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1.发明申请Transcoder for speech codecs of different CELP type and method therefor 有权用于不同CELP类型的语音编解码器的转码器及其方法公开(公告)号：US20050010403A1
公开(公告)日：2005-01-13
申请号：US10749748
申请日：2003-12-30
CPC分类号： G10L19/173 , G10L19/12
摘要： A transcoder for use between speech codecs using different Code-Excited Linear Prediction (CELP) type and a method therefor are disclosed. The transcoder includes a decoding unit of an input CELP codec, a transcoding filter, a transcoding filter design unit, and an encoding unit of an output CELP codec. By substituting a post-filter and a perceptual weighting filter of a prior art with one transcoding filter, the calculation amount of the transcoder is reduced, and speech quality decoded at a receiving end is improved.
摘要翻译： 公开了使用不同的代码激励线性预测（CELP）类型的语音编解码器及其方法之间使用的代码转换器。 代码转换器包括输入CELP编解码器，代码转换滤波器，代码转换滤波器设计单元和输出CELP编解码器的编码单元的解码单元。 通过使用一个代码转换滤波器代替现有技术的后置滤波器和感知加权滤波器，降低了代码转换器的计算量，并提高了在接收端解码的语音质量。
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公开(公告)号：US20050143979A1
公开(公告)日：2005-06-30
申请号：US11006447
申请日：2004-12-06
申请人： Mi Lee , Do Kim , Jongmo Sung , Hyun Woo Kim , Hong Kang , Sung Jung , Dae Youn , Hong Kim
发明人： Mi Lee , Do Kim , Jongmo Sung , Hyun Woo Kim , Hong Kang , Sung Jung , Dae Youn , Hong Kim
CPC分类号： G10L19/24
摘要： There is provided a speech coding/decoding apparatus and method, in which the input speech signals are classified into several classes in accordance with characteristics of the input speech signals and the input speech signals are coded using frame sizes, quantizer structures, and bit assignment methods corresponding to the determined classes, or in which the frame sizes can be adjusted in accordance with network conditions or codec type of a counter part. Therefore, by optimally adjusting the frame size, the quantizer structure, and the bit assignment method in accordance with the characteristics of input speech, it is possible to improve the performance of the speech coding apparatus, and by adjusting the frame size in accordance with the speech codec type of a counter part, it is also possible to reduce the total end-to-end delay.
摘要翻译： 提供了一种语音编码/解码装置和方法，其中根据输入的语音信号的特性将输入的语音信号分为几类，并且使用帧大小，量化器结构和比特分配方法对输入的语音信号进行编码 对应于所确定的类别，或者可以根据网络条件或计数器部件的编解码器类型来调整帧大小。 因此，通过根据输入语音的特性优化调整帧大小，量化器结构和比特分配方法，可以提高语音编码装置的性能，并且可以通过根据 语音编解码器类型的计数器部件，也可以减少总的端到端延迟。
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公开(公告)号：US20050010402A1
公开(公告)日：2005-01-13
申请号：US10749569
申请日：2003-12-30
CPC分类号： G10L19/125
摘要： A wide-band speech coder and a method thereof and a wide-band speech decoder and a method thereof are provided. The wide-band speech coder includes a speech characteristic classification unit, which stipulates a characteristic of speech corresponding to a current frame statistically using an open-circuit pitch value and a linear prediction coefficient in which a wide-code speech signal to be coded is perceptual weigh filtered, an adaptive codebook retrieving unit, which retrieves a pitch delay value around the open-circuit pitch value, calculates a pitch gain value, generates an adaptive codebook contribution signal corresponding to the retrieved pitch delay value, and outputs a difference between the generated adaptive codebook contribution signal and the perceptual weigh filtered signal as a first fixed codebook target signal, a first fixed codebook retrieving unit, which obtains a first fixed codebook index that can express the first fixed codebook target signal most properly, and a first fixed codebook gain value, generates a first fixed codebook contribution signal corresponding to the retrieved index, and outputs a difference between the first generated fixed codebook contribution signal and the first fixed codebook target signal as a second fixed codebook target signal, a second fixed codebook retrieving unit, which includes at least two second fixed codebooks according to a speech characteristic, selects a second fixed codebook according to the speech characteristic, and retrieves second fixed codebook indices that can express the second fixed codebook target signal most properly, and second fixed codebook gain values, and a parameter multiplexer, which quantizes and multiplexes the speech characteristic information, the pitch delay value, the pitch gain value, the first fixed codebook index, the first fixed codebook gain value, the second fixed codebook indices, and the second fixed codebook gain values, makes them as a bit stream, and transmits the bit stream to an external speech decoding terminal.
