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21. 发明授权

EP1194924B3 ADAPTIVE TILT COMPENSATION FOR SYNTHESIZED SPEECH RESIDUAL 无效
标题翻译：合成SPRACHRESIDUUMS的谱失真自适应补偿
公开(公告)号：EP1194924B3
公开(公告)日：2012-07-18
申请号：EP99948061.9
申请日：1999-08-24
申请人： WiAV SOLUTIONS LLC
发明人： YANG, Gao , HUAN-YU, Su
IPC分类号： G10L19/14
CPC分类号： G10L19/005 , G10L19/002 , G10L19/012 , G10L19/08 , G10L19/083 , G10L19/09 , G10L19/10 , G10L19/12 , G10L19/125 , G10L19/18 , G10L19/265 , G10L21/0364 , G10L2019/0005 , G10L2019/0007 , G10L2019/0011
摘要： A multi-rate speech codec supports a plurality of encoding bit rate modes by adaptively selecting encoding bit rate modes to match communication channel restrictions. In higher bit rate encoding modes, an accurate representation of speech through CELP (code excited linear prediction) and other associated modeling parameters are generated for higher quality decoding and reproduction. To achieve high quality in lower bit rate encoding modes, the speech encoder departs from the strict waveform matching criteria of regular CELP coders and strives to identify significant perceptual features of the input signal. To support lower bit rate encoding modes, a variety of techniques are applied many of which involve the classification of the input signal. For each bit rate mode selected, pluralities of fixed or innovation subcodebooks are selected for use in generating innovation vectors. At lower encoding bit rates, a decoder utilizes adaptive compensation to attempt to correct for spectral variations in the weighted synthesized residual. Although many approaches are possible, a long asymmetric window is applied to the synthesized residual to generate a reflection coefficient that is smoothed, scaled and used in a first order filter. Because the content of the window varies over time, the coefficient and therefore the filter varies (or adapts) to remove at least a portion of the spectral tilt. As a result, the synthesized speech signal sounds brighter without having introduced significant coding noise.

22. 发明授权

EP2080193B1 PITCH LAG ESTIMATION 有权转让
标题翻译：基音延迟估计
公开(公告)号：EP2080193B1
公开(公告)日：2012-06-06
申请号：EP07826610.3
申请日：2007-10-01
申请人： Nokia Corporation
发明人： LAAKSONEN, Lasse , RAMO, Anssi , VASILACHE, Adriana
IPC分类号： G10L11/04
CPC分类号： G10L25/90 , G10L19/09 , G10L25/06
摘要： Autocorrelation values are determined as a basis for an estimation of a pitch lag in a segment of an audio signal. A first considered delay range for the autocorrelation computations is divided into a first set of sections, and first autocorrelation values are determined for delays in a plurality of sections of this first set of sections. A second considered delay range for the autocorrelation computations is divided into a second set of sections such that sections of the first set and sections of the second set are overlapping. Second autocorrelation values are determined for delays in a plurality of sections of this second set of sections.

23. 发明公开

EP2420999A2 Method for pitch search of speech signals 审中-公开
标题翻译： Verfahren zur Emittlung der Grundfrequenz von Sprachsignalen
公开(公告)号：EP2420999A2
公开(公告)日：2012-02-22
申请号：EP11188232.0
申请日：2009-12-30
申请人： Huawei Technologies Co., Ltd.
发明人： Zhang, Dejun , Xu, Jianfeng , Miao, Lei , Qi, Fengyan , Zhang, Qing , Li, Lixiong , Ma, Fuwei , Gao, Yang
IPC分类号： G10L11/04 , G10L19/08
CPC分类号： G10L25/90 , G10L19/09 , G10L25/06
摘要： The present invention relates to a method and apparatus for pitch search. One method includes: obtaining a characteristic function value of a residual signal, where the residual signal is a result of removing a Long-Term Prediction (LTP) contribution signal from input speech signals; and obtaining a pitch according to the characteristic function value of the residual signal.
摘要翻译：一种用于音调搜索的方法，包括对输入语音信号进行下采样; 在预设的音调范围内计算对应于每个音调的下采样信号的残差信号; 计算在预设音调范围内对应于每个音调的残余信号的残余信号能量，其中残留信号是从下采样输入语音信号中去除长期预测贡献信号的结果; 在所计算的残差信号能量值中选择最小值，并将对应于最小值的音调设置为音高。

