公开(公告)号：US08275610B2

公开(公告)日：2012-09-25

申请号：US11855500

申请日：2007-09-14

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

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

IPC: G10L19/14

CPC classification number: H04S3/008 ,  G10L19/008 ,  G10L21/0232 ,  H04S5/00 ,  H04S2400/05 ,  H04S2420/03 ,  H04S2420/07

Abstract: A plural-channel audio signal (e.g., a stereo audio) is processed to modify a gain (e.g., a volume or loudness) of a speech component signal (e.g., dialogue spoken by actors in a movie) relative to an ambient component signal (e.g., reflected or reverberated sound) or other component signals. In one aspect, the speech component signal is identified and modified. In one aspect, the speech component signal is identified by assuming that the speech source (e.g., the actor currently speaking) is in the center of a stereo sound image of the plural-channel audio signal and by considering the spectral content of the speech component signal.

Abstract translation: 处理多声道音频信号（例如，立体声音频）以修改相对于环境分量信号的语音分量信号（例如，电影中的演员所说的对话）的增益（例如，音量或响度） 例如，反射或混响的声音）或其他分量信号。 在一个方面，识别和修改语音分量信号。 在一个方面，通过假设语音源（例如，当前演员）位于多声道音频信号的立体声图像的中心并且通过考虑语音分量的频谱内容来识别语音分量信号 信号。

公开(公告)号：US08275150B2

公开(公告)日：2012-09-25

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: H04B15/00

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。

公开(公告)号：US08200500B2

公开(公告)日：2012-06-12

申请号：US13046947

申请日：2011-03-14

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：US07742592B2

公开(公告)日：2010-06-22

申请号：US11912068

申请日：2006-04-19

Applicant: Christof Faller

Inventor： Christof Faller

IPC: H04M9/08

CPC classification number: H04M9/082

Abstract: Acoustic echo control and noise suppression is an important part of any “handsfree” telecommunication system, such as telephony or audio or video conferencing systems. Bandwidth and computational complexity constraints have prevented that stereo or multi-channel telecommunication systems have been widely applied. The advantages are very low complexity, high robustness, scalability to multi-channel audio without a need for loudspeaker signal distortion, and efficient integration of echo and noise control in the same algorithm. The proposed method for processing audio signals, comprises the steps of: —receiving an input signal, wherein the input signal is applied to a loudspeaker; —receiving a microphone signal generated by a microphone; —estimating the delay between the loudspeaker and the microphone signals and obtaining a delayed loudspeaker signal, —estimating a coloration correction values of the echo path on the delayed loudspeaker signal, —using information of the delayed loudspeaker signal, microphone signal, and coloration correction values to determine gain filter values, —apply the gain filter values to the microphone signal to remove the echo.

Abstract translation: 声学回声控制和噪声抑制是任何“免提”电信系统的重要组成部分，例如电话或音频或视频会议系统。 带宽和计算复杂度约束阻止了立体声或多声道电信系统已被广泛应用。 其优点是非常低的复杂性，高鲁棒性，多通道音频的可扩展性，而不需要扬声器信号失真，并且在相同的算法中有效地整合回波和噪声控制。 所提出的用于处理音频信号的方法包括以下步骤： - 接收输入信号，其中所述输入信号被施加到扬声器; - 接收由麦克风产生的麦克风信号; - 确定扬声器和麦克风信号之间的延迟并获得延迟的扬声器信号， - 估计延迟的扬声器信号上的回波路径的着色校正值， - 使用延迟的扬声器信号，麦克风信号和着色校正值的信息 以确定增益滤波器值，将增益滤波器值应用于麦克风信号以消除回波。

