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51.发明申请AUDIO CODING APPARATUS, AUDIO DECODING APPARATUS, AUDIO CODING AND DECODING APPARATUS, AND TELECONFERENCING SYSTEM 有权音频编码设备，音频解码设备，音频编码和解码设备以及电话系统公开(公告)号：US20100198589A1
公开(公告)日：2010-08-05
申请号：US12679814
申请日：2009-07-28
IPC分类号： G10L19/00
CPC分类号： G10L19/008
摘要： The delay in a multi-channel audio coding apparatus and a multi-channel audio decoding apparatus is reduced. The audio coding apparatus includes: a downmix signal generating unit (410) that generates, in a time domain, a first downmix signal that is one of a 1-channel audio signal and a 2-channel audio signal from an input multi-channel audio signal; a downmix signal coding unit (404) that codes the first downmix signal; a first t-f converting unit (401) that converts the input multi-channel audio signal into a multi-channel audio signal in a frequency domain; and a spatial information calculating unit (409) that generates spatial information for generating a multi-channel audio signal from a downmix signal.
摘要翻译： 减少了多声道音频编码装置和多声道音频解码装置中的延迟。 音频编码装置包括：下混合信号生成单元，其在时域中生成作为来自输入多声道音频的1声道音频信号和2声道音频信号之一的第1降混信号 信号; 编码第一下混信号的下混信号编码单元（404） 第一t-f转换单元，其将输入的多声道音频信号转换成频域中的多声道音频信号; 以及空间信息计算单元（409），其生成用于从降混信号生成多声道音频信号的空间信息。
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52.发明授权Bandwidth extension method, bandwidth extension apparatus, program, integrated circuit, and audio decoding apparatus 有权带宽扩展方法，带宽扩展装置，程序，集成电路和音频解码装置公开(公告)号：US09093080B2
公开(公告)日：2015-07-28
申请号：US13389276
申请日：2011-06-06
IPC分类号： G10L19/00 , G10L21/038 , G10L21/04
CPC分类号： G10L19/02 , G10L21/038 , G10L21/04
摘要： To provide a bandwidth extension method which allows reduction of computation amount in bandwidth extension and suppression of deterioration of quality in the bandwidth to be extended. In the bandwidth extension method: a low frequency bandwidth signal is transformed into a QMF domain to generate a first low frequency QMF spectrum; pitch-shifted signals are generated by applying different shifting factors on the low frequency bandwidth signal; a high frequency QMF spectrum is generated by time-stretching the pitch-shifted signals in the QMF domain; the high frequency QMF spectrum is modified; and the modified high frequency QMF spectrum is combined with the first low frequency QMF spectrum.
摘要翻译： 提供一种允许减少带宽扩展中的计算量并抑制要扩展的带宽中的质量劣化的带宽扩展方法。 在带宽扩展方法中：将低频带宽信号变换成QMF域以产生第一低频QMF频谱; 通过在低频带宽信号上应用不同的移位因子来生成音调移位信号; 通过对QMF域中的音调移位信号进行时间扩展来产生高频QMF频谱; 改变高频QMF频谱; 并且修改的高频QMF频谱与第一低频QMF频谱组合。
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53.发明申请BANDWIDTH EXTENSION METHOD, BANDWIDTH EXTENSION APPARATUS, PROGRAM, INTEGRATED CIRCUIT, AND AUDIO DECODING APPARATUS 有权带宽扩展方法，带宽扩展设备，程序，集成电路和音频解码设备公开(公告)号：US20120136670A1
公开(公告)日：2012-05-31
申请号：US13389276
申请日：2011-06-06
IPC分类号： G10L19/00
CPC分类号： G10L19/02 , G10L21/038 , G10L21/04
摘要： To provide a bandwidth extension method which allows reduction of computation amount in bandwidth extension and suppression of deterioration of quality in the bandwidth to be extended. In the bandwidth extension method: a low frequency bandwidth signal is transformed into a QMF domain to generate a first low frequency QMF spectrum; pitch-shifted signals are generated by applying different shifting factors on the low frequency bandwidth signal; a high frequency QMF spectrum is generated by time-stretching the pitch-shifted signals in the QMF domain; the high frequency QMF spectrum is modified; and the modified high frequency QMF spectrum is combined with the first low frequency QMF spectrum.
