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11. 发明授权

US09672832B2 Audio encoder, audio encoding method and program 有权
公开(公告)号：US09672832B2
公开(公告)日：2017-06-06
申请号：US13493850
申请日：2012-06-11
申请人： Yasuhiro Toguri , Yuuji Maeda , Jun Matsumoto , Shiro Suzuki , Yuuki Matsumura
发明人： Yasuhiro Toguri , Yuuji Maeda , Jun Matsumoto , Shiro Suzuki , Yuuki Matsumura
IPC分类号： H04R5/00 , G10L19/008 , H04S1/00 , H04B1/00
CPC分类号： G10L19/008 , H04S1/007 , H04S2400/09
摘要： There is provided an audio encoder comprising a determination part determining, based on frequency spectra of audio signals of a plurality of channels, a mixing ratio as a ratio, relative to a frequency spectrum after mixing for each channel of the plurality of channels, of the frequency spectrum for another channel, a mixing part mixing the frequency spectra of the plurality of channels for each channel based on the mixing ratio determined by the determination part, and an encoding part encoding the frequency spectra of the plurality of channels after mixing by the mixing part.

12. 发明授权

US08626501B2 Encoding apparatus, encoding method, decoding apparatus, decoding method, and program 有权
标题翻译：编码装置，编码方法，解码装置，解码方法和程序
公开(公告)号：US08626501B2
公开(公告)日：2014-01-07
申请号：US13303443
申请日：2011-11-23
申请人： Yasuhiro Toguri , Jun Matsumoto , Yuuji Maeda , Shiro Suzuki , Yuuki Matsumura
发明人： Yasuhiro Toguri , Jun Matsumoto , Yuuji Maeda , Shiro Suzuki , Yuuki Matsumura
IPC分类号： G10L21/00
CPC分类号： G10L19/035
摘要： An encoding apparatus includes a time-frequency transform unit that performs a time-frequency transform on an audio signal, a normalization unit that normalizes a frequency spectral coefficient obtained by the time-frequency transform in order to generate encoded data of the audio signal, a level calculation unit that calculates a level of the audio signal, a scale factor changing unit that changes a concealment scale factor included in encoded concealment data obtained by performing, on the basis of the level of the audio signal, a time-frequency transform and normalization on a minute noise signal, the concealment scale factor being a scale factor relating to a coefficient used for the normalization, and an output unit that outputs the encoded data of the audio signal generated by the normalization unit or outputs, as encoded data of the audio signal, the encoded concealment data whose concealment scale factor has been changed.
摘要翻译：编码装置包括对音频信号执行时间 - 频率变换的时间 - 频率变换单元，对通过时间 - 频率变换得到的频谱系数进行归一化的归一化单元，生成音频信号的编码数据，电平计算单元，其计算音频信号的电平;比例因子改变单元，其改变通过基于音频信号的电平执行时间 - 频率变换和归一化而获得的编码隐藏数据中包括的隐藏比例因子在微小噪声信号上，隐藏比例因子是与用于归一化的系数相关的比例因子，以及输出单元，其输出由归一化单元或输出产生的音频信号的编码数据作为音频的编码数据信号，其隐藏比例因子已经改变的编码隐藏数据。

13. 发明申请

US20130003980A1 AUDIO ENCODER, AUDIO ENCODING METHOD AND PROGRAM 有权
标题翻译：音频编码器，音频编码方法和程序
公开(公告)号：US20130003980A1
公开(公告)日：2013-01-03
申请号：US13493850
申请日：2012-06-11
申请人： Yasuhiro Toguri , Yuuji Maeda , Jun Matsumoto , Shiro Suzuki , Yuuki Matsumura
发明人： Yasuhiro Toguri , Yuuji Maeda , Jun Matsumoto , Shiro Suzuki , Yuuki Matsumura
IPC分类号： H04R5/00
CPC分类号： G10L19/008 , H04S1/007 , H04S2400/09
摘要： There is provided an audio encoder comprising a determination part determining, based on frequency spectra of audio signals of a plurality of channels, a mixing ratio as a ratio, relative to a frequency spectrum after mixing for each channel of the plurality of channels, of the frequency spectrum for another channel, a mixing part mixing the frequency spectra of the plurality of channels for each channel based on the mixing ratio determined by the determination part, and an encoding part encoding the frequency spectra of the plurality of channels after mixing by the mixing part.
摘要翻译：提供了一种音频编码器，其包括确定部件，其基于多个通道的音频信号的频谱，确定混合比率作为相对于多个通道中的每个通道混合后的频谱的比率用于另一通道的频谱，混合部分基于由确定部分确定的混合比混合用于每个通道的多个通道的频谱;以及编码部分，其通过混合混合之后编码多个通道的频谱部分。

