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1. 发明申请

WO2006136901A3 SYSTEM AND METHOD FOR ADAPTIVE TRANSMISSION OF COMFORT NOISE PARAMETERS DURING DISCONTINUOUS SPEECH TRANSMISSION 审中-公开
标题翻译：在不连续语音传输期间适应传输舒适噪声参数的系统和方法
公开(公告)号：WO2006136901A3
公开(公告)日：2007-03-08
申请号：PCT/IB2006001604
申请日：2006-06-15
申请人： NOKIA CORP , GREER STEVEN CRAIG , GOURNAY PHILIPPE , JELINEK MILAN
发明人： GREER STEVEN CRAIG , GOURNAY PHILIPPE , JELINEK MILAN
IPC分类号： G10L19/00
CPC分类号： G10L19/012 , G10L19/24
摘要： Apparatus is provided that includes at least one entity for transmitting speech signals in a discontinuous transmission mode including transmitting speech frames interspersed with frames including comfort noise parameters during periods of speech pauses. The entit(ies) include a first entity for estimating a current noise value. In addition, the apparatus includes a second entity for selectively controlling a rate at which the frames including comfort noise parameters are transmitted during the periods of speech pauses based upon the estimated current noise value.
摘要翻译：提供了包括用于以不连续传输模式发送语音信号的至少一个实体的装置，包括在语音暂停期间包括散布有包括舒适噪声参数的帧的语音帧。该权限包括用于估计当前噪声值的第一实体。此外，该设备包括第二实体，用于根据估计的当前噪声值选择性地控制在语音暂停期间发送包括舒适噪声参数的帧的速率。

2. 发明申请

WO2007073604A8 METHOD AND DEVICE FOR EFFICIENT FRAME ERASURE CONCEALMENT IN SPEECH CODECS 审中-公开
标题翻译：方法和设备在语音编码中有效的帧消除隐藏
公开(公告)号：WO2007073604A8
公开(公告)日：2007-12-21
申请号：PCT/CA2006002146
申请日：2006-12-28
申请人： VOICEAGE CORP , VAILLANCOURT TOMMY , JELINEK MILAN , GOURNAY PHILIPPE , SALAMI REDWAN
发明人： VAILLANCOURT TOMMY , JELINEK MILAN , GOURNAY PHILIPPE , SALAMI REDWAN
IPC分类号： G10L19/00 , G10L21/02
CPC分类号： G10L19/005
摘要： A method and device for concealing frame erasures caused by frames of an encoded sound signal erased during transmission from an encoder to a decoder and for recovery of the decoder after frame erasures comprise, in the encoder, determining concealment/recovery parameters including at least phase information related to frames of the encoded sound signal. The concealment/recovery parameters determined in the encoder are transmitted to the decoder and, in the decoder, frame erasure concealment is conducted in response to the received concealment/recovery parameters. The frame erasure concealment comprises resynchronizing, in response to the received phase information, the erasure-concealed frames with corresponding frames of the sound signal encoded at the encoder. When no concealment/recovery parameters are transmitted to the decoder, a phase information of each frame of the encoded sound signal that has been erased during transmission from the encoder to the decoder is estimated in the decoder. Also, frame erasure concealment is conducted in the decoder in response to the estimated phase information, wherein the frame erasure concealment comprises resynchronizing, in response to the estimated phase information, each erasure-concealed frame with a corresponding frame of the sound signal encoded at the encoder.
摘要翻译：一种用于隐藏在从编码器到解码器的传输期间被擦除的编码声音信号的帧引起的帧擦除和在帧擦除之后恢复解码器的方法和装置，在编码器中包括确定包括至少相位信息的隐藏/恢复参数与编码的声音信号的帧相关。在编码器中确定的隐藏/恢复参数被发送到解码器，并且在解码器中，响应于接收的隐藏/恢复参数进行帧擦除隐藏。帧擦除隐藏包括响应于接收到的相位信息，重新同步擦除隐藏的帧与在编码器处编码的声音信号的相应帧。当没有隐藏/恢复参数被发送到解码器时，在解码器中估计在从编码器到解码器的传输期间被擦除的编码声音信号的每一帧的相位信息。此外，响应于估计的相位信息在解码器中进行帧擦除隐藏，其中帧擦除隐藏包括响应于估计的相位信息重新同步每个被擦除隐藏的帧与在该编码的声音信号的相应帧编码器。