公开(公告)号：US20100085117A1

公开(公告)日：2010-04-08

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: H03F1/26

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：US20090319282A1

公开(公告)日：2009-12-24

申请号：US12550519

申请日：2009-08-31

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

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

IPC: G10L19/00

CPC classification number: G10L19/008 ,  H04S3/02

Abstract: In one embodiment, C input audio channels are encoded to generate E transmitted audio channel(s), where one or more cue codes are generated for two or more of the C input channels, and the C input channels are downmixed to generate the E transmitted channel(s), where C>E≧1. One or more of the C input channels and the E transmitted channel(s) are analyzed to generate a flag indicating whether or not a decoder of the E transmitted channel(s) should perform envelope shaping during decoding of the E transmitted channel(s). In one implementation, envelope shaping adjusts a temporal envelope of a decoded channel generated by the decoder to substantially match a temporal envelope of a corresponding transmitted channel.

Abstract translation: 在一个实施例中，对C个输入音频信道进行编码以产生E个发送的音频信道，其中为两个或更多个C个输入信道生成一个或多个提示码，并且将C个输入信道下混合以产生E个发送的 通道，其中C> E> = 1。 分析C个输入信道和E个发送信道中的一个或多个，以产生一个标志，该标志指示E个被发送的信道的解码器是否应在E个发送的信道的解码期间执行包络整形， 。 在一个实现中，包络整形调整由解码器产生的解码信道的时间包络，以使其对应的传输信道的时间包络基本匹配。

公开(公告)号：US20090150161A1

公开(公告)日：2009-06-11

申请号：US11719358

申请日：2005-11-22

Applicant: Christof Faller

Inventor： Christof Faller

IPC: G10L19/00

CPC classification number: G10L19/008

Abstract: Embodiments of the present invention are directed to a binaural cue coding (BCC) scheme in which an externally provided audio signal (e.g., a studio engineering audio signal) is transmitted, along with derived cue codes, to a receiver instead of an automatically downmixcd audio signal. The cue codes are (adaptively) synchronized with the externally provided audio signal to compensate for time lags (and changes in those time lags) between the externally downmixed audio signal and the multi-channel signal used to generate the cue codes. If the receiver is a legacy receiver, then the studio engineered audio signal will typically provide a higher-quality playback than would be provided by the automatically downmixed audio signal. If the receiver is a BCC-capable receiver, then the synchronization of the cue codes with the externally provided audio signal will typically improve the quality of the synthesized playback.

Abstract translation: 本发明的实施例涉及一种双耳提示编码（BCC）方案，其中将外部提供的音频信号（例如，工作室工程音频信号）连同导出的提示码一起发送到接收机而不是自动缩混音频 信号。 提示码与外部提供的音频信号（自适应地）同步，以补偿外部下混音频信号与用于生成提示码的多声道信号之间的时间滞后（和那些时间滞后的变化）。 如果接收机是传统接收机，则工作室设计的音频信号通常将提供比由自动缩混音频信号提供的更高质量的回放。 如果接收机是具有BCC能力的接收机，则提示码与外部提供的音频信号的同步通常将提高合成回放的质量。

公开(公告)号：US20080170728A1

公开(公告)日：2008-07-17

申请号：US11652615

申请日：2007-01-12

Applicant: Christof Faller

Inventor： Christof Faller

IPC: H04R5/027

CPC classification number: H04S5/005 ,  H04R5/027

Abstract: Surround sound recording is a tedious task requiring the use of many microphones. The invention aims at enabling the use of two-channel microphones (or stereo microphones) for multi-channel surround recording. A conventional stereo microphone, or a two-channel microphone specifically optimized for use with the proposed algorithm, is used to generate two signals. A post-processor is applied to the microphone generated signals to convert them to multi-channel surround.This aim is achieved through a method to generate multiple output audio channels (y1, . . . , yM) from two microphone generated audio channels (x1, x2), in which the number of output channels is equal or higher than two, this method comprising the steps of: determine directions of sound components related to the microphone characteristics determine compensation gains of sound components related to the microphone characteristics generating the output audio channels, y1, . . . , yM, by using the microphone generated audio channels, x1, x2, directions, and compensation gains

