公开(公告)号：US20130211846A1

公开(公告)日：2013-08-15

申请号：US13396259

申请日：2012-02-14

申请人： Jonathan A. Gibbs ,  James P. Ashley ,  Udar Mittal

发明人： Jonathan A. Gibbs ,  James P. Ashley ,  Udar Mittal

IPC分类号： G10L19/00

CPC分类号： G10L19/20 ,  G10L19/26

摘要： An audio signal processing system includes parallel speech and generic audio signal processing paths. One path includes a linear predictive coder and a resampling filter having a non-linear phase characteristic. A phase compensation filter is disposed along the one of the processing paths to compensate for the non-linearity of the resampling filter thereby enabling relatively seamless switching between the coders resulting in a reduction of audio artifacts that would otherwise result from the non-linear phase characteristic of the resampling filter during playback.

摘要翻译： 音频信号处理系统包括并行语音和通用音频信号处理路径。 一条路径包括线性预测编码器和具有非线性相位特性的重采样滤波器。 相位补偿滤波器沿着一个处理路径被布置，以补偿重采样滤波器的非线性，从而使编码器之间进行相对无缝的切换，从而导致由非线性相位特性导致的音频伪影的减少 的重采样滤波器。

公开(公告)号：US08504376B2

公开(公告)日：2013-08-06

申请号：US11865671

申请日：2007-10-01

申请人： Dong Soo Kim ,  Hee Suk Pang ,  Jae Hyun Lim ,  Sung Yong Yoon ,  Hyun Kook Lee

发明人： Dong Soo Kim ,  Hee Suk Pang ,  Jae Hyun Lim ,  Sung Yong Yoon ,  Hyun Kook Lee

IPC分类号： G10L19/00

CPC分类号： G10L19/008 ,  G10L19/087 ,  G10L19/20 ,  G10L21/04 ,  H04S7/302 ,  H04S2400/03 ,  H04S2400/11 ,  H04S2420/01

摘要： An audio encoding method and apparatus and an audio decoding method and apparatus are provided. The audio signal decoding method includes extracting a downmix signal and object-based side information from an audio signal; generating a modified downmix signal based on the downmix signal and extracted information which is extracted from the object-based side information; generating channel-based side information based on the object-based side information and control data for rendering the downmix signal; and generating a multi-channel audio signal based on the modified downmix signal and the channel-based side information.

摘要翻译： 提供音频编码方法和装置以及音频解码方法和装置。 音频信号解码方法包括从音频信号中提取下混信号和基于对象的侧信息; 基于所述下混合信号生成修改的下混信号，以及从所述基于对象的侧信息中提取出的提取信息; 基于所述基于对象的侧信息和用于呈现所述下混信号的控制数据生成基于信道的侧信息; 以及基于修改的下混信号和基于信道的侧信息来生成多声道音频信号。

公开(公告)号：US08452430B2

公开(公告)日：2013-05-28

申请号：US12503456

申请日：2009-07-15

申请人： Hyen O Oh ,  Yang Won Jung

发明人： Hyen O Oh ,  Yang Won Jung

IPC分类号： G06F17/00 ,  G11B20/10 ,  G06F3/00 ,  G06F3/01

CPC分类号： G11B20/10 ,  G06F3/00 ,  G06F3/01 ,  G10L19/008 ,  G10L19/167 ,  G10L19/20 ,  H04S7/30 ,  H04S7/40

摘要： An apparatus for processing an audio signal and method thereof are disclosed. The apparatus comprises an information receiving unit receiving a downmix signal including and a plurality of preset informations; an external preset information receiving unit receiving a plurality of external preset informations being inputted from external; an external preset applying-determining unit determining whether the plurality of the external preset informations applies to the downmix signal; an external preset information selecting unit selecting one external preset information, if the plurality of external preset informations is selected; and a rendering unit controlling the object by applying the external preset information to the all data regions.Accordingly, an audio signal can be efficiently reconstructed by individually selecting and applying external preset information by a data region unit or by selecting and applying the same external preset information to a whole downmix signal. And, feedback information can be received from a user by displaying an object adjusted by having external preset rendering parameter applied thereto on a screen, according to a characteristic of an audio source.

