公开(公告)号：US06678647B1

公开(公告)日：2004-01-13

申请号：US09586070

申请日：2000-06-02

Applicant: Bernd Andreas Edler ,  Christof Faller

Inventor： Bernd Andreas Edler ,  Christof Faller

IPC: G10L1900

CPC classification number: G10L19/0204 ,  G10L25/18

Abstract: A perceptual audio coder is disclosed for encoding audio signals, such as speech or music, with different spectral and temporal resolutions for the redundancy reduction and irrelevancy reduction using cascaded filterbanks. The disclosed perceptual audio coder includes a first analysis filterbank for performing irrelevancy reduction in accordance with a psychoacoustic model and a second analysis filterbank for performing redundancy reduction. The spectral/temporal resolution of the first filterbank can be optimized for irrelevancy reduction and the spectral/temporal resolution of the second filterbank can be optimized for maximum redundancy reduction. The disclosed perceptual audio coder also includes a scaling block between the cascaded filterbank that scales the spectral coefficients, based on the employed perceptual model.

Abstract translation: 公开了一种感知音频编码器，用于对具有不同频谱和时间分辨率的音频信号（例如语音或音乐）进行编码，以便使用级联滤波器组进行冗余减少和不相关性降低。 公开的感知音频编码器包括用于根据心理声学模型执行无约束减少的第一分析滤波器组和用于执行冗余减少的第二分析滤波器组。 可以对第一滤波器组的频谱/时间分辨率进行优化，以减少无约束，并且可以优化第二滤波器组的频谱/时间分辨率以实现最大冗余度降低。 公开的感知音频编码器还包括基于所采用的感知模型的级联滤波器组之间的缩放块，其缩放频谱系数。

公开(公告)号：US06778953B1

公开(公告)日：2004-08-17

申请号：US09586071

申请日：2000-06-02

Applicant: Bernd Andreas Edler ,  Christof Faller ,  Gerald Dietrich Schuller

Inventor： Bernd Andreas Edler ,  Christof Faller ,  Gerald Dietrich Schuller

IPC: G10L1900

CPC classification number: G10L19/02 ,  G10L25/24

Abstract: A method and apparatus are disclosed for representing the masked threshold in a perceptual audio coder, using line spectral frequencies (LSF) or another representation for linear prediction (LP) coefficients. The present invention calculates LP coefficients for the masked threshold using known LPC analysis techniques. In one embodiment, the masked thresholds are optionally transformed to a non-linear frequency scale suitable for auditory properties. The LP coefficients are converted to line spectral frequencies or a similar representation in which they can be quantized for transmission. In one implementation, the masked threshold is transmitted only if the masked threshold is significantly different from the previous masked threshold. In between each transmitted masked threshold, the masked threshold is approximated using interpolation schemes. The present invention decides which masked thresholds to transmit based on the change of consecutive masked thresholds, as opposed to the variation of short-term spectra.

Abstract translation: 公开了一种用于在感知音频编码器中使用线谱频率（LSF）或用于线性预测（LP）系数的另一表示）来表示屏蔽阈值的方法和装置。 本发明使用已知的LPC分析技术来计算掩蔽阈值的LP系数。 在一个实施例中，屏蔽的阈值可选地被转换成适合于听觉特性的非线性频率标度。 LP系数被转换为线谱频率或类似的表示，其中它们可被量化以用于传输。 在一个实现中，仅当掩蔽的阈值与先前的屏蔽阈值显着不同时才发送屏蔽阈值。 在每个发射的掩蔽阈值之间，使用插值方案来近似掩蔽阈值。 与短期光谱的变化相反，本发明基于连续屏蔽阈值的变化来决定要发送的掩蔽阈值。

公开(公告)号：US08538043B2

公开(公告)日：2013-09-17

申请号：US12719489

申请日：2010-03-08

Applicant: Hyen-O Oh ,  Alexis Favrot ,  Christof Faller ,  Myung Hoon Lee ,  Jong Ha Moon

