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1. 发明申请

US20140156284A1 AUDIO-ENCODING METHOD AND APPARATUS, AUDIO-DECODING METHOD AND APPARATUS, RECODING MEDIUM THEREOF, AND MULTIMEDIA DEVICE EMPLOYING SAME 有权
标题翻译：音频编码方法和装置，音频解码方法和装置，其中介及其使用的多媒体装置
公开(公告)号：US20140156284A1
公开(公告)日：2014-06-05
申请号：US14123359
申请日：2012-06-01
申请人： Anton Porov , Konstantin Osipov , Ki-hyun Choo
发明人： Anton Porov , Konstantin Osipov , Ki-hyun Choo
IPC分类号： G10L19/008
CPC分类号： G10L19/0017 , G10L19/002 , G10L19/008 , G10L19/0204 , G10L19/032 , G10L19/167
摘要： Provided is an audio encoding method. The audio encoding method includes: acquiring envelopes based on a predetermined sub-band for an audio spectrum; quantizing the envelopes based on the predetermined sub-band; and obtaining a difference value between quantized envelopes for adjacent sub-bands and lossless encoding a difference value of a current sub-band by using a difference value of a previous sub-band as a context. Accordingly, the number of bits required to encode envelope information of an audio spectrum may be reduced in a limited bit range, thereby increasing the number of bits required to encode an actual spectral component.
摘要翻译：提供了一种音频编码方法。音频编码方法包括：基于用于音频频谱的预定子带获取信封; 基于预定的子带量化信封; 并且通过使用前一子带的差值作为上下文，获得相邻子带的量化包络之间的差值，并对当前子带的差值进行无损编码。因此，编码音频频谱的包络信息所需的比特数可以在有限的比特范围内减少，由此增加编码实际频谱分量所需的比特数。

2. 发明授权

US08086446B2 Method and apparatus for non-overlapped transforming of an audio signal, method and apparatus for adaptively encoding audio signal with the transforming, method and apparatus for inverse non-overlapped transforming of an audio signal, and method and apparatus for adaptively decoding audio signal with the inverse transforming 有权
标题翻译：用于音频信号的非重叠变换的方法和装置，用于对音频信号进行逆非重叠变换的变换，方法和装置对音频信号进行自适应编码的方法和装置，以及用于使用音频信号自适应地解码音频信号的方法和装置逆变换
公开(公告)号：US08086446B2
公开(公告)日：2011-12-27
申请号：US11295648
申请日：2005-12-07
申请人： Eunmi Oh , Junghoe Kim , Boris Kudryashov , Konstantin Osipov
发明人： Eunmi Oh , Junghoe Kim , Boris Kudryashov , Konstantin Osipov
IPC分类号： G10L19/02
CPC分类号： G10L19/022
摘要： A method and apparatus for transforming an audio signal, a method and apparatus for adaptively encoding an audio signal, a method and apparatus for inversely transforming an audio signal, and a method and apparatus for adaptively decoding an audio signal. The method of transforming an audio signal includes determining a transform unit into which the audio signal in a time domain is to be transformed into an audio signal in a frequency domain, and transforming the audio signal into an audio signal in the frequency domain according to the determined transform units using a window coefficient other than 0. Accordingly, it is possible to minimize distortion of the audio signal when encoding the audio signal even at a high bit rate while increasing efficiency of compression.
摘要翻译：用于变换音频信号的方法和装置，用于对音频信号进行自适应编码的方法和装置，用于对音频信号进行逆变换的方法和装置，以及用于对音频信号进行自适应解码的方法和装置。变换音频信号的方法包括：确定将时域中的音频信号变换为频域的音频信号的变换单元，并根据所述频域将音频信号变换为频域中的音频信号使用除了0以外的窗系数的确定的变换单元。因此，即使以高比特率对音频信号进行编码也可以最小化音频信号的失真，同时提高压缩效率。