Abstract translation: 环绕声录音是一项繁琐的任务，需要使用许多麦克风。 本发明旨在使得可以使用双声道麦克风（或立体声麦克风）进行多声道环绕录音。 专门针对所提出的算法使用的传统的立体声麦克风或特别优化的双通道麦克风用于产生两个信号。 后处理器应用于麦克风产生的信号，以将其转换为多声道环绕声。 该目的通过一种从两个麦克风产生的音频通道（x 1，x 2）产生多个输出音频通道（y 1，...，yM）的方法实现，其中输出通道的数量等于或高于两个 该方法包括以下步骤：确定与麦克风特性相关的声音分量的方向确定与产生输出音频通道的麦克风特性相关的声音分量的补偿增益，y 1，...。 。 。 ，yM，通过使用麦克风生成的音频通道，x 1，x 2，方向和补偿增益

公开(公告)号：US20080167864A1

公开(公告)日：2008-07-10

申请号：US11855500

申请日：2007-09-14

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

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

IPC: G10L19/14

CPC classification number: H04S3/008 ,  G10L19/008 ,  G10L21/0232 ,  H04S5/00 ,  H04S2400/05 ,  H04S2420/03 ,  H04S2420/07

Abstract: A plural-channel audio signal (e.g., a stereo audio) is processed to modify a gain (e.g., a volume or loudness) of a speech component signal (e.g., dialogue spoken by actors in a movie) relative to an ambient component signal (e.g., reflected or reverberated sound) or other component signals. In one aspect, the speech component signal is identified and modified. In one aspect, the speech component signal is identified by assuming that the speech source (e.g., the actor currently speaking) is in the center of a stereo sound image of the plural-channel audio signal and by considering the spectral content of the speech component signal.

Abstract translation: 处理多声道音频信号（例如，立体声音频）以修改相对于环境分量信号的语音分量信号（例如，电影中的演员所说的对话）的增益（例如，音量或响度） 例如，反射或混响的声音）或其他分量信号。 在一个方面，识别和修改语音分量信号。 在一个方面，通过假设语音源（例如，当前演员）位于多声道音频信号的立体声图像的中心并且通过考虑语音分量的频谱内容来识别语音分量信号 信号。

公开(公告)号：US07394903B2

公开(公告)日：2008-07-01

申请号：US10762100

申请日：2004-01-20

Applicant: Jürgen Herre ,  Christof Faller

Inventor： Jürgen Herre ,  Christof Faller

IPC: H04R5/00

CPC classification number: G10L19/008 ,  H04S3/02 ,  H04S2420/03

Abstract: The apparatus for constructing a multi-channel output signal using an input signal and parametric side information, the input signal including the first input channel and the second input channel derived from an original multi-channel signal, and the parametric side information describing interrelations between channels of the multi-channel original signal uses base channels for synthesizing first and second output channels on one side of an assumed listener position, which are different from each other. The base channels are different from each other because of a coherence measure. Coherence between the base channels (for example the left and the left surround reconstructed channel) is reduced by calculating a base channel for one of those channels by a combination of the input channels, the combination being determined by the coherence measure. Thus, a high subjective quality of the reconstruction can be obtained because of an approximated original front/back coherence.

Abstract translation: 用于使用输入信号和参数侧信息构造多通道输出信号的装置，包括从原始多通道信号导出的第一输入通道和第二输入通道的输入信号以及描述通道之间的相互关系的参数侧信息 多信道原始信号使用用于合成彼此不同的假定收听者位置的一侧上的第一和第二输出声道的基本通道。 由于一致性测量，基本通道彼此不同。 通过输入通道的组合计算这些通道中的一个通道的基本通道来减小基本通道（例如左和左环绕重建通道）之间的相干性，该组合由相干性测量确定。 因此，由于近似的原始前/后相干性，可以获得重建的高主观质量。