3. 发明申请

WO2004034379A3 METHODS AND DEVICES FOR SOURCE CONTROLLED VARIABLE BIT-RATE WIDEBAND SPEECH CODING 审中-公开
标题翻译：用于源控制的可变比特率宽带语音编码的方法和设备
公开(公告)号：WO2004034379A3
公开(公告)日：2004-12-23
申请号：PCT/CA0301571
申请日：2003-10-09
申请人： NOKIA CORP , JELINEK MILAN
发明人： JELINEK MILAN
IPC分类号： G01L19/14 , G10L11/04 , G10L19/02 , G10L19/14 , G10L21/02
CPC分类号： G10L19/24 , G10L19/012 , G10L19/173
摘要： Speech signal classification and encoding systems and methods are disclosed herein. The signal classification is done in three steps each of them discriminating a specific signal class. First, a voice activity detector (VAD) discriminates between active and inactive speech frames. If an inactive speech frame is detected (background noise signal) then the classification chain ends and the frame is encoded with comfort noise generation (CNG). If an active speech frame is detected, the frame is subjected to a second classifier dedicated to discriminate unvoiced frames. If the classifier classifies the frame as unvoiced speech signal, the classification chain ends, and the frame is encoded using a coding method optimized for unvoiced signals. Otherwise, the speech frame is passed through to the "stable voiced" classification module. If the frame is classified as stable voiced frame, then the frame is encoded using a coding method optimized for stable voiced signals. Otherwise, the frame is likely to contain a non-stationary speech segment such as a voiced onset or rapidly evolving voiced speech signal. In this case a general-purpose speech coder is used at a high bit rate for sustaining good subjective quality .
摘要翻译：在此公开了语音信号分类和编码系统和方法。信号分类分三个步骤完成，每个步骤区分特定的信号类别。首先，语音活动检测器（VAD）区分活动和非活动语音帧。如果检测到不活动的语音帧（背景噪声信号），则分类链结束，并且该帧被编码以舒适噪声产生（CNG）。如果检测到活动语音帧，则该帧经受专用于区分无声帧的第二分类器。如果分类器将该帧分类为清音语音信号，则分类链结束，并且使用针对清音信号优化的编码方法对帧进行编码。否则，语音帧被传递到“稳定浊音”分类模块。如果帧被分类为稳定浊音帧，则使用针对稳定浊音信号优化的编码方法对帧进行编码。否则，帧可能包含非平稳的语音片段，例如浊音起始或快速演变的浊音语音信号。在这种情况下，通用语音编码器以高比特率使用，以维持良好的主观质量。

4. 发明申请

WO2009000073A8 METHOD AND DEVICE FOR SOUND ACTIVITY DETECTION AND SOUND SIGNAL CLASSIFICATION 审中-公开
标题翻译：用于声音活动检测和声音信号分类的方法和装置
公开(公告)号：WO2009000073A8
公开(公告)日：2009-03-26
申请号：PCT/CA2008001184
申请日：2008-06-20
申请人： VOICEAGE CORP , MALENOVSKY VLADIMIR , JELINEK MILAN , VAILLANCOURT TOMMY , SALAMI REDWAN
发明人： MALENOVSKY VLADIMIR , JELINEK MILAN , VAILLANCOURT TOMMY , SALAMI REDWAN
IPC分类号： G10L11/00 , G10L19/02 , G10L21/02
CPC分类号： G10L25/78 , G10L19/22
摘要： A device and method for estimating a tonality of a sound signal comprise: calculating a current residual spectrum of the sound signal; detecting peaks in the current residual spectrum; calculating a correlation map between the current residual spectrum and a previous residual spectrum for each detected peak; and calculating a long-term correlation map based on the calculated correlation map, the long-term correlation map being indicative of a tonality in the sound signal.
摘要翻译：用于估计声音信号的音调的装置和方法包括：计算声音信号的当前残余频谱; 检测当前残留谱中的峰; 计算每个检测到的峰值的当前残差谱和先前残差谱之间的相关图; 以及基于所计算的相关图计算长期相关图，所述长期相关图表示所述声音信号中的音调。

