公开(公告)号：US20080133242A1

公开(公告)日：2008-06-05

申请号：US11843289

申请日：2007-08-22

Applicant: Ho-sang SUNG ,  Ki-hyun Choo ,  Jung-hoe Kim ,  Eun-mi Oh ,  Chang-yong Son ,  Kang-eun Lee

Inventor： Ho-sang SUNG ,  Ki-hyun Choo ,  Jung-hoe Kim ,  Eun-mi Oh ,  Chang-yong Son ,  Kang-eun Lee

IPC: G10L13/00

CPC classification number: G10L19/0017 ,  G10L19/005 ,  G10L19/022

Abstract: Provided are a frame error concealment method and apparatus and an error concealment scheme construction method and apparatus. The frame error concealment method includes generating a new signal by synthesizing a plurality of previous signals that are similar to a signal of an error frame and reconstructing the signal of the error frame using the generated signal.

Abstract translation: 提供了帧错误隐藏方法和装置以及错误隐藏方案的构造方法和装置。 帧错误隐藏方法包括通过合成与错误帧的信号相似的多个先前信号并使用所生成的信号重建错误帧的信号来生成新信号。

公开(公告)号：US20080133223A1

公开(公告)日：2008-06-05

申请号：US11930844

申请日：2007-10-31

Applicant: Chang-yong SON ,  Bun-mi Oh ,  Ho-sang Sung ,  Ki-hyun Choo ,  Jung-hue Kim ,  Kang-eun Lee

Inventor： Chang-yong SON ,  Bun-mi Oh ,  Ho-sang Sung ,  Ki-hyun Choo ,  Jung-hue Kim ,  Kang-eun Lee

IPC: G10L19/00

CPC classification number: G10L19/0204 ,  G10L19/093

Abstract: A method and apparatus to extract an important frequency component of an audio signal and a method and apparatus to encode and/or decode an audio signal by using the same. The method of extracting an important frequency component of an audio signal includes converting an audio signal of a time domain into an audio signal of a frequency domain, selecting a frequency band having a harmonic feature from the converted audio signal of the frequency domain, and extracting an important frequency component from the selected frequency band having the harmonic feature.

Abstract translation: 一种用于提取音频信号的重要频率分量的方法和装置，以及通过使用该方法和装置对音频信号进行编码和/或解码的方法和装置。 提取音频信号的重要频率分量的方法包括将时域的音频信号转换为频域的音频信号，从频域的转换的音频信号中选择具有谐波特征的频带，以及提取 来自具有谐波特征的所选频带的重要频率分量。

公开(公告)号：US20080010062A1

公开(公告)日：2008-01-10

申请号：US11774664

申请日：2007-07-09

Applicant: Chang-yong Son ,  Eun-mi Oh ,  Ki-hyun Choo ,  Jung-hoe Kim

Inventor： Chang-yong Son ,  Eun-mi Oh ,  Ki-hyun Choo ,  Jung-hoe Kim

IPC: G10L19/00

CPC classification number: G10L19/18 ,  G10L19/0212 ,  G10L19/06 ,  G10L19/09 ,  G10L19/20 ,  G10L25/18

Abstract: An adaptive encoding method includes splitting an input signal into a low-frequency band signal and a high-frequency band signal; performing forward adaptive linear prediction on the low-frequency band signal and thus filtering the low-frequency band signal; selectively performing backward adaptive linear prediction or long-term prediction on the filtered low-frequency band signal according to the analysis result of the low-frequency band signal; transforming the low-frequency band signal, on which backward adaptive linear prediction or long-term prediction has been performed, into a signal in a frequency domain and quantizing the signal; and encoding the high-frequency band signal using the low-frequency band signal, on which backward adaptive linear prediction or long-term prediction has been performed, or the quantized signal. Therefore, compression efficiency of both speech and music signals can be enhanced, and a robust compression method can be provided for various audio contents at a low bit rate.

Abstract translation: 一种自适应编码方法，包括将输入信号分成低频带信号和高频带信号; 对低频带信号执行前向自适应线性预测，从而对低频带信号进行滤波; 根据低频带信号的分析结果，对经滤波的低频带信号选择性地执行反向自适应线性预测或长期预测; 将已经进行了反向自适应线性预测或长期预测的低频带信号变换为频域中的信号并量化信号; 并且使用已经执行了反向自适应线性预测或长期预测的低频带信号或量化信号对高频带信号进行编码。 因此，可以提高语音和音乐信号的压缩效率，并且可以以低比特率为各种音频内容提供鲁棒的压缩方法。

公开(公告)号：US20070296614A1

公开(公告)日：2007-12-27

申请号：US11766322

申请日：2007-06-21

Applicant: Kang-eun Lee ,  Eun-mi Oh ,  Ho-sang Sung ,  Chang-yong Son ,  Ki-hyun Choo ,  Jung-hoe Kim

Inventor： Kang-eun Lee ,  Eun-mi Oh ,  Ho-sang Sung ,  Chang-yong Son ,  Ki-hyun Choo ,  Jung-hoe Kim

IPC: H03M7/00

CPC classification number: G10L19/0208 ,  G10L21/038

Abstract: Encoding and/or decoding a wideband signal produces high frequency band spectra from low frequency band spectral information. Linear prediction filter coefficients are determined for the entire wideband spectrum of an input signal. An energy value in each of a plurality of sub-bands in the high frequency band is determined and encoded. The short-term correlation removed input signal is then down-sampled to form a low frequency band signal. At a decoder, the high frequency band signal is generated using the encoded low frequency band signal. The energy in each sub-band of the high frequency band is adjusted using the encoded energy value. Thus, the spectral envelope for the entire wideband signal is synthesized and decoded using linear predictive synthesis.