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公开(公告)号：US20060253276A1
公开(公告)日：2006-11-09
申请号：US11395838
申请日：2006-03-31
申请人： Tae Kang , Jin Choi , Keun Lee , Young Park , Dae Youn
发明人： Tae Kang , Jin Choi , Keun Lee , Young Park , Dae Youn
IPC分类号： G10L19/00
摘要： An audio coding method and apparatus capable of improving efficiency of a MPEG-4 AAC (Moving Picture Expert Group-4 Advanced Audio Coding) process are disclosed. The audio coding method and apparatus reduce the number of calculations of an audio coding algorithm to improve efficiency of an audio coding process. Specifically, the audio coding method and apparatus reduce the number of calculations required for a Psychoacoustic model process of the MPEG-4 AAC algorithm capable of coding an audio signal.
摘要翻译： 公开了能够提高MPEG-4AAC（运动图像专家组-4高级音频编码）处理效率的音频编码方法和装置。 音频编码方法和装置减少音频编码算法的计算次数，以提高音频编码处理的效率。 具体地，音频编码方法和装置减少了能够编码音频信号的MPEG-4AAC算法的心理声学模型处理所需的计算次数。
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公开(公告)号：US20050010401A1
公开(公告)日：2005-01-13
申请号：US10615268
申请日：2003-07-07
CPC分类号： G10L19/005 , G10L19/04 , G10L25/93
摘要： Provided are a speech restoration system and method for concealing packet losses. The system includes a demultiplexer that demultiplexes an input bit stream and divides the input bit stream into several packets; a packet loss concealing unit that produces and outputs a linear spectrum pair (LSP) coefficient representing the vocal tract of voice and an excitation signal corresponding to a lost frame, when a packet loss occurs; and a speech restoring unit that synthesizes voice using the packets input from the demultiplexer, outputs the result as restored voice, and synthesizes voice corresponding to a lost packet using the LSP coefficient and the excitation signal input from the packet loss concealing unit and outputs the result as restored voice when the lost packet is detected, wherein the packet loss concealing unit repeats linear prediction coefficients (LPCs) of a last-received valid frame, produces a first excitation signal for the lost frame using a time scale modification (TSM) method, when the lost frame is voiceless, and produces a second excitation signal by re-estimating a gain parameter based on the first excitation signal, when the lost frame is voiced.
摘要翻译： 提供了一种用于隐藏分组丢失的语音恢复系统和方法。 该系统包括解复用器，其对输入比特流进行解复用并将输入比特流分成若干分组; 分组丢失隐藏单元，当发生分组丢失时，产生并输出表示语音的声道的线性频谱对（LSP）系数和对应于丢失帧的激励信号; 以及语音恢复单元，其使用从解复用器输入的分组合成语音，将结果输出为恢复的语音，并且使用从分组丢失隐藏单元输入的LSP系数和激励信号来合成与丢失分组对应的语音，并输出结果 作为在检测到丢失分组时的恢复语音，其中分组丢失隐藏单元重复最后接收的有效帧的线性预测系数（LPC），使用时间缩放修改（TSM）方法产生丢失帧的第一激励信号， 当丢失的帧是无声的，并且当丢失的帧被发音时，通过基于第一激励信号重新估计增益参数来产生第二激励信号。
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