公开(公告)号：US08275154B2

公开(公告)日：2012-09-25

申请号：US12511669

申请日：2009-07-29

Applicant: Jong Ha Moon ,  Hyen O Oh ,  Joon Il Lee ,  Myung Hoon Lee ,  Yang Won Jung ,  Alexis Favrot ,  Christof Faller

Inventor： Jong Ha Moon ,  Hyen O Oh ,  Joon Il Lee ,  Myung Hoon Lee ,  Yang Won Jung ,  Alexis Favrot ,  Christof Faller

IPC: H03G3/00

CPC classification number: H03G9/025 ,  H03G7/007 ,  H03G9/005

Abstract: An apparatus for processing an audio signal and method thereof are disclosed, by which a local dynamic range of an audio signal can be adaptively normalized as well as a maximum dynamic range of the audio signal. The present invention includes receiving, by an audio processing apparatus, a signal, and feedback information estimated based on a normalizing gain; generating a noise estimation based on the signal; computing a gain filter for noise canceling, based on the noise estimation and the signal; and, obtaining a restricted gain filter by applying the feedback information to the gain filter.

Abstract translation: 公开了一种用于处理音频信号的装置及其方法，通过该装置可以将音频信号的局部动态范围自适应地归一化以及音频信号的最大动态范围。 本发明包括由音频处理装置接收基于标准化增益估计的信号和反馈信息; 基于该信号生成噪声估计; 基于噪声估计和信号计算噪声消除的增益滤波器; 并且通过将所述反馈信息应用于所述增益滤波器来获得受限增益滤波器。

公开(公告)号：US08213623B2

公开(公告)日：2012-07-03

申请号：US11652614

申请日：2007-01-12

Applicant: Christof Faller

Inventor： Christof Faller

IPC: H04R5/00 ,  H04R3/00

CPC classification number: H04R3/005

Abstract: The directionality of microphones is often not high enough, resulting in compromised music recording. Beamforming for getting a signal with a higher directional response is limited due to spatial aliasing, dependence of beamwidth on frequency, and a requirement of a high number of microphones. The invention proposes a method to generate an output audio signal y from two or more input audio signals (x1, x2, . . . ), this method comprising the steps of: define one input signal as reference signal for each of the other input signals compute gain factors related to how much of the input signal is contained in the reference signal adjust the gain factors using a limiting function compute the output signal by subtracting from the reference signal the other input signals multiplied by the corresponding adjusted gain factors.

Abstract translation: 麦克风的方向性往往不够高，导致音乐录音受损。 用于获得具有较高方向响应的信号的波束成形由于空间混叠，波束宽度对频率的依赖性以及大量麦克风的需求而受到限制。 本发明提出一种从两个或多个输入音频信号（x1，x2，...）生成输出音频信号y的方法，该方法包括以下步骤：将一个输入信号定义为每个其它输入信号的参考信号 计算与参考信号中包含多少输入信号相关的增益因子使用限制函数调整增益因子，通过从参考信号中减去乘以相应的调整增益因子的其他输入信号来计算输出信号。

公开(公告)号：US07941320B2

公开(公告)日：2011-05-10

申请号：US12548773

申请日：2009-08-27

Applicant: Frank Baumgarte ,  Jiashu Chen ,  Christof Faller

Inventor： Frank Baumgarte ,  Jiashu Chen ,  Christof Faller

IPC: G10L11/04 ,  G10L19/00 ,  G10L21/00 ,  H04R5/00 ,  H04B14/04

CPC classification number: H04S1/007 ,  G10L15/22 ,  G10L19/008 ,  G10L19/0204 ,  G10L19/0212 ,  H04S3/002 ,  H04S3/02 ,  H04S5/00 ,  H04S5/005 ,  H04S2420/03

Abstract: Generic and specific C-to-E binaural cue coding (BCC) schemes are described, including those in which one or more of the input channels are transmitted as unmodified channels that are not downmixed at the BCC encoder and not upmixed at the BCC decoder. The specific BCC schemes described include 5-to-2, 6-to-5, 7-to-5, 6.1-to-5.1, 7.1-to-5.1, and 6.2-to-5.1, where “0.1” indicates a single low-frequency effects (LFE) channel and “0.2” indicates two LFE channels.