3. 发明申请

US20070016404A1 Method and apparatus to extract important spectral component from audio signal and low bit-rate audio signal coding and/or decoding method and apparatus using the same 失效
标题翻译：从音频信号和低比特率音频信号编码和/或解码方法和装置中提取重要频谱分量的方法和装置
公开(公告)号：US20070016404A1
公开(公告)日：2007-01-18
申请号：US11480897
申请日：2006-07-06
申请人： Junghoe Kim , Eunmi Oh , Konstantin Osipov , Boris Kudryashov
发明人： Junghoe Kim , Eunmi Oh , Konstantin Osipov , Boris Kudryashov
IPC分类号： G10L19/00
CPC分类号： G10L19/035 , G10L19/0017 , G10L19/02 , G10L19/0204 , G10L19/0212
摘要： An method and apparatus to extract an audio signal having an important spectral component (ISC) and a low bit-rate audio signal coding/decoding method using the method and apparatus to extract the ISC. The method of extracting the ISC includes calculating perceptual importance including an SMR (signal-to-mark ratio) value of transformed spectral audio signals by using a psychoacoustic model, selecting spectral signals having a masking threshold value smaller than that of the spectral audio signals using the SMR value as first ISCs, and extracting a spectral peak from the audio signals selected as the ISCs according to a predetermined weighting factor to select second ISCs. Accordingly, the perceptual important spectral components can be efficiently coded so as to obtain high sound quality at a low bit-rate. In addition, it is possible to extract the perceptual important spectral component by using the psychoacoustic model, to perform coding without phase information, and to efficiently represent a spectral signal at a low bit-rate. In addition, the methods and apparatus can be employed in all the applications requiring a low bit-rate audio coding scheme and in a next generation audio scheme.
摘要翻译：一种提取具有重要频谱分量（ISC）的音频信号和使用该方法和装置提取ISC的低比特率音频信号编码/解码方法的方法和装置。提取ISC的方法包括通过使用心理声学模型来计算感知重要性，包括变换的频谱音频信号的SMR（信号到标记比）值，选择具有小于频谱音频信号的掩蔽阈值的光谱信号，使用 SMR值作为第一ISC，并且根据预定加权因子从被选择为ISC的音频信号中提取频谱峰值以选择第二ISC。因此，感知重要的频谱分量可以被有效地编码，以便以低比特率获得高音质。此外，可以通过使用心理声学模型来提取感知重要的频谱分量，以执行没有相位信息的编码，并且以低比特率有效地表示频谱信号。此外，该方法和装置可以用于需要低比特率音频编码方案和下一代音频方案的所有应用中。

4. 发明授权

US08861738B2 Method, medium, and system encoding/decoding multi-channel signal 有权
标题翻译：方法，中，系统编码/解码多通道信号
公开(公告)号：US08861738B2
公开(公告)日：2014-10-14
申请号：US13563808
申请日：2012-08-01
申请人： Jung-hoe Kim , Eun-mi Oh , Konstantin Osipov , Ki-hyun Choo
发明人： Jung-hoe Kim , Eun-mi Oh , Konstantin Osipov , Ki-hyun Choo
IPC分类号： H04H20/47 , H04H40/36 , H04H20/88 , G10L19/008
CPC分类号： H04H40/36 , G10L19/008 , H04H20/47 , H04H20/88
摘要： A multi-channel signal decoding method is provided. A down-mixed signal representative of a multi-channel signal is decoded, and parameters representing characteristic relations between channels of the multi-channel signal are decoded. An additional parameter is estimated by using the decoded parameters, and the decoded down-mixed signal is up-mixed by using the decoded parameters and the estimated parameter so as to decode the multi-channel signal.
摘要翻译：提供了一种多通道信号解码方法。解码表示多声道信号的下混信号，并对表示多声道信号的声道之间的特征关系的参数进行解码。通过使用解码参数来估计附加参数，并且通过使用解码参数和估计参数对解码的下混合信号进行混合，以解码多声道信号。

5. 发明授权

US07965206B2 Apparatus and method of lossless coding and decoding 有权
标题翻译：无损编码和解码的装置和方法
公开(公告)号：US07965206B2
公开(公告)日：2011-06-21
申请号：US12432811
申请日：2009-04-30
申请人： Ki Hyun Choo , Konstantin Osipov , Boris Kudryashov
发明人： Ki Hyun Choo , Konstantin Osipov , Boris Kudryashov
IPC分类号： H03M7/40
CPC分类号： H03M7/40 , H03M7/4006
摘要： An apparatus and a method of lossless coding and decoding are provided. The apparatus to perform lossless coding may selectively perform an arithmetic coding scheme or a Huffman coding scheme with respect to a symbol. The apparatus to perform lossless coding may generate a bitstream including a first coding bit, generated according to the Huffman coding scheme. Such bitstream may include a reserved bit for the arithmetic coding scheme.
摘要翻译：提供了无损编码和解码的装置和方法。执行无损编码的装置可以选择性地执行关于符号的算术编码方案或霍夫曼编码方案。执行无损编码的装置可以生成包括根据霍夫曼编码方案生成的第一编码比特的比特流。这样的比特流可以包括用于算术编码方案的保留位。

