘要翻译：扩展处理器具有用于将两个缓冲器之一定义为当前的反量化缓冲器并且定义两个缓冲器之一作为当前恢复缓冲器的缓冲器定义单元，用于从DCT数据中逆量化从每个样本读取的量化值的逆量化处理器缓冲器，用于对逆量化数据进行IDCT处理的IDCT处理器，用于从频域数据中恢复时域音频数据;低通滤波处理器，用于从恢复的音频数据中去除高频分量;以及音频数据输出单元，用于将连续恢复的音频数据样本输出到DAC以从扬声器输出声音。

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