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公开(公告)号：US20170366913A1
公开(公告)日：2017-12-21
申请号：US15625913
申请日：2017-06-16
申请人： Edward Stein , Martin Walsh , Guangji Shi , David Corsello
发明人： Edward Stein , Martin Walsh , Guangji Shi , David Corsello
CPC分类号： H04S7/304 , G10L19/008 , G10L19/167 , H04S3/008 , H04S7/303 , H04S7/305 , H04S2400/01 , H04S2400/03 , H04S2420/01 , H04S2420/03 , H04S2420/11
摘要： The method and apparatus described herein make use of multiple sets of head related transfer functions (HRTFs) that have been synthesized or measured at various distances from a reference head, spanning from the near-field to the boundary of the far-field. Additional synthetic or measured transfer functions maybe used to extend to the interior of the head, i.e., for distances closer than near-field. In addition, the relative distance-related gains of each set of HRTFs are normalized to the far-field HRTF gains.
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公开(公告)号：US08879750B2
公开(公告)日：2014-11-04
申请号：US12901330
申请日：2010-10-08
申请人： Martin Walsh , Edward Stein , Jean-Marc Jot
发明人： Martin Walsh , Edward Stein , Jean-Marc Jot
摘要： There are provided methods and an apparatus for conditioning an audio signal. According to one aspect of the present invention there is included a method for conditioning an audio signal having the steps of: receiving at least one audio signal, each audio signal having at least one channel, each channel being segmented into a plurality of frames over a series of time; calculating at least one measure of dynamic excursion of the audio signal for a plurality of successive segments of time; filtering the audio signal into a plurality of subbands, each frame being represented by at least one subband; deriving a dynamic gain factor from the successive segments of time; analyzing at least one subband of the frame to determine if a transient exists in the frame; and applying the dynamic gain factor to each frame having a transient.
摘要翻译： 提供了用于调节音频信号的方法和装置。 根据本发明的一个方面，包括一种用于调节音频信号的方法，该方法具有以下步骤：接收至少一个音频信号，每个音频信号具有至少一个信道，每个信道被分割成多个帧 一系列时间; 计算多个连续的时间段的音频信号的动态偏移的至少一个度量; 将音频信号过滤成多个子带，每个帧由至少一个子带表示; 从连续的时间段导出动态增益因子; 分析帧的至少一个子带以确定帧中是否存在瞬态; 以及将动态增益因子应用于具有瞬态的每个帧。
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公开(公告)号：US20090092259A1
公开(公告)日：2009-04-09
申请号：US12246491
申请日：2008-10-06
IPC分类号： H04R5/00
CPC分类号： H04S3/02 , G10L19/008
摘要： A two-channel phase-amplitude stereo encoding and decoding scheme enabling flexible and spatially accurate interactive 3-D audio reproduction via standard audio-only two-channel transmission. The encoding scheme allows associating a 2-D or 3-D positional localization to each of a plurality of sound sources by use of frequency independent inter-channel phase and amplitude differences. The decoder is based on frequency-domain spatial analysis of 2-D or 3-D directional cues in a two-channel stereo signal and re-synthesis of these cues using any preferred spatialization technique, thereby allowing faithful reproduction of positional audio cues and reverberation or ambient cues over arbitrary multi-channel loudspeaker reproduction formats or over headphones, while preserving source separation despite the intermediate encoding over only two audio channels.
摘要翻译： 双通道相位立体声编码和解码方案，通过标准的仅音频双通道传输实现灵活和空间准确的交互式3-D音频再现。 编码方案允许通过使用频率无关的信道间相位和幅度差来将多维声源中的每一个与二维或三维位置定位相关联。 解码器基于双声道立体声信号中的2-D或3-D方向提示的频域空间分析，并且使用任何优选的空间化技术重新合成这些线索，从而允许忠实地再现位置音频线索和混响 或环境提示超过任意多声道扬声器再现格式或通过耳机，同时保留源分离，尽管中间编码只有两个音频通道。
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公开(公告)号：US09697844B2
公开(公告)日：2017-07-04
申请号：US12350047
申请日：2009-01-07
申请人： Martin Walsh , Jean-Marc Jot , Edward Stein
发明人： Martin Walsh , Jean-Marc Jot , Edward Stein
IPC分类号： H04R5/00 , G10L19/16 , H04S5/00 , G10L19/008
CPC分类号： G10L19/16 , G10L19/008 , H04R2205/024 , H04R2420/07 , H04S5/005
摘要： This invention describes a method for decentralized decoding of a multichannel audio signal by broadcasting the original encoded data and distributing the decoding process between a plurality of receiving units. This allows for the design and manufacture of scalable multichannel audio reproduction systems having an arbitrary number of output channels, composed of a plurality of generic decoder and loudspeaker units each generating fewer output channels. With distributed decoding, a manufacturer can use “off-the-shelf” stereo or mono signal processors, digital-to-analog converters and amplifier components in each generic decoding module, thus reducing manufacturing costs and complexity requirements for each module while offering unlimited scalability in the total number of output channels.
