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1. 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。

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。

4. 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。

5. 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。

6. 发明授权

US08396707B2 Method and device for efficient quantization of transform information in an embedded speech and audio codec 失效
标题翻译：用于在嵌入式语音和音频编解码器中对变换信息进行有效量化的方法和装置
公开(公告)号：US08396707B2
公开(公告)日：2013-03-12
申请号：US12676399
申请日：2008-09-25
申请人： Tommy Vaillancourt , Redwan Salami
发明人： Tommy Vaillancourt , Redwan Salami
IPC分类号： G10L19/00
CPC分类号： G10L19/12 , G10L19/032 , G10L19/24
摘要： A method and device for coding an input sound signal in at least one lower layer and at least one upper layer of an embedded codec comprises, in the at least one lower layer, coding the input sound signal to produce coding parameters, wherein coding the input sound signal comprises producing a synthesized sound signal. An error signal is computed as a difference between the input sound signal and the synthesized sound signal and a spectral mask is calculated as a function of a minima of a spectrum related to the input sound signal. In the at least one upper layer, the error signal is coded to produce coding coefficients, the spectral mask is applied to the coding coefficients, and the masked coding coefficients are quantized. Applying the spectral mask to the coding coefficients reduces the quantization noise produced upon quantizing the coding coefficients.
摘要翻译：用于对至少一个下层和嵌入式编解码器的至少一个上层编码输入声音信号的方法和装置包括：在所述至少一个下层中编码所述输入声音信号以产生编码参数，其中编码所述输入声音信号包括产生合成的声音信号。误差信号被计算为输入声音信号和合成声音信号之间的差，并且根据与输入声音信号相关的频谱的最小值来计算频谱屏蔽。在至少一个上层中，对误差信号进行编码以产生编码系数，将频谱掩模应用于编码系数，并对掩蔽的编码系数进行量化。将频谱掩模应用于编码系数减少了量化编码系数时产生的量化噪声。

7. 发明申请

US20100292993A1 Method and Device for Efficient Quantization of Transform Information in an Embedded Speech and Audio Codec 失效
标题翻译：用于在嵌入式语音和音频编解码器中有效量化变换信息的方法和装置
公开(公告)号：US20100292993A1
公开(公告)日：2010-11-18
申请号：US12676399
申请日：2008-09-25
申请人： Tommy Vaillancourt , Redwan Salami
发明人： Tommy Vaillancourt , Redwan Salami
IPC分类号： G10L21/00
CPC分类号： G10L19/12 , G10L19/032 , G10L19/24
摘要： A method and device for coding an input sound signal in at least one lower layer and at least one upper layer of an embedded codec while reducing a quantization noise comprises, in the at least one lower layer, coding the input sound signal to produce coding parameters, wherein coding the input sound signal comprises producing a synthesized sound signal. An error signal is computed as a difference between the input sound signal and the synthesized sound signal and a spectral mask is calculated as a function of a spectrum related to the input sound signal. In the at least one upper layer, the error signal is coded to produce coding coefficients, the spectral mask is applied to the coding coefficients, and the masked coding coefficients are quantized. Applying the spectral mask to the coding coefficients reduces the quantization noise produced upon quantizing the coding coefficients. Therefore, a method and device for reducing the quantization noise produced during coding of the error signal in the at least one upper layer comprises providing the spectral mask and, in the at least one upper layer, applying the spectral mask to the coding coefficients prior to quantizing the coding coefficients.
摘要翻译：一种用于在减少量化噪声的同时，在至少一个下层和至少一个嵌入式编解码器的上层中对输入声音信号进行编码的方法和装置包括：在所述至少一个下层中编码所述输入声音信号以产生编码参数其中编码所述输入声音信号包括产生合成声音信号。误差信号被计算为输入声音信号和合成声音信号之间的差，并且根据与输入声音信号相关的频谱的函数计算频谱屏蔽。在至少一个上层中，对误差信号进行编码以产生编码系数，将频谱掩模应用于编码系数，并对掩蔽的编码系数进行量化。将频谱掩模应用于编码系数减少了量化编码系数时产生的量化噪声。因此，用于降低在至少一个上层中的误差信号的编码期间产生的量化噪声的方法和装置包括提供频谱掩模，并且在至少一个上层中，将频谱掩模应用于量化编码系数。

8. 发明授权

US09015038B2 Coding generic audio signals at low bitrates and low delay 有权
标题翻译：以低比特率和低延迟编码通用音频信号
公开(公告)号：US09015038B2
公开(公告)日：2015-04-21
申请号：US13280707
申请日：2011-10-25
申请人： Tommy Vaillancourt , Milan Jelinek
发明人： Tommy Vaillancourt , Milan Jelinek
IPC分类号： G10L11/04 , G10L19/20 , G10L19/02 , G10L19/08
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.
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。

9. 发明申请

US20120101813A1 Coding Generic Audio Signals at Low Bitrates and Low Delay 有权
标题翻译：以低比特率和低延迟编码通用音频信号
公开(公告)号：US20120101813A1
公开(公告)日：2012-04-26
申请号：US13280707
申请日：2011-10-25
申请人： Tommy Vaillancourt , Milan Jelinek
发明人： Tommy Vaillancourt , Milan Jelinek
IPC分类号： G10L11/06 , G10L19/00
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.
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。

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