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1. 发明申请

US20120123770A1 METHOD AND APPARATUS FOR IMPROVING SOUND QUALITY 有权
标题翻译：改善声音质量的方法和装置
公开(公告)号：US20120123770A1
公开(公告)日：2012-05-17
申请号：US13230824
申请日：2011-09-12
申请人： Hong Goo Kang , Min Seok Choi , Ho Seon Shin
发明人： Hong Goo Kang , Min Seok Choi , Ho Seon Shin
IPC分类号： G10L21/02
CPC分类号： G10L21/0364
摘要： Disclosed is a method of improving a sound quality, including: receiving a transmission signal of a first user equipment; removing noise in the transmission signal using noise information of the first user equipment side; performing speech reinforcement with respect to the noise removed transmission signal using noise information of a second user equipment side; and transmitting the speech reinforced transmission signal to the second user equipment.
摘要翻译：公开了一种改善声音质量的方法，包括：接收第一用户设备的发送信号; 使用第一用户设备侧的噪声信息去除传输信号中的噪声; 使用第二用户设备侧的噪声信息对噪声消除的传输信号执行语音强化; 以及将所述语音加强的发送信号发送到所述第二用户设备。

2. 发明申请

US20110153335A1 METHOD AND APPARATUS FOR PROCESSING AUDIO SIGNALS 有权
标题翻译：处理音频信号的方法和装置
公开(公告)号：US20110153335A1
公开(公告)日：2011-06-23
申请号：US12994425
申请日：2009-05-25
申请人： Hyen-O Oh , Chang Heon Lee , Jeongook Song , Yang Won Jung , Hong Goo Kang
发明人： Hyen-O Oh , Chang Heon Lee , Jeongook Song , Yang Won Jung , Hong Goo Kang
IPC分类号： G10L19/00
CPC分类号： G10L19/08
摘要： An audio signal processing method is disclosed. The audio signal processing method includes receiving a residual and long term prediction information, performing inverse frequency mapping with respect to the residual to generate a synthesized residual, and performing long term synthesis based on the synthesized residual and the long term prediction information to generate a synthesized audio signal of a current frame, wherein the long term prediction information comprises a final prediction gain and a final pitch lag, the final pitch lag has a range starting with 0, and the long term synthesis is performed based on a synthesized audio signal of a frame comprising a preceding frame.
摘要翻译：公开了一种音频信号处理方法。音频信号处理方法包括接收残留和长期预测信息，对残差进行逆频率映射以产生合成残差，并且基于合成残差和长期预测信息执行长期合成以产生合成的音频信号，其中所述长期预测信息包括最终预测增益和最终音调滞后，所述最终音调滞后具有从0开始的范围，并且所述长期合成基于合成音频信号帧包括前一帧。

3. 发明申请

US20110040557A1 TRANSMITTER AND RECEIVER FOR SPEECH CODING AND DECODING BY USING ADDITIONAL BIT ALLOCATION METHOD 失效
标题翻译：通过使用附加位分配方法进行语音编码和解码的发送器和接收器
公开(公告)号：US20110040557A1
公开(公告)日：2011-02-17
申请号：US12915656
申请日：2010-10-29
申请人： Ho-Sang SUNG , Dae-Hwan Hwang , Dae-Hee Youn , Hong-Goo Kang , Young-Cheol Park , Ki-Seung Lee , Sung-Kyo Jung , Kyung-Tae Kim
发明人： Ho-Sang SUNG , Dae-Hwan Hwang , Dae-Hee Youn , Hong-Goo Kang , Young-Cheol Park , Ki-Seung Lee , Sung-Kyo Jung , Kyung-Tae Kim
IPC分类号： G10L19/00
CPC分类号： G10L19/24 , G10L21/038
摘要： The present invention relates to a transmitter and a receiver for speech coding and decoding by using an additional bit allocation method. The transmitter and the receiver according to the present invention realize a voice communication service of high quality by using additional bits permitted in system requirements while using a conventional speech coder as it is. In addition, the transmitter and the receiver according to the present invention have an advantage in that they enable insertion of additional quantization blocks while not changing the structure of the conventional standard speech coder, since they allocate additional bits by applying a multi-stage quantization procedure not in a speech signal domain but in a parameter domain.
摘要翻译：本发明涉及通过使用附加比特分配方法进行语音编码和解码的发射机和接收机。根据本发明的发射机和接收机按照原来使用传统语音编码器，通过使用系统要求中允许的附加位来实现高质量的语音通信服务。此外，根据本发明的发射机和接收机具有的优点在于，它们能够插入附加量化块，同时不改变常规标准语音编码器的结构，因为它们通过应用多级量化过程来分配附加比特不在语音信号域，而是在参数域中。

