公开(公告)号：US08805694B2

公开(公告)日：2014-08-12

申请号：US13201517

申请日：2010-02-16

申请人： Mi-Suk Lee ,  Hyun-Joo Bae ,  Byung-Sun Lee

发明人： Mi-Suk Lee ,  Hyun-Joo Bae ,  Byung-Sun Lee

IPC分类号： G10L21/00 ,  G10L19/00 ,  G10L19/02 ,  G10L25/90 ,  G10L25/93 ,  H03M7/00

CPC分类号： G10L19/008 ,  G10L19/0204 ,  G10L19/093 ,  G10L19/24

摘要： A method and an apparatus for encoding and decoding audio signals using adaptive sinusoidal coding are provided. The audio signal encoding method includes the steps of dividing a synthesized audio signal into a plurality of sub-bands, calculating the energy of each sub-band, selecting a predetermined number of sub-bands having a relatively large amount of energy from the sub-bands, and performing sinusoidal coding with regard to the selected sub-bands. Application of sinusoidal coding based on consideration of the amount of energy of each sub-band of the synthesized signal improves the quality of the synthesized signal more efficiently.

摘要翻译： 提供了一种使用自适应正弦编码对音频信号进行编码和解码的方法和装置。 音频信号编码方法包括以下步骤：将合成的音频信号分成多个子带，计算每个子带的能量，从子帧中选择具有相对大量能量的预定数量的子带， 并且对于所选择的子带执行正弦编码。 基于考虑合成信号的每个子带的能量量的正弦编码的应用更有效地提高了合成信号的质量。

公开(公告)号：US20130191134A1

公开(公告)日：2013-07-25

申请号：US13876691

申请日：2011-09-28

申请人： Mi-Suk Lee

发明人： Mi-Suk Lee

IPC分类号： G10L19/008

CPC分类号： G10L19/008 ,  G10L19/005 ,  G10L19/12

摘要： The present invention relates to a method and apparatus for decoding an audio signal using a shaping function. According to one embodiment of the present invention, the method for decoding an audio signal comprises the following steps: taking frame data of the audio signal as an input; restoring a fixed codebook of the frame data using a random function; calculating a shaping function using an adaptive codebook of the frame data; shaping the restored fixed codebook using the shaping function; and synthesizing the audio signal from the frame data using the shaped fixed codebook and adaptive codebook. According to the present invention, the fixed codebook may be restored using the shaping function calculated on the basis of the adaptive codebook upon the occurrence of frame data loss, thus emphasizing a pitch period and reducing the influence of the fixed codebook between the pitch periods so as to reduce the degradation in the quality of the synthesized signal.

摘要翻译： 本发明涉及使用整形功能对音频信号进行解码的方法和装置。 根据本发明的一个实施例，用于解码音频信号的方法包括以下步骤：将音频信号的帧数据作为输入; 使用随机函数恢复帧数据的固定码本; 使用所述帧数据的自适应码本来计算整形函数; 使用整形功能对恢复的固定码本进行整形; 并使用成形固定码本和自适应码本从帧数据合成音频信号。 根据本发明，可以使用在发生帧数据丢失时基于自适应码本计算的整形函数来恢复固定码本，从而强调音调周期并减少固定码本在音调周期之间的影响 以减少合成信号的质量下降。

公开(公告)号：US20110178807A1

公开(公告)日：2011-07-21

申请号：US13011273

申请日：2011-01-21

申请人： Heesik YANG ,  Mi-Suk LEE ,  Hyun-Woo KIM ,  Jongmo SUNG ,  Hyun-Joo BAE ,  Byung-Sun LEE

发明人： Heesik YANG ,  Mi-Suk LEE ,  Hyun-Woo KIM ,  Jongmo SUNG ,  Hyun-Joo BAE ,  Byung-Sun LEE

IPC分类号： G10L19/00

CPC分类号： G10L19/093 ,  G10L19/24

摘要： Provided are a method and an apparatus for decoding an audio signal. A method for decoding an audio signal encoded by a layered sinusoidal pulse coding scheme using one or more sinusoidal pulses includes decoding the encoded audio signal, setting a smoothing frequency band of the decoded audio signal according to a layer structure of the layered sinusoidal pulse coding scheme, dividing the smoothing frequency band into one or more subbands, and smoothing the decoded audio signal on a subband-by-subband basis. Accordingly, a decoding operation time can be reduced and the quality of a synthesized signal can be improved by variably setting a frequency band to be smoothed, when decoding an audio signal encoded by a layered sinusoidal pulse coding scheme using one or more sinusoidal pulses.

