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1. 发明申请

WO2008151408A8 DEVICE AND METHOD FOR FRAME ERASURE CONCEALMENT IN A PCM CODEC INTEROPERABLE WITH THE ITU-T RECOMMENDATION G.711 审中-公开
标题翻译：与ITU-T建议相互配合的PCM编解码器中的帧擦除保护的设备和方法G.711
公开(公告)号：WO2008151408A8
公开(公告)日：2009-03-05
申请号：PCT/CA2007002357
申请日：2007-12-24
申请人： VOICEAGE CORP , MALENOVSKY VLADIMIR , SALAMI REDWAN
发明人： MALENOVSKY VLADIMIR , SALAMI REDWAN
IPC分类号： G10L19/00 , G10L21/02
CPC分类号： G10L19/26 , G10L19/005 , G10L19/24 , G10L25/93
摘要： A device and method for resynchronization and recovery after frame erasure concealment of an encoded sound signal comprise decoding, in a current frame, a correctly received signal after the frame erasure. Frame erasure concealment is extended in the current frame using an erasure-concealed signal from a previous frame to produce an extended erasure-concealed signal. The extended erasure-concealed signal is correlated with the decoded signal in the current frame and the extended erasure- concealed signal is synchronized with the decoded signal in response to the correlation. A smooth transition is produced in the current frame from the synchronized extended erasure-concealed signal to the decoded signal.
摘要翻译：在编码声音信号的帧擦除隐藏之后重新同步和恢复的装置和方法包括在帧擦除之后在当前帧中解码正确接收的信号。帧擦除隐藏在当前帧中使用来自前一帧的擦除隐藏信号来扩展以产生扩展的擦除隐藏信号。扩展的擦除隐藏信号与当前帧中的解码信号相关，并且扩展的擦除隐藏信号响应于相关而与解码信号同步。在当前帧中从同步的扩展消隐隐藏信号到解码信号产生平滑的转变。

2. 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。

3. 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。

4. 发明申请

WO2010003253A1 VARIABLE BIT RATE LPC FILTER QUANTIZING AND INVERSE QUANTIZING DEVICE AND METHOD 审中-公开
标题翻译：可变位速率LPC滤波器量子和反相量化器件及方法
公开(公告)号：WO2010003253A1
公开(公告)日：2010-01-14
申请号：PCT/CA2009000980
申请日：2009-07-10
申请人： VOICEAGE CORP , BESSETTE BRUNO , GOURNAY PHILIPPE , SALAMI REDWAN
发明人： BESSETTE BRUNO , GOURNAY PHILIPPE , SALAMI REDWAN
IPC分类号： G10L19/12
CPC分类号： G10L19/06 , G10L19/18 , H03M7/3082
摘要： A device and a method for quantizing a LPC filter in the form of an input vector in a quantization domain, comprises a calculator of a first-stage approximation of the input vector, a subtractor of the first-stage approximation from the input vector to produce a residual vector, a calculator of a weighting function from the first-stage approximation, a warper of the residual vector with the weighting function, and a quantizer of the weighted residual vector to supply a quantized weighted residual vector. A device and a method for inverse quantizing of a LPC filter, comprises means for receiving coded indices representative of a first-stage approximation of a vector representative of the LPC filter in a quantization domain and of a quantized weighted residual version of the vector, a calculator of an inverse weighting function from the first-stage approximation, an inverse quantizer of the quantized weighted residual version of the vector to produce a weighted residual vector, a multiplier of the weighted residual vector by the inverse weighting function to produce a residual vector, and an adder of the first-stage approximation with the residual vector to produce the vector representative of the LPC filter in the quantization domain.
摘要翻译：在量化域中以输入向量的形式对LPC滤波器进行量化的装置和方法包括：输入向量的第一级逼近的计算器，从输入向量产生的第一级逼近的减法器一个残差矢量，一个来自第一阶段逼近的加权函数的计算器，具有加权函数的残差向量的整经器，以及加权残差向量的量化器，以提供量化的加权残差向量。用于LPC滤波器的逆量化的装置和方法包括用于接收代表代表量化域中的LPC滤波器的矢量的第一级逼近的编码索引和矢量的量化加权残差版本的装置，来自第一阶段近似的逆加权函数的计算器，用于产生加权残差向量的向量的量化加权残差版本的逆量化器，通过逆加权函数的加权残差向量的乘数以产生残差向量，以及具有残差矢量的第一级近似的加法器，以产生代表量化域中的LPC滤波器的向量。

