公开(公告)号：US20090067634A1

公开(公告)日：2009-03-12

申请号：US12190534

申请日：2008-08-12

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

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

IPC: H04R5/00

CPC classification number: H04S3/008

Abstract: One or more attributes (e.g., pan, gain, etc.) associated with one or more objects (e.g., an instrument) of a stereo or multi-channel audio signal can be modified to provide remix capability.

Abstract translation: 可以修改与立体声或多声道音频信号的一个或多个对象（例如，仪器）相关联的一个或多个属性（例如，平移，增益等）以提供混音能力。

公开(公告)号：US07460629B2

公开(公告)日：2008-12-02

申请号：US09895926

申请日：2001-06-29

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：US20080267413A1

公开(公告)日：2008-10-30

申请号：US12065502

申请日：2006-09-01

Applicant: Christof Faller

Inventor： Christof Faller

IPC: H04R5/00

CPC classification number: H04S3/002 ,  H04S5/00

Abstract: A perceptually motivated spatial decomposition for two-channel stereo audio signals, capturing the information about the virtual sound stage, is proposed. The spatial decomposition allows to re-synthesize audio signals for playback over other sound systems than two-channel stereo. With the use of more front loudspeakers, the width of the virtual sound stage can be increased beyond +/−30° and the sweet spot region is extended. Optionally, lateral independent sound components can be played back separately over loudspeakers on the two sides of a listener to increase listener envelopment. It is also explained how the spatial decomposition can be used with surround sound and wavefield synthesis based audio system. According to the main embodiment of the invention applying to multiple audio signals, it is proposed to generate multiple output audio signals (y1 . . . yM) from multiple input audio signals (x1, . . . , xL), in which the number of output is equal or higher than the number of input signals, this method comprising the steps of: —by means of linear combinations of the input subbands X1(i), . . . , XL(i), computing one or more independent sound subbands representing signal components which are independent between the input subbands, —by means of linear combinations of the input subbands X1(i), . . . , XL(i), computing one or more localized direct sound subbands representing signal components which are contained in more than one of the input subbands and direction factors representing the ratios with which these signal components are contained in two or more input subbands, —generating the output subband signals, Y1(i) . . . YM(i), where each output subband signal is a linear combination of the independent sound subbands and the localized direct sound subbands—converting the output subband signals, Y1(i) . . . YM(i), to time domain audio signals, y1 . . . yM.

公开(公告)号：US20080267333A1

公开(公告)日：2008-10-30

申请号：US12171004

申请日：2008-07-10

Applicant: Christof Faller

Inventor： Christof Faller

IPC: H04L25/00

CPC classification number: H04H60/27 ,  H04B14/04 ,  H04H60/11 ,  H04H2201/20

Abstract: A method and apparatus are disclosed for controlling a buffer in a digital audio broadcasting (DAB) communication system. An audio encoder marks a frame as “dropped” whenever a buffer overflow might occur. Only a small number of bits are utilized to process a lost frame, thereby preventing the buffer from overflowing and allowing the encoder buffer-level to quickly recover from the potential overflow condition. The audio encoder optionally sets a flag that provides an indication to the receivers that a frame has been lost. If a “frame lost” condition is detected by a receiver, the receiver can optionally employ mitigation techniques to reduce the impact of the lost frame(s).

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。 每当发生缓冲区溢出时，音频编码器将帧标记为“丢弃”。 仅使用少量的比特来处理丢失的帧，从而防止缓冲器溢出，并允许编码器缓冲器级从潜在的溢出状态快速恢复。 音频编码器可选择地设置向接收机提供帧丢失的指示。 如果接收机检测到“帧丢失”条件，则接收机可以可选地使用缓解技术来减少丢失帧的影响。

公开(公告)号：US20080201153A1

公开(公告)日：2008-08-21

申请号：US11995700

申请日：2006-07-12

Applicant: Dirk Jeroen Breebaart ,  Lars Falck Villemoes ,  Heiko Purnhagen ,  Christof Faller

Inventor： Dirk Jeroen Breebaart ,  Lars Falck Villemoes ,  Heiko Purnhagen ,  Christof Faller

IPC: G10L19/00

CPC classification number: H04S3/008 ,  G10L19/008

Abstract: A decoder (115) generates a multi channel audio signal, such as a surround sound signal, from a received first signal. The multi-channel signal comprises a second set of audio channels and the first signal comprises a first set of audio channels. The decoder (115) comprises a receiver (401) which receives the first signal. The receiver (401) is coupled to an estimate processor (405) which generates estimated parametric data for the second set of audio channels in response to characteristics of the first set of audio channels. The estimated parametric data relates characteristics of the second set of audio channels to characteristics of the first set of audio channels. The decoder (115) furthermore comprises a spatial audio decoder (403) which decodes the first signal in response to the estimated parametric data to generate the multi-channel signal comprising the second set of channels. The invention allows use of spatial audio decoding with signals that are not encoded by a spatial audio encoder.

