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    • 1. 发明申请
    • CODING GENERIC AUDIO SIGNALS AT LOW BITRATES AND LOW DELAY
    • 编码低频和低延迟的一般音频信号
    • WO2012055016A1
    • 2012-05-03
    • PCT/CA2011001182
    • 2011-10-24
    • VOICEAGE CORPVAILLANCOURT TOMMYJELINEK MILAN
    • VAILLANCOURT TOMMYJELINEK MILAN
    • G10L19/12
    • G10L19/20G10L19/02G10L19/08
    • 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.
    • 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。
    • 2. 发明申请
    • METHODS AND DEVICES FOR SOURCE CONTROLLED VARIABLE BIT-RATE WIDEBAND SPEECH CODING
    • 用于源控制的可变比特率宽带语音编码的方法和设备
    • WO2004034379A3
    • 2004-12-23
    • PCT/CA0301571
    • 2003-10-09
    • NOKIA CORPJELINEK MILAN
    • JELINEK MILAN
    • G01L19/14G10L11/04G10L19/02G10L19/14G10L21/02
    • G10L19/24G10L19/012G10L19/173
    • 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 .
    • 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。
    • 3. 发明申请
    • METHODS AND DEVICES FOR SOURCE CONTROLLED VARIABLE BIT-RATE WIDEBAND SPEECH CODING
    • 源控制可变比特率宽带语音编码的方法和设备
    • WO2004034379A2
    • 2004-04-22
    • PCT/CA2003/001571
    • 2003-10-09
    • VOICEAGE CORPORATIONJELINEK, Milan
    • JELINEK, Milan
    • G10L19/00
    • G10L19/24G10L19/012G10L19/173
    • 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 .
    • 本文公开了语音信号分类和编码系统和方法。 信号分类通过三个步骤完成,每个步骤区分特定的信号类别。 首先,语音活动检测器(VAD)在有效和无效的语音帧之间进行区分。 如果检测到无效语音帧(背景噪声信号),则分类链结束,并且以舒适噪声产生(CNG)编码该帧。 如果检测到活动语音帧,则该帧经受专用于区分清音帧的第二分类器。 如果分类器将帧分类为无声语音信号,则分类链结束,并且使用针对无声信号优化的编码方法对帧进行编码。 否则,将语音帧传递到“稳定浊音”分类模块。 如果帧被分类为稳定的有声帧,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,该帧可能包含诸如有声开始或快速演进的有声语音信号之类的非平稳语音段。 在这种情况下,通用语音编码器以高比特率被使用以维持良好的主观质量。
    • 4. 发明申请
    • SIGNAL MODIFICATION METHOD FOR EFFICIENT CODING OF SPEECH SIGNALS
    • WO2003052744A3
    • 2003-06-26
    • PCT/CA2002/001948
    • 2002-12-13
    • VOICEAGE CORPORATIONTAMMI, MikkoJELINEK, MilanLAFLAMME, ClaudeRUOPPILA, Vesa
    • TAMMI, MikkoJELINEK, MilanLAFLAMME, ClaudeRUOPPILA, Vesa
    • G10L19/08
    • 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.
    • 5. 发明申请
    • METHOD AND DEVICE FOR EFFICIENT FRAME ERASURE CONCEALMENT IN SPEECH CODECS
    • 方法和设备在语音编码中有效的帧消除隐藏
    • WO2007073604A8
    • 2007-12-21
    • PCT/CA2006002146
    • 2006-12-28
    • VOICEAGE CORPVAILLANCOURT TOMMYJELINEK MILANGOURNAY PHILIPPESALAMI REDWAN
    • VAILLANCOURT TOMMYJELINEK MILANGOURNAY PHILIPPESALAMI REDWAN
    • G10L19/00G10L21/02
    • G10L19/005
    • 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.
    • 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。
    • 6. 发明申请
    • METHOD AND DEVICE FOR EFFICIENT FRAME ERASURE CONCEALMENT IN SPEECH CODECS
    • 语音编解码器中有效帧擦除隐藏的方法和装置
    • WO2007073604A1
    • 2007-07-05
    • PCT/CA2006/002146
    • 2006-12-27
    • VOICEAGE CORPORATIONVAILLANCOURT, TommyJELINEK, MilanGOURNAY, PhilippeSALAMI, Redwan
    • VAILLANCOURT, TommyJELINEK, MilanGOURNAY, PhilippeSALAMI, Redwan
    • G10L19/00G10L21/02
    • G10L19/005
    • 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.