14. 发明授权

US09015053B2 Audio coding device and audio coding method, audio decoding device and audio decoding method, and program 有权
标题翻译：音频编码装置和音频编码方法，音频解码装置和音频解码方法以及程序
公开(公告)号：US09015053B2
公开(公告)日：2015-04-21
申请号：US13616682
申请日：2012-09-14
申请人： Shiro Suzuki , Yuuki Matsumura , Yasuhiro Toguri , Yuuji Maeda , Jun Matsumoto
发明人： Shiro Suzuki , Yuuki Matsumura , Yasuhiro Toguri , Yuuji Maeda , Jun Matsumoto
IPC分类号： G10L19/00 , G10L19/008 , G10L19/022
CPC分类号： G10L19/008 , G10L19/022
摘要： There is provided an audio coding device including a first windowing part that multiplies an audio signal by a first window function, a second windowing part that multiplies the audio signal by a second window function having a characteristic different from a characteristic of the first window function, a window selecting part that selects the first window function or the second window function as an optimum window function based on the audio signal multiplied by the first windowing part and the audio signal multiplied by the second windowing part, a coding part that codes a frequency spectrum of the audio signal multiplied by the optimum window function, and a transmitting part that transmits the frequency spectrum coded by the coding part and window function information representing the optimum window function.
摘要翻译：提供了一种音频编码装置，包括：第一窗口部分，其将音频信号乘以第一窗口功能;第二窗口部分，其将音频信号乘以具有与第一窗口功能的特性不同的特征的第二窗口功能; 窗口选择部，其基于乘以第一加窗部的音频信号和乘以第二加窗部的音频信号，选择第一窗口功能或第二窗口功能作为最佳窗口功能;编码部，其对频谱进行编码乘以最佳窗口功能的发送部分，以及发送由编码部分编码的频谱的发送部分和表示最佳窗口功能的窗口功能信息。

15. 发明授权

US08977541B2 Speech processing apparatus, speech processing method and program 有权
标题翻译：语音处理装置，语音处理方法和程序
公开(公告)号：US08977541B2
公开(公告)日：2015-03-10
申请号：US13583839
申请日：2011-03-08
申请人： Yasuhiro Toguri , Shiro Suzuki , Jun Matsumoto , Yuuji Maeda , Yuuki Matsumura
发明人： Yasuhiro Toguri , Shiro Suzuki , Jun Matsumoto , Yuuji Maeda , Yuuki Matsumura
IPC分类号： G10L19/02 , G10L19/008
CPC分类号： G10L19/008 , G10L19/0212
摘要： The present invention relates to a speech processing apparatus, a speech processing method and a program which, when multichannel audio signals are downmixed and coded, prevent delay and an increase in the computation amount upon decoding of the audio signals. An inverse multiplexing unit (101) acquires coded data on which a BC parameter is multiplexed. An uncorrelated frequency-time transform unit (102) performs IMDCT transform and IMDST transform of frequency spectrum coefficients of a monaural signal (XM) obtained from this coded data to generate the monaural signal XM) which is a time domain signal and a signal (XD′) which is substantially uncorrelated with this monaural signal (XM). The stereo synthesis unit (103) generates a stereo signal by synthesizing the monaural signal (XM) and the signal (XD′) using the BC parameter. The present invention is applicable to, for example, a speech processing apparatus which decodes a downmixed and coded stereo signal.
摘要翻译：本发明涉及一种语音处理装置，语音处理方法和程序，当多声道音频信号被混合和编码时，防止音频信号解码时的延迟和计算量的增加。逆多路复用单元（101）获取多路复用BC参数的编码数据。不相关的频率 - 时间变换单元（102）执行从该编码数据获得的单声道信号（XM）的频谱系数的IMDCT变换和IMDST变换，以生成作为时域信号和信号（XD）的单声道信号XM '），其与该单声道信号（XM）基本上不相关。立体声合成单元（103）通过使用BC参数合成单声道信号（XM）和信号（XD'）来产生立体声信号。本发明可应用于例如解码下混合和编码的立体声信号的语音处理装置。