Abstract translation: 描述了通用和特定的C-to-E双耳提示编码（BCC）方案，包括其中一个或多个输入通道作为未修改的通道发送的未修改的通道，其未在BCC编码器下混合，而不在BCC解码器处混合。 所描述的具体的BCC方案包括5到2,6到5,7到5,6.1到5.1,7.1到5.1和6.2到5.1，其中“0.1”表示单个 低频效应（LFE）通道和“0.2”表示两个LFE通道。

公开(公告)号：US07903824B2

公开(公告)日：2011-03-08

申请号：US11032689

申请日：2005-01-10

Applicant: Christof Faller ,  Juergen Herre

Inventor： Christof Faller ,  Juergen Herre

IPC: H04R5/00

CPC classification number: G10L19/008

Abstract: At an audio encoder, cue codes are generated for one or more audio channels, wherein a combined cue code (e.g., a combined inter-channel correlation (ICC) code) is generated by combining two or more estimated cue codes, each estimated cue code estimated from a group of two or more channels. At an audio decoder, E transmitted audio channel(s) are decoded to generate C playback audio channels. Received cue codes include a combined cue code (e.g., a combined ICC code). One or more transmitted channel(s) are upmixed to generate one or more upmixed channels. One or more playback channels are synthesized by applying the cue codes to the one or more upmixed channels, wherein two or more derived cue codes are derived from the combined cue code, and each derived cue code is applied to generate two or more synthesized channels.

Abstract translation: 在音频编码器处，为一个或多个音频通道生成提示码，其中通过组合两个或多个估计的提示码来生成组合的提示码（例如，组合的信道间相关（ICC）码），每个估计的提示码 从一组两个或多个渠道估计。 在音频解码器处，E个发送的音频信道被解码以产生C个播放音频信道。 接收的提示码包括组合的提示码（例如，组合的ICC码）。 一个或多个传输的信道被混合以产生一个或多个上混频道。 通过将提示码应用于一个或多个上混合信道来合成一个或多个播放频道，其中从组合提示码导出两个或更多个派生提示码，并且应用每个派生的提示码以生成两个或多个合成频道。

公开(公告)号：US20110019833A1

公开(公告)日：2011-01-27

申请号：US12864890

申请日：2009-01-16

Applicant: Fabian Kuech ,  Markus Kallinger ,  Christof Faller ,  Alexis Favrot

Inventor： Fabian Kuech ,  Markus Kallinger ,  Christof Faller ,  Alexis Favrot

IPC: H04B3/23

CPC classification number: H04M9/082 ,  G10L21/0208 ,  G10L2021/02082 ,  H04B3/237

Abstract: A preferred embodiment of an apparatus for computing filter coefficients for an adaptive filter for filtering a microphone signal so as to suppress an echo due to a loudspeaker signal includes an extractor for extracting a stationary component signal or a non-stationary component signal from the loudspeaker signal or from a signal derived from the loudspeaker signal, and a computer for computing the filter coefficients for the adaptive filter on the basis of the extracted stationary component signal or the extracted non-stationary component signal.