摘要翻译： 公开了一种用于处理音频信号的装置及其方法。 该装置包括：信息接收单元，接收包含多个预置信息的下混信号; 接收从外部输入的多个外部预设信息的外部预设信息接收单元; 外部预置应用确定单元，确定所述多个外部预设信息是否适用于所述降混信号; 如果选择了多个外部预设信息，则外部预设信息选择单元选择一个外部预设信息; 以及渲染单元，通过将外部预设信息应用于所有数据区域来控制对象。 因此，通过数据区域单元单独选择和应用外部预设信息或者通过选择和应用相同的外部预设信息到整个降混信号，可以有效地重构音频信号。 并且，可以根据音频源的特征，通过显示通过在屏幕上施加了外部预设渲染参数而调整的对象，从用户接收反馈信息。

公开(公告)号：US20130124215A1

公开(公告)日：2013-05-16

申请号：US13736762

申请日：2013-01-08

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

发明人： Jeremie LECOMTE ,  Patrick WARMBOLD ,  Stefan BAYER

IPC分类号： G10L19/04

CPC分类号： G10L19/04 ,  G10L19/005 ,  G10L19/0212 ,  G10L19/20

摘要： A codec supporting switching between time-domain aliasing cancellation transform coding mode and time-domain coding mode is made less liable to frame loss by adding a further syntax portion to the frames, depending on which the parser of the decoder may select between a first action of expecting the current frame to have, and thus reading forward aliasing cancellation data from the current frame and a second action of not-expecting the current frame to have, and thus not reading forward aliasing cancellation data from the current frame. In other words, while a bit of coding efficiency is lost due to the provision of the new syntax portion, it is merely the new syntax portion which provides for the ability to use the codec in case of a communication channel with frame loss. Without the new syntax portion, the decoder would not be capable of decoding any data stream portion after a loss and will crash in trying to resume parsing. Thus, in an error prone environment, the coding efficiency is prevented from vanishing by the introduction of the new syntax portion.

摘要翻译： 支持在时域混叠取消变换编码模式和时域编码模式之间进行切换的编解码器通过向帧中添加另外的语法部分而不易造成帧丢失，这取决于解码器的解析器可以在第一动作 期望当前帧具有并且因此从当前帧中转发混叠取消数据，以及不期望当前帧具有并且因此不从当前帧读取前向混叠取消数据的第二动作。 换句话说，由于提供新的语法部分而导致编码效率的一部分丢失，仅仅是在具有帧丢失的通信信道的情况下提供使用编解码器的能力的新语法部分。 没有新的语法部分，解码器将不能在丢失之后解码任何数据流部分，并且在尝试恢复解析时将崩溃。 因此，在容易出错的环境中，通过引入新的语法部分，防止了编码效率的消失。

公开(公告)号：US08438019B2

公开(公告)日：2013-05-07

申请号：US11063664

申请日：2005-02-22

申请人： Janne Vainio ,  Hannu Mikkola ,  Pasi Ojala ,  Jari Mäkinen

发明人： Janne Vainio ,  Hannu Mikkola ,  Pasi Ojala ,  Jari Mäkinen

IPC分类号： G10L19/12 ,  G10L21/00

CPC分类号： G10L19/20

摘要： An encoder comprising an input for inputting frames of an audio signal in a frequency band, at least a first excitation block for performing a first excitation for a speech like audio signal, and a second excitation block for performing a second excitation for a non-speech like audio signal. The encoder further comprises a filter for dividing the frequency band into a plurality of sub bands each having a narrower bandwidth than the frequency band. The encoder also comprises an excitation selection block for selecting one excitation block among the at least first excitation block and the second excitation block for performing the excitation for a frame of the audio signal on the basis of the properties of the audio signal at least at one of the sub bands. The invention also relates to a device, a system, a method and a storage medium for a computer program.

