公开(公告)号：US08560306B2

公开(公告)日：2013-10-15

申请号：US11457251

申请日：2006-07-13

申请人： Hosang Sung ,  Kungeun Lee ,  Sang-won Kang ,  Thomas R. Fischer ,  Ja-kyoung Jun

发明人： Hosang Sung ,  Kungeun Lee ,  Sang-won Kang ,  Thomas R. Fischer ,  Ja-kyoung Jun

IPC分类号： G10L21/00 ,  G10L19/12 ,  G10L19/10 ,  G10L19/107

CPC分类号： G10L19/0212 ,  G10L15/08 ,  G10L19/10 ,  G10L19/107 ,  G10L19/12 ,  G10L2019/0013

摘要： A method and apparatus to search a codebook including pulses that model a predetermined component of a speech signal. The method includes the operations of selecting a predetermined number of paths corresponding to a predetermined number of pulse locations that are most consistent with the predetermined component, from among paths corresponding to pulse locations of a predetermined pulse location set allocated to at least one branch that connects one state of a predetermined Trellis structure to another state, performing the path selecting operation on each of states other than the one state, and selecting a path corresponding to pulse locations that are most consistent with the predetermined component from among paths including the selected paths, wherein each path corresponds to a union of plural tracks of an algebraic codebook. Accordingly, a number of calculations required during a codebook search is reduced.

摘要翻译： 一种搜索包括对语音信号的预定分量进行建模的脉冲的码本的方法和装置。 该方法包括从对应于分配给连接至少一个分支的预定脉冲位置集的脉冲位置的路径中选择对应于与预定分量最一致的预定数量的脉冲位置的预定数量的路径的操作 将预定网格结构的一个状态转换到另一状态，对除了一个状态之外的每个状态执行路径选择操作，并且从包括所选择的路径的路径中选择与预定分量最一致的脉冲位置对应的路径， 其中每个路径对应于代数码本的多个轨道的并集。 因此，减少了码本搜索期间所需的一些计算。

公开(公告)号：US08019600B2

公开(公告)日：2011-09-13

申请号：US11128432

申请日：2005-05-13

申请人： Changyong Son ,  Hosang Sung ,  Hochong Park ,  Byounghak Jeong ,  Youngyo Kim

发明人： Changyong Son ,  Hosang Sung ,  Hochong Park ,  Byounghak Jeong ,  Youngyo Kim

IPC分类号： G10L19/00 ,  G10L19/02 ,  G10L19/14

CPC分类号： G10L19/025

摘要： A speech signal compression and/or decompression method, medium, and apparatus in which the speech signal is transformed into the frequency domain for quantizing and dequantizing information of frequency coefficients. The speech signal compression apparatus includes a transform unit to transform a speech signal into the frequency domain and obtain frequency coefficients, a magnitude quantization unit to transform magnitudes of the frequency coefficients, quantize the transformed magnitudes and obtain magnitude quantization indices, a sign quantization unit to quantize signs of the frequency coefficients and obtain sign quantization indices, and a packetizing unit to generate the magnitude and sign quantization indices as a speech packet.

摘要翻译： 一种语音信号压缩和/或解压缩方法，介质和装置，其中语音信号被变换成频域以量化和去量化频率系数的信息。 语音信号压缩装置包括将语音信号变换为频域并获得频率系数的变换单元，变换频率系数的幅度量化单位，量化变换幅度并获得幅度量化索引，符号量化单元， 量化频率系数的符号并获得符号量化索引，以及分组单元，用于生成幅度和符号量化索引作为语音分组。

公开(公告)号：US07805314B2

公开(公告)日：2010-09-28

申请号：US11471635

申请日：2006-06-21

申请人： Hosang Sung ,  Sangwook Kim ,  Rakesh Taori ,  Kangeun Lee ,  Shihwa Lee

发明人： Hosang Sung ,  Sangwook Kim ,  Rakesh Taori ,  Kangeun Lee ,  Shihwa Lee

IPC分类号： G10L19/00 ,  G10L19/02

CPC分类号： G10L19/035 ,  G10L19/0212 ,  G10L19/24

摘要： A method and apparatus to quantize/dequantize frequency amplitude data and a method and apparatus to audio encode/decode using the method and apparatus to quantize/dequantize the frequency amplitude data. The method includes calculating and quantizing power of frequency amplitudes for each of a plurality of bands constituting an audio frame, normalizing frequency amplitude data for each of the bands using the quantized power, and quantizing a first one of even-numbered or odd-numbered data among the normalized frequency amplitude data. The method may further include interpolating frequency amplitude data that corresponds to a second one of the even-numbered or odd-numbered frequency amplitude that is not quantized from among the normalized frequency amplitude data using the quantized first one of the even-numbered or odd-numbered data, and quantizing an interpolation error corresponding to a difference between the second frequency amplitude data that is not quantized and the interpolated frequency amplitude data.