摘要翻译： 提供了一种用于对音频信号进行解码的方法和装置。 一种使用一个或多个正弦脉冲对由分层正弦脉冲编码方案编码的音频信号进行解码的方法包括解码编码音频信号，根据分层正弦脉冲编码方案的层结构设置解码音频信号的平滑频带 将平滑频带划分成一个或多个子带，并且以逐个子带为基础平滑解码的音频信号。 因此，当使用一个或多个正弦脉冲对由分层正弦脉冲编码方案编码的音频信号进行解码时，可以通过可变地设置要平滑的频带来改善解码操作时间并且可以提高合成信号的质量。

公开(公告)号：US20110153337A1

公开(公告)日：2011-06-23

申请号：US12957027

申请日：2010-11-30

申请人： Hyun-Woo KIM ,  Jong-Mo SUNG ,  Mi-Suk LEE ,  Hee-Sik YANG ,  Hyun-Joo BAE ,  Byung-Sun LEE

发明人： Hyun-Woo KIM ,  Jong-Mo SUNG ,  Mi-Suk LEE ,  Hee-Sik YANG ,  Hyun-Joo BAE ,  Byung-Sun LEE

IPC分类号： G10L19/00

CPC分类号： G10L19/10

摘要： An encoding apparatus is provided. The encoding apparatus includes a track structure determiner determining a track structure using frequency coefficients, a frequency coefficient allocator allocating the frequency coefficients to each track according to the determined track structure, and a quantizer quantizing one or more pulses in each track based on a number of frequency coefficients allocated to a corresponding track. The encoding apparatus can prevent the degradation of sound quality by avoiding the problem faced by most sinusoidal quantization techniques using a fixed track structure, i.e., a failure to quantize all pulses due to mismatches between the pulse distribution of frequency coefficients and a track structure.

摘要翻译： 提供一种编码装置。 编码装置包括：轨道结构确定器，使用频率系数确定轨迹结构;频率系数分配器，根据所确定的轨道结构将频率系数分配给每个轨道，以及量化器，基于多个轨道结构的数量量化每个轨道中的一个或多个脉冲 分配给相应轨道的频率系数。 编码装置可以通过避免使用固定轨道结构的大多数正弦量化技术所面临的问题，即由于频率系数的脉冲分布与轨道结构之间的不匹配造成的所有脉冲的量化，从而可以防止音质的恶化。

公开(公告)号：US20060122830A1

公开(公告)日：2006-06-08

申请号：US11297686

申请日：2005-12-07

申请人： Mi-Suk Lee ,  Do-Young Kim ,  JongMo Sung ,  Hyun-Woo Kim

发明人： Mi-Suk Lee ,  Do-Young Kim ,  JongMo Sung ,  Hyun-Woo Kim

IPC分类号： G10L19/02

CPC分类号： G10L19/24 ,  G10L19/083 ,  G10L19/10

摘要： Provides is an embedded code-excited linear prediction speech coding/decoding apparatus and method that can deal with the capacity change of speech transmission channel by modeling an error signal not coded at a core speech coder based on a transmission rate in a multiple pulse search mode- or gain compensation mode and then transmitting it in an optimum mode. The apparatus includes a core speech coding unit for coding an input speech signal with spectral envelop and an excitation signal, a transmission rate determination unit for allocating the number of bits additionally allowed depending on a capacity of a transmission channel, and an embedded excitation signal coding unit for coding a residual excitation signal that is not coded in the core speech coding unit based on the number of additionally allowed bits using one of a multiple pulse excitation coding mode and a gain compensation mode.

