公开(公告)号：US20080130904A1

公开(公告)日：2008-06-05

申请号：US11667747

申请日：2005-11-22

Applicant: Christof Faller

Inventor： Christof Faller

IPC: H04R5/00

CPC classification number: H04S1/002 ,  G10L19/008

Abstract: A binaural cue coding scheme involving one or more object-based cue codes, wherein an object-based cue code directly represents a characteristic of an auditory scene corresponding to the audio channels, where the characteristic is independent of number and positions of loudspeakers used to create the auditory scene. Examples of object-based cue codes include the angle of an auditory event, the width of the auditory event, the degree of envelopment of the auditory scene, and the directionality of the auditory scene.

Abstract translation: 一种涉及一个或多个基于对象的提示码的双耳提示编码方案，其中基于对象的提示码直接表示对应于音频信道的听觉场景的特征，其中该特征独立于用于创建的扬声器的数量和位置 听觉场面。 基于对象的提示代码的示例包括听觉事件的角度，听觉事件的宽度，听觉场景的包络程度以及听觉场景的方向性。

公开(公告)号：US07062429B2

公开(公告)日：2006-06-13

申请号：US09948431

申请日：2001-09-07

Applicant: Christof Faller

Inventor： Christof Faller

IPC: G10L19/00

CPC classification number: H04B1/665 ,  H04B1/66

Abstract: A method and apparatus are disclosed for controlling a buffer in a communication system, such as a digital audio broadcasting (DAB) communication system. A more consistent perceptual quality over time provides for a more pleasing auditory experience to a listener. Thus, the disclosed bit allocation process determines, for each frame, a distortion d[k] at which the frame is to be encoded. Generally, the distortion d[k] is determined to minimize (i) the probability for a buffer overflow, and (ii) the variation of perceived distortion over time. A buffer level is controlled by partitioning a signal into a sequence of successive frames; estimating a distortion rate for a number of frames; and selecting a distortion such that the variance of the buffer level is bounded by a specified value. In one implementation, a signal is coded by partitioning the signal into a sequence of successive frames; encoding each frame k for each of a plurality of distortions Di to compute a frame bitrate; estimating an average bitrate Ri[k] for each of said plurality of distortions Di given current and past frame bitrates; interpolating between each of said pair of values for said average bitrate Ri[k] and said plurality of distortions Di to obtain an approximation of a function that maps a distortion to an estimated average bitrate; and encoding each frame with a distortion level determined from said function.

Abstract translation: 公开了一种用于控制诸如数字音频广播（DAB）通信系统的通信系统中的缓冲器的方法和装置。 随着时间的推移，更一致的感知质量可以为聆听者提供更愉悦的听觉体验。 因此，所公开的比特分配处理针对每个帧确定要对其进行编码的失真d [k]。 通常，确定失真d [k]使（i）缓冲器溢出的概率最小化，以及（ii）随时间的感知失真的变化。 通过将信号分成连续帧序列来控制缓冲器级; 估计多个帧的失真率; 并选择一个失真，使得缓冲器级别的方差被指定的值限制。 在一个实现中，通过将信号划分为连续帧序列来对信号进行编码; 针对多个失真D i i中的每一个对每个帧k进行编码以计算帧比特率; 针对给定当前和过去帧比特率的所述多个失真D i i中的每一个估计平均比特率R i i [k] 对于所述平均比特率R i i [k]和所述多个失真D i i i的所述一对值中的每一个之间内插，以获得将失真映射到 估计平均比特率; 并且从由所述功能确定的失真水平对每个帧进行编码。