5. 发明申请

WO2012055016A1 CODING GENERIC AUDIO SIGNALS AT LOW BITRATES AND LOW DELAY 审中-公开
标题翻译：编码低频和低延迟的一般音频信号
公开(公告)号：WO2012055016A1
公开(公告)日：2012-05-03
申请号：PCT/CA2011001182
申请日：2011-10-24
申请人： VOICEAGE CORP , VAILLANCOURT TOMMY , JELINEK MILAN
发明人： VAILLANCOURT TOMMY , JELINEK MILAN
IPC分类号： G10L19/12
CPC分类号： G10L19/20 , G10L19/02 , G10L19/08
摘要： A mixed time-domain / frequency-domain coding device and method for coding an input sound signal, wherein a time-domain excitation contribution is calculated in response to the input sound signal. A cut-off frequency for the time-domain excitation contribution is also calculated in response to the input sound signal, and a frequency extent of the time-domain excitation contribution is adjusted in relation to this cut-off frequency. Following calculation of a frequency-domain excitation contribution in response to the input sound signal, the adjusted time-domain excitation contribution and the frequency-domain excitation contribution are added to form a mixed time-domain / frequency-domain excitation constituting a coded version of the input sound signal. In the calculation of the time-domain excitation contribution, the input sound signal may be processed in successive frames of the input sound signal and a number of sub-frames to be used in a current frame may be calculated. Corresponding encoder and decoder using the mixed time-domain / frequency-domain coding device are also described.
摘要翻译：一种用于编码输入声音信号的混合时域/频域编码装置和方法，其中响应于输入声音信号计算时域激励贡献。还响应于输入声音信号计算时域激励贡献的截止频率，并且相对于该截止频率调整时域激励贡献的频率范围。在响应于输入声音信号计算频域激励贡献之后，调整调整的时域激励贡献和频域激励贡献以形成构成编码版本的混合时域/频域激励输入声音信号。在时域激励贡献的计算中，可以在输入声音信号的连续帧中处理输入声音信号，并且可以计算要在当前帧中使用的多个子帧。还描述了使用混合时域/频域编码装置的对应编码器和解码器。

6. 发明申请

WO03052744A2 SIGNAL MODIFICATION METHOD FOR EFFICIENT CODING OF SPEECH SIGNALS 审中-公开
标题翻译：用于语音信号有效编码的信号修改方法
公开(公告)号：WO03052744A2
公开(公告)日：2003-06-26
申请号：PCT/CA0201948
申请日：2002-12-13
申请人： VOICEAGE CORP , TAMMI MIKKO , JELINEK MILAN , LAFLAMME CLAUDE , RUOPPILA VESA
发明人： TAMMI MIKKO , JELINEK MILAN , LAFLAMME CLAUDE , RUOPPILA VESA
IPC分类号： G10L19/12 , G10L19/08
CPC分类号： G10L19/08
摘要： For determining a long-term-prediction delay parameter characterizing a long term prediction in a technique using signal modification for digitally encoding a sound signal, the sound signal is divided into a series of successive frames, a feature of the sound signal is located in a previous frame, a corresponding feature of the sound signal is located in a current frame, and the long-term-prediction delay parameter is determined for the current frame while mapping, with the long term prediction, the signal feature of the previous frame with the corresponding signal feature of the current frame. In a signal modification method for implementation into a technique for digitally encoding a sound signal, the sound signal is divided into a series of successive frames, each frame of the sound signal is partitioned into a plurality of signal segments, and at least a part of the signal segments of the frame are warped while constraining the warped signal segments inside the frame. For searching pitch pulses in a sound signal, a residual signal is produced by filtering the sound signal through a linear prediction analysis filter, a weighted sound signal is produced by processing the sound signal through a weighting filter, the weighted sound signal being indicative of signal periodicity, a synthesized weighted sound signal is produced by filtering a synthesized speech signal produced during a last subframe of a previous frame of the sound signal through the weighting filter, a last pitch pulse of the sound signal of the previous frame is located from the residual signal, a pitch pulse prototype of given length is extracted around the position of the last pitch pulse of the sound signal of the previous frame using the synthesized weighted sound signal, and the pitch pulses are located in a current frame using the pitch pulse prototype.
摘要翻译：为了确定在使用用于数字编码声音信号的信号修改的技术中表征长期预测的长期预测延迟参数，声音信号被分成一系列连续的帧，声音信号的特征位于前一帧，声音信号的对应特征位于当前帧中，并且为当前帧确定长期预测延迟参数，同时长期预测将前一帧的信号特征与当前帧的相应信号特征。在用于实现用于对声音信号进行数字编码的技术的信号修改方法中，声音信号被分成一系列连续的帧，声音信号的每个帧被划分为多个信号段，并且至少部分框架的信号段扭曲，同时约束框架内的翘曲的信号段。为了在声音信号中搜索音调脉冲，通过线性预测分析滤波器对声音信号进行滤波来产生残留信号，通过加权滤波器处理声音信号产生加权声音信号，加权声音信号表示信号通过对通过加权滤波器的声音信号的先前帧的最后一个子帧产生的合成语音信号进行滤波，产生合成加权声音信号，前一帧的声音信号的最后音调脉冲位于剩余信号，使用合成的加权声音信号在前一帧的声音信号的最后音调脉冲的位置周围提取给定长度的音调脉冲原型，并且使用音调脉冲原型将音调脉冲位于当前帧中。