24. 发明授权

EP2071565B1 Coding apparatus and decoding apparatus 有权转让
标题翻译：编码装置和解码装置
公开(公告)号：EP2071565B1
公开(公告)日：2011-05-04
申请号：EP09156970.7
申请日：2004-09-15
申请人： Panasonic Corporation
发明人： Oshikiri, Masahiro
IPC分类号： G10L21/02 , G10L19/02
CPC分类号： G10L21/038 , G10L19/0208 , G10L19/09
摘要： A coding apparatus capable of coding a spectrum at a low bit rate and with high quality without producing any disturbance in a harmonic structure of the spectrum. In this apparatus, internal state setting section (106) sets an internal state of filtering section (107) using first spectrum S1(k). Pitch coefficient setting section (109) outputs pitch coefficient T by gradually changing it. Filtering section (107) calculates estimated value S' 2 (k) of second spectrum S2 (k) based on pitch coefficient T. Search section (108) calculates the degree of similarity between S2 (k) and S' 2 (k). At this time, pitch coefficient T' corresponding to the maximum calculated degree of similarity is given to filter coefficient calculation section (110). Filter coefficient calculation section (110) determines filter coefficient ² i using this pitch coefficient T'.

25. 发明公开

EP1391879A3 Speech coding apparatus, linear prediction coefficient analyzing apparatus and noise reducing apparatus 失效
标题翻译：语音编码装置，线性预测系数分析装置和噪声降低装置
公开(公告)号：EP1391879A3
公开(公告)日：2011-02-16
申请号：EP03024767.0
申请日：1995-11-27
申请人： Panasonic Corporation
发明人： Morii, Toshiyuki
IPC分类号： G10L19/14 , G10L19/06
CPC分类号： G10L19/06 , G10L15/02 , G10L15/16 , G10L19/09 , G10L19/18 , G10L21/0208 , G10L21/0216 , G10L25/06 , G10L25/12 , G10L25/27 , G10L2021/02168
摘要： There is provided a linear prediction coefficient analyzing apparatus comprising extracting means (63) for extracting a plurality of partial analyzing periods from an analyzing period for a digital signal and extracting a plurality of partial digital signals at the partial analyzing periods from the digital signal, window coefficient multiplying means (64) for multiplying each of the partial digital signals extracted by the extracting means (63) by a window coefficient to generate a window-processed partial digital signal for each of the partial analyzing periods extracted by the extracting means, auto-correlation analyzing means (65) for analyzing an auto-correlation of each of the window-processed partial digital signals generated by the window coefficient multiplying means (64) to generate an auto-correlation function from each of the window-processed partial digital signals for each of the partial analyzing periods, auto-correlation function synthesizing means (66) for weighting each of the auto-correlation functions generated by the auto-correlation analyzing means (65) with a weighting factor to generate a weighted auto-correlation function for each of the partial analyzing periods and adding the weighted auto-correlation functions to each other to generate a synthesized auto-correlation function, and linear prediction coefficient analyzing means (67) for performing a linear prediction analysis for the synthesized auto-correlation function generated by the auto-correlation function synthesizing means (66) to obtain a linear prediction coefficient for the digital signal.
摘要翻译：提供了一种线性预测系数分析装置，包括：提取装置（63），用于从数字信号的分析周期中提取多个部分分析周期，并从数字信号提取部分分析周期的多个部分数字信号;窗口系数乘法装置（64），用于将由提取装置（63）提取的每个部分数字信号乘以窗系数，以产生由提取装置提取的每个部分分析周期的经窗处理的部分数字信号; 相关性分析装置（65），用于分析窗口系数乘法装置（64）产生的每个窗口处理的部分数字信号的自相关，以从每个窗口处理的部分数字信号产生自相关函数，每个部分分析周期，用于对ea进行加权的自相关函数合成装置（66）由自相关分析装置（65）利用加权因子生成的自相关函数的自相关函数以生成用于每个部分分析周期的加权自相关函数并将加权自相关函数相加以生成合成自相关函数和线性预测系数分析装置（67），用于对由自相关函数合成装置（66）生成的合成自相关函数执行线性预测分析，以获得数字的线性预测系数信号。