4. 发明申请

US20100070284A1 METHOD AND AN APPARATUS FOR PROCESSING A SIGNAL 有权
标题翻译：方法和处理信号的装置
公开(公告)号：US20100070284A1
公开(公告)日：2010-03-18
申请号：US12497375
申请日：2009-07-02
申请人： Hyen O. OH , Jeong Ook Song , Chang Heon Lee , Yang Won Jung , Hong Goo Kang
发明人： Hyen O. OH , Jeong Ook Song , Chang Heon Lee , Yang Won Jung , Hong Goo Kang
IPC分类号： G10L19/00
CPC分类号： G10L19/22 , G11B2020/00014
摘要： An apparatus for processing an encoded signal and method thereof are disclosed, by which an audio signal can be compressed and reconstructed in higher efficiency.An audio signal processing method includes the steps of identifying whether a type of an audio signal is a music using first type information, if the type of the audio signal is not the music signal, identifying whether the type of the audio signal is a speech signal or a mixed signal using second type information, and if the type of the audio signal is determined as either the speech signal or the mixed signal, reconstructing the audio signal according to a coding scheme applied per frame using coding identification information. If the type of the audio signal is the music signal, the first type information is received only. If the type of the audio signal is the speech signal or the mixed signal, both of the first type information and the second type information are received.Accordingly, various kinds of audio signals can be encoded/decoded in higher efficiency.
摘要翻译：公开了一种用于处理编码信号的装置及其方法，通过该装置可以以更高的效率对音频信号进行压缩和重构。音频信号处理方法包括以下步骤：如果音频信号的类型不是音乐信号，则识别音频信号的类型是否是使用第一类型信息的音乐，识别音频信号的类型是否是语音信号或使用第二类型信息的混合信号，并且如果音频信号的类型被确定为语音信号或混合信号，则根据使用编码识别信息的每帧应用的编码方案重建音频信号。如果音频信号的类型是音乐信号，则仅接收第一类型信息。如果音频信号的类型是语音信号或混合信号，则接收第一类型信息和第二类型信息。因此，可以更高效率地对各种音频信号进行编码/解码。

5. 发明授权

US07269553B2 Pseudo-cepstral adaptive short-term post-filters for speech coders 失效
标题翻译：用于语音编码器的伪倒频自适应短期后置滤波器
公开(公告)号：US07269553B2
公开(公告)日：2007-09-11
申请号：US10684852
申请日：2003-10-14
申请人： Hong-Goo Kang , Kim Hong Kook
发明人： Hong-Goo Kang , Kim Hong Kook
IPC分类号： G10L19/04 , G10L21/00
CPC分类号： G10L21/0208 , G10L19/04 , G10L21/0232
摘要： Methods and systems for filtering synthesized or reconstructed speech are implemented. A filter based on a set of linear predictive coding (LPC) coefficients is constructed by transforming the LPC coefficients to the pseudo-cepstrum, a domain existing between LPC domain and the line spectral frequency (LSF) domain. The resulting filter can emphasize spectral frequencies associated with various formants, or spectral peaks, of an inverse transfer function relating to the LPC coefficients, and can de-emphasize spectral frequencies associated with various spectral minima, or spectral valleys, of the inverse transfer function relating to the LPC coefficients.
摘要翻译：实现滤波合成或重构语音的方法和系统。通过将LPC系数变换到伪倒频谱（LPC域和线谱频率（LSF）域之间存在的域）来构造基于一组线性预测编码（LPC）系数的滤波器。所得到的滤波器可以强调与与LPC系数相关的逆传递函数的各种共振峰或者谱峰相关的频谱频率，并且可以去除与相关的逆传递函数的各种频谱最小值或频谱谷相关联的频谱频率到LPC系数。