Abstract translation: 编码和/或解码宽带信号从低频带频谱信息产生高频带频谱。 对于输入信号的整个宽带谱确定线性预测滤波器系数。 确定并编码高频带中的多个子带中的每一个中的能量值。 然后将短期相关去除输入信号进行下采样以形成低频带信号。 在解码器处，使用编码的低频带信号产生高频带信号。 使用编码的能量值来调整高频带的每个子带中的能量。 因此，使用线性预测合成来合成和解码整个宽带信号的频谱包络。

公开(公告)号：US09378750B2

公开(公告)日：2016-06-28

申请号：US13584170

申请日：2012-08-13

Applicant: Chang Yong Son ,  Kang Eun Lee ,  Do Hyung Kim ,  Shi Hwa Lee

Inventor： Chang Yong Son ,  Kang Eun Lee ,  Do Hyung Kim ,  Shi Hwa Lee

IPC: G10L19/00 ,  G10L19/16 ,  G06F17/00

CPC classification number: G10L19/16

Abstract: A method and apparatus for reproducing audio data using low power are provided. The apparatus may reproduce the audio data by determining a power mode based on a memory resource of an internal memory, and an amount of a memory required for reproducing the audio data, controlling a power based on the determined power mode, and decoding the audio data.

Abstract translation: 提供了一种使用低功率再现音频数据的方法和装置。 该装置可以通过基于内部存储器的存储器资源确定功率模式和再现音频数据所需的存储器的量来再现音频数据，基于所确定的功率模式来控制功率，以及对音频数据进行解码 。

公开(公告)号：US09154895B2

公开(公告)日：2015-10-06

申请号：US12805121

申请日：2010-07-13

Applicant: Chang Yong Son ,  Do-Hyung Kim ,  Kang Eun Lee

Inventor： Chang Yong Son ,  Do-Hyung Kim ,  Kang Eun Lee

IPC: H04S3/00

CPC classification number: H04S3/008

Abstract: An apparatus of generating a multi-channel sound signal is provided. The apparatus may include a sound separator to determine a number (N) of sound signals based on at least one of a mixing characteristic and a spatial characteristic of a multi-channel sound signal when receiving the multi-channel sound signal, and to separate the multi-channel sound signal into N sound signals, the sound signals being generated such that the multi-channel sound signal is separated, and a sound synthesizer to synthesize N sound signals to be M sound signals.

Abstract translation: 提供一种产生多声道声音信号的装置。 该装置可以包括声音分离器，用于当接收多声道声音信号时，基于多声道声音信号的混合特性和空间特性中的至少一个来确定声音信号的数量（N），并且将 将多声道声音信号转换为N个声音信号，生成声音信号，使得多声道声音信号被分离，并且声音合成器将N个声音信号合成为M个声音信号。

公开(公告)号：US08676569B2

公开(公告)日：2014-03-18

申请号：US13544203

申请日：2012-07-09

Applicant: Eun-mi Oh ,  Ki-hyun Choo ,  Ho-sang Sung ,  Chang-yong Son ,  Jung-hoe Kim ,  Kang-eun Lee

Inventor： Eun-mi Oh ,  Ki-hyun Choo ,  Ho-sang Sung ,  Chang-yong Son ,  Jung-hoe Kim ,  Kang-eun Lee

IPC: G10L19/00

CPC classification number: G10L19/005 ,  G10L19/0212 ,  G10L19/022 ,  H03M13/00

Abstract: An error concealment method and apparatus for an audio signal and a decoding method and apparatus for an audio signal using the error concealment method and apparatus. The error concealment method includes selecting one of an error concealment in a frequency domain and an error concealment in a time domain as an error concealment scheme for a current frame based on a predetermined criteria when an error occurs in the current frame, selecting one of a repetition scheme and an interpolation scheme in the frequency domain as the error concealment scheme for the current frame based on a predetermined criteria when the error concealment in the frequency domain is selected, and concealing the error of the current frame using the selected scheme.