16. 发明授权

US08892429B2 Encoding device and encoding method, decoding device and decoding method, and program 有权
标题翻译：编码装置和编码方法，解码装置及解码方法及程序
公开(公告)号：US08892429B2
公开(公告)日：2014-11-18
申请号：US13583994
申请日：2011-03-08
申请人： Shiro Suzuki , Yuuki Matsumura , Yasuhiro Toguri , Yuuji Maeda
发明人： Shiro Suzuki , Yuuki Matsumura , Yasuhiro Toguri , Yuuji Maeda
IPC分类号： G10L19/02 , G10L19/035
CPC分类号： G10L19/035 , G10L19/0212
摘要： The present invention relates to an encoding device and an encoding method, a decoding device and a decoding method, and a program that reduce deterioration of sound quality due to encoding of audio signals.An envelope emphasis part (51) emphasizes an envelope (ENV). A noise shaping part (52) divides an emphasized envelope (D) formed by emphasis of the envelope (ENV) by a value larger than 1, and subtracts noise shaping (G) specified by information (NS) from a result of the division. A quantization part (14) sets a result of the subtraction as a quantization bit count (WL), and quantizes a normalized spectrum (S1) formed by normalization of a spectrum (S0) based on the quantization bit count (WL). A multiplexing part (53) multiplexes the information (NS), a quantized spectrum (QS) formed by quantization of the normalized spectrum (S1), and the envelope (ENV). The present invention can be applied to an encoding device encoding audio signals, for example.
摘要翻译：编码装置和编码方法，解码装置和解码方法技术领域本发明涉及一种减少音频信号编码导致的音质劣化的程序。信封重点部分（51）强调信封（ENV）。噪声整形部分（52）将由包络（ENV）的强调形成的强调包络（D）除以大于1的值，并从分割结果中减去由信息（NS）指定的噪声整形（G）。量化部分（14）将减法的结果设置为量化位计数（WL），并且通过基于量化位计数（WL）对通过频谱归一化形成的归一化频谱（S1）进行量化。复用部分（53）复用信息（NS），通过归一化频谱（S1）的量化形成的量化频谱（QS）和信封（ENV）。例如，本发明可以应用于编码音频信号的编码装置。

17. 发明申请

US20130096927A1 AUDIO CODING DEVICE AND AUDIO CODING METHOD, AUDIO DECODING DEVICE AND AUDIO DECODING METHOD, AND PROGRAM 有权
标题翻译：音频编码设备和音频编码方法，音频解码设备和音频解码方法及程序
公开(公告)号：US20130096927A1
公开(公告)日：2013-04-18
申请号：US13616682
申请日：2012-09-14
申请人： Shiro SUZUKI , Yuuki Matsumura , Yasuhiro Toguri , Yuuji Maeda , Jun Matsumoto
发明人： Shiro SUZUKI , Yuuki Matsumura , Yasuhiro Toguri , Yuuji Maeda , Jun Matsumoto
IPC分类号： G10L19/008
CPC分类号： G10L19/008 , G10L19/022
摘要： There is provided an audio coding device including a first windowing part that multiplies an audio signal by a first window function, a second windowing part that multiplies the audio signal by a second window function having a characteristic different from a characteristic of the first window function, a window selecting part that selects the first window function or the second window function as an optimum window function based on the audio signal multiplied by the first windowing part and the audio signal multiplied by the second windowing part, a coding part that codes a frequency spectrum of the audio signal multiplied by the optimum window function, and a transmitting part that transmits the frequency spectrum coded by the coding part and window function information representing the optimum window function.
摘要翻译：提供了一种音频编码装置，包括：第一窗口部分，其将音频信号乘以第一窗口功能;第二窗口部分，其将音频信号乘以具有与第一窗口功能的特性不同的特征的第二窗口功能; 窗口选择部，其基于乘以第一加窗部的音频信号和乘以第二加窗部的音频信号，选择第一窗口功能或第二窗口功能作为最佳窗口功能;编码部，其对频谱进行编码乘以最佳窗口功能的发送部分，以及发送由编码部分编码的频谱的发送部分和表示最佳窗口功能的窗口功能信息。

18. 发明申请

US20130006618A1 SPEECH PROCESSING APPARATUS, SPEECH PROCESSING METHOD AND PROGRAM 有权
标题翻译：语音处理设备，语音处理方法和程序
公开(公告)号：US20130006618A1
公开(公告)日：2013-01-03
申请号：US13583839
申请日：2011-03-08
申请人： Yasuhiro Toguri , Shiro Suzuki , Jun Matsumoto , Yuuji Maeda , Yuuki Matsumura
发明人： Yasuhiro Toguri , Shiro Suzuki , Jun Matsumoto , Yuuji Maeda , Yuuki Matsumura
IPC分类号： G10L19/02
CPC分类号： G10L19/008 , G10L19/0212
摘要： The present invention relates to a speech processing apparatus, a speech processing method and a program which, when multichannel audio signals are downmixed and coded, prevent delay and an increase in the computation amount upon decoding of the audio signals. An inverse multiplexing unit (101) acquires coded data on which a BC parameter is multiplexed. An uncorrelated frequency-time transform unit (102) performs IMDCT transform and IMDST transform of frequency spectrum coefficients of a monaural signal (XM) obtained from this coded data to generate the monaural signal XM) which is a time domain signal and a signal (XD′) which is substantially uncorrelated with this monaural signal (XM). The stereo synthesis unit (103) generates a stereo signal by synthesizing the monaural signal (XM) and the signal (XD′) using the BC parameter. The present invention is applicable to, for example, a speech processing apparatus which decodes a downmixed and coded stereo signal.
摘要翻译：本发明涉及一种语音处理装置，语音处理方法和程序，当多声道音频信号被混合和编码时，防止音频信号解码时的延迟和计算量的增加。逆多路复用单元（101）获取多路复用BC参数的编码数据。不相关的频率 - 时间变换单元（102）执行从该编码数据获得的单声道信号（XM）的频谱系数的IMDCT变换和IMDST变换，以生成作为时域信号和信号（XD）的单声道信号XM '），其与该单声道信号（XM）基本上不相关。立体声合成单元（103）通过使用BC参数合成单声道信号（XM）和信号（XD'）来产生立体声信号。本发明可应用于例如解码下混合和编码的立体声信号的语音处理装置。