摘要翻译： 一种用于量化/去量化频率幅度数据的方法和装置以及使用该方法和装置对频率振幅数据进行量化/去量化的音频编码/解码的方法和装置。 该方法包括：计算和量化构成音频帧的多个频带中的每个频带的频率幅度的功率，使用量化功率归一化每个频带的频率振幅数据，以及量化偶数或奇数数据中的第一个 在归一化的频率振幅数据中。 该方法可以进一步包括使用偶数或奇数编号的量化的第一个量化的归一化频率幅度数据中对应于未被量化的偶数或奇数频率振幅中的第二频率振幅数据， 量化与未量化的第二频率振幅数据和内插频率振幅数据之间的差对应的内插误差。

公开(公告)号：US20100191523A1

公开(公告)日：2010-07-29

申请号：US12659943

申请日：2010-03-25

申请人： Hosang Sung ,  Seungho Choi ,  Kihyun Choo

发明人： Hosang Sung ,  Seungho Choi ,  Kihyun Choo

IPC分类号： G10L11/04

CPC分类号： G10L19/005 ,  G10L19/07

摘要： A method and an apparatus for recovering a line spectrum pair (LSP) parameter of a spectrum region when frame loss occurs during speech decoding and a speech decoding apparatus adopting the same are provided. The method of recovering an LSP parameter in speech decoding includes: if it is determined that a received speech packet has an erased frame, converting an LSP parameter of a previous good frame (PGF) of the erased frame or LSP parameters of the PGF and a next good frame (NGF) of the erased frame into a spectrum region and obtaining a spectrum envelope of the PGF or spectrum envelopes of the PGF and NGF; recovering a spectrum envelope of the erased frame using the spectrum envelope of the PGF or the spectrum envelopes of the PGF and NGF; and converting the recovered spectrum envelope of the erased frame into an LSP parameter of the erased frame. The method and apparatus can improve the quality of a recovered speech signal, be applied to a variety of technologies, and provide a method of recovering an LSP parameter for development of an algorithm for speech decoding.

摘要翻译： 提供了一种用于在语音解码期间发生帧丢失时恢复频谱区的线谱对（LSP）参数的方法和装置，以及采用该频谱对参数的语音解码装置。 在语音解码中恢复LSP参数的方法包括：如果确定接收到的语音分组具有已擦除的帧，则将已擦除帧的先前好帧（PGF）的LSP参数或PGF的LSP参数和 将擦除的帧的下一个良好帧（NGF）进入频谱区域并获得PGF和NGF的PGF或频谱包络的​​频谱包络; 使用PGF的频谱包络或PGF和NGF的频谱包络来恢复被擦除帧的频谱包络; 以及将所述已擦除帧的所恢复的频谱包络转换为所述已擦除帧的LSP参数。 该方法和装置可以提高恢复的语音信号的质量，应用于各种技术，并提供一种恢复用于语音解码算法开发的LSP参数的方法。

公开(公告)号：US20070291835A1

公开(公告)日：2007-12-20

申请号：US11709254

申请日：2007-02-22

申请人： Hosang Sung ,  Eunmi Oh ,  Junghoe Kim ,  Kangeun Lee

发明人： Hosang Sung ,  Eunmi Oh ,  Junghoe Kim ,  Kangeun Lee

IPC分类号： H04B1/66 ,  G11B11/00

CPC分类号： G10L19/12 ,  G10L19/24 ,  H04N19/115 ,  H04N19/164 ,  H04N19/187 ,  H04N19/30 ,  H04N21/234327 ,  H04N21/2402 ,  H04N21/2662 ,  H04N21/6437

摘要： An encoder and decoder to encode one or more input signals into a scalable codec and to decode the scalable codec, and encoding and decoding methods using a bitstream with a layered structure in the scalable codec change a top coding bit rate to encode the input signals according to a network status, and the bitstream is decoded by analyzing the top coding bit rate included in the bitstream.

