公开(公告)号：US09042558B2

公开(公告)日：2015-05-26

申请号：US13122143

申请日：2008-10-01

申请人： Yousuke Takada

发明人： Yousuke Takada

IPC分类号： H04R5/00 ,  G10L19/022 ,  G10L19/008

CPC分类号： G10L19/022 ,  G10L19/008 ,  H04S2400/03

摘要： A decoding apparatus (10) is disclosed which includes: a storing means (11) for storing encoded audio signals including multi-channel audio signals; a transforming means (40) for transforming the encoded audio signals to generate transform block-based audio signals in a time domain; a window processing means (41) for multiplying the transform block-based audio signals by a product of a mixture ratio of the audio signals and a first window function, the product being a second window function; a synthesizing means (43) for overlapping the multiplied transform block-based audio signals to synthesize audio signals of respective channels; and a mixing means (14) for mixing audio signals of the respective channels between the channels to generate a downmixed audio signal. Furthermore, an encoding apparatus is also disclosed which downmixes the multi-channel audio signals, encodes the downmixed audio signals, and generates the encoded, downmixed audio signals.

摘要翻译： 公开了一种解码装置（10），其包括：存储装置（11），用于存储包括多声道音频信号的编码音频信号; 用于变换编码音频信号以在时域中产生基于变换块的音频信号的变换装置（40）; 窗口处理装置（41），用于将基于变换块的音频信号乘以音频信号的混合比和第一窗口函数的乘积，乘积作为第二窗口函数; 一个合成装置，用于重叠相乘的基于变换块的音频信号，以合成各个声道的音频信号; 以及混合装置（14），用于混合通道之间各个通道的音频信号，以产生下混音频信号。 此外，还公开了一种编码装置，其降低多声道音频信号的混合，对下混合音频信号进行编码，并产生经编码的下混频音频信号。

公开(公告)号：US08880411B2

公开(公告)日：2014-11-04

申请号：US13120473

申请日：2009-10-05

申请人： Pierrick Philippe ,  David Virette

发明人： Pierrick Philippe ,  David Virette

IPC分类号： G10L21/00 ,  G10L19/022 ,  H04R5/00 ,  H04B1/00 ,  G10L19/02 ,  G10L19/107 ,  G10L19/20 ,  G10L19/04

CPC分类号： G10L19/022 ,  G10L19/0212 ,  G10L19/04 ,  G10L19/107 ,  G10L19/20

摘要： A method for encoding and decoding a digital audio signal is provided, said method comprising the steps of: encoding a first sequence of samples of the digital signal according to a transform encoding; encoding a second sequence of samples of the digital signal according to a predictive encoding; wherein the second sequence starts before the end of the first sequence, a subsequence common to the first and second sequences being thus encoded both by predictive encoding and by transform encoding.

摘要翻译： 提供了一种对数字音频信号进行编码和解码的方法，所述方法包括以下步骤：根据变换编码对数字信号的第一采样序列进行编码; 根据预测编码对数字信号的第二采样序列进行编码; 其中所述第二序列在所述第一序列结束之前开始，因此通过预测编码和通过变换编码对所述第一和第二序列共同的子序列进行编码。

公开(公告)号：US08838441B2

公开(公告)日：2014-09-16

申请号：US13766945

申请日：2013-02-14

申请人： Dolby International AB

发明人： Lars Villemoes

IPC分类号： G10L19/00 ,  G10L11/04 ,  G10L19/02 ,  G10L13/00 ,  G10L11/00 ,  G10L21/04 ,  G10L19/002 ,  G10L19/022

CPC分类号： G10L19/002 ,  G10L19/0212 ,  G10L19/022

摘要： A representation of an audio signal having a first, a second and a third frame is derived by estimating first warp information for the first and second frames and second warp information for the second and third frames, the warp information describing pitch information of the audio signal. First or second spectral coefficients for first and second frames or second and third frames are derived using first or second warp information and a first or second weighted representation of the first and second frames or second and third frames, the first or second weighted representation derived by applying a first or second window function to the first and second frames or second and third frames, wherein the first or second window function depends on the first or second warp information. The representation of the audio signal is generated including the first and the second spectral coefficients.

摘要翻译： 具有第一，第二和第三帧的音频信号的表示通过估计第一和第二帧的第一扭曲信息和第二和第三帧的第二扭曲信息得出，扭曲信息描述音频信号的音调信息 。 使用第一或第二扭曲信息以及第一和第二帧或第二和第三帧的第一或第二加权表示来推导第一和第二帧或第二和第三帧的第一或第二频谱系数​​，第一或第二加权表示由 将第一或第二窗口功能应用于第一和第二帧或第二和第三帧，其中第一或第二窗口功能取决于第一或第二扭曲信息。 产生音频信号的表示，包括第一和第二频谱系数​​。