    • 一种用于隐藏由在从编码器到解码器的传输期间擦除的编码声音信号的帧引起的帧擦除以及在帧擦除之后恢复解码器的方法和设备包括在编码器中确定隐藏 /恢复参数至少包括与编码声音信号的帧有关的相位信息。 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,响应于所接收的隐藏/恢复参数进行帧删除隐藏。 帧擦除隐藏包括响应于接收到的相位信息而将隐藏了擦除的帧与在编码器处编码的声音信号的对应帧再同步。 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已被擦除的编码声音信号的每帧的相位信息。 另外,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,将每个擦除隐藏帧与在该帧中编码的声音信号的对应帧重新同步 编码器。
    • 8. 发明申请
    • METHOD FOR INTEROPERATION BETWEEN ADAPTIVE MULTI-RATE WIDEBAND (AMR-WB) AND MULTI-MODE VARIABLE BIT-RATE WIDEBAND (VMR-WB) CODECS
    • 自适应多速率宽带(AMR-WB)与多模式可变比特率宽带(VMR-WB)编解码器之间的交互方法
    • WO2004034376A2
    • 2004-04-22
    • PCT/CA2003/001572
    • 2003-10-10
    • VOICEAGE CORPORATIONJELINEK, MilanSALAMI, Redwan
    • JELINEK, MilanSALAMI, Redwan
    • G10L
    • G10L19/24G10L19/012G10L19/173
    • A source-controlled Variable bit-rate Multi-mode WideBand (VMR-WB) 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 an 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) codec for interoperation with an adaptative multi-rate wideband (AMR-WB) codec, ii) translating a Variable bit rate multi-mode wideband (VMR-WB) codecsignal frame into an Adaptive Multi-Rate wideband (AMR-WB) signal frame, iii) translating an Adaptive Multi-Rate wideband (AMR-WB) signal frame into a Variable bit rate multi-mode wideband (VMR-WB) signal frame, and iv) translating an Adaptive Multi-Rate wideband (AMR-WB) signal frame into a Variable bit rate multi-mode wideband (VMR-WB) signal frame are also provided.
    • 源控制的可变比特率多模宽带(VMR-WB)编解码器,具有可与自适应多速率宽带(AMR-WB)编解码器互操作的操作模式,编解码器包括:至少一个可互操作的全 (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; 以及至少一种用于基于AMR-WB SID_UPDATE编码类型对具有第二比特分配结构的无效语音帧进行编码的舒适噪声发生器(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)编解码器对适应性多速率宽带(AMR-WB)编解码器进行互操作的数字编码声音的方法,ii)将可变比特率多 (VMR-WB)编解码信号帧转换为自适应多速率宽带(AMR-WB)信号帧,iii)将自适应多速率宽带(AMR-WB)信号帧转换为可变比特率多模宽带 (VMR-WB)信号帧,以及iv)将自适应多速率宽带(AMR-WB)信号帧转换为可变比特率多模宽带(VMR-WB)信号帧。
    • 9. 发明申请
    • SIGNAL MODIFICATION METHOD FOR EFFICIENT CODING OF SPEECH SIGNALS
    • 用于语音信号有效编码的信号修改方法
    • WO03052744A2
    • 2003-06-26
    • PCT/CA0201948
    • 2002-12-13
    • VOICEAGE CORPTAMMI MIKKOJELINEK MILANLAFLAMME CLAUDERUOPPILA VESA
    • TAMMI MIKKOJELINEK MILANLAFLAMME CLAUDERUOPPILA VESA
    • G10L19/12G10L19/08
    • G10L19/08
    • 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.
    • 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。
    • 10. 发明申请
    • METHOD AND DEVICE FOR CODING TRANSITION FRAMES IN SPEECH SIGNALS
    • 用于编码语音信号中的过渡帧的方法和装置
    • WO2008049221A1
    • 2008-05-02
    • PCT/CA2007/001896
    • 2007-10-24
    • VOICEAGE CORPORATIONEKSLER, VaclavJELINEK, MilanSALAMI, Redwan
    • EKSLER, VaclavJELINEK, MilanSALAMI, Redwan
    • G01L19/08G01L19/12
    • G10L19/08G10L19/12
    • There is provided a transition mode device and method for use in a predictive-type sound signal codec for producing a transition mode excitation replacing an adaptive codebook excitation in a transition frame and/or a frame following the transition in the sound signal, comprising an input for receiving a codebook index and a transition mode codebook for generating a set of codevectors independent from past excitation. The transition mode codebook is responsive to the index for generating, in the transition frame and/or frame following the transition, one of the codevectors of the set corresponding to the transition mode excitation. There is also provided an encoding device and method and a decoding device and method using the above described transition mode device and method.
    • 提供了一种用于预测型声音信号编解码器中的过渡模式装置和方法,用于产生在过渡帧和/或声音信号中的转变之后的帧中替换自适应码本激励的转换模式激励,包括输入 用于接收码本索引和用于生成独立于过去激励的一组码矢量的转换模式码本。 转换模式码本响应于索引,用于在转换之后的转换帧和/或帧中生成对应于转换模式激励的集合的码矢量之一。 还提供了一种编码装置和方法以及使用上述转换模式装置和方法的解码装置和方法。