摘要翻译： 提供一种允许减少带宽扩展中的计算量并抑制要扩展的带宽中的质量劣化的带宽扩展方法。 在带宽扩展方法中：将低频带宽信号变换成QMF域以产生第一低频QMF频谱; 通过在低频带宽信号上应用不同的移位因子来生成音调移位信号; 通过对QMF域中的音调移位信号进行时间扩展来产生高频QMF频谱; 改变高频QMF频谱; 并且修改的高频QMF频谱与第一低频QMF频谱组合。
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54.发明申请AUDIO SIGNAL PROCESSING APPARATUS, AUDIO CODING APPARATUS, AND AUDIO DECODING APPARATUS 有权音频信号处理装置，音频编码装置和音频解码装置公开(公告)号：US20120022676A1
公开(公告)日：2012-01-26
申请号：US13256055
申请日：2010-10-19
IPC分类号： G06F17/00
CPC分类号： G10L21/04 , G10L21/0388
摘要： To provide an audio signal processing apparatus which can perform, with low operation amount, audio signal processing that is either time stretch and/or compression processing or frequency modulation processing. The audio signal processing apparatus is intended to transform an input audio signal sequence using a predetermined adjustment factor. The audio signal processing apparatus includes a filter bank (2601) which transforms the input audio signal sequence into Quadrature Mirror Filter (QMF) coefficients using a filter for Quadrature Mirror Filter analysis (a QMF analysis filter) and an adjusting unit (2602) which adjusts the QMF coefficients based on a predetermined adjustment factor.
摘要翻译： 提供一种音频信号处理装置，其可以以低操作量执行时间延伸和/或压缩处理或频率调制处理的音频信号处理。 音频信号处理装置旨在使用预定的调节因子来变换输入音频信号序列。 音频信号处理装置包括滤波器组（2601），其使用用于正交镜滤波器分析的滤波器（QMF分析滤波器）和调整单元（2602）将输入音频信号序列变换为正交镜像滤波器（QMF）系数，调整单元（2602） 基于预定调整因子的QMF系数。
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公开(公告)号：US20100063828A1
公开(公告)日：2010-03-11
申请号：US12519248
申请日：2008-10-16
IPC分类号： G10L19/00
CPC分类号： H04M3/568 , G10L19/008 , G10L25/12 , H04L65/607
摘要： To provide an enhanced true-to-life atmosphere enjoyed in multipoint connecting, and reduce a calculation load at a multipoint connection unit, as well.A stream synthesizing device includes an input unit which inputs at least two coded signals each including a first downmix acoustic signal and an extended signal, each of first downmix acoustic signals being obtained by coding an acoustic signal into which at least two sound signals are downmixed, and the extended signal being for obtaining the at least two sound signals out of the first downmix acoustic signal; a coded signal generating unit which generates: a second downmix acoustic signal and an extended signal based on each of coded signals inputted by the input unit, the second downmix acoustic signal being for obtaining each of the first downmix acoustic signals, and the generated extended signal being for obtaining each of the first downmix acoustic signals out of the second downmix acoustic signal; and generate a coded signal including the generated second downmix acoustic signal, the generated extended signal, and each of extended signals included in the corresponding inputted coded signal; and an output unit which outputs the generated coded signal.