Abstract translation: 一种用于音频信号的错误隐藏方法和装置，以及使用错误隐藏方法和装置的用于音频信号的解码方法和装置。 错误隐藏方法包括：当在当前帧中发生错误时，基于预定标准，选择频域中的错误隐藏和时域中的错误隐藏中的一个作为当前帧的错误隐藏方案，选择 重复方案和在频域中的内插方案作为当前帧的错误隐藏方案，当选择频域中的错误隐藏时，基于预定标准，并且使用所选择的方案来隐藏当前帧的错误。

公开(公告)号：US20130103392A1

公开(公告)日：2013-04-25

申请号：US13584170

申请日：2012-08-13

Applicant: Chang Yong SON ,  Kang Eun LEE ,  Do Hyung KIM ,  Shi Hwa LEE

Inventor： Chang Yong SON ,  Kang Eun LEE ,  Do Hyung KIM ,  Shi Hwa LEE

IPC: G10L19/00

CPC classification number: G10L19/16

Abstract: A method and apparatus for reproducing audio data using low power are provided. The apparatus may reproduce the audio data by determining a power mode based on a memory resource of an internal memory, and an amount of a memory required for reproducing the audio data, controlling a power based on the determined power mode, and decoding the audio data.

Abstract translation: 提供了一种使用低功率再现音频数据的方法和装置。 该装置可以通过基于内部存储器的存储器资源确定功率模式和再现音频数据所需的存储器的量来再现音频数据，基于所确定的功率模式来控制功率，以及对音频数据进行解码 。

公开(公告)号：US08050913B2

公开(公告)日：2011-11-01

申请号：US11930750

申请日：2007-10-31

Applicant: Kang-eun Lee ,  Do-hyung Kim ,  Chang-yong Son

Inventor： Kang-eun Lee ,  Do-hyung Kim ,  Chang-yong Son

IPC: G10L19/12

CPC classification number: G10L19/16 ,  G10L19/12

Abstract: A method and apparatus for implementing fixed codebooks as a common module are provided. In the method of implementing fixed codebooks of a plurality of speech codecs as a common module, it is possible to include only a part excluding fixed codebooks in a communication terminal or communication system, support various speech codecs without using a chip with high price and high performance, and reduce a memory space that is occupied by the speech codecs by generating a track of a fixed codebook corresponding to a speech codec based on information on the speech codec among the plurality of speech codecs and selecting a codebook vector corresponding to a target signal among codebook vectors constructed with combinations of pulses represented by the generated track. In addition, it is possible to reduce processing complexity as compared with a case of embodying the common fixed codebook module in software by embodying the common fixed codebook module in hardware. In addition, it is possible to improve the entire voice processing performance by applying the latest fixed codebook searching algorithm only to the common fixed codebook, thereby easily applying the latest fixed codebook searching algorithm to the entire voice codec.

Abstract translation: 提供了一种用于实现固定码本作为公共模块的方法和装置。 在将多个语音编解码器的固定码本实现为公共模块的方法中，可以在通信终端或通信系统中仅包括不包括固定码本的部分，而不使用高价格高的芯片支持各种语音编解码器 通过基于多个语音编解码器中的语音编解码器的信息生成与语音编解码器对应的固定码本的轨道，并且选择与目标信号相对应的码本矢量，来降低语音编解码器所占用的存储器空间 在由所生成的轨道表示的脉冲的组合构成的码本矢量中。 此外，与通过在硬件中实施公共固定码本模块的软件中体现公共固定码本模块的情况相比，可以降低处理复杂度。 此外，通过将最新的固定码本搜索算法应用于公共固定码本，可以提高整个语音处理性能，从而容易地将最新的固定码本搜索算法应用于整个语音编解码器。

公开(公告)号：US20080270124A1

公开(公告)日：2008-10-30

申请号：US11872116

申请日：2007-10-15

Applicant: Chang-yong SON ,  Eun-mi Oh ,  Jung-hoe Kim ,  Ho-sang Sung ,  Kang-eun Lee ,  Ki-hyun Choo

Inventor： Chang-yong SON ,  Eun-mi Oh ,  Jung-hoe Kim ,  Ho-sang Sung ,  Kang-eun Lee ,  Ki-hyun Choo

IPC: G10L21/00

CPC classification number: G10L19/00 ,  G10L19/025

Abstract: Provided is a method of encoding an audio/speech signal, the method including determining a variable length of a frame, that is, a processing unit of an input signal in accordance with a position of an attack in the input signal; transforming each frame of the input signal to a frequency domain and dividing the frame into a plurality of sub frequency bands; and, if a signal of a sub frequency band is determined to be encoded in the frequency domain, encoding the signal of the sub frequency band in the frequency domain, and if the signal of the sub frequency band is determined to be encoded in a time domain, inverse transforming the signal of the sub frequency band to the time domain and encoding the inverse transformed signal in the time domain. According to the present invention, the audio/speech signal may be efficiently encoded by controlling time resolution and frequency resolution.

Abstract translation: 提供了一种对音频/语音信号进行编码的方法，该方法包括根据输入信号中的攻击位置来确定帧的可变长度，即输入信号的处理单元; 将所述输入信号的每个帧变换为频域并将所述帧划分为多个子频带; 并且如果确定在频域中编码子频带的信号，则对频域中的子频带的信号进行编码，并且如果确定子频带的信号被编码在一个时间 域，将子频带的信号逆变换到时域，并对时域中的逆变换信号进行编码。 根据本发明，可以通过控制时间分辨率和频率分辨率来有效地编码音频/语音信号。