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公开(公告)号：US20110251704A1
公开(公告)日：2011-10-13
申请号：US13084298
申请日：2011-04-11
申请人： Martin Walsh , Edward Stein , Jean-Marc Jot , James D. Johnston
发明人： Martin Walsh , Edward Stein , Jean-Marc Jot , James D. Johnston
IPC分类号： G06F17/00
CPC分类号： H04S7/307 , G10L21/0232 , G10L21/0364 , H04R3/04 , H04S7/301
摘要： The present invention counterbalances background noise by applying dynamic equalization. A psychoacoustic model representing the perception of masking effects of background noise relative to a desired foreground soundtrack is used to accurately counterbalance background noise. A microphone samples what the listener is hearing and separates the desired soundtrack from the interfering noise. The signal and noise components are analyzed from a psychoacoustic perspective and the soundtrack is equalized such that the frequencies that were originally masked are unmasked. Subsequently, the listener may hear the soundtrack over the noise. Using this process the EQ can continuously adapt to the background noise level without any interaction from the listener and only when required. When the background noise subsides, the EQ adapts back to its original level and the user does not experience unnecessarily high loudness levels.
摘要翻译： 本发明通过应用动态均衡来抵消背景噪声。 使用表示背景噪声相对于期望的前景音轨的掩蔽效应的感知的心理声学模型来精确地抵消背景噪声。 麦克风采样听众听到的内容，并将所需的音轨与干扰噪声分开。 信号和噪声分量从心理声学的角度进行分析，并且音轨被均衡，使得最初被屏蔽的频率未被屏蔽。 随后，听众可能会听到声音的声音。 使用这个过程，EQ可以连续适应背景噪声水平，而不需要听众的任何交互，只有在需要的时候。 当背景噪声消失时，EQ适应于其原始水平，并且用户不会遇到不必要的高响度水平。
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公开(公告)号：US20090110204A1
公开(公告)日：2009-04-30
申请号：US12350047
申请日：2009-01-07
申请人： Martin WALSH , Jean-Marc Jot , Edward Stein
发明人： Martin WALSH , Jean-Marc Jot , Edward Stein
IPC分类号： H04R5/00
CPC分类号： G10L19/16 , G10L19/008 , H04R2205/024 , H04R2420/07 , H04S5/005
摘要： This invention describes a method for decentralized decoding of a multichannel audio signal by broadcasting the original encoded data and distributing the decoding process between a plurality of receiving units. This allows for the design and manufacture of scalable multichannel audio reproduction systems having an arbitrary number of output channels, composed of a plurality of generic decoder and loudspeaker units each generating fewer output channels. With distributed decoding, a manufacturer can use “off-the-shelf” stereo or mono signal processors, digital-to-analog converters and amplifier components in each generic decoding module, thus reducing manufacturing costs and complexity requirements for each module while offering unlimited scalability in the total number of output channels.
摘要翻译： 本发明描述了一种通过广播原始编码数据并在多个接收单元之间分配解码过程来对多声道音频信号进行分散解码的方法。 这允许设计和制造具有任意数量的输出通道的可扩展多声道音频再现系统，该多路音频再现系统由多个通用解码器和每个产生较少输出通道的扬声器单元组成。 通过分布式解码，制造商可以在每个通用解码模块中使用“现成”的立体声或单声道信号处理器，数模转换器和放大器组件，从而降低每个模块的制造成本和复杂性要求，同时提供无限的可扩展性 在输出通道的总数。
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7.发明授权Dynamic compensation of audio signals for improved perceived spectral imbalances 有权音频信号的动态补偿，用于改善感知频谱不平衡公开(公告)号：US09391579B2
公开(公告)日：2016-07-12
申请号：US13228272
申请日：2011-09-08
申请人： Martin Walsh , Edward Stein , Jean-Marc Jot
发明人： Martin Walsh , Edward Stein , Jean-Marc Jot
摘要： An input audio signal is equalized to form an output audio signal on the basis of an intended listening sound pressure level, the output capabilities of a particular playback device, and unique hearing characteristics of a listener. An intended listening level is first determined based on the properties of the audio signal and a mastering sound level. The intended listening level is used to determine an optimal sound pressure level for the particular playback device based on its capabilities and any master volume gain. These two levels are used to determine how much louder to make individual frequencies based on data pertaining to human auditory perception, either standardized or directly measured. The audio is further compensated on the basis of hearing loss data, again either standardized or directly measured, after optionally extending the signal bandwidth. The final, compensated audio signal is sent to the playback device for playback.