26. 发明公开

EP1391878A3 Speech coding apparatus, linear prediction coefficient analyzing apparatus and noise reducing apparatus 失效
标题翻译：一种语音编码装置，所述线性预测分析和降噪
公开(公告)号：EP1391878A3
公开(公告)日：2011-02-09
申请号：EP03024766.2
申请日：1995-11-27
申请人： Panasonic Corporation
发明人： Morii, Toshiyuki
IPC分类号： G10L21/02
CPC分类号： G10L19/06 , G10L15/02 , G10L15/16 , G10L19/09 , G10L19/18 , G10L21/0208 , G10L21/0216 , G10L25/06 , G10L25/12 , G10L25/27 , G10L2021/02168
摘要： There is provided a noise reducing apparatus comprising speech signal receiving means (12,72) for receiving a plurality of frames of analog speech signals in which a noise exists and converting the frames of analog speech signals into a plurality of frames of digital speech signals one after another, Fourier transforming means (75) for performing a discrete Fourier transformation for each of the frames of digital speech signals obtained by the speech signal receiving means and producing an input spectrum and a phase spectrum corresponding to each of the frames of digital speech signals for each of frequency values, noise assuming means (76) for selecting a particular input spectrum having a minimum value from among a current input spectrum, produced by the Fourier transforming means, corresponding to a current frame of digital speech signals and a predetermined number of past input spectra, produced by the Fourier transforming means, corresponding to past frames of digital speech signals preceding to the current frame and assuming the particular input spectrum as a noise spectrum corresponding to the current frame of digital speech signals for each of the frequency values, noise reducing degree determining means (79) for determining a degree of a noise reduction according to each of the frames of digital speech signals obtained by the speech signal receiving means, noise reducing means (80) for adjusting a value of each of the noise spectra assumed by the noise assuming means according to the degree of the noise reduction determined by the noise reducing degree determining means to produce an adjusted noise spectrum having an adjusted value corresponding to the current frame of digital speech signals for each of the frequency values, subtracting the adjusted noise spectrum from the current input spectrum produced by the Fourier transforming means (75) for each of the frequency values to reduce the noise existing in the current frame of digital speech signals, and producing a noise-reduced input spectrum corresponding to the current input spectrum for each of the frequency values, and inverse Fourier transforming means (83) for performing an inverse Fourier transformation for the noise-reduced input spectra produced by the noise reducing means according to the phase spectra, produced by the Fourier transforming means, corresponding to the current frame of digital speech signals, producing a current frame of first-order output signals corresponding to the current frame of digital speech signals, and outputting a plurality of frames of first-order output signals corresponding to the frames of analog speech signals received by the speech signal receiving means, one after another as a plurality of frames of output signals.

27. 发明公开

EP2099025A4 AUDIO ENCODING DEVICE AND AUDIO ENCODING METHOD 审中-公开
公开(公告)号：EP2099025A4
公开(公告)日：2010-12-22
申请号：EP07850636
申请日：2007-12-14
申请人： PANASONIC CORP
发明人： MORII TOSHIYUKI
IPC分类号： G10L19/08 , G10L19/09 , G10L19/12
CPC分类号： G10L19/12 , G10L19/09
摘要： Provided is an audio encoding device which performs a closed loop search of a gain and a sound source vector without significantly increasing the calculation amount as compared to an open loop search. In the audio encoding device, firstly, a first parameter decision unit (121) performs a sound source search by an adaptive sound source codebook and then a second parameter decision unit (122) simultaneously performs by a closed loop, the sound source and the gain search by using a fixed sound source codebook. More specifically, for a combination of a fixed sound source vector and gain, the sum of a value obtained by multiplying a candidate fixed sound source vector by a candidate gain and a value obtained by multiplying an adaptive sound source vector by a candidate gain is subjected to a combination filter formed by a filter coefficient based on a quantization linear prediction coefficient so as to generate a combined signal. An encoded distortion as a distance between the combined signal and the input signal is calculated so as to search for the code and the gain of the fixed sound source vector which minimizes the encoded distortion.

