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21. 发明申请

US20110099009A1 NETWORK/PEER ASSISTED SPEECH CODING 有权
标题翻译：网络/对等辅助语音编码
公开(公告)号：US20110099009A1
公开(公告)日：2011-04-28
申请号：US12901832
申请日：2010-10-11
申请人： Robert W. Zopf , Kelly Hale
发明人： Robert W. Zopf , Kelly Hale
IPC分类号： G10L19/00 , G10L21/00
CPC分类号： G10L21/00 , G10L17/00 , G10L19/0018 , G10L19/16
摘要： A communications network is used to transfer user attribute information about participants in a communication session to their respective communication terminals for storage and use thereon to configure a speech codec to operate in a speaker-dependent manner, thereby improving speech coding efficiency. In a network-assisted model, the user attribute information is stored on the communications network and selectively transmitted to the communication terminals while in a peer-assisted model, the user attribute information is derived by and transferred between communication terminals.
摘要翻译：通信网络用于将通信会话中的参与者的用户属性信息传送到其各自的通信终端以进行存储和使用，以将语音编解码器配置为以说话者相关的方式操作，从而提高语音编码效率。在网络辅助模型中，用户属性信息存储在通信网络上，并且在对等辅助模型中选择性地发送到通信终端，用户属性信息由通信终端之间导出和传送。

22. 发明授权

US07647067B2 Information processing apparatus and a cellular phone 有权
标题翻译：信息处理装置和蜂窝电话
公开(公告)号：US07647067B2
公开(公告)日：2010-01-12
申请号：US10995854
申请日：2004-11-23
申请人： Takashi Fujioka
发明人： Takashi Fujioka
IPC分类号： H04M1/00 , G06F9/46
CPC分类号： G06F9/4825 , G10L19/16 , H04M1/72522 , H04M1/72558
摘要： When a specified task process may mix with another more highly prioritized task process, it is aimed at accurately calculating a process time for the specified task process. A CPU 11 can simultaneously perform a plurality of task processes such as a task process to display the reproduction time during an audio reproduction process on a display 20 and a task process concerning a telephone call. A DSP 12 performs the audio reproduction process and changes a timing to notify the CPU 11 of information about the audio reproduction time in accordance with loads on the task process in the CPU 11.
摘要翻译：当指定的任务进程可能与另一个更高优先级的任务进程混合时，其目的是准确地计算指定任务进程的进程时间。 CPU11可以同时执行多个任务处理，例如任务处理，以在显示器20上的音频再现处理期间显示再现时间，以及关于电话呼叫的任务处理。 DSP12执行音频再现处理，并且根据CPU11中的任务处理的负载改变定时以通知CPU 11关于音频再现时间的信息。

23. 发明授权

US07627623B2 Inverse modified discrete cosine transform (IMDCT) co-processor and audio decoder having the same 失效
标题翻译：协处理器和音频解码器的逆修正离散余弦变换（IMDCT）具有相同的功能
公开(公告)号：US07627623B2
公开(公告)日：2009-12-01
申请号：US11432100
申请日：2006-05-11
申请人： Suk Ho Lee , Nak Woong Eum , Hee Bum Jung
发明人： Suk Ho Lee , Nak Woong Eum , Hee Bum Jung
IPC分类号： G06F17/14
CPC分类号： G06F17/147 , G10L19/16
摘要： Provided are an IMDCT co-processor and an audio decoder having the same. The IMDCT co-processor includes: an input buffer for storing an input inverse-quantized frequency spectrum sample value; an I/Q separator for dividing the sample value stored in the input buffer into real data I and imaginary data Q; a first operation unit for performing complex-multiplication on the data divided by the I/Q separator and a given twiddle factor; an IFFT unit for performing an inverse fast Fourier transform on the operation result value of the first operation unit; a second operation unit for performing complex-multiplication on the result value from the IFFT unit and a given twiddle factor; a deinterleaver for receiving the operation result value from the second operation unit to arrange data and performing inverse-mapping on a positive value (+) and a negative value (−) of a certain portion of the data to each other to output a final IMDCT time sample value; and a control register for selecting the input inverse-quantized frequency spectrum sample value according to a given window sequence value to determine the final IMDCT time sample value.
摘要翻译：提供了一种IMDCT协处理器和具有其的音频解码器。 IMDCT协处理器包括：输入缓冲器，用于存储输入的反量化频谱样本值; I / Q分离器，用于将存储在输入缓冲器中的采样值分成实数数据I和虚数据Q; 第一操作单元，用于对由I / Q分离器划分的数据和给定的旋转因子进行复数乘法; IFFT单元，用于对第一操作单元的操作结果值进行快速傅立叶逆变换; 第二操作单元，用于对来自IFFT单元的结果值和给定的旋转因子进行复数乘法; 去交织器，用于从第二操作单元接收操作结果值以排列数据并且对数据的某一部分的正值（+）和负值（ - ）进行反向映射，以输出最终的IMDCT 时间样本值; 以及控制寄存器，用于根据给定的窗口序列值选择输入的反量化频谱采样值，以确定最终的IMDCT时间采样值。