6. 发明申请

US20120294448A1 METHOD, MEDIUM, AND SYSTEM ENCODING/DECODING MULTI-CHANNEL SIGNAL 有权
公开(公告)号：US20120294448A1
公开(公告)日：2012-11-22
申请号：US13563808
申请日：2012-08-01
申请人： Jung-hoe KIM , Eun-mi OH , Konstantin OSIPOV , Ki-hyun CHOO
发明人： Jung-hoe KIM , Eun-mi OH , Konstantin OSIPOV , Ki-hyun CHOO
IPC分类号： H04R5/00
CPC分类号： H04H40/36 , G10L19/008 , H04H20/47 , H04H20/88
摘要： A multi-channel signal decoding method is provided. A down-mixed signal representative of a multi-channel signal is decoded, and parameters representing characteristic relations between channels of the multi-channel signal are decoded. An additional parameter is estimated by using the decoded parameters, and the decoded down-mixed signal is up-mixed by using the decoded parameters and the estimated parameter so as to decode the multi-channel signal.

7. 发明授权

US07973683B2 Apparatus and method for lossless coding and decoding 有权
标题翻译：无损编码和解码的装置和方法
公开(公告)号：US07973683B2
公开(公告)日：2011-07-05
申请号：US12320288
申请日：2009-01-22
申请人： Ki Hyun Choo , Konstantin Osipov , Anton V. Porov
发明人： Ki Hyun Choo , Konstantin Osipov , Anton V. Porov
IPC分类号： H03M7/00
CPC分类号： H03M7/40
摘要： A lossless coding and/or decoding apparatus and method. The lossless coding apparatus may read a probability model corresponding to each of a plurality of context groups. Here, the probability model stored in a memory may be generated by grouping a context. The lossless coding apparatus may code a symbol using the probability model and generate a bitstream. The lossless coding apparatus may enhance coding efficiency and reduce an amount of space utilized by the memory.
摘要翻译：一种无损编码和/或解码装置和方法。无损编码装置可以读取与多个上下文组中的每一个对应的概率模型。这里，可以通过对上下文进行分组来生成存储在存储器中的概率模型。无损编码装置可以使用概率模型对符号进行编码，并生成比特流。无损编码装置可以提高编码效率并减少存储器利用的空间量。

8. 发明授权

US07835907B2 Method and apparatus for low bit rate encoding and decoding 有权
标题翻译：用于低比特率编码和解码的方法和装置
公开(公告)号：US07835907B2
公开(公告)日：2010-11-16
申请号：US11312457
申请日：2005-12-21
申请人： Junghoe Kim , Eunmi Oh , Boris Kudryashov , Konstantin Osipov
发明人： Junghoe Kim , Eunmi Oh , Boris Kudryashov , Konstantin Osipov
IPC分类号： G10L19/00 , G10L11/00 , G10L21/02
CPC分类号： G10L19/028 , G10L19/0017 , G10L19/02
摘要： An apparatus and method of low bit rate encoding and reproducing. The method includes transforming input audio signals in a time domain into spectral signals in a frequency domain, extracting important-spectrum components from the spectral signals in the frequency domain, and quantizing the important-spectrum components, extracting residual-spectrum components other than the important-spectrum components from the spectral signals in the frequency domain, and calculating and quantizing a noise level of the residual-spectrum components, and encoding the quantized important-spectrum components and the quantized noise level losslessly, and outputting encoded bitstreams.
摘要翻译：低比特率编码和再现的装置和方法。该方法包括将时域中的输入音频信号变换为频域的频谱信号，从频域的频谱信号中提取重要的频谱分量，并对重要的频谱成分进行量化，提取重要的频谱成分频域中的频谱信号的频谱分量，以及计算和量化残余频谱分量的噪声电平，以及对量化的重要频谱分量和量化噪声电平进行无损编码，并输出编码比特流。

