公开(公告)号：US20110164756A1

公开(公告)日：2011-07-07

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: H04R5/00

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 “.1” indicates a single low-frequency effects (LFE) channel and “.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“.1”表示一个 单低频效应（LFE）通道，“.2”表示两个LFE通道。

公开(公告)号：US20110044461A1

公开(公告)日：2011-02-24

申请号：US12864240

申请日：2009-01-12

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

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

IPC: H04B3/20 ,  H03G3/00

CPC classification number: H04M9/082

Abstract: An embodiment of an apparatus for computing control information for a suppression filter for filtering a second audio signal to suppress an echo based on a first audio signal includes a computer having a value determiner for determining at least one energy-related value for a band-pass signal of at least two temporally successive data blocks of at least one signal of a group of signals. The computer further includes a mean value determiner for determining at least one mean value of the at least one determined energy-related value for the band-pass signal. The computer further includes a modifier for modifying the at least one energy-related value for the band-pass signal on the basis of the determined mean value for the band-pass signal. The computer further includes a control information computer for computing the control information for the suppression filter on the basis of the at least one modified energy-related value.

Abstract translation: 用于计算用于滤波第二音频信号以抑制基于第一音频信号的回波的抑制滤波器的控制信息的装置的实施例包括具有用于确定带通的至少一个能量相关值的值确定器的计算机 一组信号的至少一个信号的至少两个时间上连续的数据块的信号。 该计算机还包括一个平均值确定器，用于确定该带通信号的至少一个确定的能量相关值的至少一个平均值。 该计算机还包括修改器，用于基于确定的带通信号的平均值修改带通信号的至少一个能量相关值。 计算机还包括控制信息计算机，用于基于至少一个修改的能量相关值来计算抑制滤波器的控制信息。

公开(公告)号：US20100250259A1

公开(公告)日：2010-09-30

申请号：US12676730

申请日：2008-09-08

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

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

IPC: G10L19/00

CPC classification number: G10L21/0272 ,  G10L19/008 ,  G10L21/00

Abstract: The present invention includes an audio signal receiving unit receiving the audio signal having a plurality of channel signals including an ambient component signal and a source component signal; an ambient component signal extracting unit extracting the ambient component signal of each of the channels based on correlation between the channel signals; an ambient component signal modifying unit modifying the ambient component signal using surround effect information; a source component signal extracting unit extracting the source component signal of each of the channels based on the correlation between the channel signals; a first signal output unit outputting the modified ambient component signal and the source component signal; and a second signal output unit outputting the audio signal or the source component signal. Accordingly, in an apparatus for decoding an audio signal and method thereof according to the present invention, an ambient component signal is extracted and modified based on correlation and the modified ambient and source component signals are outputted using different signal output units, respectively. Therefore, the present invention enhances a stereo effect of the audio signal. And, a signal output unit for outputting an ambient component signal is arranged to have an output direction different from that of another signal output unit for outputting a source component signal, whereby a listener can be provided with an audio signal of which ambient sound is enhanced.

Abstract translation: 本发明包括音频信号接收单元，其接收具有包括环境分量信号和源分量信号的多个信道信号的音频信号; 环境分量信号提取单元，基于信道信号之间的相关性提取每个信道的环境分量信号; 环境分量信号修改单元，使用环绕效果信息修改环境分量信号; 源分量信号提取单元，基于信道信号之间的相关性提取每个信道的源分量信号; 输出修正的环境分量信号和源分量信号的第一信号输出单元; 以及输出音频信号或源分量信号的第二信号输出单元。 因此，在根据本发明的用于解码音频信号的装置及其方法中，基于相关性提取和修改环境分量信号，并且分别使用不同的信号输出单元输出修改的环境和源分量信号。 因此，本发明增强了音频信号的立体效果。 并且，用于输出环境分量信号的信号输出单元布置成具有与用于输出源分量信号的另一个信号输出单元的输出方向不同的输出方向，从而可以向收听者提供其环境声音被增强的音频信号 。

公开(公告)号：US07805313B2

公开(公告)日：2010-09-28

申请号：US10827900

申请日：2004-04-20

Applicant: Christof Faller ,  Juergen Herre

Inventor： Christof Faller ,  Juergen Herre

IPC: G10L19/00 ,  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：US20100241438A1