公开(公告)号：US20050058304A1

公开(公告)日：2005-03-17

申请号：US10936464

申请日：2004-09-08

Applicant: Frank Baumgarte ,  Jiashu Chen ,  Christof Faller

Inventor： Frank Baumgarte ,  Jiashu Chen ,  Christof Faller

IPC: G10L15/22 ,  G10L19/02 ,  H04S1/00 ,  H04S3/00 ,  H04S5/00 ,  H03G5/00 ,  H04R5/00 ,  H04B15/00

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：US06499010B1

公开(公告)日：2002-12-24

申请号：US09477314

申请日：2000-01-04

Applicant: Christof Faller

Inventor： Christof Faller

IPC: G10L1902

CPC classification number: G10L19/002 ,  G10L19/0208

Abstract: A method (and apparatus) for coding an audio signal, the method comprising the steps of partitioning the audio signal into a sequence of successive frames; calculating one or more noise thresholds for each of a plurality of frames in the sequence, each noise threshold for a particular one of the frames corresponding to a different perceptual coding quality for the particular frame; estimating a bit demand for each of a corresponding one or more perceptual coding qualities for each frame, wherein each estimated bit demand comprises a number of bits which would be used to code a given frame at the corresponding perceptual coding quality; selecting one of the perceptual coding qualities for the coding of a particular frame based upon the estimated bit demand for the perceptual coding quality for the particular frame, and further based on one or more bit demands estimated for one or more other frames; and coding the particular frame based on the noise threshold corresponding to the selected perceptual coding quality for the particular frame. In particular, and in accordance with one illustrative embodiment of the present invention, the average bit demand for coding each of a plurality of frames at each of a plurality of different perceptual coding qualities is advantageously estimated, and based on these estimates, each frame is coded so as to maintain a relatively consistent perceptual coding quality from one frame to the next.

Abstract translation: 一种用于对音频信号进行编码的方法（和装置），该方法包括以下步骤：将音频信号划分为连续帧序列; 为所述序列中的多个帧中的每一个计算一个或多个噪声阈值，对于所述特定帧的不同感知编码质量的每个帧中的特定一个的每个噪声阈值; 针对每个帧估计对应的一个或多个感知编码质量中的每一个的比特请求，其中每个估计的比特请求包括将被用于以相应的感知编码质量对给定帧进行编码的比特数; 基于针对特定帧的感知编码质量的估计比特需求，并且还基于针对一个或多个其他帧估计的一个或多个比特请求，选择用于特定帧的编码的感知编码质量之一; 并且基于与特定帧的选择的感知编码质量相对应的噪声阈值对特定帧进行编码。 特别地，根据本发明的一个说明性实施例，有利地估计以多个不同的感知编码质量中的每一个编码多个帧中的每一个的平均比特请求，并且基于这些估计，每个帧是 编码，以便保持从一帧到下一帧的相对一致的感知编码质量。

公开(公告)号：US08594320B2

公开(公告)日：2013-11-26

申请号：US11912082

申请日：2006-04-19

Applicant: Christof Faller

Inventor： Christof Faller

IPC: H04M9/08 ,  H04M1/00 ,  H04M9/00

CPC classification number: H03G9/005 ,  H04M9/082

Abstract: Acoustic echo control and noise suppression in telecommunication systems. The proposed method of processing multi-channels audio loudspeakers signals and at least one microphone signal, comprises the steps of: transforming the input microphone signals (y1 (n), y2 (n), . . . , yM (n)) into input microphone short-time spectra, computing a combined loudspeaker signal short-time spectrum [X(i,k)] from the loudspeaker signals, (x1 (n), x2 (n), . . . , xL (n)), computing a combined microphone signal short-time spectrum [Y(i,k)] from the input microphone signal, (y1 (n), y2 (n), . . . , yM (n)), estimating a magnitude or power spectrum of the echo in the combined microphone signal short-time spectrum, computing a gain filter (G(i,k)) for magnitude modification of the input microphone short-time spectra, applying the gain filter to at least one of the input microphone spectra, converting the filtered input microphone spectra into the time domain (e1 (n), e2 (n), . . . , eM (n)).

