公开(公告)号：US08214203B2

公开(公告)日：2012-07-03

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： G10L11/00 ,  G10L19/14 ,  G10L13/00 ,  G10L13/06 ,  G10L19/00

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。

公开(公告)号：US07765100B2

公开(公告)日：2010-07-27

申请号：US11347429

申请日：2006-02-06

申请人： Hosang Sung ,  Seungho Choi ,  Kihyun Choo

发明人： Hosang Sung ,  Seungho Choi ,  Kihyun Choo

IPC分类号： G10L11/00 ,  G10L19/14 ,  G10L13/00 ,  G10L13/06 ,  G10L19/00

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.

公开(公告)号：US08473284B2

公开(公告)日：2013-06-25

申请号：US11097319

申请日：2005-04-04

申请人： Kangeun Lee ,  Hosang Sung ,  Kihyun Choo

发明人： Kangeun Lee ,  Hosang Sung ,  Kihyun Choo

IPC分类号： G10L19/12

CPC分类号： G10L19/07

摘要： A voice encoding/decoding method and apparatus. A voice encoder includes: a quantization selection unit generating a quantization selection signal; and a quantization unit extracting a linear prediction coding (LPC) coefficient from an input signal, converting the extracted LPC coefficient into a line spectral frequency (LSF), quantizing the LSF with a first LSF quantization unit or a second LSF quantization unit based on the quantization selection signal, and converting the quantized LSF into a quantized LPC coefficient. The quantization selection signal selects the first LSF quantization unit or second LSF quantization unit based on characteristics of a synthesized voice signal in previous frames of the input signal.

摘要翻译： 语音编码/解码方法和装置。 语音编码器包括：量化选择单元，生成量化选择信号; 以及量化单元，从输入信号提取线性预测编码（LPC）系数，将所提取的LPC系数转换为线谱频率（LSF），基于所述线频谱频率（LSF）对第一LSF量化单元或第二LSF量化单元量化LSF 量化选择信号，并将量化的LSF转换成量化的LPC系数。 量化选择信号基于输入信号的先前帧中的合成语音信号的特性来选择第一LSF量化单元或第二LSF量化单元。

公开(公告)号：US08271267B2

公开(公告)日：2012-09-18

申请号：US11490139

申请日：2006-07-21

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

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

IPC分类号： G10L21/00

CPC分类号： G10L19/24 ,  G10L19/0212 ,  G10L25/18

摘要： Provided are a scalable wide-band speech coding/decoding apparatus, method, and medium. An input wide-band speech input signal is first divided into a low-band signal and a high-band signal. The divided low-band signal is then coded using a code excited linear prediction (CELP) method. The divided high-band signal is coded using a harmonic method. A signal representing a difference between a synthetic signal obtained from the low-band and the high band, and a signal input to the low-band and the high-band is then coded using a modified discrete cosine transform (MDCT) method. The coded signal is then multiplexed. The multiplexed signal is then output. Accordingly, high quality speech can be achieved for all layers.

摘要翻译： 提供了一种可扩展的宽带语音编码/解码装置，方法和媒体。 输入宽带语音输入信号首先被分成低频带信号和高频带信号。 然后使用码激励线性预测（CELP）方法对分频的低频带信号进行编码。 分频高频信号采用谐波法编码。 然后，使用修正的离散余弦变换（MDCT）方法对表示从低频带和高频带获得的合成信号之间的差异以及输入到低频带和高频带的信号进行编码的信号。 然后对编码信号进行多路复用。 然后输出复用的信号。 因此，可以实现对所有层的高质量语音。

公开(公告)号：US08015017B2

公开(公告)日：2011-09-06

申请号：US11337487

申请日：2006-01-24

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： G10L19/0208 ,  G10L19/093

摘要： Audio coding and decoding apparatuses and methods which support fine granularity scalability (FGS) using harmonic information of a high-band audio signal or wideband error audio signal when performing wideband audio coding and decoding, and recording mediums on which the methods are stored. The audio coding method includes detecting harmonics of a high-band audio signal or wideband error audio signal of an input audio signal; determining an order of the detected harmonics; and coding the detected harmonics based on the determined order.

摘要翻译： 在执行宽带音频编码和解码时使用高频带音频信号或宽带误差音频信号的谐波信息支持精细粒度可伸缩性（FGS）的音频编码和解码装置和方法，以及存储方法的记录介质。 音频编码方法包括检测输入音频信号的高频带音频信号或宽带误差音频信号的谐波; 确定检测到的谐波的顺序; 并根据确定的顺序对检测到的谐波进行编码。

公开(公告)号：US20070040709A1

公开(公告)日：2007-02-22

申请号：US11485468

申请日：2006-07-13

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

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

IPC分类号： H03M7/00

CPC分类号： G10L19/0208

摘要： A method and apparatus to scalably encode and/or decode an audio signal includes encoding a specific band signal included in an input signal, encoding a frequency envelope of an excited signal in which the encoded specific band signal is removed from the input signal, encoding a residual signal in which the encoded frequency envelope is removed from the excited signal, and forming a bit-stream by scalably packing the encoded specific band signal, frequency envelop, and residual signal.

摘要翻译： 一种用于对音频信号进行可缩放编码和/或解码的方法和装置包括编码包括在输入信号中的特定频带信号，对其中去除编码的特定频带信号的激励信号的频率包络进行编码， 残留信号，其中编码的频率包络从激励信号中去除，以及通过可扩展地打包编码的特定频带信号，频率包络和残余信号来形成比特流。

公开(公告)号：US20060074643A1

公开(公告)日：2006-04-06

申请号：US11097319

申请日：2005-04-04

申请人： Kangeun Lee ,  Hosang Sung ,  Kihyun Choo

发明人： Kangeun Lee ,  Hosang Sung ,  Kihyun Choo

IPC分类号： G10L19/12

CPC分类号： G10L19/07

摘要： A voice encoding/decoding method and apparatus. A voice encoder includes: a quantization selection unit generating a quantization selection signal; and a quantization unit extracting a linear prediction coding (LPC) coefficient from an input signal, converting the extracted LPC coefficient into a line spectral frequency (LSF), quantizing the LSF with a first LSF quantization unit or a second LSF quantization unit based on the quantization selection signal, and converting the quantized LSF into a quantized LPC coefficient. The the quantization selection signal selects the first LSF quantization unit or second LSF quantization unit based on characteristics of a synthesized voice signal in previous frames of the input signal.

摘要翻译： 语音编码/解码方法和装置。 语音编码器包括：量化选择单元，生成量化选择信号; 以及量化单元，从输入信号提取线性预测编码（LPC）系数，将所提取的LPC系数转换为线谱频率（LSF），基于所述线频谱频率（LSF）对第一LSF量化单元或第二LSF量化单元量化LSF 量化选择信号，并将量化的LSF转换成量化的LPC系数。 量化选择信号基于输入信号的先前帧中的合成语音信号的特性来选择第一LSF量化单元或第二LSF量化单元。

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：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：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。