VIRTUAL AUDIO ENVIRONMENT FOR MULTIDIMENSIONAL CONFERENCING
    1.
    发明申请
    VIRTUAL AUDIO ENVIRONMENT FOR MULTIDIMENSIONAL CONFERENCING 有权
    多媒体会议的虚拟音频环境

    公开(公告)号:US20120155680A1

    公开(公告)日:2012-06-21

    申请号:US12970964

    申请日:2010-12-17

    IPC分类号: H04R5/02

    摘要: The disclosed architecture employs signal processing techniques to provide audio perception only, or audio perception that matches the visual perception. This also provides spatial audio reproduction for multiparty teleconferencing such that the teleconferencing participants perceive themselves as if they were sitting in the same room. The solution is based on the premise that people perceive sounds as a reconstructed wavefront, and hence, the wavefronts are used to provide the spatial perceptual cues. The differences between the spatial perceptual cues derived from the reconstructed wavefront of sound waves and the ideal wavefront of sound waves form an objective metric for spatial perceptual quality, and provide the means of evaluating the overall system performance. Additionally, compensation filters are employed to improve the spatial perceptual quality of stereophonic systems by optimizing the objective metrics.

    摘要翻译: 所公开的架构采用信号处理技术来仅提供音频感知,或者与视觉感知匹配的音频感知。 这也为多方电话会议提供了空间音频再现,使得电话会议参与者将自己视为坐在同一个房间中。 解决方案是基于人们将声音视为重建波前的前提,因此波前用于提供空间感知线索。 从声波重构波前衍生的空间感知线索与声波理想波阵面之间的差异形成了空间感知质量的客观指标,并提供了评估整体系统性能的手段。 另外,通过优化客观指标,采用补偿滤波器来提高立体声系统的空间感知质量。

    Virtual audio environment for multidimensional conferencing
    2.
    发明授权
    Virtual audio environment for multidimensional conferencing 有权
    用于多维会议的虚拟音频环境

    公开(公告)号:US08693713B2

    公开(公告)日:2014-04-08

    申请号:US12970964

    申请日:2010-12-17

    IPC分类号: H04R5/02

    摘要: The disclosed architecture employs signal processing techniques to provide audio perception only, or audio perception that matches the visual perception. This also provides spatial audio reproduction for multiparty teleconferencing such that the teleconferencing participants perceive themselves as if they were sitting in the same room. The solution is based on the premise that people perceive sounds as a reconstructed wavefront, and hence, the wavefronts are used to provide the spatial perceptual cues. The differences between the spatial perceptual cues derived from the reconstructed wavefront of sound waves and the ideal wavefront of sound waves form an objective metric for spatial perceptual quality, and provide the means of evaluating the overall system performance. Additionally, compensation filters are employed to improve the spatial perceptual quality of stereophonic systems by optimizing the objective metrics.

    摘要翻译: 所公开的架构采用信号处理技术来仅提供音频感知,或者与视觉感知匹配的音频感知。 这也为多方电话会议提供了空间音频再现,使得电话会议参与者将自己视为坐在同一个房间中。 解决方案是基于人们将声音视为重建波前的前提,因此波前用于提供空间感知线索。 从声波重构波前衍生的空间感知线索与声波理想波阵面之间的差异形成了空间感知质量的客观指标,并提供了评估整体系统性能的手段。 另外,通过优化客观指标,采用补偿滤波器来提高立体声系统的空间感知质量。

    HARMONICITY-BASED SINGLE-CHANNEL SPEECH QUALITY ESTIMATION
    3.
    发明申请
    HARMONICITY-BASED SINGLE-CHANNEL SPEECH QUALITY ESTIMATION 有权
    基于谐波的单通道语音质量估计

    公开(公告)号:US20130151244A1

    公开(公告)日:2013-06-13

    申请号:US13316430

    申请日:2011-12-09

    IPC分类号: G10L19/14

    CPC分类号: G10L25/69

    摘要: Speech quality estimation technique embodiments are described which generally involve estimating the human speech quality of an audio frame in a single-channel audio signal. A representation of a harmonic component of the frame is synthesized and used to compute a non-harmonic component of the frame. The synthesized harmonic component representation and the non-harmonic component are then used to compute a harmonic to non-harmonic ratio (HnHR). This HnHR is indicative of the quality of a user's speech and is designated as an estimate of the speech quality of the frame. In one implementation, the HnHR is used to establish a minimum speech quality threshold below which the quality of the user's speech is considered unacceptable. Feedback to the user is then provided based on whether the HnHR falls below the threshold.