Abstract translation: 电信系统中的声学回声控制和噪声抑制。 所提出的处理多声道音频扬声器信号和至少一个麦克风信号的方法包括以下步骤：将输入麦克风信号（y1（n），y2（n），...，yM（n））变换为输入 麦克风短时频谱，从扬声器信号（x1（n），x2（n），...，xL（n））计算组合的扬声器信号短时频谱[X（i，k）]，计算 来自输入麦克风信号的组合麦克风信号短时频谱[Y（i，k）]，（y1（n），y2（n），...，yM（n）），估计幅度或功率谱 在组合的麦克风信号短时间频谱中的回波，计算用于输入麦克风短时频谱的幅度修改的增益滤波器（G（i，k）），将增益滤波器应用于至少一个输入麦克风频谱， 将滤波后的输入麦克风频谱转换为时域（e1（n），e2（n），...，eM（n））。

公开(公告)号：US08462958B2

公开(公告)日：2013-06-11

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: H04B3/20

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：US08340306B2

公开(公告)日：2012-12-25

申请号：US11667747

申请日：2005-11-22

Applicant: Christof Faller

Inventor： Christof Faller

IPC: H04R5/00

CPC classification number: H04S1/002 ,  G10L19/008

Abstract: A binaural cue coding scheme involving one or more object-based cue codes, wherein an object-based cue code directly represents a characteristic of an auditory scene corresponding to the audio channels, where the characteristic is independent of number and positions of loudspeakers used to create the auditory scene. Examples of object-based cue codes include the angle of an auditory event, the width of the auditory event, the degree of envelopment of the auditory scene, and the directionality of the auditory scene.

Abstract translation: 一种涉及一个或多个基于对象的提示码的双耳提示编码方案，其中基于对象的提示码直接表示对应于音频信道的听觉场景的特征，其中该特征独立于用于创建的扬声器的数量和位置 听觉场面。 基于对象的提示代码的示例包括听觉事件的角度，听觉事件的宽度，听觉场景的包络程度以及听觉场景的方向性。

公开(公告)号：US08238562B2

公开(公告)日：2012-08-07

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: H04R5/00

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：US08204261B2

公开(公告)日：2012-06-19

申请号：US11006492

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: H04R5/02

CPC classification number: G10L19/008 ,  H04S3/02

Abstract: An input audio signal having an input temporal envelope is converted into an output audio signal having an output temporal envelope. The input temporal envelope of the input audio signal is characterized. The input audio signal is processed to generate a processed audio signal, wherein the processing de-correlates the input audio signal. The processed audio signal is adjusted based on the characterized input temporal envelope to generate the output audio signal, wherein the output temporal envelope substantially matches the input temporal envelope.

Abstract translation: 具有输入时间包络的输入音频信号被转换成具有输出时间包络的输出音频信号。 表征输入音频信号的输入时间包络。 输入音频信号被处理以产生经处理的音频信号，其中该处理使输入音频信号去相关。 经处理的音频信号基于表征的输入时间包络被调整以产生输出音频信号，其中输出时间包络基本上与输入的时间包络相匹配。

公开(公告)号：US08180061B2

公开(公告)日：2012-05-15

申请号：US11458646

申请日：2006-07-19

Applicant: Johannes Hilpert ,  Christof Faller ,  Karsten Linzmeier ,  Ralph Sperschneider

Inventor： Johannes Hilpert ,  Christof Faller ,  Karsten Linzmeier ,  Ralph Sperschneider

IPC: H04R5/00

CPC classification number: G06F12/0815 ,  G10L19/008 ,  H04S3/02

Abstract: The purpose of the invention is to bridge the gap between parametric multi-channel audio coding and matrixed-surround multi-channel coding by gradually improving the sound of an up-mix signal while raising the bit-rate consumed by the side-information starting from 0 up to the bit-rates of the parametric methods. More specifically, it provides a method of flexibly choosing an “operating point” somewhere between matrixed-surround (no side-information, limited audio quality) and fully parametric reconstruction (full side-information rate required, good quality). This operating point can be chosen dynamically (i.e. varying over time) and in response to the permissible side-information rate, as it is dictated by the individual application.

Abstract translation: 本发明的目的是通过逐渐改善上混合信号的声音来弥合参数多声道音频编码和矩阵环绕多声道编码之间的差距，同时提高从侧信息开始消耗的比特率 0到参数方法的比特率。 更具体地说，它提供了在矩阵环绕（无侧信息，有限音频质量）和完全参数重建（所需的全侧信息速率，良好质量）之间的某处的灵活选择“操作点”的方法。 该操作点可以动态地选择（即随着时间而变化）并且响应于由个体应用所规定的允许的侧信息速率。