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1. 发明申请

US20160005414A1 SYSTEM AND METHOD FOR COMPRESSED DOMAIN ESTIMATION OF THE SIGNAL TO NOISE RATIO OF A CODED SPEECH SIGNAL 有权
标题翻译：信号的压缩域估计系统和方法对编码语音信号的噪声比
公开(公告)号：US20160005414A1
公开(公告)日：2016-01-07
申请号：US14322369
申请日：2014-07-02
申请人： Nuance Communications, Inc.
发明人： Jose Lainez , Daniel A. Barreda , Dushyant Sharma , Patrick Naylor , Sridhar Pilli
IPC分类号： G10L19/028 , G10L25/18 , G10L25/90 , G10L19/002
CPC分类号： G10L19/028 , G10L19/002 , G10L21/0216 , G10L25/03 , G10L25/12 , G10L25/18 , G10L25/21 , G10L25/60 , G10L25/69 , G10L25/90 , G10L2019/0002 , G10L2019/0006
摘要： The present disclosure is directed towards a process for estimating the signal to noise ratio of a speech signal. The process may include receiving, at a computing device, a speech signal having a bitstream and a signal-to-noise ratio (“SNR”) associated therewith. The process may further include estimating the SNR directly from the bitstream or using a partial decoder that is configured to extract one or more parameters, the parameters including at least one of a fixed codebook gain, an adaptive codebook gain, a pitch lag, and a line spectral frequency (“LSF”) coefficient.
摘要翻译：本公开涉及用于估计语音信号的信噪比的处理。该过程可以包括在计算设备处接收具有与其相关联的比特流和信噪比（“SNR”）的语音信号。该过程还可以包括直接从比特流估计SNR，或者使用被配置为提取一个或多个参数的部分解码器，该参数包括固定码本增益，自适应码本增益，音调滞后和线谱频率（“LSF”）系数。

2. 发明申请

US20150302865A1 SYSTEM AND METHOD FOR AUDIO CONFERENCING 有权
标题翻译：用于音频会议的系统和方法
公开(公告)号：US20150302865A1
公开(公告)日：2015-10-22
申请号：US14256561
申请日：2014-04-18
申请人： Nuance Communications, Inc.
发明人： Sridhar Pilli , Mahesh Godavarti
IPC分类号： G10L21/0232 , G10L21/007 , H04L29/06
CPC分类号： H04L65/604 , G10L21/02 , H04L65/403 , H04L65/80 , H04M3/562 , H04M3/568 , H04M3/569
摘要： The present disclosure is directed towards an audio conferencing method. Some embodiments may include receiving, at a first mixing device, an audio signal from a first user associated with an audio conference. Embodiments may further include processing the audio signal at the first mixing device to generate a processed audio signal and transmitting the processed audio signal to a second mixing device, wherein the first mixing device and the second mixing device are distributed over a network in a cascaded configuration. Embodiments may also include receiving, at the second mixing device, a third audio signal from a second user associated with the audio conference and processing the third audio signal at the second mixing device to generate a second processed audio signal.
摘要翻译：本公开针对音频会议方法。一些实施例可以包括在第一混合设备处接收来自与音频会议相关联的第一用户的音频信号。实施例还可以包括处理在第一混合装置处的音频信号以产生经处理的音频信号并将经处理的音频信号发送到第二混合装置，其中第一混合装置和第二混合装置以级联配置在网络上分布。实施例还可以包括在第二混合设备处接收来自与音频会议相关联的第二用户的第三音频信号，并且在第二混合设备处处理第三音频信号以产生第二处理音频信号。

3. 发明授权

US09361899B2 System and method for compressed domain estimation of the signal to noise ratio of a coded speech signal 有权
标题翻译：用于编码语音信号的信噪比的压缩域估计的系统和方法
公开(公告)号：US09361899B2
公开(公告)日：2016-06-07
申请号：US14322369
申请日：2014-07-02
申请人： Nuance Communications, Inc.
发明人： Jose Lainez , Daniel A. Barreda , Dushyant Sharma , Patrick Naylor , Sridhar Pilli
IPC分类号： G10L21/00 , G10L19/028 , G10L19/002 , G10L25/18 , G10L25/90 , G10L19/00
CPC分类号： G10L19/028 , G10L19/002 , G10L21/0216 , G10L25/03 , G10L25/12 , G10L25/18 , G10L25/21 , G10L25/60 , G10L25/69 , G10L25/90 , G10L2019/0002 , G10L2019/0006
摘要： The present disclosure is directed towards a process for estimating the signal to noise ratio of a speech signal. The process may include receiving, at a computing device, a speech signal having a bitstream and a signal-to-noise ratio (“SNR”) associated therewith. The process may further include estimating the SNR directly from the bitstream or using a partial decoder that is configured to extract one or more parameters, the parameters including at least one of a fixed codebook gain, an adaptive codebook gain, a pitch lag, and a line spectral frequency (“LSF”) coefficient.
摘要翻译：本公开涉及用于估计语音信号的信噪比的处理。该过程可以包括在计算设备处接收具有与其相关联的比特流和信噪比（“SNR”）的语音信号。该过程还可以包括直接从比特流估计SNR，或者使用被配置为提取一个或多个参数的部分解码器，该参数包括固定码本增益，自适应码本增益，音调滞后和线谱频率（“LSF”）系数。