6. 发明授权

US09020812B2 Audio signal processing method and device 有权
标题翻译：音频信号处理方法及装置
公开(公告)号：US09020812B2
公开(公告)日：2015-04-28
申请号：US13511331
申请日：2010-11-24
申请人： Hye Jeong Jeon , Dae Hwan Kim , Hong Goo Kang , Min Ki Lee , Byung Suk Lee , Gyu Hyeok Jeong
发明人： Hye Jeong Jeon , Dae Hwan Kim , Hong Goo Kang , Min Ki Lee , Byung Suk Lee , Gyu Hyeok Jeong
IPC分类号： G10L11/00 , G10L19/14 , G10L13/00 , G10L13/06 , G10L19/00 , G10L11/06 , G10L19/005 , G10L19/12
CPC分类号： G10L19/00 , G10L19/005 , G10L19/12 , G10L2019/0001
摘要： Disclosed is an audio signal processing method comprising the steps of: receiving an audio signal containing current frame data; generating a first temporary output signal for the current frame when an error occurs in the current frame data, by carrying out frame error concealment with respect to the current frame data a random codebook; generating a parameter by carrying out one or more of short-term prediction, long-term prediction and a fixed codebook search based on the first temporary output signal; and memory updating the parameter for the next frame; wherein the parameter comprises one or more of pitch gain, pitch delay, fixed codebook gain and a fixed codebook.
摘要翻译：公开了一种音频信号处理方法，包括以下步骤：接收包含当前帧数据的音频信号; 通过对当前帧数据执行随机码本的帧错误隐藏，当前帧数据发生错误时，产生当前帧的第一临时输出信号; 通过基于第一临时输出信号执行短期预测，长期预测和固定码本搜索中的一个或多个来产生参数; 和存储器更新下一帧的参数; 其中所述参数包括音调增益，音调延迟，固定码本增益和固定码本中的一个或多个。

7. 发明授权

US08930199B2 Method and an apparatus for processing an audio signal 有权
标题翻译：用于处理音频信号的方法和装置
公开(公告)号：US08930199B2
公开(公告)日：2015-01-06
申请号：US13391992
申请日：2010-09-17
申请人： Hyen-O Oh , Chang Heon Lee , Hong Goo Kang , Jeongook Song
发明人： Hyen-O Oh , Chang Heon Lee , Hong Goo Kang , Jeongook Song
IPC分类号： G10L19/04 , G10L19/00 , G10L19/18 , G10L19/022 , G10L19/16 , G11B20/00 , G11B20/10
CPC分类号： G10L19/00 , G10L19/022 , G10L19/04 , G10L19/167 , G10L19/18 , G11B20/00007 , G11B2020/00028 , G11B2020/10546
摘要： A method of processing an audio signal is disclosed. The present invention includes receiving, by an audio processing apparatus, coding identification information indicating whether to apply a first coding scheme or a second coding scheme to a current frame; when the coding identification information indicates that the second coding scheme is applied to the current frame, receiving window type information indicating a particular window for the current frame, from among a plurality of windows; identifying that a current window is a long stop window based on the window type information, wherein the long stop window is followed by only long window of a following frame, wherein the long stop window includes a gentle long stop window and a steep long stop window; and, when the first coding scheme is applied to a previous frame, applying the gentle long stop window to the current frame, wherein: the gentle long stop window comprise an ascending line with first slope, the steep long stop window comprise an ascending line with second slope, and, the first slope is gentler than the second slope.
摘要翻译：公开了一种处理音频信号的方法。本发明包括由音频处理装置接收指示是否对当前帧应用第一编码方案或第二编码方案的编码标识信息; 当编码识别信息指示第二编码方案被应用于当前帧时，从多个窗口中接收指示当前帧的特定窗口的窗口类型信息; 基于所述窗口类型信息识别当前窗口是长时间停止窗口，其中所述长停止窗口后面仅是后续框架的长窗口，其中所述长停止窗口包括平缓的长停止窗口和陡峭的长停止窗口 ; 并且当将第一编码方案应用于先前帧时，将柔和的长时间窗口应用于当前帧，其中：平缓的长停止窗口包括具有第一斜率的上行线，陡峭的长停止窗口包括具有第二斜坡，第一斜坡比第二斜坡慢。