7. 发明申请

WO2009109050A1 SYSTEM AND METHOD FOR ENHANCING A DECODED TONAL SOUND SIGNAL 审中-公开
标题翻译：用于增强解码的声音信号的系统和方法
公开(公告)号：WO2009109050A1
公开(公告)日：2009-09-11
申请号：PCT/CA2009000276
申请日：2009-03-05
申请人： VOICEAGE CORP , VAILLANCOURT TOMMY , JELINEK MILAN , MALENOVSKY VLADIMIR , SALAMI REDWAN
发明人： VAILLANCOURT TOMMY , JELINEK MILAN , MALENOVSKY VLADIMIR , SALAMI REDWAN
IPC分类号： G10L21/02 , G10L19/12
CPC分类号： G10L19/26 , G10L25/18
摘要： A system and method for enhancing a tonal sound signal decoded by a decoder of a speech-specific codec in response to a received coded bit stream, in which a spectral analyser is responsive to the decoded tonal sound signal to produce spectral parameters representative of the decoded tonal sound signal. A quantization noise in low-energy spectral regions of the decoded tonal sound signal is reduced in response to the spectral parameters produced by the spectral analyser. The spectral analyser divides a spectrum resulting from spectral analysis into a set of critical frequency bands each comprising a number of frequency bins, and the reducer of quantization noise comprises a noise attenuator that scales the spectrum of the decoded tonal sound signal per critical frequency band, per frequency bin, or per both critical frequency band and frequency bin.
摘要翻译：一种用于响应于接收的编码比特流来增强由语音专用编解码器的解码器解码的音调声音信号的系统和方法，其中频谱分析仪响应于解码的音调声音信号以产生表示解码的频谱参数音调声信号。响应于由光谱分析仪产生的光谱参数，解码的音调声音信号的低能谱区域中的量化噪声被减小。光谱分析仪将由光谱分析得到的光谱分成一组包括多个频率仓的临界频带，并且量化噪声的衰减器包括噪声衰减器，其对每个关键频带的解码音调声音信号的频谱进行缩放，每个频率仓，或每个临界频带和频率仓。

8. 发明申请

WO2004034376A3 METHODS FOR INTEROPERATION BETWEEN ADAPTIVE MULTI-RATE WIDEBAND (AMR-WB) AND MULTI-MODE VARIABLE BIT-RATE WIDEBAND (WMR-WB) SPEECH CODECS 审中-公开
标题翻译：用于自适应多速率宽带（AMR-WB）和多模式可变比特率宽带（WMR-WB）语音编码器之间的交互的方法
公开(公告)号：WO2004034376A3
公开(公告)日：2004-06-10
申请号：PCT/CA0301572
申请日：2003-10-10
申请人： VOICEAGE CORP , JELINEK MILAN , SALAMI REDWAN
发明人： JELINEK MILAN , SALAMI REDWAN
IPC分类号： G01L19/14 , G10L11/04 , G10L19/02 , G10L19/14 , G10L21/02
CPC分类号： G10L19/24 , G10L19/012 , G10L19/173
摘要： A source-controlled Variable bit-rate Multi-mode WideBand (VMR-WB) speech codec, having a mode of operation that is interoperable with the Adaptive Multi-Rate wideband (AMR-WB) codec, the codec comprising: at least one Interoperable full-rate (1-FR) mode, having a first bit allocation structure based an one of a AMR-WB codec coding types; and at least one comfort noise generator (CNG) coding type for encoding inactive speech frame having a second bit allocation structure based on AMR-WB SID_UPDATE coding type. Methods for i) digitally encoding a sound using a source-controlled Variable bit rate multi-mode wideband (VMR-WB) speech codec for interoperation with an adaptative multi-rate wideband (AMR-WB) codec, ii) translating a Variable bit rate multi-mode wideband (VMR-WB) speech codec-signal frame into an Adaptive Multi-Rate wideband (AMR-WB) speech signal frame, iii) translating an Adaptive Multi-Rate wideband (AMR-WB) speech signal frame into a Variable bit rate multi-mode wideband (VMR-WB) speech signal frame, and iv) translating an Adaptive Multi-Rate wideband (AMR-WB) speech signal frame into a Variable bit rate multi-mode wideband (VMR-WB) speech signal frame are also provided.
摘要翻译：一种具有与自适应多速率宽带（AMR-WB）编解码器相互操作的操作模式的源控制的可变比特率多模式宽带（VMR-WB）语音编解码器，所述编解码器包括：至少一个可互操作的全速率（1-FR）模式，具有基于AMR-WB编解码器类型之一的第一比特分配结构; 以及至少一种用于编码基于AMR-WB SID_UPDATE编码类型的具有第二位分配结构的无效语音帧的舒适噪声发生器（CNG）编码类型。用于i）使用源控制的可变比特率多模宽带（VMR-WB）语音编解码器对数字编码声音的方法，用于与适应性多速率宽带（AMR-WB）编解码器进行互操作，ii）将可变比特率多模宽带（VMR-WB）语音编解码信号帧转换为自适应多速率宽带（AMR-WB）语音信号帧，iii）将自适应多速率宽带（AMR-WB）语音信号帧转换为变量比特率多模宽带（VMR-WB）语音信号帧，以及iv）将自适应多速率宽带（AMR-WB）语音信号帧转换为可变比特率多模宽带（VMR-WB）语音信号帧也提供。