摘要翻译： 编码器和解码器将一个或多个输入信号编码成可扩展编解码器并对可伸缩编解码器进行解码，以及使用具有可分级编解码器中的分层结构的比特流进行编码和解码方法，改变顶编码比特率以对输入信号进行编码 通过分析比特流中包括的顶部编码比特率来解码比特流。

公开(公告)号：US20070038440A1

公开(公告)日：2007-02-15

申请号：US11480449

申请日：2006-07-05

申请人： Hosang Sung ,  Rakesh Taori ,  Kangeun Lee

发明人： Hosang Sung ,  Rakesh Taori ,  Kangeun Lee

IPC分类号： G10L19/00

CPC分类号： G10L19/22 ,  G10L19/022

摘要： A method, apparatus, and medium for classifying a speech signal and a method, apparatus, and medium for encoding the speech signal using the same are provided. The method for classifying a speech signal includes calculating classification parameters from an input signal having block units, calculating a plurality of classification criteria from the classification parameters, and classifying the level of the input signal using the plurality of classification criteria. The classification parameters include at least one of an energy parameter of the input signal, a cross-correlation parameter between a specific block of a present frame and the input signal, and an integrated cross-correlation parameter obtained by accumulating the cross-correlation parameter.

摘要翻译： 提供了一种用于对语音信号进行分类的方法，装置和媒体，以及使用该语音信号编码语音信号的方法，装置和媒体。 用于分类语音信号的方法包括从具有块单位的输入信号计算分类参数，从分类参数计算多个分类标准，以及使用多个分类标准对输入信号的等级进行分类。 分类参数包括输入信号的能量参数，当前帧的特定块与输入信号之间的互相关参数，以及通过累加互相关参数而获得的积分互相关参数中的至少一个。

公开(公告)号：US20070027683A1

公开(公告)日：2007-02-01

申请号：US11417165

申请日：2006-05-04

申请人： Hosang Sung ,  Kangeun Lee ,  Seungho Choi

发明人： Hosang Sung ,  Kangeun Lee ,  Seungho Choi

IPC分类号： G10L19/00

CPC分类号： G10L19/0017 ,  G10L19/005 ,  G10L19/06 ,  G10L19/07 ,  G10L19/083 ,  G10L19/12 ,  G10L2019/0002 ,  G10L2019/0016

摘要： An apparatus and method for concealing frame erasure and a voice decoding apparatus and method using the same. The frame erasure concealment apparatus includes: a parameter extraction unit determining whether there is an erased frame in a voice packet, and extracting an excitement signal parameter and a line spectrum pair parameter of a previous good frame; and an erasure frame concealment unit, if there is an erased frame, restoring the excitement signal and line spectrum pair parameter of the erased frame by using a regression analysis from the excitement signal and line spectrum pair parameter of the previous good frame. According to the method and apparatus, by predicting and restoring the parameter of the erased frame through the regression analysis, the quality of the restored voice signal can be enhanced and the algorithm can be simplified.

摘要翻译： 一种用于隐藏帧擦除的装置和方法，以及使用该帧擦除的语音解码装置和方法。 帧擦除隐藏装置包括：参数提取单元，确定语音分组中是否存在被擦除的帧，以及提取先前好帧的兴奋信号参数和线谱对参数; 以及擦除帧隐藏单元，如果存在擦除帧，则通过使用来自先前好帧的兴奋信号和线谱对参数的回归分析来恢复被擦除帧的兴奋信号和线谱对参数。 根据该方法和装置，通过回归分析预测和恢复被擦除的帧的参数，可以提高恢复的语音信号的质量，并且可以简化算法。

公开(公告)号：US20070016417A1

公开(公告)日：2007-01-18

申请号：US11471635

申请日：2006-06-21

申请人： Hosang Sung ,  Sangwook Kim ,  Rakesh Taori ,  Kangeun Lee ,  Shihwa Lee

发明人： Hosang Sung ,  Sangwook Kim ,  Rakesh Taori ,  Kangeun Lee ,  Shihwa Lee

IPC分类号： G10L19/00

CPC分类号： G10L19/035 ,  G10L19/0212 ,  G10L19/24

摘要： A method and apparatus to quantize/dequantize frequency amplitude data and a method and apparatus to audio encode/decode using the method and apparatus to quantize/dequantize the frequency amplitude data. The method includes calculating and quantizing power of frequency amplitudes for each of a plurality of bands constituting an audio frame, normalizing frequency amplitude data for each of the bands using the quantized power, and quantizing a first one of even-numbered or odd-numbered data among the normalized frequency amplitude data. The method may further include interpolating frequency amplitude data that corresponds to a second one of the even-numbered or odd-numbered frequency amplitude that is not quantized from among the normalized frequency amplitude data using the quantized first one of the even-numbered or odd-numbered data, and quantizing an interpolation error corresponding to a difference between the second frequency amplitude data that is not quantized and the interpolated frequency amplitude data.