摘要翻译： 为了提供在多点连接中享受的增强的真实生活氛围，并减少多点连接单元的计算负荷。 流合成装置包括：输入单元，其输入至少两个编码信号，每个编码信号包括第一下混音声信号和扩展信号，每个第一下混音声信号是通过对至少两个声音信号进行混合的声信号进行编码而获得的， 并且所述扩展信号用于从所述第一下混声音信号中获得所述至少两个声音信号; 编码信号生成单元，其基于由所述输入单元输入的每个编码信号生成第二下混声音信号和扩展信号，所述第二下混声音信号用于获得所述第一下混声音信号中的每一个，并且所生成的扩展信号 用于从所述第二下混声音信号中获得所述第一下混声音信号中的每一个; 并生成包括所生成的第二下混声音信号，所生成的扩展信号和包括在相应的输入编码信号中的每个扩展信号的编码信号; 以及输出单元，其输出所生成的编码信号。
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56.发明授权公开(公告)号：US08886548B2
公开(公告)日：2014-11-11
申请号：US13141169
申请日：2010-10-21
CPC分类号： G10L19/265 , G10L19/008 , G10L19/0212 , G10L19/09 , G10L25/90
摘要： Provided is an encoding device (1) including: a pitch contour analysis unit (101) which detects information, a dynamic time-warping unit (102) which generates, based on the information, pitch change ratios (Tw_ratio in FIG. 18) within a range (86) including a range (86a) of the pitch change ratios corresponding to absolute pitch differences of 42 cents or larger; a first lossless coding unit (103) which codes the generated pitch parameters (102x); a time-warping unit (104) which shifts a pitch of a signal according to the information; and a second encoding unit which codes a signal (104x) obtained by the shifting.
摘要翻译： 提供了一种编码装置（1），包括：检测信息的音调轮廓分析单元（101），动态时间扭曲单元（102），其基于该信息生成内部的音调变化率（图18中的Tw_ratio） 包括对应于42分或以上的绝对音调差的音调变化率的范围（86a）的范围（86） 编码所生成的音调参数（102x）的第一无损编码单元（103）。 时间扭曲单元（104），其根据该信息移动信号的音调; 以及第二编码单元，其对通过移位获得的信号（104x）进行编码。
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公开(公告)号：US20130144611A1
公开(公告)日：2013-06-06
申请号：US13816741
申请日：2011-10-05
IPC分类号： G10L25/90
CPC分类号： G10L25/90 , G10L19/0212 , G10L19/09 , G10L19/26 , G10L19/265 , G10L2025/906
摘要： A coding device includes: a pitch contour detection unit which detects a pitch contour of an input audio signal; a dynamic time warping unit which determines the number of pitch nodes based on the pitch contour and generates a first time warping parameter including information indicating the determined number of pitch nodes, a pitch change position, and a pitch change ratio; a first encoder which codes the first time warping parameter; a time warping unit which corrects pitch, using the information obtained from the first time warping parameter, to approximate the pitches of the number of pitch nodes to a predetermined reference value; a second encoder which codes the input audio signal at the corrected pitch; and a multiplexer which multiplexes the coded time warping parameter and the coded audio signal to generate a bitstream.
摘要翻译： 编码装置包括：音调轮廓检测单元，其检测输入音频信号的音高轮廓; 动态时间扭曲单元，其基于所述音调轮廓来确定音调节点的数量，并且生成包括指示所确定的音调节点数量的信息的第一时间扭曲参数，音高改变位置和音高变化率; 编码第一时间扭曲参数的第一编码器; 使用从第一时间扭曲参数获得的信息来校正音调的时间扭曲单元将音调节点数的音调近似为预定参考值; 第二编码器，以校正的音调对输入音频信号进行编码; 以及复用器，其对编码的时间扭曲参数和编码的音频信号进行复用以生成比特流。
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公开(公告)号：US20130090929A1
公开(公告)日：2013-04-11
申请号：US13703044
申请日：2011-06-14
IPC分类号： G10L19/04
CPC分类号： G10L19/04 , G10L19/0212 , G10L19/022 , G10L19/107 , G10L19/20
摘要： Provided are a new hybrid audio decoder and a new hybrid audio encoder having block switching for speech signals and audio signals. Currently, very low bitrate audio coding methods for speech and audio signal are proposed. These audio coding methods cause very long delay. Generally, in coding an audio signal, algorithm delay tends to be long to achieve higher frequency resolution. In coding a speech signal, the delay needs to be reduced because the speech signal is used for telecommunication. To balance fine coding quality for these two kinds of input signals with very low bitrate, this invention provides a combination of a low delay filter bank like AAC-ELD and a CELP coding method.