摘要翻译： 一种编码器，包括用于输入频带中的音频信号的帧的输入，用于对诸如音频信号的语音执行第一激励的至少第一激励块，以及用于对非语音进行第二激励的第二激励块 像音频信号。 编码器还包括滤波器，用于将频带划分成多个子频带，每个子带具有比频带窄的带宽。 编码器还包括激励选择块，用于在至少第一激励块和第二激励块之间选择一个激励块，用于根据至少一个音频信号的属性对音频信号的帧执行激励 的子带。 本发明还涉及一种用于计算机程序的装置，系统，方法和存储介质。

公开(公告)号：US20130110523A1

公开(公告)日：2013-05-02

申请号：US13722176

申请日：2012-12-20

申请人： Electronics and Telecommunications Research Institute

发明人： Seung-Kwon BEACK ,  Jeong-Il Seo ,  Tae-Jin Lee ,  Yong-Ju Lee ,  In-Seon Jang ,  Jae-Hyoun Yoo ,  Dae-Young Jang ,  Jin-Woo Hong ,  Jin-Woong Kim ,  Kyeong-Ok Kang

IPC分类号： G10L19/00

CPC分类号： G10L19/00 ,  G10L19/008 ,  G10L19/20

摘要： Provided are an apparatus and method for coding and decoding a multi-object audio signal. The apparatus includes a down-mixer for down-mixing the audio signals into one down-mixed audio signal and extracting supplementary information including header information and spatial cue information for each of the audio signals, a coder for coding the down-mixed audio signal, and a supplementary information coder for generating the supplementary information as a bit stream. The header information includes identification information for each of the audio signals and channel information for the audio signals.

摘要翻译： 提供了一种用于对多对象音频信号进行编码和解码的装置和方法。 该装置包括一个下混合器，用于将音频信号下混合为一个下混合音频信号，并提取包括用于每个音频信号的标题信息和空间提示信息的补充信息，用于对混合音频信号进行编码的编码器， 以及用于将补充信息生成为比特流的补充信息编码器。 标题信息包括用于音频信号的每个音频信号和频道信息的识别信息。

公开(公告)号：US08423355B2

公开(公告)日：2013-04-16

申请号：US12844199

申请日：2010-07-27

申请人： Udar Mittal ,  Jonathan A. Gibbs ,  James P. Ashley

发明人： Udar Mittal ,  Jonathan A. Gibbs ,  James P. Ashley

IPC分类号： G10L19/02

CPC分类号： G10L19/18 ,  G10L19/0212 ,  G10L19/12 ,  G10L19/20

摘要： A method for encoding audio frames by producing a first frame of coded audio samples by coding a first audio frame in a sequence of frames, producing at least a portion of a second frame of coded audio samples by coding at least a portion of a second audio frame in the sequence of frames, and producing parameters for generating audio gap filler samples, wherein the parameters are representative of either a weighted segment of the first frame of coded audio samples or a weighted segment of the portion of the second frame of coded audio samples.

摘要翻译： 一种通过以帧序列编码第一音频帧来产生编码音频样本的第一帧来对音频帧进行编码的方法，通过编码第二音频的至少一部分来产生编码音频采样的第二帧的至少一部分 帧，并产生用于产生音频间隙填充样本的参数，其中参数表示编码音频样本的第一帧的加权段或编码音频样本的第二帧的部分的加权段 。

公开(公告)号：US20130030798A1

公开(公告)日：2013-01-31

申请号：US13190517

申请日：2011-07-26

申请人： Udar Mittal ,  James P. Ashley ,  Jonathan A. Gibbs

发明人： Udar Mittal ,  James P. Ashley ,  Jonathan A. Gibbs

IPC分类号： G10L19/00

CPC分类号： G10L19/20

摘要： An encoder and decoder for processing an audio signal including generic audio and speech frames are provided herein. During operation, two encoders are utilized by the speech coder, and two decoders are utilized by the speech decoder. The two encoders and decoders are utilized to process speech and non-speech (generic audio) respectively. During a transition between generic audio and speech, parameters that are needed by the speech decoder for decoding frame of speech are generated by processing the preceding generic audio (non-speech) frame for the necessary parameters. Because necessary parameters are obtained by the speech coder/decoder, the discontinuities associated with prior-art techniques are reduced when transitioning between generic audio frames and speech frames.