Abstract translation: 用于计算用于滤波麦克风信号的滤波器系数的滤波器系数的装置的优选实施例，以便抑制由于扬声器信号引起的回波，包括用于从扬声器信号中提取静止分量信号或非平稳分量信号的提取器 或者根据从扬声器信号得到的信号，以及计算机，用于基于所提取的固定分量信号或所提取的非平稳分量信号来计算用于自适应滤波器的滤波器系数。

公开(公告)号：US07787631B2

公开(公告)日：2010-08-31

申请号：US11058747

申请日：2005-02-15

Applicant: Christof Faller

Inventor： Christof Faller

IPC: H04R5/00 ,  G10L19/00

CPC classification number: G10L19/008 ,  H04S3/008

Abstract: A binaural cue coding scheme in which cue codes are derived from the transmitted audio signal. In one embodiment, an encoder downmixes C input channels to generate E transmitted channels, where C>E>1. A decoder derives cue codes from the transmitted channels and uses those cue codes to synthesize playback channels. For example, in one 5-to-2 BCC embodiment, the encoder downmixes a 5-channel surround signal to generate left and right channels of a stereo signal. The decoder derives stereo cues from the transmitted stereo signal, maps those stereo cues to surround cues, and applies the surround cues to the transmitted stereo channels to generate playback channels of a 5-channel synthesized surround signal.

Abstract translation: 一种双声道提示编码方案，其中从发送的音频信号导出提示码。 在一个实施例中，编码器将混合C个输入信道以产生E个发送的信道，其中C> E> 1。 解码器从发送的频道中导出提示码，并使用这些提示码来合成播放频道。 例如，在一个5对2的BCC实施例中，编码器将5声道环绕信号下混合以产生立体声信号的左声道和右声道。 解码器从传输的立体声信号中导出立体声提示，将这些立体声提示视为环绕声，并将环绕线索应用于传输的立体声通道，以产生5声道合成环绕声信号的播放通道。

公开(公告)号：US20100189266A1

公开(公告)日：2010-07-29

申请号：US12530615

申请日：2008-03-07

Applicant: Hyen O. Oh ,  Yang Won Jung ,  Christof Faller

Inventor： Hyen O. Oh ,  Yang Won Jung ,  Christof Faller

IPC: H04R5/00

CPC classification number: H04S3/002 ,  G10L19/008 ,  H04S3/008 ,  H04S2400/01 ,  H04S2400/11 ,  H04S2420/03

Abstract: A method of processing an audio signal is disclosed. The present invention comprises receiving a downmix signal, object information and preset information, generating downmix processing information using the object information and the preset information, processing the downmix signal using the downmix processing information, and generating multi-channel information using the object information and the preset information, wherein the preset information is extracted from a bitstream. Accordingly, a gain and panning of an object can be easily controlled without user's setting for each object using preset information set in advance. And, a gain and panning of an object can be controlled using preset information modified based on a selection made by a user.

Abstract translation: 公开了一种处理音频信号的方法。 本发明包括接收下混信号，对象信息和预设信息，使用对象信息和预设信息生成下混处理信息，使用下混处理信息处理下混信号，并使用对象信息生成多声道信息， 预置信息，其中从比特流中提取预设信息。 因此，可以容易地控制物体的增益和平移，而无需使用预先设定的预设信息对每个物体进行用户设定。 并且，可以使用基于用户进行的选择修改的预设信息来控制对象的增益和平移。

公开(公告)号：US07720230B2

公开(公告)日：2010-05-18

申请号：US11006482

申请日：2004-12-07

Applicant: Eric Allamanche ,  Sascha Disch ,  Christof Faller ,  Juergen Herre

Inventor： Eric Allamanche ,  Sascha Disch ,  Christof Faller ,  Juergen Herre

IPC: H04R5/00

CPC classification number: G10L19/008

Abstract: At an audio encoder, cue codes are generated for one or more audio channels, wherein an envelope cue code is generated by characterizing a temporal envelope in an audio channel. At an audio decoder, E transmitted audio channel(s) are decoded to generate C playback audio channels, where C>E≧1. Received cue codes include an envelope cue code corresponding to a characterized temporal envelope of an audio channel corresponding to the transmitted channel(s). One or more transmitted channel(s) are upmixed to generate one or more upmixed channels. One or more playback channels are synthesized by applying the cue codes to the one or more upmixed channels, wherein the envelope cue code is applied to an upmixed channel or a synthesized signal to adjust a temporal envelope of the synthesized signal based on the characterized temporal envelope such that the adjusted temporal envelope substantially matches the characterized temporal envelope.