9. 发明申请

US20060122825A1 Method and apparatus for transforming audio signal, method and apparatus for adaptively encoding audio signal, method and apparatus for inversely transforming audio signal, and method and apparatus for adaptively decoding audio signal 有权
标题翻译：用于变换音频信号的方法和装置，用于对音频信号进行自适应编码的方法和装置，用于对音频信号进行逆变换的方法和装置，以及用于自适应地解码音频信号的方法和装置
公开(公告)号：US20060122825A1
公开(公告)日：2006-06-08
申请号：US11295648
申请日：2005-12-07
申请人： Eunmi Oh , Junghoe Kim , Boris Kudryashov , Konstantin Osipov
发明人： Eunmi Oh , Junghoe Kim , Boris Kudryashov , Konstantin Osipov
IPC分类号： G10L19/02
CPC分类号： G10L19/022
摘要： A method and apparatus for transforming an audio signal, a method and apparatus for adaptively encoding an audio signal, a method and apparatus for inversely transforming an audio signal, and a method and apparatus for adaptively decoding an audio signal. The method of transforming an audio signal includes determining a transform unit into which the audio signal in a time domain is to be transformed into an audio signal in a frequency domain, and transforming the audio signal into an audio signal in the frequency domain according to the determined transform units using a window coefficient other than 0. Accordingly, it is possible to minimize distortion of the audio signal when encoding the audio signal even at a high bit rate while increasing efficiency of compression.
摘要翻译：用于变换音频信号的方法和装置，用于对音频信号进行自适应编码的方法和装置，用于对音频信号进行逆变换的方法和装置，以及用于对音频信号进行自适应解码的方法和装置。变换音频信号的方法包括：确定将时域中的音频信号变换为频域的音频信号的变换单元，并根据所述频域将音频信号变换为频域中的音频信号使用除了0以外的窗系数的确定的变换单元。因此，即使以高比特率对音频信号进行编码也可以最小化音频信号的失真，同时提高压缩效率。

10. 发明申请

US20060004566A1 Low-bitrate encoding/decoding method and system 审中-公开
标题翻译：低比特率编码/解码方法和系统
公开(公告)号：US20060004566A1
公开(公告)日：2006-01-05
申请号：US11165569
申请日：2005-06-24
申请人： Eunmi Oh , Junghoe Kim , Sangwook Kim , Andrew Egorov , Anton Porov , Konstantin Osipov , Boris Kudryashov
发明人： Eunmi Oh , Junghoe Kim , Sangwook Kim , Andrew Egorov , Anton Porov , Konstantin Osipov , Boris Kudryashov
IPC分类号： G10L21/00
CPC分类号： G10L19/0204 , G10L19/0017 , G10L19/032
摘要： A low-bitrate encoding system includes: a time-frequency transform unit transforming an input time-domain audio signal into a frequency-domain audio signal; a frequency component processor unit decimating frequency components in the frequency-domain audio signal; a psychoacoustic model unit modeling the received time-domain audio signal on the basis of human auditory characteristics, and calculating encoding bit allocation information; a quantizer unit quantizing the frequency-domain audio signal input from the frequency component processor unit to have a bitrate based on the encoding bit allocation information input from the psychoacoustic model unit; and a lossless encoder unit encoding the quantized audio signal losslessly, and outputting the encoded audio signal in a bitstream format. Using the low-bitrate encoding system, it is possible to effectively compress data at a low bitrate, and thus to provide a high quality audio signal.
摘要翻译：低比特率编码系统包括：时间 - 频率变换单元，将输入的时域音频信号变换为频域音频信号; 频率分量处理器单元抽取频域音频信号中的频率分量; 心理声学模型单元，基于人类听觉特征对所接收的时域音频信号进行建模，以及计算编码比特分配信息; 量化器单元，根据从心理声学模型单元输入的编码比特分配信息量化从频率分量处理器单元输入的频域音频信号以具有比特率; 以及无损地编码量化音频信号的无损编码器单元，并以比特流格式输出编码音频信号。使用低比特率编码系统，可以以低比特率有效地压缩数据，从而提供高质量的音频信号。

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