摘要翻译： 提供一种嵌入式码激励线性预测语音编码/解码装置和方法，可以通过对基于多脉冲搜索模式中的传输速率的核心语音编码器对未编码的误差信号进行建模来处理语音传输信道的容量变化 - 或增益补偿模式，然后以最佳模式发送。 该装置包括：核心语音编码单元，用于对具有频谱包络和激励信号的输入语音信号进行编码;传输速率确定单元，用于根据传输信道的容量分配额外容许的比特数;以及嵌入式激励信号编码 用于使用多脉冲激励编码模​​式和增益补偿模式中的一种，基于使用附加允许位的数量来编码不在核心语音编码单元中编码的残余激励信号的单元。

公开(公告)号：US09111535B2

公开(公告)日：2015-08-18

申请号：US13011273

申请日：2011-01-21

申请人： Heesik Yang ,  Mi-Suk Lee ,  Hyun-Woo Kim ,  Jongmo Sung ,  Hyun-Joo Bae ,  Byung-Sun Lee

发明人： Heesik Yang ,  Mi-Suk Lee ,  Hyun-Woo Kim ,  Jongmo Sung ,  Hyun-Joo Bae ,  Byung-Sun Lee

IPC分类号： G10L21/02 ,  G10L19/093 ,  G10L19/24

CPC分类号： G10L19/093 ,  G10L19/24

摘要： Provided are a method and an apparatus for decoding an audio signal. A method for decoding an audio signal encoded by a layered sinusoidal pulse coding scheme using one or more sinusoidal pulses includes decoding the encoded audio signal, setting a smoothing frequency band of the decoded audio signal according to a layer structure of the layered sinusoidal pulse coding scheme, dividing the smoothing frequency band into one or more subbands, and smoothing the decoded audio signal on a subband-by-subband basis. Accordingly, a decoding operation time can be reduced and the quality of a synthesized signal can be improved by variably setting a frequency band to be smoothed, when decoding an audio signal encoded by a layered sinusoidal pulse coding scheme using one or more sinusoidal pulses.

摘要翻译： 提供了一种用于对音频信号进行解码的方法和装置。 一种使用一个或多个正弦脉冲对由分层正弦脉冲编码方案编码的音频信号进行解码的方法包括解码编码音频信号，根据分层正弦脉冲编码方案的层结构设置解码音频信号的平滑频带 将平滑频带划分成一个或多个子带，并且以逐个子带为基础平滑解码的音频信号。 因此，当使用一个或多个正弦脉冲对由分层正弦脉冲编码方案编码的音频信号进行解码时，可以通过可变地设置要平滑的频带来改善解码操作时间并且可以提高合成信号的质量。

公开(公告)号：US09087510B2

公开(公告)日：2015-07-21

申请号：US13876768

申请日：2011-09-28

申请人： Mi-Suk Lee

发明人： Mi-Suk Lee

IPC分类号： G10L19/005 ,  G10L19/09

CPC分类号： G10L19/005 ,  G10L19/09

摘要： Disclosed are a method and apparatus for decoding a an audiospeech signal using an adaptive codebook update. The method for decoding speech an audio signal includes: receiving an N+1-th normal frame data that is a normal frame transmitted after an N-th frame that is a loss frame data loss; determining whether an adaptive codebook of a final subframe of the N-th frame is updated or not by using the N-th frame and the N+1-th frame; updating the adaptive codebook of the final subframe of the N-th frame by using a the pitch index of the N+1-the frame; and synthesizing an audio a speech signal of by using the N+1-th frame.

摘要翻译： 公开了一种使用自适应码本更新来解码听力信号的方法和装置。 用于对语音解码音频信号的方法包括：接收作为丢失帧数据丢失的第N帧之后发送的正常帧的N + 1个正常帧数据; 通过使用第N帧和第N + 1帧来确定第N帧的最后子帧的自适应码本是否被更新; 通过使用N + 1个帧的音调索引来更新第N帧的最后子帧的自适应码本; 以及通过使用第N + 1帧来合成音频语音信号。

公开(公告)号：US20130066627A1

公开(公告)日：2013-03-14

申请号：US13613792

申请日：2012-09-13

申请人： Mi-Suk LEE ,  Do-Young KIM ,  Byung-Sun LEE

发明人： Mi-Suk LEE ,  Do-Young KIM ,  Byung-Sun LEE

IPC分类号： G10L21/00

CPC分类号： G10L21/0208 ,  G10L19/24 ,  G10L21/0316 ,  G10L21/038 ,  G11B20/10527 ,  G11B2020/10555 ,  H03G3/341

摘要： An apparatus and method of improving the quality of a speech codec are provided. In the method, a first energy of a signal decoded by a low-band codec is calculated, and a second energy of a signal decoded by a low-band enhancement mode is calculated. Then, when the first energy is less than a first threshold value or less than a product of the second energy and a second threshold value, a size of the decoded signal is scaled. Accordingly, generation of a quantization error with respect to a silence segment is reduced.

