公开(公告)号：US08798172B2

公开(公告)日：2014-08-05

申请号：US11749249

申请日：2007-05-16

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

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

IPC: H04N7/12 ,  G06F11/00

CPC classification number: G10L19/005

Abstract: A method and apparatus to decode audio data constructed with a plurality of layers. An error concealment method of process a decoded bitstream selects one of a frequency domain and a time domain in order to conceal the errors, detects a position where the errors exist in a frame when the error concealment method in the frequency domain is selected, and conceals the errors only in a segment after the detected position.

Abstract translation: 一种解码由多层构成的音频数据的方法和装置。 处理解码比特流的错误隐藏方法选择频域和时域中的一个以隐藏错误，当选择频域中的错误隐藏方法时，检测在帧中存在错误的位置，并且隐藏 检测到位置之后的错误。

公开(公告)号：US08706481B2

公开(公告)日：2014-04-22

申请号：US11608956

申请日：2006-12-11

Applicant: Kang Eun Lee ,  Eun Mi Oh ,  Ho Sang Sung ,  Chang Yong Son

Inventor： Kang Eun Lee ,  Eun Mi Oh ,  Ho Sang Sung ,  Chang Yong Son

IPC: G10L19/12

CPC classification number: G10L19/032

Abstract: A method of multi-path trellis coded quantization (TCQ) usable in a speech coding system, and a quantizer using the method. Specifically the method includes calculating accumulated distortions corresponding to 2N survivor paths, wherein N indicates an integer greater than two, each of the 2N survivor paths is going towards one of nodes at an i th stage of a trellis, and i indicates an integer greater than zero, comparing the accumulated distortions respectively corresponding to the 2N survivor paths to select N paths among the 2N survivor paths, wherein the accumulated distortions corresponding to selected N paths are smaller than the accumulated distortions corresponding to unselected N paths establishing the selected N paths as survivor paths going toward an i+1 th stage, and selecting an optimal path among the 2N survivor paths corresponding to each node of a last stage.

Abstract translation: 一种可用于语音编码系统的多径网格编码量化（TCQ）的方法，以及使用该方法的量化器。 具体地，该方法包括计算对应于2N个幸存路径的累积失真，其中N表示大于2的整数，2N个幸存路径中的每一个正在网格的第i阶段中的一个节点，i表示大于 比较分别对应于2N个幸存路径的累积失真以选择2N个幸存路径中的N个路径，其中对应于所选N个路径的累积失真小于对应于建立所选择的N个路径的未选择的N个路径的累积失真作为幸存者 路径朝向第i + 1级，并且在对应于最后一级的每个节点的2N个幸存路径中选择最佳路径。

公开(公告)号：US08630863B2

公开(公告)日：2014-01-14

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: G10L19/00 ,  G10L19/02

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。

公开(公告)号：US08630849B2

公开(公告)日：2014-01-14

申请号：US11911775

申请日：2006-11-15

Applicant: Chang-Yong Son ,  Eun-Mi Oh ,  Ho-Sang Sung ,  Kang-Eun Lee ,  Ki-Hyun Choo ,  Jung-Hoe Kim

Inventor： Chang-Yong Son ,  Eun-Mi Oh ,  Ho-Sang Sung ,  Kang-Eun Lee ,  Ki-Hyun Choo ,  Jung-Hoe Kim

IPC: G10L19/12 ,  G10L21/00

CPC classification number: G10L19/07 ,  G10L2019/0005

Abstract: A method and apparatus to convert a linear predictive coding (LPC) coefficient into a coefficient having order characteristics, such as a line spectrum frequency (LSF), and to vector quantize the coefficient having the order characteristics when a speech signal is encoded. The method and apparatus split the vector of the coefficient having the order characteristics into a plurality of subvectors, select a codebook in which an available bit is variably allocated to each subvector according to distribution of elements of each subvector, and quantize each subvector according to the selected codebook. The method and apparatus use normalized codebooks.

