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1. 发明授权

US08401845B2 System and method for enhancing a decoded tonal sound signal 有权
标题翻译：用于增强解码音调声音信号的系统和方法
公开(公告)号：US08401845B2
公开(公告)日：2013-03-19
申请号：US12918586
申请日：2009-03-05
申请人： Tommy Vaillancourt , Milan Jelinek , Vladimir Malenovsky , Redwan Salami
发明人： Tommy Vaillancourt , Milan Jelinek , Vladimir Malenovsky , Redwan Salami
IPC分类号： G10L21/02 , G10L19/14 , G10L19/00 , H04B15/00
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。光谱分析仪将从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，并且量化噪声的减法器包括噪声衰减器，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。

2. 发明授权

US08990073B2 Method and device for sound activity detection and sound signal classification 有权
标题翻译：声音检测和声音信号分类的方法和装置
公开(公告)号：US08990073B2
公开(公告)日：2015-03-24
申请号：US12664934
申请日：2008-06-20
申请人： Vladimir Malenovsky , Milan Jelinek , Tommy Vaillancourt , Redwan Salami
发明人： Vladimir Malenovsky , Milan Jelinek , Tommy Vaillancourt , Redwan Salami
IPC分类号： G10L21/00 , G10L25/78 , G10L25/00 , H04B15/00 , G10L19/22
CPC分类号： G10L25/78 , G10L19/22
摘要： A device and method for estimating a tonal stability of a sound signal include: 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 tonal stability in the sound signal.
摘要翻译：用于估计声音信号的音调稳定性的装置和方法包括：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来计算长期相关图，所述长期相关图表示所述声音信号中的音调稳定性。

3. 发明申请

US20110035213A1 Method and Device for Sound Activity Detection and Sound Signal Classification 有权
标题翻译：声音检测和声音信号分类方法和装置
公开(公告)号：US20110035213A1
公开(公告)日：2011-02-10
申请号：US12664934
申请日：2008-06-20
申请人： Vladimir Malenovsky , Milan Jelinek , Tommmy Vaillancourt , Redwan Salami
发明人： Vladimir Malenovsky , Milan Jelinek , Tommmy Vaillancourt , Redwan Salami
IPC分类号： G10L11/06
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。

4. 发明授权

US08255207B2 Method and device for efficient frame erasure concealment in speech codecs 有权
标题翻译：语音编解码器中高效帧擦除隐藏的方法和设备
公开(公告)号：US08255207B2
公开(公告)日：2012-08-28
申请号：US12095224
申请日：2006-12-27
申请人： Tommy Vaillancourt , Milan Jelinek , Philippe Gournay , Redwan Salami
发明人： Tommy Vaillancourt , Milan Jelinek , Philippe Gournay , Redwan Salami
IPC分类号： G10L15/00
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，响应于接收到的隐藏/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。

5. 发明申请

US20110125505A1 Method and Device for Efficient Frame Erasure Concealment in Speech Codecs 有权
标题翻译：用于语音编解码器中高效帧擦除隐藏的方法和设备
公开(公告)号：US20110125505A1
公开(公告)日：2011-05-26
申请号：US12095224
申请日：2006-12-28
申请人： Tommy Vaillancourt , Milan Jelinek , Philleppe Gournay , Redwan Salami
发明人： Tommy Vaillancourt , Milan Jelinek , Philleppe Gournay , Redwan Salami
IPC分类号： G10L19/00
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并且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，响应于接收到的隐藏/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

US20110046947A1 System and Method for Enhancing a Decoded Tonal Sound Signal 有权
标题翻译：用于增强解码音调声音信号的系统和方法
公开(公告)号：US20110046947A1
公开(公告)日：2011-02-24
申请号：US12918586
申请日：2009-03-05
申请人： Tommy Vaillancourt , Milan Jelinek , Vladimir Malenvosky , Redwan Salami
发明人： Tommy Vaillancourt , Milan Jelinek , Vladimir Malenvosky , Redwan Salami
IPC分类号： G10L21/02
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。光谱分析仪将从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，并且量化噪声的减法器包括噪声衰减器，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。