Abstract translation: 在音频编码器中，为一个或多个音频通道生成提示码，其中通过表征音频通道中的时间包络来产生包络线索码。 在音频解码器处，对E个发送的音频信道进行解码以生成C个回放音频信道，其中C>E≥1。 接收的提示码包括与对应于所发送的频道的音频信道的特征化时间包络对应的信封提示码。 一个或多个传输的信道被混合以产生一个或多个上混频道。 通过将提示码应用于一个或多个上混合通道来合成一个或多个回放通道，其中，将包络提示码应用于上混合通道或合成信号，以基于表征的时间包络线来调整合成信号的时间包络 使得调整的时间包络基本上与所表征的时间包络相匹配。

公开(公告)号：US20100106270A1

公开(公告)日：2010-04-29

申请号：US12530604

申请日：2008-03-07

Applicant: Hyen O. Oh ,  Yang Won Jung ,  Christof Faller

Inventor： Hyen O. Oh ,  Yang Won Jung ,  Christof Faller

IPC: G06F17/00

CPC classification number: H04S3/002 ,  G10L19/008 ,  H04S3/008 ,  H04S2400/01 ,  H04S2400/11 ,  H04S2420/03

Abstract: A method of processing an audio signal is disclosed. The present invention comprises receiving downmix signal including object signals, transforming the downmix signal per frequency band, determining a direction of an object signal from the transformed downmix signal, and determining blind information by estimating a level of the object signal corresponding to the direction. Accordingly, the present invention generates blind information in case of using an encoder incapable of generating object information, thereby enabling a gain and panning of object to be controlled using the blind information.

Abstract translation: 公开了一种处理音频信号的方法。 本发明包括接收包括对象信号的下混信号，对每个频带变换下混信号，从变换的下混信号确定对象信号的方向，以及通过估计对应于该方向的对象信号的电平来确定盲信息。 因此，本发明在使用不能产生对象信息的编码器的情况下产生盲信息，从而能够利用盲信息来对要被控制的对象进行增益和平移。

公开(公告)号：US20100061566A1

公开(公告)日：2010-03-11

申请号：US12511589

申请日：2009-07-29

Applicant: Jong Ha Moon ,  Hyen O. Oh ,  Joon Il Lee ,  Myung Hoon Lee ,  Yang Won Jung ,  Alexis Favrot ,  Christof Faller

Inventor： Jong Ha Moon ,  Hyen O. Oh ,  Joon Il Lee ,  Myung Hoon Lee ,  Yang Won Jung ,  Alexis Favrot ,  Christof Faller

IPC: G10K11/16 ,  H03G3/00

CPC classification number: H03G9/025 ,  H03G7/007 ,  H03G9/005

Abstract: An apparatus for processing an audio signal and method thereof are disclosed, by which a local dynamic range of an audio signal can be adaptively normalized as well as a maximum dynamic range of the audio signal. The present invention includes receiving a signal, by an audio processing apparatus; computing a long-term power and a short-term power by estimating power of the signal; generating a slow gain based on the long-term power; generating a fast gain based on the short-term power; obtaining a final gain by combining the slow gain and the fast gain; and, modifying the signal using the final gain.

Abstract translation: 公开了一种用于处理音频信号的装置及其方法，通过该装置可以将音频信号的局部动态范围自适应地归一化以及音频信号的最大动态范围。 本发明包括由音频处理装置接收信号; 通过估计信号的功率来计算长期功率和短期功率; 基于长期权力产生缓慢的收益; 基于短期电力产生快速增长; 通过组合慢增益和快速增益获得最终增益; 并使用最终增益修改信号。