Inventor： Hyen-O Oh ,  Alexis Favrot ,  Christof Faller ,  Myung Hoon Lee ,  Jong Ha Moon

IPC: H03G1/02

CPC classification number: G10K11/1784 ,  G10K11/178 ,  G10L21/0208 ,  H03G1/02 ,  H03G3/001 ,  H03G5/165 ,  H03G7/002 ,  H03G9/005 ,  H03G9/025

Abstract: A method of processing an audio signal is disclosed. The present invention includes receiving, by an audio processing apparatus, an input signal; receiving user gain input; generating a linear gain factor and a non-linear gain factor using the user gain input; modifying the non-linear gain factor using absolute threshold of hearing and power of the input signal to generate a modified non-linear gain factor; and, applying the linear gain factor and the modified non-linear gain factor to the audio signal.

Abstract translation: 公开了一种处理音频信号的方法。 本发明包括由音频处理装置接收输入信号; 接收用户增益输入; 使用用户增益输入产生线性增益因子和非线性增益因子; 使用输入信号的听觉和功率的绝对阈值来修改非线性增益因子以产生修改的非线性增益因子; 以及将线性增益因子和修改的非线性增益因子应用于音频信号。

公开(公告)号：US08515087B2

公开(公告)日：2013-08-20

申请号：US12719388

申请日：2010-03-08

Applicant: Hyen-O Oh ,  Alexis Favrot ,  Christof Faller ,  Myung Hoon Lee ,  Jong Ha Moon

Inventor： Hyen-O Oh ,  Alexis Favrot ,  Christof Faller ,  Myung Hoon Lee ,  Jong Ha Moon

IPC: G10K11/1784 ,  H03G5/165 ,  H03G3/001 ,  H03G1/02 ,  H03G7/002 ,  G10K11/1788 ,  G10L21/0208 ,  G10K11/178 ,  H03G5/16 ,  H03G3/00 ,  H03G7/00

CPC classification number: G10K11/1784 ,  G10K11/178 ,  G10L21/0208 ,  H03G1/02 ,  H03G3/001 ,  H03G5/165 ,  H03G7/002 ,  H03G9/005 ,  H03G9/025

Abstract: A method of processing an audio signal is disclosed. The present invention includes receiving, by an audio processing apparatus, an input signal; estimating indicator function using a signal power of the input signal; obtaining an adapted filter using the indicator function and an equalization filter; and, generating an output signal by applying the adapted filter to the input signal.

Abstract translation: 公开了一种处理音频信号的方法。 本发明包括由音频处理装置接收输入信号; 使用输入信号的信号功率来估计指标函数; 使用所述指示器功能获得适配滤波器和均衡滤波器; 并且通过将适配滤波器应用于输入信号来产生输出信号。

公开(公告)号：US08359113B2

公开(公告)日：2013-01-22

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 ,  H04B1/00

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：US20110286609A1

公开(公告)日：2011-11-24

申请号：US13147940

申请日：2010-02-09

Applicant: Christof Faller

Inventor： Christof Faller

IPC: H04R3/00

CPC classification number: H04R3/005 ,  G10K11/178 ,  H04R2410/01 ,  H04R2430/20

Abstract: A system and method for use in filtering of an acoustic signal are provided for producing an output signal of attenuated amount of diffuse sound in accordance with predetermined parameters of desired output directional response and required attenuation of diffuse sound. The system includes a filtration module and a filter generation module including a directional analysis module and filter construction module.