摘要翻译： 提供了一种新的混合音频解码器和具有用于语音信号和音频信号的块切换的新的混合音频编码器。 目前，提出了用于语音和音频信号的非常低的比特率音频编码方法。 这些音频编码方法造成很长的延迟。 通常，在对音频信号进行编码时，算法延迟往往很长以达到更高的频率分辨率。 在对语音信号进行编码时，由于语音信号用于通信，所以需要减小延迟。 为了平衡具有非常低的比特率的这两种输入信号的精细编码质量，本发明提供了像AAC-ELD这样的低延迟滤波器组和CELP编码方法的组合。
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59.发明授权Reduced delay spatial coding and decoding apparatus and teleconferencing system 有权减少延迟空间编码和解码装置和电话会议系统公开(公告)号：US08311810B2
公开(公告)日：2012-11-13
申请号：US12679814
申请日：2009-07-28
CPC分类号： G10L19/008
摘要： The delay in a multi-channel audio coding apparatus and a multi-channel audio decoding apparatus is reduced. The audio coding apparatus includes: a downmix signal generating unit that generates, in a time domain, a first downmix signal that is one of a 1-channel audio signal and a 2-channel audio signal from an input multi-channel audio signal; a downmix signal coding unit that codes the first downmix signal; a first t-f converting unit that converts the input multi-channel audio signal into a multi-channel audio signal in a frequency domain; and a spatial information calculating unit that generates spatial information for generating a multi-channel audio signal from a downmix signal.
摘要翻译： 减少了多声道音频编码装置和多声道音频解码装置中的延迟。 音频编码装置包括：降混信号生成单元，其在时域中生成来自输入多声道音频信号的1声道音频信号和2声道音频信号之一的第1下混合信号; 下混合信号编码单元，对第一下混信号进行编码; 第一t-f转换单元，其将输入的多声道音频信号转换成频域中的多声道音频信号; 以及空间信息计算单元，其生成用于从降混信号生成多声道音频信号的空间信息。
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公开(公告)号：US20110182432A1
公开(公告)日：2011-07-28
申请号：US13121991
申请日：2010-07-30
IPC分类号： H04R5/00
CPC分类号： G10L19/008 , H04S2420/03
摘要： A coding apparatus which suppresses an extreme increase in a bit rate, includes: a downmixing and coding unit (301) that downmixes audio signals that have been provided, to reduce the number of channels to be fewer than the number of the provided audio signals, and to code the downmix signals; an object parameter extracting unit (304) that extracts parameters indicating correlation between the audio signals; and a multiplexing circuit (309) that multiplexes the extracted parameters with the generated downmix coded signals. The object parameter extracting unit (304) includes: an object classifying unit (305) that classifies each of the provided audio signals into a predetermined one of types based on audio characteristics; and an object parameter extracting circuit (308) that extracts parameters using a temporal granularity and a frequency granularity each of which is determined for a corresponding one of the types.
摘要翻译： 抑制比特率的极大增加的编码装置包括：降混和编码单元（301），其降低已经提供的音频信号，以将频道数减少到少于所提供的音频信号的数量， 并对下混合信号进行编码; 提取表示音频信号之间的相关性的参数的对象参数提取单元（304） 和多路复用电路（309），将所提取的参数与生成的下混合编码信号进行复用。 对象参数提取单元（304）包括：对象分类单元（305），其基于音频特性将所提供的音频信号分类为预定类型; 以及目标参数提取电路（308），其使用时间粒度和频率粒度提取参数，每个时间粒度和频率粒度被确定为相应的一种类型。
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