摘要翻译： 本文提供了一种用于处理包括通用音频和语音帧的音频信号的编码器和解码器。 在操作期间，语音编码器利用两个编码器，语音解码器利用两个解码器。 两个编码器和解码器分别用于处理语音和非语音（通用音频）。 在通用音频和语音之间的转换期间，语音解码器为解码语音帧所需的参数是通过处理前面的通用音频（非语音）帧来产生的，用于必要的参数。 由于必要的参数是由语音编码器/解码器获得的，当在通用音频帧和语音帧之间转换时，与现有技术相关联的不连续性被减少。

公开(公告)号：US20120253797A1

公开(公告)日：2012-10-04

申请号：US13449890

申请日：2012-04-18

申请人： Ralf Geiger ,  Guillaume Fuchs ,  Markus Multrus ,  Bernhard Grill

发明人： Ralf Geiger ,  Guillaume Fuchs ,  Markus Multrus ,  Bernhard Grill

IPC分类号： G10L19/04

CPC分类号： G10L19/12 ,  G10L19/03 ,  G10L19/04 ,  G10L19/083 ,  G10L19/20 ,  G10L2019/0002

摘要： In an embodiment, bitstream elements of sub-frames are encoded differentially to a global gain value so that a change of the global gain value results in an adjustment of an output level of the decoded representation of the audio content. Concurrently, the differential coding saves bits. Even further, the differential coding enables the lowering of the burden of globally adjusting the gain of an encoded bitstream. In another embodiment, a global gain control across CELP coded frames and transform coded frames is achieved by co-controlling the gain of the codebook excitation of the CELP codec, along with a level of the transform or inverse transform of the transform coded frames. In another embodiment, the gain value determination in CELP coding is performed in the weighted domain of the excitation signal.

摘要翻译： 在一个实施例中，子帧的比特流元素被差分地编码为全局增益值，使得全局增益值的改变导致音频内容的解码表示的输出电平的调整。 同时，差分编码保存位。 此外，差分编码能够降低全局调整编码比特流的增益的负担。 在另一个实施例中，跨CELP编码帧和变换编码帧的全局增益控制与CELP编解码器的码本激励的增益以及变换编码帧的变换或逆变换的级别一起共同实现。 在另一实施例中，在激励信号的加权域中执行CELP编码中的增益值确定。

公开(公告)号：US08203930B2

公开(公告)日：2012-06-19

申请号：US12083081

申请日：2006-10-04

申请人： Hyen O Oh ,  Hee Suk Pang ,  Dong Soo Kim ,  Jae Hyun Lim ,  Yang Won Jung

发明人： Hyen O Oh ,  Hee Suk Pang ,  Dong Soo Kim ,  Jae Hyun Lim ,  Yang Won Jung

IPC分类号： H04J11/00 ,  H04J3/24 ,  H04W4/00 ,  H04B1/66 ,  H04B14/04 ,  H04L25/34 ,  H04L27/00

CPC分类号： G10L19/18 ,  G10L19/0017 ,  G10L19/008 ,  G10L19/04 ,  G10L19/167 ,  G10L19/20 ,  G10L19/22 ,  H03M7/30 ,  H03M7/40 ,  H03M7/42 ,  H04L27/2602 ,  H04N21/235 ,  H04N21/435 ,  H04S1/007

摘要： A method and apparatus for signal processing which enable data compression and recovery with high transmission efficiency are disclosed. Data coding and entropy coding are performed with correlation and grouping is used to increase coding efficiency. A method for signal processing according to this invention, the method includes decapsulating the signal received over an Internet protocol network, obtaining a pilot reference value corresponding to a plurality of data and a pilot difference value corresponding the pilot reference value from the decapsulated signal and obtaining the data using the pilot reference value and the pilot difference value.

摘要翻译： 公开了一种能够实现高传输效率的数据压缩和恢复的信号处理方法和装置。 数据编码和熵编码通过相关性进行，分组用于提高编码效率。 一种根据本发明的信号处理方法，该方法包括：对通过因特网协议网络接收的信号进行解封装，从解封装信号中获取对应于多个数据的导频参考值和对应于导频参考值的导频差值，并获得 使用导频参考值和导频差值的数据。