公开(公告)号：US20080049943A1

公开(公告)日：2008-02-28

申请号：US11744156

申请日：2007-05-03

Applicant: Christof Faller ,  Hyen Oh ,  Yang Jung

Inventor： Christof Faller ,  Hyen Oh ,  Yang Jung

IPC: H04S3/00

CPC classification number: H04S3/008 ,  G10L19/0018 ,  G10L19/008 ,  H04S3/00 ,  H04S2420/03

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：US20070055510A1

公开(公告)日：2007-03-08

申请号：US11323965

申请日：2005-12-29

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

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

IPC: G10L19/00

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：US07116787B2

公开(公告)日：2006-10-03

申请号：US09848877

申请日：2001-05-04

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. In another embodiment, the combined audio signal corresponds to the combination of the left and right audio signals of a binaural signal corresponding to an input auditory scene. In this embodiment, the different sets of spatial parameters are applied to reconstruct the input auditory scene. In either case, transmission bandwidth requirements are reduced by reducing to one the number of different audio signals that need to be transmitted to a receiver configured to synthesize/reconstruct the auditory scene.

公开(公告)号：US20060184358A1

公开(公告)日：2006-08-17

申请号：US11403530

申请日：2006-04-13

Applicant: Christof Faller

Inventor： Christof Faller

IPC: G10L19/00

CPC classification number: H04B1/665 ,  H04B1/66

Abstract: Methods 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. The disclosed bit allocation process determines, for each frame, a distortion d[k] at which the frame is to be encoded. 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.

公开(公告)号：US08917797B2

公开(公告)日：2014-12-23

申请号：US12171004

申请日：2008-07-10

Applicant: Christof Faller

Inventor： Christof Faller

IPC: H03K9/00 ,  H04L23/00 ,  H04H60/27 ,  H04H60/11 ,  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：US08442819B2

公开(公告)日：2013-05-14

申请号：US11403530

申请日：2006-04-13

Applicant: Christof Faller

Inventor： Christof Faller

IPC: G10L19/02

CPC classification number: H04B1/665 ,  H04B1/66

Abstract: Methods 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. The disclosed bit allocation process determines, for each frame, a distortion d[k] at which the frame is to be encoded. 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.

Abstract translation: 公开了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。 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]。 确定失真d [k]以最小化（i）缓冲器溢出的概率，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。

公开(公告)号：US08355509B2

公开(公告)日：2013-01-15

申请号：US11837123

申请日：2007-08-10

Applicant: Christof Faller

Inventor： Christof Faller

IPC: H04R5/00 ,  H04B1/00

CPC classification number: G10L19/02 ,  G10L19/008 ,  G10L19/0204 ,  H04S3/00 ,  H04S3/008 ,  H04S7/00 ,  H04S7/307 ,  H04S2420/03 ,  H04S2420/13

Abstract: The following coding scenario is addressed: A number of audio source signals need to be transmitted or stored for the purpose of mixing wave field synthesis, multi-channel surround, or stereo signals after decoding the source signals. The proposed technique offers significant coding gain when jointly coding the source signals, compared to separately coding them, even when no redundancy is present between the source signals. This is possible by considering statistical properties of the source signals, the properties of mixing techniques, and spatial hearing. The sum of the source signals is transmitted plus the statistical properties of the source signals which mostly determine the perceptually important spatial cues of the final mixed audio channels. Source signals are recovered at the receiver such that their statistical properties approximate the corresponding properties of the original source signals. Subjective evaluations indicate that high audio quality is achieved by the proposed scheme.

Abstract translation: 解决以下编码场景：需要传输或存储多个音频源信号，以在解码源信号之后混合波场合成，多声道环绕或立体声信号。 与单独编码它们相比，即使在源信号之间不存在冗余的情况下，所提出的技术在联合编码源信号时提供了显着的编码增益。 这可以通过考虑源信号的统计特性，混合技术的性质和空间听觉来实现。 源信号的和被传送加上源信号的统计特性，这主要决定最终混合音频通道的感知重要的空间线索。 源信号在接收机处被恢复，使得它们的统计特性近似于原始信号源的对应属性。 主观评价表明，提出的方案可以实现高音质。

公开(公告)号：US08295494B2

公开(公告)日：2012-10-23

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. An audio decoding apparatus obtains an audio signal having a set of objects and side information. The apparatus obtains a set of mix parameters from a user input and an attenuation factor from the set of mix parameters. The apparatus then generates a plural-channel audio signal using at least one of the side information, the attenuation factor or the set of mix parameters.

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。 音频解码装置获得具有一组对象和边信息的音频信号。 该装置从用户输入获得一组混合参数，并从混合参数集中获得衰减因子。 该装置然后使用边信息，衰减因子或混合参数集合中的至少一个来生成多声道音频信号。

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。