摘要翻译： 输入音频信号被均衡以根据预期的听力声压级，特定播放设备的输出能力和收听者的独特听觉特征来形成输出音频信号。 首先根据音频信号的属性和母带声级别来确定预期的听音级别。 根据其能力和任何主音量增益，目标聆听电平用于确定特定播放设备的最佳声压级。 这两个级别用于确定基于与人类听觉知觉相关的数据（标准化或直接测量）来制作单个频率的响度。 在可选地扩展信号带宽之后，基于听力损失数据进一步补偿音频，再次被标准化或直接测量。 最后的补偿音频信号被发送到播放设备进行播放。
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公开(公告)号：US08712061B2
公开(公告)日：2014-04-29
申请号：US12246491
申请日：2008-10-06
IPC分类号： H04R5/00
CPC分类号： H04S3/02 , G10L19/008
摘要： A two-channel phase-amplitude stereo encoding and decoding scheme enabling flexible and spatially accurate interactive 3-D audio reproduction via standard audio-only two-channel transmission. The encoding scheme allows associating a 2-D or 3-D positional localization to each of a plurality of sound sources by use of frequency independent inter-channel phase and amplitude differences. The decoder is based on frequency-domain spatial analysis of 2-D or 3-D directional cues in a two-channel stereo signal and re-synthesis of these cues using any preferred spatialization technique, thereby allowing faithful reproduction of positional audio cues and reverberation or ambient cues over arbitrary multi-channel loudspeaker reproduction formats or over headphones, while preserving source separation despite the intermediate encoding over only two audio channels.
摘要翻译： 双通道相位立体声编码和解码方案，通过标准的仅音频双通道传输实现灵活和空间准确的交互式3-D音频再现。 编码方案允许通过使用频率无关的信道间相位和幅度差来将多维声源中的每一个与二维或三维位置定位相关联。 解码器基于双声道立体声信号中的2-D或3-D方向提示的频域空间分析，并且使用任何优选的空间化技术重新合成这些线索，从而允许忠实地再现位置音频线索和混响 或环境提示超过任意多声道扬声器再现格式或通过耳机，同时保留源分离，尽管中间编码只有两个音频通道。
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公开(公告)号：US08619998B2
公开(公告)日：2013-12-31
申请号：US11835403
申请日：2007-08-07
申请人： Martin Walsh , Jean Marc Jot , Edward Stein
发明人： Martin Walsh , Jean Marc Jot , Edward Stein
IPC分类号： H04S1/00
CPC分类号： H04S1/002 , G10L19/008 , H04S3/02 , H04S5/00 , H04S5/005 , H04S7/00 , H04S2400/01 , H04S2420/01
摘要： The present invention describes techniques that can be used to provide novel methods of spatial audio rendering using adapted M-S matrix shuffler topologies. Such techniques include headphone and loudspeaker-based binaural signal simulation and rendering, stereo expansion, multichannel upmix and pseudo multichannel surround rendering.
摘要翻译： 本发明描述了可以用于使用适应的M-S矩阵洗牌器拓扑来提供空间音频呈现的新颖方法的技术。 这些技术包括耳机和基于扬声器的双耳信号模拟和渲染，立体声扩展，多声道上混和伪多声道环绕渲染。
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10.发明申请DYNAMIC COMPENSATION OF AUDIO SIGNALS FOR IMPROVED PERCEIVED SPECTRAL IMBALANCES 有权音频信号的动态补偿，用于改进的频谱不平衡公开(公告)号：US20120063616A1
公开(公告)日：2012-03-15
申请号：US13228272
申请日：2011-09-08
申请人： Martin Walsh , Edward Stein , Jean-Marc Jot
发明人： Martin Walsh , Edward Stein , Jean-Marc Jot
IPC分类号： H03G5/00
摘要： An input audio signal is equalized to form an output audio signal on the basis of an intended listening sound pressure level, the output capabilities of a particular playback device, and unique hearing characteristics of a listener. An intended listening level is first determined based on the properties of the audio signal and a mastering sound level. The intended listening level is used to determine an optimal sound pressure level for the particular playback device based on its capabilities and any master volume gain. These two levels are used to determine how much louder to make individual frequencies based on data pertaining to human auditory perception, either standardized or directly measured. The audio is further compensated on the basis of hearing loss data, again either standardized or directly measured, after optionally extending the signal bandwidth. The final, compensated audio signal is sent to the playback device for playback.
摘要翻译： 输入音频信号被均衡以根据预期的听力声压级，特定播放设备的输出能力和收听者的独特听觉特征来形成输出音频信号。 首先根据音频信号的属性和母带声级别来确定预期的听音级别。 根据其能力和任何主音量增益，目标聆听电平用于确定特定播放设备的最佳声压级。 这两个级别用于确定基于与人类听觉知觉相关的数据（标准化或直接测量）来制作单个频率的响度。 在可选地扩展信号带宽之后，基于听力损失数据进一步补偿音频，再次被标准化或直接测量。 最后的补偿音频信号被发送到播放设备进行播放。
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