公开(公告)日：2010-09-23

申请号：US12676848

申请日：2008-09-08

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

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

IPC: G10L19/00

CPC classification number: G10L21/0272 ,  G10L19/008 ,  G10L21/00

Abstract: A method of decoding an audio signal is disclosed, The present invention includes the steps of receiving the audio signal having a plurality of channel signals including an ambient component signal and a source component signal, extracting the ambient component signal and the source component signal of each of the channels based on correlation between the channel signals, modifying the ambient component signal using surround effect information, and generating the audio signal including a plurality of channels using the modified ambient component signal and the source component signal. Accordingly, in an apparatus for decoding an audio signal and method thereof according to the present invention, an ambient component signal is extracted and modified based on correlation and the modified ambient and source component signals are outputted using different signal output units, respectively. Therefore, the present invention enhances a stereo effect of the audio signal. And, a signal output unit for outputting a ambient component signal is arranged to have an output direction different from that of another signal output unit for outputting a source component signal, whereby a listener can be provided with an audio signal of which ambient sound is enhanced.

Abstract translation: 公开了一种解码音频信号的方法，本发明包括以下步骤：接收具有多个信道信号的音频信号，包括环境分量信号和源分量信号，提取每个信号的环境分量信号和源分量信号 基于信道信号之间的相关性，使用环绕效果信息修改环境分量信号，以及使用修改的环境分量信号和源分量信号来生成包括多个信道的音频信号。 因此，在根据本发明的用于解码音频信号的装置及其方法中，基于相关性提取和修改环境分量信号，并且分别使用不同的信号输出单元输出修改的环境和源分量信号。 因此，本发明增强了音频信号的立体效果。 并且，用于输出环境分量信号的信号输出单元被布置成具有与用于输出源分量信号的另一信号输出单元的输出方向不同的输出方向，从而可以向收听者提供其环境声音被增强的音频信号 。

公开(公告)号：US07583805B2

公开(公告)日：2009-09-01

申请号：US10815591

申请日：2004-04-01

Applicant: Frank Baumgarte ,  Christof Faller

Inventor： Frank Baumgarte ,  Christof Faller

IPC: H03G3/00 ,  H04R5/00 ,  G10L19/00

CPC classification number: H04S3/002 ,  G10L19/008 ,  H04S3/004 ,  H04S7/305 ,  H04S2420/03

Abstract: A scheme for stereo and multi-channel synthesis of inter-channel correlation (ICC) (normalized cross-correlation) cues for parametric stereo and multi-channel coding. The scheme synthesizes ICC cues such that they approximate those of the original. For that purpose, diffuse audio channels are generated and mixed with the transmitted combined (e.g., sum) signal(s). The diffuse audio channels are preferably generated using relatively long filters with exponentially decaying Gaussian impulse responses. Such impulse responses generate diffuse sound similar to late reverberation. An alternative implementation for reduced computational complexity is proposed, where inter-channel level difference (ICLD), inter-channel time difference (ICTD), and ICC synthesis are all carried out in the domain of a single short-time Fourier transform (STFT), including the filtering for diffuse sound generation.

Abstract translation: 用于参数立体声和多声道编码的信道间相关（ICC）（归一化互相关）提示的立体声和多声道合成的方案。 该方案合成了ICC提示，使它们与原始图像近似。 为此，生成扩散音频通道并与发送的组合（例如，和）信号混合。 优选地，使用具有指数衰减高斯脉冲响应的相对长的滤波器生成扩散音频信道。 这种脉冲响应产生类似于后期混响的漫射声。 提出了一种用于降低计算复杂度的替代实现方式，其中，在单个短时傅立叶变换（STFT）的领域中，都进行信道间电平差（ICLD），信道间时差（ICTD）和ICC合成， ，包括扩散声产生的滤波。

公开(公告)号：US07412004B2

公开(公告)日：2008-08-12

申请号：US09895927

申请日：2001-06-29

Applicant: Christof Faller

Inventor： Christof Faller

IPC: H04B14/04

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：US20070019813A1

公开(公告)日：2007-01-25

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。

公开(公告)号：US20060115100A1

公开(公告)日：2006-06-01

申请号：US11058747

申请日：2005-02-15

Applicant: Christof Faller

Inventor： Christof Faller

IPC: H04B1/00 ,  H04B1/20

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：US20060085200A1

公开(公告)日：2006-04-20

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: G10L21/00

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。