公开(公告)号：US07076421B2

公开(公告)日：2006-07-11

申请号：US11077809

申请日：2005-03-09

申请人： Carlo Murgia ,  Jeffrey D. Klein ,  Huan-Yu Su

发明人： Carlo Murgia ,  Jeffrey D. Klein ,  Huan-Yu Su

IPC分类号： G10L21/02

CPC分类号： G10L19/008

摘要： An exemplary multi-channel speech processor comprises a controller capable of interfacing with a plurality of channels, and at least one signal processing unit (SPU) coupled to the controller, where the multi-channel speech processor has a maximum execution time for processing all frames, one channel at a time, by processing a single frame from each of the plurality of channels. The signal processing unit encodes each of the single frames from each of the plurality of channels, one channel at a time, to generate encoded frames until the maximum execution time elapses or is about to elapse. The controller also transmits a predetermined frame for each of the plurality of channels not processed during the encoding step, due to the maximum execution time elapsing or being about to elapse, such that the predetermined frame causes a decoder which receives the predetermined frame to generate a frame erase frame.

摘要翻译： 示例性多声道语音处理器包括能够与多个通道接口的控制器以及耦合到控制器的至少一个信号处理单元（SPU），其中多声道语音处理器具有用于处理所有帧的最大执行时间 通过处理来自多个信道中的每一个的单个帧，一次一个信道。 信号处理单元从多个信道中的每一个，每次一个信道中对每个单个帧进行编码，以生成经编码的帧，直到经过或将要经过最大执行时间。 由于最大的执行时间经过或将要经过，所以控制器还为编码步骤期间未被处理的多个通道中的每一个发送预定的帧，使得预定帧使得接收预定帧的解码器产生 帧擦除帧。

公开(公告)号：US06873956B2

公开(公告)日：2005-03-29

申请号：US10464307

申请日：2003-06-17

申请人： Carlo Murgia ,  Jeffrey D. Klein ,  Huan-Yu Su

发明人： Carlo Murgia ,  Jeffrey D. Klein ,  Huan-Yu Su

IPC分类号： G01L21/04 ,  G10L19/00 ,  G10L19/14 ,  G10L21/04 ,  H04L12/56

CPC分类号： G10L19/008

摘要： An exemplary multi-channel speech processor comprises a controller capable of interfacing with a plurality of channels, and at least one signal processing unit (SPU) coupled to the controller, where the multi-channel speech processor has a maximum execution time for processing all frames, one channel at a time, by processing a single frame from each of the plurality of channels. The signal processing unit encodes each of the single frames from each of the plurality of channels, one channel at a time, to generate encoded frames until the maximum execution time elapses or is about to elapse. The controller also transmits a pre-determined frame for each of the plurality of channels not processed during the encoding step, due to the maximum execution time elapsing or being about to elapse, such that the predetermined frame causes a decoder which receives the predetermined frame to generate a frame erase frame.

摘要翻译： 示例性多声道语音处理器包括能够与多个通道接口的控制器以及耦合到控制器的至少一个信号处理单元（SPU），其中多声道语音处理器具有用于处理所有帧的最大执行时间 通过处理来自多个信道中的每一个的单个帧，一次一个信道。 信号处理单元从多个信道中的每一个，每次一个信道中对每个单个帧进行编码，以生成经编码的帧，直到经过或将要经过最大执行时间。 由于最大执行时间经过或将要经过，控制器还为编码步骤期间未被处理的多个通道中的每一个发送预定帧，使得预定帧使接收预定帧的解码器 生成帧擦除帧。

公开(公告)号：US20050220133A1

公开(公告)日：2005-10-06

申请号：US11077809

申请日：2005-03-09

申请人： Carlo Murgia ,  Jeffrey Klein ,  Huan-Yu Su

发明人： Carlo Murgia ,  Jeffrey Klein ,  Huan-Yu Su

IPC分类号： G01L21/04 ,  G10L19/00 ,  G10L19/14 ,  G10L21/04 ,  H04L12/56

CPC分类号： G10L19/008

摘要： An exemplary multi-channel speech processor comprises a controller capable of interfacing with a plurality of channels, and at least one signal processing unit (SPU) coupled to the controller, where the multi-channel speech processor has a maximum execution time for processing all frames, one channel at a time, by processing a single frame from each of the plurality of channels. The signal processing unit encodes each of the single frames from each of the plurality of channels, one channel at a time, to generate encoded frames until the maximum execution time elapses or is about to elapse. The controller also transmits a predetermined frame for each of the plurality of channels not processed during the encoding step, due to the maximum execution time elapsing or being about to elapse, such that the predetermined frame causes a decoder which receives the predetermined frame to generate a frame erase frame.