7. 发明授权

US07203638B2 Method for interoperation between adaptive multi-rate wideband (AMR-WB) and multi-mode variable bit-rate wideband (VMR-WB) codecs 有权
标题翻译：自适应多速率宽带（AMR-WB）和多模式可变比特率宽带（VMR-WB）编解码器之间的互操作方法
公开(公告)号：US07203638B2
公开(公告)日：2007-04-10
申请号：US11039540
申请日：2005-01-19
申请人： Milan Jelinek , Redwan Salami
发明人： Milan Jelinek , Redwan Salami
IPC分类号： G10L19/00
CPC分类号： G10L19/24 , G10L19/012 , G10L19/173
摘要： 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 (I-FR) mode, having a first bit allocation structure based on 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) 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.
摘要翻译：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（I-FR）模式，具有基于AMR-WB编解码编码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比特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）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。

8. 发明申请

US20100280831A1 Method and Device for Fast Algebraic Codebook Search in Speech and Audio Coding 有权
标题翻译：用于语音和音频编码的快速代数码本搜索的方法和设备
公开(公告)号：US20100280831A1
公开(公告)日：2010-11-04
申请号：US12676004
申请日：2008-09-11
申请人： Redwan Salami , Vaclav Eksler , Milan Jelinek
发明人： Redwan Salami , Vaclav Eksler , Milan Jelinek
IPC分类号： G10L21/00
CPC分类号： G10L19/107
摘要： A method and device for searching an algebraic codebook during encoding of a sound signal, wherein the algebraic codebook comprises a set of codevectors formed of a number of pulse positions and a number of pulses distributed over the pulse positions. In the algebraic codebook searching method and device, a reference signal for use in searching the algebraic codebook is calculated. In a first stage, a position of a first pulse is determined in relation with the reference signal and among the number of pulse positions. In each of a number of stages subsequent to the first stage, (a) an algebraic codebook gain is recomputed, (b) the reference signal is updated using the recomputed algebraic codebook gain and (c) a position of another pulse is determined in relation with the updated reference signal and among the number of pulse positions. A codevector of the algebraic codebook is computed using the positions of the pulses determined in the first and subsequent stages, wherein a number of the first and subsequent stages corresponds to the number of pulses in the codevectors of the algebraic codebook.
摘要翻译：一种用于在编码声音信号期间搜索代数码本的方法和装置，其中代数码本包括由多个脉冲位置和分布在脉冲位置上的多个脉冲形成的一组码矢。在代数码本搜索方法和装置中，计算用于搜索代数码本的参考信号。在第一阶段中，相对于参考信号和脉冲位置数确定第一脉冲的位置。在第一阶段之后的多个阶段的每个阶段中，（a）代数码本增益被重新计算，（b）使用重新计算的代数码本增益来更新参考信号，并且（c）另一个脉冲的位置被确定具有更新的参考信号和脉冲位置的数量。使用在第一级和后级中确定的脉冲的位置来计算代数码本的码矢量，其中第一级和后级的数量对应于代数码本的代码矢量中的脉冲数。

9. 发明申请

US20060100859A1 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 有权
标题翻译：用于cdma无线系统的可变比特率宽带语音编码中的有效带内暗和信令信令和半速率最大运算的方法和装置
公开(公告)号：US20060100859A1
公开(公告)日：2006-05-11
申请号：US10520374
申请日：2003-06-27
申请人： Milan Jelinek , Redwan Salami
发明人： Milan Jelinek , Redwan Salami
IPC分类号： G10L19/00
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. 发明申请

US20050251387A1 Method and device for gain quantization in variable bit rate wideband speech coding 有权
标题翻译：用于可变比特率宽带语音编码中增益量化的方法和装置
公开(公告)号：US20050251387A1
公开(公告)日：2005-11-10
申请号：US11039538
申请日：2005-01-19
申请人： Milan Jelinek , Redwan Salami
发明人： Milan Jelinek , Redwan Salami
IPC分类号： G10L19/08 , G10L19/14 , G10L19/12
CPC分类号： G10L19/083 , G10L19/24
摘要： The present invention relates to a gain quantization method and device for implementation in a technique for coding a sampled sound signal processed, during coding, by successive frames of L samples, wherein each frame is divided into a number of subframes and each subframe comprises a number N of samples, where N
摘要翻译：增益量化方法和装置技术领域本发明涉及一种增益量化方法和装置，用于在由L个采样的连续帧在编码期间对采样的声音信号进行编码的技术中实现，其中每个帧被划分成多个子帧，并且每个子帧包括数字 N个样本，其中N

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