Abstract translation: 将线性预测编码（LPC）系数转换成具有诸如线谱频率（LSF）等阶特征的系数的方法和装置，并且在语音信号被编码时向矢量量化具有阶特征的系数。 该方法和装置将具有有序特征的系数的向量分解为多个子向量，根据每个子向量的元素的分布，选择可用比特被可变地分配给每个子向量的码本，并且根据 所选码本。 该方法和装置使用归一化码本。

公开(公告)号：US08542839B2

公开(公告)日：2013-09-24

申请号：US12382562

申请日：2009-03-18

Applicant: Chang Yong Son ,  Do Hyung Kim ,  Sang Oak Woo ,  Kang Eun Lee

Inventor： Chang Yong Son ,  Do Hyung Kim ,  Sang Oak Woo ,  Kang Eun Lee

IPC: H04R5/00 ,  G10L13/00

CPC classification number: G10L25/48 ,  G10L21/0264

Abstract: An audio processing apparatus and method for a mobile device are provided. The audio processing apparatus and method may appropriately determine sound source localizations corresponding to a voice signal and an audio signal, and thereby may simultaneously provide a voice call service and a multimedia service. Also, the audio processing apparatus and method may guarantee quality of the voice call service even when simultaneously providing the voice call service and the multimedia service.

Abstract translation: 提供了一种用于移动设备的音频处理设备和方法。 音频处理装置和方法可以适当地确定对应于语音信号和音频信号的声源定位，从而可以同时提供语音呼叫服务和多媒体服务。 此外，即使在同时提供语音呼叫服务和多媒体服务的情况下，音频处理装置和方法也可以保证语音呼叫服务的质量。

公开(公告)号：US08010348B2

公开(公告)日：2011-08-30

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: G10L19/02 ,  G10L19/04 ,  G10L19/08

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：US20110116638A1

公开(公告)日：2011-05-19

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: H04R5/00

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：US07864843B2

公开(公告)日：2011-01-04

申请号：US11757528

申请日：2007-06-04

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

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

IPC: H04N11/02 ,  G10L13/00

CPC classification number: G10L19/0208 ,  G10L21/038

Abstract: A method and apparatus to perform bandwidth extension encoding and decoding encodes and/or decodes a high frequency signal using an excitation signal for a low frequency signal encoded in a time domain or a frequency domain or using an excitation spectrum for the low frequency signal. Accordingly, although an audio signal is encoded or decoded using a small number of bits, the quality of sound corresponding to a signal in a high frequency band does not degrade. Therefore, a coding efficiency of the audio signal can be maximized.

Abstract translation: 执行带宽扩展编码和解码的方法和装置使用用于在时域或频域中编码的低频信号的激励信号或者使用低频信号的激励频谱对高频信号进行编码和/或解码。 因此，尽管使用少量的比特对音频信号进行编码或解码，但是与高频带中的信号相对应的声音质量不会降低。 因此，可以使音频信号的编码效率最大化。

公开(公告)号：US20100104106A1

公开(公告)日：2010-04-29

申请号：US12382562

申请日：2009-03-18

Applicant: Chang Yong Son ,  Do Hyung Kim ,  Sang Oak Woo ,  Kang Eun Lee

Inventor： Chang Yong Son ,  Do Hyung Kim ,  Sang Oak Woo ,  Kang Eun Lee

IPC: H04R5/00 ,  G10L13/00

CPC classification number: G10L25/48 ,  G10L21/0264

Abstract: An audio processing apparatus and method for a mobile device are provided. The audio processing apparatus and method may appropriately determine sound source localizations corresponding to a voice signal and an audio signal, and thereby may simultaneously provide a voice call service and a multimedia service. Also, the audio processing apparatus and method may guarantee quality of the voice call service even when simultaneously providing the voice call service and the multimedia service.

Abstract translation: 提供了一种用于移动设备的音频处理设备和方法。 音频处理装置和方法可以适当地确定对应于语音信号和音频信号的声源定位，从而可以同时提供语音呼叫服务和多媒体服务。 此外，即使在同时提供语音呼叫服务和多媒体服务的情况下，音频处理装置和方法也可以保证语音呼叫服务的质量。

公开(公告)号：US20080162121A1

公开(公告)日：2008-07-03

申请号：US11964963

申请日：2007-12-27

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: G10L19/22

Abstract: Provided are a classifying method and apparatus for an audio signal, and an encoding/decoding method and apparatus for an audio signal using the classifying method and apparatus. In the classification method, an audio signal is classified by adaptively adjusting a classification threshold for a frame of the audio signal that is to be classified according to a long-term feature of the audio signal, thereby improving a hit rate of signal classification, suppressing frequent mode switching per frame, improving noise tolerance, and providing smooth reconstruction of the audio signal.

Abstract translation: 提供了一种用于音频信号的分类方法和装置，以及使用分类方法和装置的用于音频信号的编码/解码方法和装置。 在分类方法中，通过根据音频信号的长期特征自适应地调整要分类的音频信号的帧的分类阈值来分类音频信号，从而提高信号分类的命中率，抑制 每帧频繁模式切换，提高噪声容限，并提供音频信号的平滑重建。