摘要翻译： 示例性多声道语音处理器包括能够与多个通道接口的控制器以及耦合到控制器的至少一个信号处理单元（SPU），其中多声道语音处理器具有用于处理所有帧的最大执行时间 通过处理来自多个信道中的每一个的单个帧，一次一个信道。 信号处理单元从多个信道中的每一个，每次一个信道中对每个单个帧进行编码，以生成经编码的帧，直到经过或将要经过最大执行时间。 由于最大的执行时间经过或将要经过，所以控制器还为编码步骤期间未被处理的多个通道中的每一个发送预定的帧，使得预定帧使得接收预定帧的解码器产生 帧擦除帧。

公开(公告)号：US10366701B1

公开(公告)日：2019-07-30

申请号：US15681395

申请日：2017-08-20

申请人： Huan-Yu Su

发明人： Huan-Yu Su

IPC分类号： G10L15/20 ,  G10L21/0216 ,  G10L21/0224 ,  H04R1/32 ,  H04R29/00 ,  G10L21/0264

摘要： Provided is a method and computer program product for producing an enhanced audio signal for an output device from audio signals received by 2 or more microphones in close proximity to each other. For example, one embodiment of the present invention comprises the steps of receiving a first input audio signal from the first microphone, digitizing the first input audio signal to produce a first digitized audio input signal, receiving a second input audio input signal from the second microphone, digitizing the second input audio input signal to produce a second digitized audio input signal, using the first digitized audio input signal as a reference signal to an adaptive prediction filter, using the second digitized audio input signal as input to said adaptive prediction filter and finally adding a prediction result signal from the adaptive prediction filter to the first digitized audio input signal to produce the enhanced audio signal. In other embodiments, any number of microphones can be used, and in all embodiments there is no requirement to detect or locate the source or direction of arrival of the input audio signals.

公开(公告)号：US07796623B2

公开(公告)日：2010-09-14

申请号：US12384019

申请日：2009-03-30

申请人： Michael M. Metzger ,  Huan-Yu Su ,  Armin Abold

发明人： Michael M. Metzger ,  Huan-Yu Su ,  Armin Abold

IPC分类号： H04L12/28 ,  H04M3/22

CPC分类号： H04M3/2236 ,  H04L41/5009 ,  H04L41/5087 ,  H04L65/80 ,  H04M7/1205

摘要： There is provided a method of detecting and reporting poor voice quality for use by a gateway device. The method comprises facilitating a connection between a telephone and a remote telephone via a network, and detecting a poor voice quality indictor during the connection. The method further comprises capturing, for a pre-determined period of time, telephone voice data being exchanged between the gateway and the telephone, network voice data being exchanged between the gateway and the network, and gateway parameters. The method also comprises packetizing the telephone voice data, the network voice data and the gateway parameters into a plurality packets having a network address of a network storage, and transmitting the plurality packets destined for the network storage via the network. In one aspect, the poor voice quality indictor may be generated by a user of the telephone in response to a poor voice quality of the connection.

摘要翻译： 提供了一种检测和报告由网关设备使用的较差语音质量的方法。 该方法包括通过网络促进电话和远程电话之间的连接，以及在连接期间检测不良语音质量指示符。 该方法还包括：在预定时间段内，捕获在网关与电话之间交换的电话语音数据，网关和网络之间交换的网络语音数据以及网关参数。 该方法还包括将电话语音数据，网络语音数据和网关参数分组成具有网络存储器的网络地址的多个分组，并且经由网络发送去往网络存储的多个分组。 在一个方面，响应于连接的差的语音质量，可能由电话的用户产生差的语音质量指示符。

公开(公告)号：US20080147384A1

公开(公告)日：2008-06-19

申请号：US12069973

申请日：2008-02-14

申请人： Huan-Yu Su ,  Yang Gao

发明人： Huan-Yu Su ,  Yang Gao

IPC分类号： G10L11/04

CPC分类号： G10L19/12 ,  G10L19/0204 ,  G10L19/09 ,  G10L19/18 ,  G10L19/20 ,  G10L25/90 ,  G10L2019/0002 ,  G10L2019/0016

摘要： There is provided a method of selecting a pitch lag value for a portion of a speech signal, the method comprising: computing a weighted correlation function of the portion of the speech signal for a range of delay times, wherein the weighting of the correlation function depends on both the delay time and a characteristic of one or more previous portions of the speech signal; and selecting the pitch lag value based on a delay time from the range of delay times that maximizes the weighted correlation function.