24. 发明申请

US20090037169A1 METHOD AND APPARATUS FOR IMPLEMENTING FIXED CODEBOOKS OF SPEECH CODECS AS COMMON MODULE 有权
标题翻译：用于实现语音编码的固定代码作为通用模块的方法和装置
公开(公告)号：US20090037169A1
公开(公告)日：2009-02-05
申请号：US11930750
申请日：2007-10-31
申请人： Kang-eun LEE , Do-hyung Kim , Chang-yong Son
发明人： Kang-eun LEE , Do-hyung Kim , Chang-yong Son
IPC分类号： G10L19/12
CPC分类号： G10L19/16 , G10L19/12
摘要： A method and apparatus for implementing fixed codebooks as a common module are provided. In the method of implementing fixed codebooks of a plurality of speech codecs as a common module, it is possible to include only a part excluding fixed codebooks in a communication terminal or communication system, support various speech codecs without using a chip with high price and high performance, and reduce a memory space that is occupied by the speech codecs by generating a track of a fixed codebook corresponding to a speech codec based on information on the speech codec among the plurality of speech codecs and selecting a codebook vector corresponding to a target signal among codebook vectors constructed with combinations of pulses represented by the generated track. In addition, it is possible to reduce processing complexity as compared with a case of embodying the common fixed codebook module in software by embodying the common fixed codebook module in hardware. In addition, it is possible to improve the entire voice processing performance by applying the latest fixed codebook searching algorithm only to the common fixed codebook, thereby easily applying the latest fixed codebook searching algorithm to the entire voice codec.
摘要翻译：提供了一种用于实现固定码本作为公共模块的方法和装置。在将多个语音编解码器的固定码本实现为公共模块的方法中，可以在通信终端或通信系统中仅包括不包括固定码本的部分，而不使用高价格高的芯片支持各种语音编解码器通过基于多个语音编解码器中的语音编解码器的信息生成与语音编解码器对应的固定码本的轨道，并且选择与目标信号相对应的码本矢量，来降低语音编解码器所占用的存储器空间在由所生成的轨道表示的脉冲的组合构成的码本矢量中。此外，与通过在硬件中实施公共固定码本模块的软件中体现公共固定码本模块的情况相比，可以降低处理复杂度。此外，通过将最新的固定码本搜索算法应用于公共固定码本，可以提高整个语音处理性能，从而容易地将最新的固定码本搜索算法应用于整个语音编解码器。

25. 发明申请

US20080091438A1 AUDIO SIGNAL DECODER AND RESOURCE ACCESS CONTROL METHOD 审中-公开
标题翻译：音频信号解码器和资源访问控制方法
公开(公告)号：US20080091438A1
公开(公告)日：2008-04-17
申请号：US11871336
申请日：2007-10-12
申请人： Yukio MASUDA , Yasuaki HASHIMOTO , Takahisa KOUDA
发明人： Yukio MASUDA , Yasuaki HASHIMOTO , Takahisa KOUDA
IPC分类号： G10L19/00
CPC分类号： G10L19/16 , G11B2020/00014 , G11B2020/10546 , G11B2020/10638 , G11B2020/10759 , G11B2020/10768
摘要： The audio signal decoder of the present invention includes a first decoding unit which decodes a first encoded audio signal, a second decoding unit which decodes a second encoded audio signal, a memory which stores data under decoding as well as data decoded by the first and second decoding units, and a selection unit which selects one of the first and second decoding units, as a device having access to said memory.
摘要翻译：本发明的音频信号解码器包括对第一编码音频信号进行解码的第一解码单元，对第二编码音频信号进行解码的第二解码单元，存储解码数据的存储器，以及由第一和第二解码器解码单元以及选择第一和第二解码单元之一的选择单元作为具有访问所述存储器的设备。