4. 发明申请

US20160112153A1 SYSTEM AND METHOD FOR PACKET SCHEDULING 有权
标题翻译：用于分组调度的系统和方法
公开(公告)号：US20160112153A1
公开(公告)日：2016-04-21
申请号：US14518622
申请日：2014-10-20
申请人： Nuance Communications, Inc.
发明人： Mahesh Godavarti , Sridhar Pilli , Biswaranjan Panigrahi
IPC分类号： H04J99/00 , H04L12/931 , H04L12/841
CPC分类号： H04J15/00 , H04L47/562
摘要： The present disclosure is directed towards a method for scheduling data packets in a multi-channel packet processing environment. The method may include receiving one or more data packets associated with an incoming signal and inserting the one or more data packets into a queue. The method may further include monitoring a time delay associated with each of the one or more data packets, wherein the time delay indicates a difference between packet arrival and packet departure times. The method may also include sorting the time delay results based upon an increasing order of time delay and determining a total number of data packets associated with each of a plurality of channels. The method may also include scheduling a data packet for processing based upon, at least in part, at least one of the sorted time delay results and the total number of data packets associated with each channel.
摘要翻译：本公开涉及一种用于在多信道分组处理环境中调度数据分组的方法。该方法可以包括接收与输入信号相关联的一个或多个数据分组，并将一个或多个数据分组插入到队列中。所述方法还可以包括监视与所述一个或多个数据分组中的每一个相关联的时间延迟，其中所述时间延迟指示分组到达和分组离开时间之间的差异。该方法还可以包括基于时间延迟的递增顺序对时间延迟结果进行排序，并且确定与多个信道中的每个信道相关联的数据分组的总数。该方法还可以包括至少部分地基于排序的时间延迟结果和与每个信道相关联的数据分组的总数中的至少一个来调度用于处理的数据分组。

5. 发明授权

US10237412B2 System and method for audio conferencing 有权
公开(公告)号：US10237412B2
公开(公告)日：2019-03-19
申请号：US14256567
申请日：2014-04-18
申请人： Nuance Communications, Inc.
发明人： Sridhar Pilli , Mahesh Godavarti
IPC分类号： H04M3/56 , H04W4/06 , H04L29/06 , H04M3/42
摘要： The present disclosure is directed towards an audio conferencing method. Some embodiments may include receiving, at a first mixing device, a first audio stream from one or more participant conferencing devices. The method may further include generating a top-N voice stream at the first mixing device, wherein the top-N voice stream corresponds with at least one top-N talker and wherein the identification of the at least one top-N talker is based upon, at least in part, an activity ranking. The method may also include receiving the top-N voice stream at a centralized mixing device and generating at least one mixed audio stream at the centralized mixing device.

6. 发明授权

US09979769B2 System and method for audio conferencing 有权
公开(公告)号：US09979769B2
公开(公告)日：2018-05-22
申请号：US14256561
申请日：2014-04-18
申请人： Nuance Communications, Inc.
发明人： Sridhar Pilli , Mahesh Godavarti
IPC分类号： G10L21/00 , H04M3/42 , H04L29/06 , H04M3/56 , G10L21/02
CPC分类号： H04L65/604 , G10L21/02 , H04L65/403 , H04L65/80 , H04M3/562 , H04M3/568 , H04M3/569
摘要： The present disclosure is directed towards an audio conferencing method. Some embodiments may include receiving, at a first mixing device, an audio signal from a first user associated with an audio conference. Embodiments may further include processing the audio signal at the first mixing device to generate a processed audio signal and transmitting the processed audio signal to a second mixing device, wherein the first mixing device and the second mixing device are distributed over a network in a cascaded configuration. Embodiments may also include receiving, at the second mixing device, a third audio signal from a second user associated with the audio conference and processing the third audio signal at the second mixing device to generate a second processed audio signal.