28. 发明公开

EP2259255A1 Speech encoding method and system 审中-公开
标题翻译： Sprachkodierverfahren und Sprachkodiersystem
公开(公告)号：EP2259255A1
公开(公告)日：2010-12-08
申请号：EP10180379.9
申请日：1999-08-24
申请人： Mindspeed Technologies Inc
发明人： Gao, Yang
IPC分类号： G10L19/14
CPC分类号： G10L19/265 , G10L19/002 , G10L19/005 , G10L19/012 , G10L19/08 , G10L19/083 , G10L19/09 , G10L19/10 , G10L19/12 , G10L19/125 , G10L19/18 , G10L19/20 , G10L21/0364 , G10L2019/0005 , G10L2019/0007 , G10L2019/0011
摘要： A multi-rate speech codec supports a plurality of encoding bit rate modes by adaptively selecting encoding bit rate modes to match communication channel restrictions. In higher bit rate encoding models, an accurate representation of speech through CELP (code excited linear prediction) and other associated modeling parameters are generated for higher quality decoding and reproduction. For each bit rate mode selected, pluralities of fixed or innovation subcodebooks are selected for use in generating innovation vectors. The speech coder distinguishes various voice signals as a function of their voice content. For example, a Voice Activity Detection (VAD) algorithm selects an appropriate coding scheme depending on whether the speech signal comprises active or inactive speech. The encoder may consider varying characteristics of the speech signal including sharpness, a delay correlation, a zero-crossing rate, and a residual energy. In another embodiment of the present invention, code excited linear prediction is used for voice active signals whereas random excitation is used for voice inactive signals; the energy level and spectral content of the voice inactive signal may also be used for noise coding. The multi-rate speech codec may employ distributed detection and compensation processing the speech signal. For high quality perceptual speech reproduction, the speech codec may perform noise detection in both an encoder and decoder. The noise detection may be coordinated between the encoder and decoder. Similarly, noise compensation may be performed in a distributed manner among both the decoder and the encoder.
摘要翻译：多速率语音编解码器通过自适应地选择编码比特率模式来匹配通信信道限制，支持多种编码比特率模式。在较高比特率编码模型中，通过CELP（代码激励线性预测）和其他相关联的建模参数准确表达语音，以实现更高质量的解码和再现。对于所选择的每个比特率模式，选择多个固定或创新子码本来用于产生创新向量。语音编码器将各种语音信号区分为其语音内容的函数。例如，语音活动检测（VAD）算法根据语音信号是否包括有源或非活动语音来选择适当的编码方案。编码器可以考虑包括锐度，延迟相关性，过零率和剩余能量的语音信号的变化特性。在本发明的另一实施例中，代码激励线性预测用于语音有源信号，而随机激励用于语音无效信号; 语音不活动信号的能级和频谱内容也可用于噪声编码。多速率语音编解码器可以采用语音信号的分布式检测和补偿处理。对于高质量的感知语音再现，语音编解码器可以在编码器和解码器中执行噪声检测。可以在编码器和解码器之间协调噪声检测。类似地，可以在解码器和编码器之间以分布式方式执行噪声补偿。

29. 发明授权

EP1918909B1 Sampling error compensation 有权
标题翻译：采样误差补偿
公开(公告)号：EP1918909B1
公开(公告)日：2010-07-07
申请号：EP06123492.8
申请日：2006-11-03
申请人： Psytechnics Ltd
发明人： Barrett, Paul , Malfait, Ludovic
IPC分类号： G10L19/00 , G10L21/04
CPC分类号： G10L21/04 , G10L19/09 , G10L25/69
摘要： This invention relates to generation of a sample error coefficient suitable for use in an audio signal quality assessment system. The invention provides a method of determining a sample error coefficient between a first signal and a similar second signal comprising the steps of: determining a first periodicity measure from the first signal; determining a second periodicity measure from the second signal; generating a ratio in dependence upon said first periodicity measure and said second periodicity measure; and determining a sampling rate error coefficient in dependence upon said ratio.

30. 发明公开

EP2153436A1 GENERATING A FRAME OF AUDIO DATA 有权
标题翻译：创建音频数据的框架
公开(公告)号：EP2153436A1
公开(公告)日：2010-02-17
申请号：EP07735889.3
申请日：2007-05-14
申请人： FREESCALE SEMICONDUCTOR, INC.
发明人： SUSAN, Adrian , NEGHINA, Mihai
IPC分类号： G10L19/00 , G11B20/18
CPC分类号： G10L19/005 , G10L19/09 , G10L19/167
摘要： A method of generating a frame of audio data for an audio signal from preceding audio data for the audio signal that precede the frame of audio data, the method comprising the steps of: predicting a predetermined number of data samples for the frame of audio data based on the preceding audio data, to form predicted data samples; identifying a section of the preceding audio data for use in generating the frame of audio data; and forming the audio data of the frame of audio data as a repetition (602) of at least part of the identified section to span the frame of audio data, wherein the beginning of the frame of audio data comprises a combination of a subset of the repetition (602) of the at least part of the identified section and the predicted data samples.

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