26. 发明授权

US07333034B2 Data processing device, encoding device, encoding method, decoding device decoding method, and program 失效
标题翻译：数据处理装置，编码装置，编码方法，解码装置解码方法和程序
公开(公告)号：US07333034B2
公开(公告)日：2008-02-19
申请号：US10557557
申请日：2004-05-20
申请人： Jun Matsumoto , Masayuki Nishiguchi
发明人： Jun Matsumoto , Masayuki Nishiguchi
IPC分类号： H03M7/00
CPC分类号： G10L19/16
摘要： The present invention relates to a data processing apparatus, a method and apparatus for encoding, a method and apparatus for decoding, and a program, that allow a reduction in an algorithm delay. An interpolator 51 produces interpolated PCM data by performing R-times oversampling on original PCM data. A frame encoder 54 fetches a predetermined number of samples of the oversampled data as one frame, encodes the oversampled data on a frame-by-frame basis, and outputs resultant encoded data. A frame decoder 55 decodes the encoded data on a frame-by-frame basis at a rate R times higher than a predetermined normal rate. A decimator 56 decimates data obtained as a result of the decoding such that the number of samples is reduced to 1/R of the number of sampled included in the original data. The present invention is applicable, for example, to an IP telephone system.
摘要翻译：本发明涉及一种用于编码的数据处理装置，方法和装置，用于解码的方法和装置以及允许减少算法延迟的程序。内插器51通过对原始PCM数据执行R次过采样来产生内插PCM数据。帧编码器54将过采样数据的预定数目的样本取出为一帧，对逐帧逐帧进行过采样数据进行编码，并输出合成编码数据。帧解码器55以比预定正常速率高R倍的速率逐帧地解码编码数据。抽取器56抽取作为解码结果获得的数据，使得样本数量减少到包含在原始数据中的采样数量的1 / R。本发明可以应用于例如IP电话系统。

27. 发明授权

US07289953B2 Apparatus and method for speech coding 有权
标题翻译：用于语音编码的装置和方法
公开(公告)号：US07289953B2
公开(公告)日：2007-10-30
申请号：US11095530
申请日：2005-04-01
申请人： Kazutoshi Yasunaga , Toshiyuki Morii
发明人： Kazutoshi Yasunaga , Toshiyuki Morii
IPC分类号： G10L19/12
CPC分类号： G10L19/16 , G10L19/083 , G10L19/09
摘要： A speech encoder includes an LPC synthesizer that obtains synthesized speech by filtering an adaptive excitation vector and a stochastic excitation vector stored in an adaptive codebook and in a stochastic codebook using LPC coefficients obtained from input speech. A gain calculator calculates gains of the adaptive excitation vector and the stochastic excitation vector and searches code of the adaptive excitation vector and code of the stochastic excitation vector by comparing distortions between the input speech and the synthesized speech obtained using the adaptive excitation vector and the stochastic excitation vector. A parameter coder performs predictive coding of gains using the adaptive excitation vector and the stochastic excitation vector corresponding to the codes obtained. The parameter coder comprises a prediction coefficient adjuster that adjusts at least one prediction coefficient used for the predictive coding according to at least one state of at least one previous subframe.
摘要翻译：语音编码器包括LPC合成器，其通过使用从输入语音获得的LPC系数滤波自适应码本和存储在自适应码本中的随机激励矢量和随机码本来获得合成语音。增益计算器通过比较输入语音和使用自适应激励矢量获得的合成语音之间的失真和随机激励矢量，随机随机激励矢量的自适应激励矢量和随机激励矢量的代码，并搜索随机激励矢量的代码，激励矢量。参数编码器使用自适应激励矢量和对应于所获得的代码的随机激励矢量来执行增益的预测编码。参数编码器包括预测系数调整器，其根据至少一个先前子帧的至少一个状态来调整用于预测编码的至少一个预测系数。