7. 发明授权

US10530519B2 System and method for packet scheduling 有权
公开(公告)号：US10530519B2
公开(公告)日：2020-01-07
申请号：US15645132
申请日：2017-07-10
申请人： Nuance Communications, Inc.
发明人： Mahesh Godavarti , Sridhar Pilli , Biswaranjan Panigrahi
IPC分类号： H04J99/00 , H04L12/875
摘要： The present disclosure is directed towards a method for scheduling data packets in a multi-channel packet processing environment. The method may include receiving one or more data packets associated with an incoming signal and inserting the one or more data packets into a queue. The method may further include monitoring a time delay associated with each of the one or more data packets, wherein the time delay indicates a difference between packet arrival and packet departure times. The method may also include sorting the time delay results based upon an increasing order of time delay and determining a total number of data packets associated with each of a plurality of channels. The method may also include scheduling a data packet for processing based upon, at least in part, at least one of the sorted time delay results and the total number of data packets associated with each channel.

8. 发明申请

US20170310418A1 SYSTEM AND METHOD FOR PACKET SCHEDULING 审中-公开
公开(公告)号：US20170310418A1
公开(公告)日：2017-10-26
申请号：US15645132
申请日：2017-07-10
申请人： Nuance Communications, Inc.
发明人： Mahesh Godavarti , Sridhar Pilli , Biswaranjan Panigrahi
IPC分类号： H04J99/00 , H04L12/875
摘要： The present disclosure is directed towards a method for scheduling data packets in a multi-channel packet processing environment. The method may include receiving one or more data packets associated with an incoming signal and inserting the one or more data packets into a queue. The method may further include monitoring a time delay associated with each of the one or more data packets, wherein the time delay indicates a difference between packet arrival and packet departure times. The method may also include sorting the time delay results based upon an increasing order of time delay and determining a total number of data packets associated with each of a plurality of channels. The method may also include scheduling a data packet for processing based upon, at least in part, at least one of the sorted time delay results and the total number of data packets associated with each channel.

9. 发明授权

US09705633B2 System and method for packet scheduling 有权
公开(公告)号：US09705633B2
公开(公告)日：2017-07-11
申请号：US14518622
申请日：2014-10-20
申请人： Nuance Communications, Inc.
发明人： Mahesh Godavarti , Sridhar Pilli , Biswaranjan Panigrahi
IPC分类号： H04L1/00 , H04L12/26 , H04J99/00 , H04L12/875
CPC分类号： H04J15/00 , H04L47/562
摘要： The present disclosure is directed towards a method for scheduling data packets in a multi-channel packet processing environment. The method may include receiving one or more data packets associated with an incoming signal and inserting the one or more data packets into a queue. The method may further include monitoring a time delay associated with each of the one or more data packets, wherein the time delay indicates a difference between packet arrival and packet departure times. The method may also include sorting the time delay results based upon an increasing order of time delay and determining a total number of data packets associated with each of a plurality of channels. The method may also include scheduling a data packet for processing based upon, at least in part, at least one of the sorted time delay results and the total number of data packets associated with each channel.

10. 发明授权

US09373342B2 System and method for speech enhancement on compressed speech 有权
标题翻译：压缩语音语音增强的系统和方法
公开(公告)号：US09373342B2
公开(公告)日：2016-06-21
申请号：US14312074
申请日：2014-06-23
申请人： Nuance Communications, Inc.
发明人： Sridhar Pilli , Mahesh Godavarti , Qian-Yu Tang , Jose Lainez , Jagadeesh Balam
IPC分类号： G10L21/0364 , G10L19/26 , G10L25/12 , G10L19/12 , G10L25/78 , G10L25/93 , G10L19/16 , G10L25/21
CPC分类号： G10L21/0364 , G10L19/12 , G10L19/173 , G10L19/26 , G10L25/12 , G10L25/21 , G10L25/78 , G10L25/93
摘要： The present disclosure is directed towards a method for speech intelligibility. The method may include receiving, at one or more computing devices, a first speech input from a first user and performing voice activity detection upon the first speech input. The method may also include analyzing a spectral tilt associated with the first speech input, wherein analyzing includes computing an impulse response of a linear predictive coding (“LPC”) synthesis filter in a linear pulse code modulation (“PCM”) domain and wherein the one or more computing devices includes an adaptive high pass filter configured to recalculate one or more linear prediction coefficients.
摘要翻译：本公开涉及一种用于语音清晰度的方法。该方法可以包括在一个或多个计算设备处接收来自第一用户的第一语音输入并且在第一语音输入时执行语音活动检测。该方法还可以包括分析与第一语音输入相关联的频谱倾斜，其中分析包括在线性脉码调制（“PCM”）域中计算线性预测编码（“LPC”）合成滤波器的脉冲响应，并且其中一个或多个计算设备包括被配置为重新计算一个或多个线性预测系数的自适应高通滤波器。

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