5. 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。

6. 发明申请

WO2010040522A2 MULTI-RESOLUTION SWITCHED AUDIO ENCODING/DECODING SCHEME 审中-公开
标题翻译：多分辨率切换音频编码/解码方案
公开(公告)号：WO2010040522A2
公开(公告)日：2010-04-15
申请号：PCT/EP2009007205
申请日：2009-10-07
申请人： FRAUNHOFER GES FORSCHUNG , VOICEAGE CORP , NEUENDORF MAX , BAYER STEFAN , LECOMTE JEREMIE , FUCHS GUILLAUME , ROBILLIARD JUIIEN , RETTELBACH NIKOLAUS , NAGEL FREDERIK , GEIGER RAIF , MULTRUS MARKUS , GRILL BERNHARD , GOURNAY PHILIPPE , SALAMI REDWAN
发明人： NEUENDORF MAX , BAYER STEFAN , LECOMTE JEREMIE , FUCHS GUILLAUME , ROBILLIARD JUIIEN , RETTELBACH NIKOLAUS , NAGEL FREDERIK , GEIGER RAIF , MULTRUS MARKUS , GRILL BERNHARD , GOURNAY PHILIPPE , SALAMI REDWAN
IPC分类号： G10L19/14 , G10L19/022 , G10L19/24
CPC分类号： G10L19/24 , G10L19/022
摘要： An audio encoder for encoding an audio signal comprises a first coding branch (400), the first coding branch comprising a first converter (410) for converting a signal from a time domain into a frequency domain. Furthermore, the audio encoder comprises a second coding branch (500) comprising a second time/frequency converter (523). Additionally, a signal analyzer (300/525) for analyzing the audio signal is provided. The signal analyzer, on the hand, determines whether an audio portion is effective in the encoder output signal as a first encoded signal from the first encoding branch or as a second encoded signal from a second encoding branch. On the other hand, the signal analyzer determines a time/frequency resolution to be applied by the converters (410, 523) when generating the encoded signals. An output interface includes, in addition to the first encoded signal and the second encoded signal, a resolution information identifying the resolution used by the first time/frequency converter and used by the second time/frequency converter.
摘要翻译：用于编码音频信号的音频编码器包括第一编码分支（400），第一编码分支包括用于将来自时域的信号转换成频域的第一转换器（410）。此外，音频编码器包括包括第二时间/频率转换器（523）的第二编码分支（500）。另外，提供了用于分析音频信号的信号分析器（300/525）。手持信号分析仪确定音频部分是否在编码器输出信号中作为来自第一编码分支的第一编码信号或来自第二编码分支的第二编码信号有效。另一方面，当生成编码信号时，信号分析器确定要由转换器（410,523）应用的时间/频率分辨率。除了第一编码信号和第二编码信号之外，输出接口包括识别由第一时间/频率转换器使用并由第二时间/频率转换器使用的分辨率的分辨率信息。

7. 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。

8. 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。

9. 发明申请

WO9804046A3 ENHANCED ENCODING OF DTMF AND OTHER SIGNALLING TONES 审中-公开
标题翻译：增强DTMF和其他信令音的编码
公开(公告)号：WO9804046A3
公开(公告)日：1998-03-26
申请号：PCT/CA9700516
申请日：1997-07-17
申请人： UNIV SHERBROOKE , SALAMI REDWAN , LAFLAMME CLAUDE , ADOUL JEAN PIERRE
发明人： SALAMI REDWAN , LAFLAMME CLAUDE , ADOUL JEAN-PIERRE
IPC分类号： H03M7/42 , H04Q7/22
CPC分类号： G10L19/07 , G10L19/20 , G10L25/78 , G10L2019/0016
摘要： In a method and device for quantizing a spectrum vector, supplied at recurrent time intervals, to produce a spectrum index, a spectrum-vector quantization codebook including a first voice-signal quantization codebook portion and a second non-voice signal quantization codebook portion is used. Whether the spectrum vector represents a voice signal or a non-voice signal is detected. When the spectrum vector represents a voice signal, the first codebook portion is searched for quantizing the spectrum vector and producing the spectrum index. The second codebook portion is searched for quantizing the spectrum vector and producing the spectrum index when the spectrum vector represents a non-voice signal. According to a first alternative, the first and second codebook portions are searched by measuring a weighted distance between the spectrum vector and the entries of these two codebook portions, and the spectrum vector is considered as representing a non-voice signal when the smallest weighted distance is the weighted distance measured between the spectrum vector and one entry of the second codebook portion. According to a second alternative, a weighted distance is measured between the spectrum vector and the entries of the second codebook portion, and the spectrum vector is considered as representing a non-voice signal when the weighted distance measured between the spectrum vector and one entry of the second codebook portion is smaller than a predetermined threshold.

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