Abstract translation: 解码器（115）从接收的第一信号生成诸如环绕声信号的多声道音频信号。 多声道信号包括第二组音频通道，第一信号包括第一组音频通道。 解码器（115）包括接收第一信号的接收机（401）。 接收器（401）耦合到估计处理器（405），该估计处理器响应于第一组音频通道的特性而产生第二组音频通道的估计参数数据。 估计的参数数据将第二组音频通道的特性与第一组音频通道的特性相关联。 解码器（115）还包括空间音频解码器（403），其响应于所估计的参数数据对第一信号进行解码，以产生包括第二组信道的多信道信号。 本发明允许对空间音频编码器未编码的信号使用空间音频解码。

公开(公告)号：US20070003069A1

公开(公告)日：2007-01-04

申请号：US11470314

申请日：2006-09-06

Applicant: Christof Faller

Inventor： Christof Faller

IPC: H04R5/00

CPC classification number: H04M3/56 ,  H04M3/568 ,  H04S3/00 ,  H04S2420/03

Abstract: An auditory scene is synthesized by applying two or more different sets of one or more spatial parameters (e.g., an inter-ear level difference (ILD), inter-ear time difference (ITD), and/or head-related transfer function (HRTF)) to two or more different frequency bands of a combined audio signal, where each different frequency band is treated as if it corresponded to a single audio source in the auditory scene. In one embodiment, the combined audio signal corresponds to the combination of two or more different source signals, where each different frequency band corresponds to a region of the combined audio signal in which one of the source signals dominates the others. In this embodiment, the different sets of spatial parameters are applied to synthesize an auditory scene comprising the different source signals.

Abstract translation: 通过应用两个或更多个不同的一个或多个空间参数集合（例如，耳朵间电平差（ILD），耳间时间差（ITD）和/或头部相关传递函数（HRTF）来合成听觉场景 ））到组合音频信号的两个或更多个不同的频带，其中每个不同的频带被视为对应于听觉场景中的单个音频源。 在一个实施例中，组合的音频信号对应于两个或更多个不同的源信号的组合，其中每个不同的频带对应于组合的音频信号的一个区域，其中源信号之一占主导地位。 在该实施例中，应用不同的空间参数集合来合成包括不同源信号的听觉场景。

公开(公告)号：US07062040B2

公开(公告)日：2006-06-13

申请号：US10251404

申请日：2002-09-20

Applicant: Christof Faller

Inventor： Christof Faller

IPC: H04M1/00

CPC classification number: H04M9/082

Abstract: In a microphone signal, the signal component corresponding to, e.g., echo is suppressed using an echo control scheme that estimates the spectral envelope of the echo signal, without having to estimate the waveform for the echo signal. In one embodiment, the input signal (to be applied to a loudspeaker) and the microphone signal are spectrally decomposed into multiple subbands, where echo suppression processing is independently performed on each subband. The echo control of the present invention can be implemented with substantially reduced (1) computational complexity and (2) phase sensitivity, as compared to traditional acoustic echo cancellation, in which the waveform for the echo signal is estimated.

Abstract translation: 在麦克风信号中，使用估计回波信号的频谱包络的​​回波控制方案来抑制对应于例如回波的信号分量，而不必估计回波信号的波形。 在一个实施例中，输入信号（要施加到扬声器）和麦克风信号被频谱分解成多个子带，其中在每个子带上独立地执行回波抑制处理。 与传统的回声消除相比，可以实质地减少（1）计算复杂度和（2）相位灵敏度来实现本发明的回波控制，其中估计回波信号的波形。

公开(公告)号：US20050195981A1

公开(公告)日：2005-09-08

申请号：US10827900

申请日：2004-04-20

Applicant: Christof Faller ,  Juergen Herre

Inventor： Christof Faller ,  Juergen Herre

IPC: G10L19/00 ,  H04S3/00 ,  H04R5/00

CPC classification number: H04S3/00 ,  G10L19/008 ,  H04S2420/03

Abstract: For a multi-channel audio signal, parametric coding is applied to different subsets of audio input channels for different frequency regions. For example, for a 5.1 surround sound signal having five regular channels and one low-frequency (LFE) channel, binaural cue coding (BCC) can be applied to all six audio channels for sub-bands at or below a specified cut-off frequency, but to only five audio channels (excluding the LFE channel) for sub-bands above the cut-off frequency. Such frequency-based coding of channels can reduce the encoding and decoding processing loads and/or size of the encoded audio bitstream relative to parametric coding techniques that are applied to all input channels over the entire frequency range.

Abstract translation: 对于多声道音频信号，参数编码被应用于不同频率区域的音频输入通道的不同子集。 例如，对于具有五个常规频道和一个低频（LFE）频道的5.1环绕声信号，可以将双耳提示编码（BCC）应用于所有六个音频通道，用于等于或小于指定截止频率的子频带 ，但对于截止频率以上的子频带，只有五个音频通道（不包括LFE通道）。 通道的这种基于频率的编码可以相对于在整个频率范围上应用于所有输入通道的参数编码技术来减少编码和解码处理负载和/或编码音频比特流的大小。

公开(公告)号：US20050157883A1

公开(公告)日：2005-07-21

申请号：US10762100

申请日：2004-01-20

Applicant: Jurgen Herre ,  Christof Faller

Inventor： Jurgen Herre ,  Christof Faller

IPC: H04S3/02 ,  H04R5/00 ,  H04B1/00

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。

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。