Abstract translation: 提供用于对声信号进行滤波的系统和方法，用于根据期望的输出方向响应的预定参数和漫反射声音的所需衰减产生衰减量的扩散声音的输出信号。 该系统包括过滤模块和包括定向分析模块和过滤器构造模块的过滤器生成模块。

公开(公告)号：US08041043B2

公开(公告)日：2011-10-18

申请号：US11652615

申请日：2007-01-12

Applicant: Christof Faller

Inventor： Christof Faller

IPC: H04R5/00 ,  H04R3/00

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（x1，x2）产生多个输出音频通道（y1，...，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，y1，...。 。 。 ，yM，通过使用麦克风生成的音频通道，x1，x2，方向和补偿增益。

公开(公告)号：US07761304B2

公开(公告)日：2010-07-20

申请号：US11719358

申请日：2005-11-22

Applicant: Christof Faller

Inventor： Christof Faller

IPC: G10L19/02

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 downmixed 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）以及派生的提示码一起发送到接收机而不是自动下混音频 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：US20090052601A1

公开(公告)日：2009-02-26

申请号：US12262239

申请日：2008-10-31

Applicant: Christof Faller ,  Raziel Haimi-Cohen

Inventor： Christof Faller ,  Raziel Haimi-Cohen

IPC: H04L7/00

CPC classification number: G10L19/167 ,  H04H60/27 ,  H04H2201/20 ,  H04J3/0632 ,  H04J3/1688 ,  H04N19/152

Abstract: A method and apparatus are disclosed for controlling a buffer in a digital audio broadcasting (DAB) communication system. The decoder buffer level limits are specified in terms of a maximum number of encoded frames (or duration). The transmitter can predict the number of encoded frames, Fpred, in the decoder buffer and transmit the value, Fpred, to the receiver with the audio data. If the transmitter determines that the decoder buffer level is becoming too high, the frames being generated by the encoder are too small and additional bits are allocated to each frame for each of the N programs. Likewise, if the transmitter determines that the decoder buffer level is becoming too low, the frames being generated by the encoder are too big and fewer bits are allocated to each frame for each of the N programs. The transmitted predicted buffer level, Fpred, can also be employed to (i) determine when the decoder should commence decoding frames; and (ii) synchronize the transmitter and the receiver. The receiver fills the decoder buffer until Fpred frames are received before commencing decoding frames when the decoder first starts up or possibly when a new audio program is selected. The transmitter and receiver clocks may be synchronized by adjusting the clock at the receiver by using a feedback loop that compares the actual level of the decoder buffer to the predicted value, Fpred, received from the transmitter (a higher number of encoded frames in the buffer indicates that the clock of the receiver is too slow and should be increased, and a lower number of encoded frames in the buffer indicates that the clock of the receiver is too fast and needs to be slowed down).

Abstract translation: 公开了一种用于控制数字音频广播（DAB）通信系统中的缓冲器的方法和装置。 解码器缓冲器电平限制是根据最大编码帧数（或持续时间）来指定的。 发射机可以在解码器缓冲器中预测编码帧数目Fpred，并将该值Fpred传送到具有音频数据的接收机。 如果发射机确定解码器缓冲器电平变得太高，则由编码器产生的帧太小，并且对于每个N个节目的每个帧分配附加比特。 类似地，如果发射机确定解码器缓冲器电平变得太低，则由编码器生成的帧太大，并且对于N个节目中的每个节目，每个帧分配更少的比特。 发送的预测缓冲器电平Fpred也可以用于（i）确定解码器何时开始解码帧; 和（ii）同步发射机和接收机。 接收器填充解码器缓冲器，直到当解码器首次启动或者当选择新的音频节目时才开始解码帧之前接收到Fpred帧。 发射机和接收机时钟可以通过使用将解码器缓冲器的实际电平与从发送器接收的预测值Fpred（缓冲器中更高数量的编码帧）进行比较的反馈回路来调整接收器处的时钟来同步 表示接收机的时钟太慢，应该增加，缓冲区中编码帧的数量越少表示接收机的时钟速度太快，需要减慢）。

公开(公告)号：US20080192946A1

公开(公告)日：2008-08-14

申请号：US11912068

申请日：2006-04-19

Applicant: Christof Faller

Inventor： Christof Faller

IPC: H04B3/20

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。