9. 发明申请

WO2004006226B1 METHOD AND DEVICE FOR EFFICIENT IN-BAND DIM-AND-BURST SIGNALING AND HALF-RATE MAX OPERATION IN VARIABLE BIT-RATE WIDEBAND SPEECH CODING FOR CDMA WIRELESS SYSTEMS 审中-公开
标题翻译：用于CDMA无线系统的可变位速率宽带语音编码中的有效带内DIM-AND-BURST信令和高达率最大值操作的方法和设备
公开(公告)号：WO2004006226B1
公开(公告)日：2004-03-04
申请号：PCT/CA0300980
申请日：2003-06-27
申请人： VOICEAGE CORP , JELINEK MILAN , SALAMI REDWAN
发明人： JELINEK MILAN , SALAMI REDWAN
IPC分类号： G10L19/12 , G10L19/24 , H03M7/30 , H04B1/707 , H04B7/24 , H04B7/26 , G10L19/14 , H04Q7/30
CPC分类号： G10L19/24
摘要： In the method and device for interoperating a first station using a first communication scheme and comprising a first coder and a first decoder with a second station using a second communication scheme and comprising a second coder and a second decoder, communication between the first and second stations is conducted by transmitting signal-coding parameters related to a sound signal from the coder of one of the first and second stations to the decoder of the other station. The sound signal is classified to determine whether the signal-coding parameters should be transmitted from the coder of one station to the decoder of the other station using a first communication mode in which full bit rate is used for transmission of the signal-coding parameters. When classification of the sound signal determines that the signal-coding parameters should be transmitted using the first communication mode and when a request to transmit the signal-coding parameters from the coder of one station to the decoder of the other station using a second communication mode designed to reduce bit rate during transmission of the signal-coding parameters is received, a portion of the signal-coding parameters from the coder one station is dropped and the remaining signal-coding parameters are transmitting to the decoder of the other station using the second communication mode. The dropped portion of the signal-coding parameters are regenerated before the decoder of the other station decodes the signal-coding parameters.
摘要翻译：在用于使用第一通信方案互操作第一站的方法和设备中，包括第一编码器和具有第二站的第一解码器，并且包括第二编码器和第二解码器，第一和第二站之间的通信通过将与来自第一和第二站中的一个的编码器的声音信号相关的信号编码参数发送到另一站的解码器来进行。声音信号被分类以确定信号编码参数是否应当使用全位比特率用于传输信号编码参数的第一通信模式从一个站的编码器发送到另一站的解码器。当声音信号的分类确定应当使用第一通信模式发送信号编码参数时，以及当使用第二通信模式从一个站的编码器向另一站的解码器发送信号编码参数的请求时被设计为在信号编码参数的传输期间降低比特率被接收到，来自编码器一个站的信号编码参数的一部分被丢弃，剩下的信号编码参数使用第二个信号编码参数传送到另一台的解码器通讯模式。信号编码参数的丢弃部分在另一站的解码器解码信号编码参数之前被再生。

10. 发明专利

CS179556B1 SIZING COMPOSITION FOR SIZING NATURAL AND/OR SYNTHETIC TEXTILE MATERIALS 未知
公开(公告)号：CS179556B1
公开(公告)日：1977-11-30
申请号：CS718874
申请日：1974-10-21
申请人： JELINEK MILAN , SMID JOSEF , HUDECEK ZDENEK , ULMER KAREL
发明人： JELINEK MILAN , SMID JOSEF , HUDECEK ZDENEK , ULMER KAREL
IPC分类号： D06M15/03 , D06M15/263 , D06M15/285 , D06M15/04 , D06M15/38

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