28. 发明申请

US20070223885A1 Playback apparatus 审中-公开
标题翻译：播放装置
公开(公告)号：US20070223885A1
公开(公告)日：2007-09-27
申请号：US11724562
申请日：2007-03-15
申请人： Shinji Kuno , Takanobu Mukaide
发明人： Shinji Kuno , Takanobu Mukaide
IPC分类号： H04N7/00
CPC分类号： H04N21/4341 , G10L19/16 , H04N21/2368 , H04N21/44029 , H04N21/8106
摘要： According to one embodiment, a playback apparatus includes first to third digital signal processors. The first digital signal processor includes decode functions corresponding to a plurality of kinds of compression-decoding schemes and decodes first audio data, which is compression-encoded by using an arbitrary one of the plurality of kinds of compression-encoding schemes, thereby generating a first digital audio signal. The second digital signal processor includes decode functions corresponding to the plurality of kinds of compression-decoding schemes and decodes second audio data, which is compression-encoded by using an arbitrary one of the plurality of kinds of compression-encoding schemes, thereby generating a second digital audio signal. The third digital signal processor executes a mixing process of mixing the first digital audio signal and the second digital audio signal, thereby generating a digital audio output signal.
摘要翻译：根据一个实施例，重放装置包括第一至第三数字信号处理器。第一数字信号处理器包括与多种压缩解码方案相对应的解码功能，并对通过使用多种压缩编码方式中的任意一种压缩编码的第一音频数据进行解码，从而产生第一数字音频信号。第二数字信号处理器包括与多种压缩解码方案相对应的解码功能，并对通过使用多种压缩编码方案中的任意一种压缩编码的第二音频数据进行解码，从而产生第二数字音频信号。第三数字信号处理器执行混合第一数字音频信号和第二数字音频信号的混合处理，从而产生数字音频输出信号。

29. 发明授权

US07275036B2 Apparatus and method for coding a time-discrete audio signal to obtain coded audio data and for decoding coded audio data 有权
标题翻译：用于编码时分离音频信号以获得编码音频数据和解码编码音频数据的装置和方法
公开(公告)号：US07275036B2
公开(公告)日：2007-09-25
申请号：US10966780
申请日：2004-10-15
申请人： Ralf Geiger , Thomas Sporer , Karlheinz Brandenburg , Juergen Herre , Juergen Koller , Joachim Deguara
发明人： Ralf Geiger , Thomas Sporer , Karlheinz Brandenburg , Juergen Herre , Juergen Koller , Joachim Deguara
IPC分类号： G10L19/00
CPC分类号： G10L19/16 , G10L19/022 , G10L19/20 , G10L2019/0005
摘要： A time-discrete audio signal is processed to provide a quantization block with quantized spectral values. Furthermore, an integer spectral representation is generated from the time-discrete audio signal using an integer transform algorithm. The quantization block having been generated using a psychoacoustic model is inversely quantized and rounded to then form a difference between the integer spectral values and the inversely quantized rounded spectral values. The quantization block alone provides a lossy psychoacoustically coded/decoded audio signal after the decoding, whereas the quantization block, together with the combination block, provides a lossless or almost lossless coded and again decoded audio signal in the decoding. By generating the differential signal in the frequency domain, a simpler coder/decoder structure results.
摘要翻译：处理时间离散音频信号以向量化块提供量化的频谱值。此外，使用整数变换算法从时间离散音频信号生成整数谱表示。已经使用心理声学模型产生的量化块被逆量化并舍入，从而形成整数频谱值和逆量化的舍入频谱值之间的差。量化块单独提供在解码之后的有损心理声学编码/解码音频信号，而量化块与组合块一起在解码中提供无损或几乎无损的编码和再次解码的音频信号。通过在频域中产生差分信号，可以得到更简单的编码器/解码器结构。

30. 发明申请

US20060036436A1 Method for dynamic selection of optimized codec for streaming audio content 失效
公开(公告)号：US20060036436A1
公开(公告)日：2006-02-16
申请号：US10916977
申请日：2004-08-12
申请人： Michael Halcrow , Dustin Kirkland
发明人： Michael Halcrow , Dustin Kirkland
IPC分类号： G10L19/00
CPC分类号： G10L19/00 , G10L19/16 , G10L19/167
摘要： Several encoders at a broadcast system encode the same audio content. Packets from the resulting streams are immediately decoded and compared against the packets of the original audio stream. The broadcast system dynamically selects the codec that performs the best for the audio in any given packet. The packet produced by the encoder of the best-performing codec devices is selected to be broadcasted/transmitted.

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