公开(公告)号：US20120095754A1

公开(公告)日：2012-04-19

申请号：US13321297

申请日：2010-05-19

申请人： Mi-Suk Lee ,  Heesik Yang ,  Hyun-Woo Kim ,  Jongmo Sung ,  Hyun-Joo Bae ,  Byung-Sun Lee

发明人： Mi-Suk Lee ,  Heesik Yang ,  Hyun-Woo Kim ,  Jongmo Sung ,  Hyun-Joo Bae ,  Byung-Sun Lee

IPC分类号： G10L19/02

CPC分类号： G10L19/02 ,  G10L19/0212 ,  G10L19/24

摘要： Provided are a method and an apparatus for encoding and decoding an audio signal. A method for encoding an audio signal includes receiving a transformed audio signal, dividing the transformed audio signal into a plurality of subbands, performing a first sinusoidal pulse coding operation on the subbands, determining a performance region of a second sinusoidal pulse coding operation among the subbands on the basis of coding information of the first sinusoidal pulse coding operation, and performing the second sinusoidal pulse coding operation on the determined performance region, wherein the first sinusoidal pulse coding operation is performed variably according to the coding information. Accordingly, it is possible to further improve the quality of a synthesized signal by considering the sinusoidal pulse coding of a lower layer when encoding or decoding an audio signal in an upper layer by a layered sinusoidal pulse coding scheme.

摘要翻译： 提供了一种用于对音频信号进行编码和解码的方法和装置。 一种对音频信号进行编码的方法包括接收经变换的音频信号，将变换的音频信号分成多个子带，对子带执行第一正弦脉冲编码操作，确定子带中的第二正弦脉冲编码操作的性能区域 基于第一正弦脉冲编码操作的编码信息，并对所确定的执行区域执行第二正弦脉冲编码操作，其中根据编码信息可变地执行第一正弦脉冲编码操作。 因此，通过考虑在通过分层正弦脉冲编码方案对上层中的音频信号进行编码或解码时下层的正弦脉冲编码，可以进一步提高合成信号的质量。

公开(公告)号：US20100153119A1

公开(公告)日：2010-06-17

申请号：US12518263

申请日：2007-12-07

申请人： Mi-Suk Lee ,  Do-Young Kim ,  Hae-Won Jung

发明人： Mi-Suk Lee ,  Do-Young Kim ,  Hae-Won Jung

IPC分类号： G10L19/00

CPC分类号： G10L19/008

摘要： Provided is an audio coding apparatus and method that can selectively apply a operation mode of a coding module for stereo or multi-channel representation according to input signal characteristics of each channel, when voice or music signals are transmitted using an audio codec in portable terminals capable of stereo or multi-channel input and output. The audio coding apparatus includes a down-mixer for down-mixing multi-channel audio signals into mono signals; a coder for coding the mono signals; an input channel correlation analyzer for deciding whether to give them stereo effect based on their signal distribution characteristics, and outputting a control signal indicating whether to perform stereo representation process; and a stereo representation unit for performing stereo representation process onto the multi-channel audio signals when the control signal indicating to perform stereo representation process.

摘要翻译： 提供了一种音频编码装置和方法，当使用便携式终端的音频编解码器传输语音或音乐信号时，可以根据每个信道的输入信号特性选择性地应用用于立体声或多声道表示的编码模块的操作模式 的立体声或多声道输入和输出。 音频编码装置包括用于将多声道音频信号降混为单声道信号的下混频器; 用于对单声道信号进行编码的编码器; 输入信道相关分析器，用于基于它们的信号分布特性决定是否给出立体声效果，并输出指示是否执行立体声表示处理的控制信号; 以及立体声表示单元，用于当指示执行立体声表示处理的控制信号时，对多声道音频信号执行立体声表示处理。