19. 发明申请

US20120137191A1 DECODING DEVICE, DECODING METHOD, AND PROGRAM 有权
标题翻译：解码设备，解码方法和程序
公开(公告)号：US20120137191A1
公开(公告)日：2012-05-31
申请号：US13301542
申请日：2011-11-21
申请人： Yuuji Maeda , Jun Matsumoto , Yasuhiro Toguri , Shiro Suzuki , Yuuki Matsumura
发明人： Yuuji Maeda , Jun Matsumoto , Yasuhiro Toguri , Shiro Suzuki , Yuuki Matsumura
IPC分类号： H03M13/00 , G06F11/08
CPC分类号： G10L19/005 , G10L19/0212 , G10L25/06 , G10L25/90
摘要： A decoding device including a decoding unit which decodes encoded data, an inverse orthogonal transformation unit which performs inverse orthogonal transformation for the encoded data and obtains a time series waveform element in a unit of blocks, a correlation calculation unit which obtains a correlation between a time series waveform element of a block arranged immediately before an error block which is a block in which an error has occurred during decoding by the decoding unit and a time series waveform element of a block arranged a predetermined number of blocks before the block, a cycle calculation unit which obtains a basic cycle of a block unit of the error block based on the correlation obtained by the correlation calculation unit, and a generation unit which generates a substitute signal of the time series waveform element of the error block.
摘要翻译：一种解码装置，包括对编码数据进行解码的解码单元，对编码数据执行逆正交变换的逆正交变换单元，以块为单位求出时间序列波形要素;相关计算单元，紧接在作为在解码单元进行解码期间发生错误的块的错误块之前布置的块的串联波形元素以及在块前排列预定数量的块的块的时间序列波形元素，周期计算单元，其基于由相关计算单元获得的相关性获得误差块的块单位的基本周期;以及生成单元，其生成误差块的时间序列波形元素的替代信号。

20. 发明授权

US08311816B2 Noise shaping for predictive audio coding apparatus 有权
标题翻译：预测音频编码装置的噪声整形
公开(公告)号：US08311816B2
公开(公告)日：2012-11-13
申请号：US12639676
申请日：2009-12-16
申请人： Yasuhiro Toguri , Jun Matsumoto
发明人： Yasuhiro Toguri , Jun Matsumoto
IPC分类号： G10L21/02 , G10L19/00
CPC分类号： H04B14/068 , H03M7/3046
摘要： An information coding apparatus includes a predictive signal generator that generates a predictive signal; a predictive residual signal generator that generates a predictive residual signal; a quantizer that quantizes a quantization input signal generated based on the predictive residual signal; a quantization error signal generator that generates a quantization error signal; a feedback signal generator that generates a feedback signal for controlling the frequency characteristic of the quantization noise after decoding based on the quantization error signal; and a quantization input signal generator that generates the quantization input signal. The feedback signal generator is configured by a pole-zero filter that includes a filter coefficient of an all-pole filter which is based on spectral envelope information estimated by the input audio signal, a parameter for adjusting a peak level in the frequency characteristic of the quantization noise caused by the all-pole filter, and the predictive filter coefficient.
摘要翻译：一种信息编码装置，包括产生预测信号的预测信号发生器; 产生预测残差信号的预测残差信号发生器; 量化器，其量化基于所述预测残差信号产生的量化输入信号; 产生量化误差信号的量化误差信号发生器; 反馈信号发生器，其基于量化误差信号生成用于控制解码之后的量化噪声的频率特性的反馈信号; 以及产生量化输入信号的量化输入信号发生器。反馈信号发生器由极零滤波器构成，该极零滤波器包括基于由输入音频信号估计的频谱包络信息的全极滤波器的滤波器系数，用于调整频率特性中的峰值电平的参数由全极滤波器引起的量化噪声和预测滤波器系数。

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