摘要翻译： 一种用于量化/去量化频率幅度数据的方法和装置以及使用该方法和装置对频率振幅数据进行量化/去量化的音频编码/解码的方法和装置。 该方法包括：计算和量化构成音频帧的多个频带中的每个频带的频率幅度的功率，使用量化功率归一化每个频带的频率振幅数据，以及量化偶数或奇数数据中的第一个 在归一化的频率振幅数据中。 该方法可以进一步包括使用偶数或奇数编号的量化的第一个量化的归一化频率幅度数据中对应于未被量化的偶数或奇数频率振幅中的第二频率振幅数据， 量化与未量化的第二频率振幅数据和内插频率振幅数据之间的差对应的内插误差。

公开(公告)号：US20060206316A1

公开(公告)日：2006-09-14

申请号：US11333342

申请日：2006-01-18

申请人： Hosang Sung ,  Rakesh Taori ,  Kangeun Lee

发明人： Hosang Sung ,  Rakesh Taori ,  Kangeun Lee

IPC分类号： G10L11/04

CPC分类号： G10L19/24 ,  G10L19/06 ,  G10L19/09 ,  G10L19/093

摘要： Audio coding and decoding apparatuses and methods that can optimize the quality of an audio signal including harmonics, and recording mediums storing the methods. An audio coding apparatus includes: a first harmonic coding module performing first harmonic coding on an input audio signal using a pitch lag of the input audio signal and producing a quantized linear prediction coding coefficient; a first detector detecting a first difference audio signal from a difference between an audio signal output from the first harmonic coding module and the input audio signal; a second harmonic coding module performing harmonic coding on the first difference audio signal using the quantized linear prediction coding coefficient and a previous harmonic coding result; a second detector detecting a second difference audio signal obtained from a difference between an audio signal output from the second harmonic coding module and the first difference audio signal; and a code excited linear prediction (CELP) module CELP coding the second difference audio signal using the quantized linear prediction coding coefficient obtained from the first harmonic coding module.

摘要翻译： 可以优化包括谐波的音频信号的质量的音频编码和解码装置和方法，以及存储方法的记录介质。 音频编码装置包括：第一谐波编码模块，使用输入音频信号的音调滞后对输入音频信号执行一次谐波编码，并产生量化的线性预测编码系数; 第一检测器，从第一谐波编码模块输出的音频信号与输入音频信号之间的差检测第一差分音频信号; 使用量化线性预测编码系数和先前的谐波编码结果对第一差分音频信号执行谐波编码的二次谐波编码模块; 第二检测器，检测从二次谐波编码模块输出的音频信号与第一差音频信号之间的差获得的第二差分音频信号; 以及使用从第一谐波编码模块获得的量化线性预测编码系数对第二差音频信号进行编码的码激励线性预测（CELP）模块CELP。

公开(公告)号：US20060178872A1

公开(公告)日：2006-08-10

申请号：US11347429

申请日：2006-02-06

申请人： Hosang Sung ,  Seunghe Choi ,  Kihyun Choo

发明人： Hosang Sung ,  Seunghe Choi ,  Kihyun Choo

IPC分类号： G10L19/14

CPC分类号： G10L19/005 ,  G10L19/07

摘要： A method and an apparatus for recovering a line spectrum pair (LSP) parameter of a spectrum region when frame loss occurs during speech decoding and a speech decoding apparatus adopting the same are provided. The method of recovering an LSP parameter in speech decoding includes: if it is determined that a received speech packet has an erased frame, converting an LSP parameter of a previous good frame (PGF) of the erased frame or LSP parameters of the PGF and a next good frame (NGF) of the erased frame into a spectrum region and obtaining a spectrum envelope of the PGF or spectrum envelopes of the PGF and NGF; recovering a spectrum envelope of the erased frame using the spectrum envelope of the PGF or the spectrum envelopes of the PGF and NGF; and converting the recovered spectrum envelope of the erased frame into an LSP parameter of the erased frame. The method and apparatus can improve the quality of a recovered speech signal, be applied to a variety of technologies, and provide a method of recovering an LSP parameter for development of an algorithm for speech decoding.

摘要翻译： 提供了一种用于在语音解码期间发生帧丢失时恢复频谱区的线谱对（LSP）参数的方法和装置，以及采用该频谱对参数的语音解码装置。 在语音解码中恢复LSP参数的方法包括：如果确定接收到的语音分组具有已擦除的帧，则将已擦除帧的先前好帧（PGF）的LSP参数或PGF的LSP参数和 将擦除的帧的下一个良好帧（NGF）进入频谱区域并获得PGF和NGF的PGF或频谱包络的​​频谱包络; 使用PGF的频谱包络或PGF和NGF的频谱包络来恢复被擦除的帧的频谱包络; 以及将所述已擦除帧的所恢复的频谱包络转换为所述已擦除帧的LSP参数。 该方法和装置可以提高恢复的语音信号的质量，应用于各种技术，并提供一种恢复用于语音解码算法开发的LSP参数的方法。