    摘要翻译: 描述了通常涉及在单声道音频信号中估计音频帧的人类语音质量的语音质量估计技术实施例。 合成帧的谐波分量的表示,并用于计算帧的非谐波分量。 然后使用合成谐波分量表示和非谐波分量来计算谐波到非谐波比(HnHR)。 该HnHR表示用户语音的质量,并且被指定为帧的语音质量的估计。 在一个实现中,HnHR用于建立最小语音质量阈值,低于该最低语音质量阈值,用户语音的质量被认为是不可接受的。 然后基于HnHR是否低于阈值来提供对用户的反馈。

    SPATIALIZED AUDIO OVER HEADPHONES
    4.
    发明申请
    SPATIALIZED AUDIO OVER HEADPHONES 有权
    耳机上的空间音频

    公开(公告)号:US20100303266A1

    公开(公告)日:2010-12-02

    申请号:US12472080

    申请日:2009-05-26

    IPC分类号: H04R5/02

    CPC分类号: H04R27/00

    摘要: A spatial element is added to communications, including over telephone conference calls heard through headphones or a stereo speaker setup. Functions are created to modify signals from different callers to create the illusion that the callers are speaking from different parts of the room.

    摘要翻译: 一个空间元素添加到通信中,包括通过耳机听到的电话会议通话或立体声扬声器设置。 创建功能来修改来自不同呼叫者的信号,以创建呼叫者从房间的不同部分讲话的错觉。

    Harmonicity-based single-channel speech quality estimation
    5.
    发明授权
    Harmonicity-based single-channel speech quality estimation 有权
    基于谐波的单通道语音质量估计

    公开(公告)号:US08731911B2

    公开(公告)日:2014-05-20

    申请号:US13316430

    申请日:2011-12-09

    IPC分类号: G10L21/00

    CPC分类号: G10L25/69

    摘要: Speech quality estimation technique embodiments are described which generally involve estimating the human speech quality of an audio frame in a single-channel audio signal. A representation of a harmonic component of the frame is synthesized and used to compute a non-harmonic component of the frame. The synthesized harmonic component representation and the non-harmonic component are then used to compute a harmonic to non-harmonic ratio (HnHR). This HnHR is indicative of the quality of a user's speech and is designated as an estimate of the speech quality of the frame. In one implementation, the HnHR is used to establish a minimum speech quality threshold below which the quality of the user's speech is considered unacceptable. Feedback to the user is then provided based on whether the HnHR falls below the threshold.

    摘要翻译: 描述了通常涉及在单声道音频信号中估计音频帧的人类语音质量的语音质量估计技术实施例。 合成帧的谐波分量的表示,并用于计算帧的非谐波分量。 然后使用合成谐波分量表示和非谐波分量来计算谐波到非谐波比(HnHR)。 该HnHR表示用户语音的质量,并且被指定为帧的语音质量的估计。 在一个实现中,HnHR用于建立最小语音质量阈值,低于该最低语音质量阈值,用户语音的质量被认为是不可接受的。 然后基于HnHR是否低于阈值来提供对用户的反馈。

    STEREOPHONIC TELECONFERENCING USING A MICROPHONE ARRAY
    6.
    发明申请
    STEREOPHONIC TELECONFERENCING USING A MICROPHONE ARRAY 审中-公开
    使用麦克风阵列的立体声电话

    公开(公告)号:US20120262536A1

    公开(公告)日:2012-10-18

    申请号:US13086632

    申请日:2011-04-14

    IPC分类号: H04N7/14 H04R5/02 H04R5/00

    摘要: Stereophonic teleconferencing system embodiments are described which advantageously employ a microphone array at a remote conference site having multiple conferencees to produce a separate output channel from the each microphone in the array. Audio data streams each representing one of the audio output channels from the microphone array are then sent to a local conference site where a local conferencee is in attendance. The voices of the aforementioned remote conferencees are spatialized within a sound-field of the local site using multiple loudspeakers. Generally, this involves receiving the monophonic audio data streams from the remote site, and processing them to generate an audio signal for each loudspeaker. Each of the generated audio signals is then played through its respective loudspeaker to produce a spatial audio sound-field which is audibly perceived by the local conferencee as having the voice of each of the remote conferencees coming from a different location.