公开(公告)号：US08812307B2

公开(公告)日：2014-08-19

申请号：US13228965

申请日：2011-09-09

申请人： Jianfeng Xu ,  Lei Miao ,  Fengyan Qi ,  Dejun Zhang ,  Qing Zhang

发明人： Jianfeng Xu ,  Lei Miao ,  Fengyan Qi ,  Dejun Zhang ,  Qing Zhang

IPC分类号： G10L19/00 ,  G10L19/022 ,  G10L19/04

CPC分类号： G10L19/022 ,  G10L19/04

摘要： The present invention relates to communication technologies and discloses a method, an apparatus and a system for Linear Prediction Coding (LPC) analysis to improve LPC prediction performance and simplify analysis operation. The method includes: obtaining signal feature information of at least one sample point of input signals; comparing and analyzing the signal feature information to obtain an analysis result; selecting a window function according to the analysis result to perform adaptive windowing for the input signals and obtain windowed signals; and processing the windowed signals to obtain an LPC coefficient for linear prediction. The embodiments of the present invention are applicable to LPC.

摘要翻译： 本发明涉及通信技术，并且公开了一种用于线性预测编码（LPC）分析的方法，装置和系统，以提高LPC预测性能并简化分析操作。 该方法包括：获取输入信号的至少一个采样点的信号特征信息; 比较和分析信号特征信息以获得分析结果; 根据分析结果选择窗口功能，对输入信号进行自适应窗口化，获得窗口信号; 并处理窗口信号以获得用于线性预测的LPC系数。 本发明的实施例可应用于LPC。

公开(公告)号：US20130268264A1

公开(公告)日：2013-10-10

申请号：US13862655

申请日：2013-04-15

申请人： HUAWEI TECHNOLOGIES CO., LTD.

发明人： Anisse Taleb ,  Fengyan Qi ,  Chen Hu

IPC分类号： G10L19/022

CPC分类号： G10L19/022 ,  G10L19/0212 ,  G10L19/20

摘要： The present disclosure relates to a signal analyzer for processing an overlapped input signal frame comprising 2N subsequent input signal values. The signal analyzer comprises: a windower adapted to window the overlapped input signal frame to obtain a windowed signal, wherein the windower is adapted to zero M+N/2 subsequent input signal values of the overlapped input signal frame, wherein M is equal or greater than 1 and smaller than N/2; and a transformer adapted to transform the remaining 3N/2−M subsequent windowed signal values of the windowed signal using N−M sets of transform parameters to obtain a transformed-domain signal comprising N−M transformed-domain signal values.

摘要翻译： 本公开涉及一种用于处理包括2N个后续输入信号值的重叠输入信号帧的信号分析器。 所述信号分析器包括：风力发电机，其适于窗口重叠的输入信号帧以获得加窗信号，其中，所述加窗器适用于重叠输入信号帧的M + N / 2个后续输入信号值，其中M等于或大于 小于1，小于N / 2; 以及变压器，其适于使用N-M组变换参数来变换所述加窗信号的剩余3N / 2-M后续窗口化信号值，以获得包括N-M个变换域信号值的变换域信号。

公开(公告)号：US5214742A

公开(公告)日：1993-05-25

申请号：US585127

申请日：1990-10-01

申请人： Bernd Edler

发明人： Bernd Edler

IPC分类号： G10L19/022 ,  H04B1/66 ,  H04B14/00

CPC分类号： H04B1/66 ,  G10L19/022

摘要： Transmitting a signal wherein the signals is segmented by means of windows into successive overlapping blocks, the partial signals contained in the blocks are converted by transformation into a spectrum, with the spectra then being coded, transmitted, decoded after transmission and converted back into partial signals by retransformation. Finally, the blocks containing the partial signals are joined in an overlapping manner, with the overlapping regions of the blocks being weighted such that the resultant of the window functions in the respective overlapped regions equals one. To2.2. In order to avoid interferences in adjacent blocks upon changes in the signal amplitude, the length of the window functions is selected as a function of signal amplitude changes. The method is suitable for the treatment of audio and video signals which are subjected to data reduction during transmission.

摘要翻译： PCT No.PCT / EP90 / 00143 Sec。 371 1990年10月1日第 102（e）1990年10月1日PCT PCT 1990年1月26日PCT公布。 公开号WO90 / 09063 日期：1990年8月9日。发送信号，其中信号通过窗口分段成连续的重叠块，块中包含的部分信号通过变换转换为频谱，然后将频谱进行编码，发送，解码 传输并通过重新转换回部分信号。 最后，包含部分信号的块以重叠的方式连接，块的重叠区域被加权，使得各个重叠区域中的窗口函数的结果等于1。 到2.2。 为了避免在信号幅度变化时相邻块中的干扰，窗口函数的长度被选择为信号幅度变化的函数。 该方法适用于在传输过程中进行数据减少的音视频信号的处理。

公开(公告)号：USRE50194E1

公开(公告)日：2024-10-29

申请号：US17845407

申请日：2022-06-21

申请人： Fraunhofer-Gesellschaft zur Foerderung der angewandten Forschung e.V.