8. 发明申请

US20120239408A1 METHOD AND AN APPARATUS FOR PROCESSING AN AUDIO SIGNAL 有权
标题翻译：用于处理音频信号的方法和装置
公开(公告)号：US20120239408A1
公开(公告)日：2012-09-20
申请号：US13391992
申请日：2010-09-17
申请人： Hyen-O Oh , Chang Heon Lee , Hong Goo Kang , Jeongook Song
发明人： Hyen-O Oh , Chang Heon Lee , Hong Goo Kang , Jeongook Song
IPC分类号： G10L19/04
CPC分类号： G10L19/00 , G10L19/022 , G10L19/04 , G10L19/167 , G10L19/18 , G11B20/00007 , G11B2020/00028 , G11B2020/10546
摘要： A method of processing an audio signal is disclosed. The present invention includes receiving, by an audio processing apparatus, coding identification information indicating whether to apply a first coding scheme or a second coding scheme to a current frame; when the coding identification information indicates that the second coding scheme is applied to the current frame, receiving window type information indicating a particular window for the current frame, from among a plurality of windows; identifying that a current window is a long stop window based on the window type information, wherein the long stop window is followed by only long window of a following frame, wherein the long stop window includes a gentle long stop window and a steep long stop window; and, when the first coding scheme is applied to a previous frame, applying the gentle long stop window to the current frame, wherein: the gentle long stop window comprise an ascending line with first slope, the steep long stop window comprise an ascending line with second slope, and, the first slope is gentler than the second slope.
摘要翻译：公开了一种处理音频信号的方法。本发明包括由音频处理装置接收指示是否对当前帧应用第一编码方案或第二编码方案的编码标识信息; 当编码识别信息指示第二编码方案被应用于当前帧时，从多个窗口中接收指示当前帧的特定窗口的窗口类型信息; 基于所述窗口类型信息识别当前窗口是长时间停止窗口，其中所述长停止窗口后面仅是后续框架的长窗口，其中所述长停止窗口包括平缓的长停止窗口和陡峭的长停止窗口 ; 并且当将第一编码方案应用于先前帧时，将柔和的长时间窗口应用于当前帧，其中：平缓的长停止窗口包括具有第一斜率的上行线，陡峭的长停止窗口包括具有第二斜坡，第一斜坡比第二斜坡慢。

9. 发明授权

US08259629B2 System and method for transmitting and receiving wideband speech signals with a synthesized signal 有权
标题翻译：用合成信号发送和接收宽带语音信号的系统和方法
公开(公告)号：US08259629B2
公开(公告)日：2012-09-04
申请号：US12631396
申请日：2009-12-04
申请人： Ho-Sang Sung , Dae-Hwan Hwang , Dae-Hee Youn , Hong-Goo Kang , Young-Cheol Park , Ki-Seung Lee , Sung-Kyo Jung , Kyung-Tae Kim
发明人： Ho-Sang Sung , Dae-Hwan Hwang , Dae-Hee Youn , Hong-Goo Kang , Young-Cheol Park , Ki-Seung Lee , Sung-Kyo Jung , Kyung-Tae Kim
IPC分类号： H04B3/20 , G10L19/00
CPC分类号： H04L12/66
摘要： A system for transmitting and receiving a wideband speech signal includes an A/D converter for receiving an analog speech signal to convert it into a digital speech signal, encoding a difference between a baseband speech signal and a standard baseband between a synthesized standard baseband signal, and an IP network interface for multiplexing the coded standard and additional baseband signals and enhancement residual band signal.
摘要翻译：用于发送和接收宽带语音信号的系统包括用于接收模拟语音信号以将其转换为数字语音信号的A / D转换器，对基带语音信号和合成标准基带信号之间的标准基带之间的差进行编码，以及用于复用编码标准和附加基带信号和增强剩余频带信号的IP网络接口。

10. 发明申请

US20120185257A1 METHOD AND AN APPARATUS FOR PROCESSING AN AUDIO SIGNAL 有权
标题翻译：用于处理音频信号的方法和装置
公开(公告)号：US20120185257A1
公开(公告)日：2012-07-19
申请号：US13387248
申请日：2010-07-27
申请人： Hyen-O Oh , Hong Goo Kang , Chang Heon Lee , Jeong Ook Song
发明人： Hyen-O Oh , Hong Goo Kang , Chang Heon Lee , Jeong Ook Song
IPC分类号： G10L19/00
CPC分类号： G10L19/04 , G10L19/005 , G10L19/022 , G10L19/18 , G10L25/45
摘要： An apparatus for processing an audio signal and method thereof are disclosed. The present invention includes receiving, by an audio processing apparatus, an audio signal including a first data of a first block encoded with rectangular coding scheme and a second data of a second block encoded with non-rectangular coding scheme; receiving a compensation signal corresponding to the second block; estimating a prediction of an aliasing part using the first data; and, obtaining a reconstructed signal for the second block based on the second data, the compensation signal and the prediction of aliasing part.
摘要翻译：公开了一种用于处理音频信号的装置及其方法。本发明包括由音频处理设备接收包括用矩形编码方案编码的第一块的第一数据和用非矩形编码方案编码的第二块的第二数据的音频信号; 接收对应于第二块的补偿信号; 使用所述第一数据估计混叠部分的预测; 并且基于第二数据，补偿信号和混叠部分的预测来获得第二块的重构信号。

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