    摘要翻译: 描述了立体声电话会议系统实施例,其有利地在具有多个会议的远程会议站采用麦克风阵列,以从阵列中的每个麦克风产生单独的输出通道。 然后将每个表示来自麦克风阵列的音频输出声道之一的音频数据流发送到本地会议室出席的本地会议现场。 使用多个扬声器,上述远程会议的声音在本地站点的声场内被空间化。 通常,这涉及从远程站点接收单声道音频数据流,并且处理它们以产生每个扬声器的音频信号。 然后通过其相应的扬声器播放所生成的每个音频信号,以产生由本地会议室听得见的具有每个远程会议的声音来自不同位置的空间音频声场。

    Spatialized audio over headphones
    7.
    发明授权
    Spatialized audio over headphones 有权
    通过耳机进行空间化音频

    公开(公告)号:US08737648B2

    公开(公告)日:2014-05-27

    申请号:US12472080

    申请日:2009-05-26

    IPC分类号: H04R5/02

    CPC分类号: H04R27/00

    摘要: A spatial element is added to communications, including over telephone conference calls heard through headphones or a stereo speaker setup. Functions are created to modify signals from different callers to create the illusion that the callers are speaking from different parts of the room.

    摘要翻译: 一个空间元素添加到通信中,包括通过耳机听到的电话会议通话或立体声扬声器设置。 创建功能来修改来自不同呼叫者的信号,以创建呼叫者从房间的不同部分讲话的错觉。

    Method and apparatus for reducing timestamp noise in audio echo cancellation
    8.
    发明授权
    Method and apparatus for reducing timestamp noise in audio echo cancellation 有权
    用于减少音频回声消除中的时间戳噪声的方法和装置

    公开(公告)号:US08259928B2

    公开(公告)日:2012-09-04

    申请号:US11788939

    申请日:2007-04-23

    IPC分类号: H04M9/08

    CPC分类号: H04M9/082

    摘要: A communication end device of a two-way communication system is shown. The device includes an audio signal capture device for capturing local audio to be transmitted to another end device, an audio signal rendering device for playing remote audio received from the other end device, and buffers for buffering the captured and rendered audio signals. The device also includes an audio echo canceller operating to predict echo from the rendered audio signal at a calculated relative offset in the captured audio signal based on an adaptive filter, and subtract the predicted echo from the signal transmitted to the other end device The calculated relative offset that is used by the audio echo canceller for a current signal sample is adjusted if a difference between it and an adjusted relative offset of a preceding sample exceeds a threshold value.

    摘要翻译: 示出了双向通信系统的通信终端设备。 该装置包括用于捕获要发送到另一终端设备的本地音频的音频信号捕获设备,用于播放从另一端设备接收的远程音频的音频信号渲染设备,以及用于缓冲所捕获和渲染的音频信号的缓冲器。 该装置还包括音频回波消除器,其操作以基于自适应滤波器在所捕获的音频信号中以计算的相对偏移来预测来自渲染音频信号的回波,并且从发送到另一端装置的信号中减去预测回波。计算的相对 当音频回波消除器用于当前信号样本的偏移量如果其与先前样本的经调整的相对偏移之间的差异超过阈值则被调整。

    Entropy coding by adapting coding between level and run-length/level modes
    9.
    发明申请
    Entropy coding by adapting coding between level and run-length/level modes 有权
    熵编码通过适应水平和游程长度/级别模式之间的编码

    公开(公告)号:US20050015249A1

    公开(公告)日:2005-01-20

    申请号:US10647923

    申请日:2003-08-25

    摘要: An audio encoder performs adaptive entropy encoding of audio data. For example, an audio encoder switches between variable dimension vector Huffman coding of direct levels of quantized audio data and run-level coding of run lengths and levels of quantized audio data. The encoder can use, for example, context-based arithmetic coding for coding run lengths and levels. The encoder can determine when to switch between coding modes by counting consecutive coefficients having a predominant value (e.g., zero). An audio decoder performs corresponding adaptive entropy decoding.

    摘要翻译: 音频编码器执行音频数据的自适应熵编码。 例如,音频编码器在量化音频数据的直接电平的可变维矢量霍夫曼编码和游程长度的游程级编码以及量化的音频数据的电平之间切换。 编码器可以使用例如用于对运行长度和电平进行编码的基于上下文的算术编码。 编码器可以通过计算具有主要值(例如,零)的连续系数来确定何时在编码模式之间切换。 音频解码器执行相应的自适应熵解码。