发明人： Markus Schnell ,  Manfred Lutzky ,  Markus Lohwasser ,  Markus Schmidt ,  Marc Gayer ,  Michael Mellar ,  Bernd Edler ,  Markus Multrus ,  Gerald Schuller ,  Ralf Geiger ,  Bernhard Grill

IPC分类号： G10L19/00 ,  G10L19/02 ,  G10L19/022 ,  G10L25/45 ,  H03H17/02 ,  G10L21/038

CPC分类号： G10L19/0204 ,  G10L19/022 ,  G10L25/45 ,  H03H17/0266 ,  G10L21/038

摘要： An embodiment of an apparatus for generating audio subband values in audio subband channels includes an analysis windower for windowing a frame of time-domain audio input samples being in a time sequence extending from an early sample to a later sample using an analysis window function including a sequence of window coefficients to obtain windowed samples. The analysis window function includes a first number of window coefficients derived from a larger window function including a sequence of a larger second number of window coefficients, wherein the window coefficients of the window function are derived by an interpolation of window coefficients of the larger window function. The apparatus further includes a calculator for calculating the audio subband values using the windowed samples.

公开(公告)号：US20240294987A1

公开(公告)日：2024-09-05

申请号：US18637832

申请日：2024-04-17

申请人： Dolby International AB

发明人： Lars VILLEMOES ,  Per EKSTRAND

IPC分类号： C12Q1/6883 ,  G10L19/02 ,  G10L19/022 ,  G10L19/26 ,  G10L21/038

CPC分类号： C12Q1/6883 ,  G10L19/0204 ,  G10L19/022 ,  G10L19/265 ,  G10L21/038 ,  C12Q2600/118 ,  C12Q2600/156

摘要： The present invention relates to coding of audio signals, and in particular to high frequency reconstruction methods including a frequency domain harmonic transposer. A system and method for generating a high frequency component of a signal from a low frequency component of the signal is described. The system comprises an analysis filter bank (501) comprising an analysis transformation unit (601) having a frequency resolution of Δf; and an analysis window (611) having a duration of DA; the analysis filter bank (501) being configured to provide a set of analysis subband signals from the low frequency component of the signal; a nonlinear processing unit (502, 650) configured to determine a set of synthesis subband signals based on a portion of the set of analysis subband signals, wherein the portion of the set of analysis subband signals is phase shifted by a transposition order T; and a synthesis filter bank (504) comprising a synthesis transformation unit (602) having a frequency resolution of QΔf; and a synthesis window (612) having a duration of DS; the synthesis filter bank (504) being configured to generate the high frequency component of the signal from the set of synthesis subband signals; wherein Q is a frequency resolution factor with Q≥1 and smaller than the transposition order T; and wherein the value of the product of the frequency resolution Δf and the duration DA of the analysis filter bank is selected based on the frequency resolution factor Q.

公开(公告)号：US20240249735A1

公开(公告)日：2024-07-25

申请号：US18628590

申请日：2024-04-05

申请人： Fraunhofer-Gesellschaft zur Foerderung der angewandten Forschung e.V.

发明人： Markus Multrus ,  Bernhard Grill ,  Guillaume Fuchs ,  Stefan Geyersberger ,  Nikolaus Rettelbach ,  Virgilio Bacigalupo

IPC分类号： G10L19/02 ,  G10L19/022 ,  H03M7/30

CPC分类号： G10L19/02 ,  G10L19/022 ,  H03M7/30

摘要： An audio encoder for encoding segments of coefficients, the segments of coefficients representing different time or frequency resolutions of a sampled audio signal, the audio encoder including a processor for deriving a coding context for a currently encoded coefficient of a current segment based on a previously encoded coefficient of a previous segment, the previously encoded coefficient representing a different time or frequency resolution than the currently encoded coefficient. The audio encoder further includes an entropy encoder for entropy encoding the current coefficient based on the coding context to obtain an encoded audio stream.

公开(公告)号：US12033648B2

公开(公告)日：2024-07-09

申请号：US18339915

申请日：2023-06-22

申请人： Fraunhofer-Gesellschaft zur Foerderung der angewandten Forschung e.V.

发明人： Emmanuel Ravelli ,  Manuel Jander ,  Grzegorz Pietrzyk ,  Martin Dietz ,  Marc Gayer

IPC分类号： G10L19/26 ,  G10L19/005 ,  G10L19/022 ,  G10L19/03 ,  G10L19/12 ,  G10L19/20 ,  H04B1/10 ,  G10L21/0364 ,  G11B27/038

CPC分类号： G10L19/26 ,  G10L19/005 ,  G10L19/022 ,  G10L19/03 ,  G10L19/12 ,  G10L19/20 ,  G10L21/0364 ,  G11B27/038 ,  H04B1/1027

摘要： A method is described that processes an audio signal. A discontinuity between a filtered past frame and a filtered current frame of the audio signal is removed using linear predictive filter. Removing the discontinuity further comprises processing a beginning portion of the filtered current frame, wherein the beginning portion of the current frame comprises a predefined number of samples being less or equal than a total number of samples in the current frame, and wherein processing the beginning portion of the current frame comprises subtracting a beginning portion of a zero-input-response (ZIR) from the beginning portion of the filtered current frame.