摘要翻译： 提供了一种为语音信号的一部分选择音调滞后值的方法，所述方法包括：在延迟时间范围内计算语音信号部分的加权相关函数，其中相关函数的权重取决于 在延迟时间和语音信号的一个或多个先前部分的特性上; 以及从加权相关函数最大化的延迟时间的范围内，基于延迟时间选择音调滞后值。

公开(公告)号：US07266493B2

公开(公告)日：2007-09-04

申请号：US11251179

申请日：2005-10-13

申请人： Huan-Yu Su ,  Yang Gao

发明人： Huan-Yu Su ,  Yang Gao

IPC分类号： G10L11/04

CPC分类号： G10L19/12 ,  G10L19/0204 ,  G10L19/09 ,  G10L19/18 ,  G10L19/20 ,  G10L25/90 ,  G10L2019/0002 ,  G10L2019/0016

摘要： There is provided a method of selecting a pitch lag value from a plurality of pitch lag candidates for coding a speech signal. The method comprises identifying the plurality of pitch lag candidates from a frame of the speech signal using correlation; classifying the speech signal to obtain a voice classification; determining whether one or more of the plurality of pitch lag candidates are in a temporal neighborhood of one or more previous pitch lag values; favoring the one or more of the plurality of pitch lag candidates determined to be in the temporal neighborhood of the one or more previous pitch lag values, by adaptive weighting, over other ones of the plurality of pitch lag candidates; and selecting the pitch lag value based on the voice classification and the one or more of the plurality of pitch lag candidates favored by the adaptive weighting.

摘要翻译： 提供了一种从用于编码语音信号的多个音调滞后候选中选择音调滞后值的方法。 该方法包括使用相关性从语音信号的帧中识别多个音调滞后候选; 对语音信号进行分类以获得语音分类; 确定所述多个音调滞后候选中的一个或多个是否在一个或多个先前音调滞后值的时间邻域中; 通过对多个音调滞后候选中的其他音调滞后候选，通过自适应加权来确定被确定为处于一个或多个先前音调滞后值的时间邻域中的多个音调滞后候选中的一个或多个; 以及基于所述语音分类和由所述自适应加权优选的所述多个音调滞后候选中的一个或多个来选择所述音调滞后值。

公开(公告)号：US07080010B2

公开(公告)日：2006-07-18

申请号：US10911118

申请日：2004-08-03

申请人： Eyal Shlomot ,  Huan-Yu Su

发明人： Eyal Shlomot ,  Huan-Yu Su

IPC分类号： G10L19/02

CPC分类号： G10L15/285

摘要： A multi-channel speech processor for encoding speech in a packet network environment is disclosed. In one illustrative aspect, a complexity resource manager (CRM) is executed by a controller or processor. The CRM manages the level of complexity of encoding which is used by a signal processing unit (SPU) to convert the speech signal into packet data. In general, the CRM determines the level of complexity of encoding based on a calculated complexity budget, where the complexity budget is determined based on the time required to process prior speech signal channels and the time available to process the remaining channels. In this way, the CRM is able to control the overall complexity of the speech processor through its ability to signal the SPU to encode speech signal in a complexity reduced mode based on the calculated complexity budget under certain conditions.

摘要翻译： 公开了一种用于在分组网络环境中编码语音的多声道语音处理器。 在一个说明性方面，复杂性资源管理器（CRM）由控制器或处理器执行。 CRM管理由信号处理单元（SPU）用于将语音信号转换成分组数据的编码的复杂程度。 通常，CRM基于计算的复杂度预算确定编码的复杂程度，其中基于处理先前语音信号信道所需的时间和可用于处理剩余信道的时间来确定复杂度预算。 以这种方式，CRM能够通过其在特定条件下基于计算的复杂度预算在复杂度降低模式下对SPU进行信号编码语音信号的能力来控制语音处理器的总体复杂性。

公开(公告)号：US06898566B1

公开(公告)日：2005-05-24

申请号：US09640841

申请日：2000-08-16

申请人： Adil Benyassine ,  Huan-Yu Su

发明人： Adil Benyassine ,  Huan-Yu Su

IPC分类号： G10L11/02 ,  G10L11/04 ,  G10L19/14 ,  G10L21/02

CPC分类号： G10L19/22 ,  G10L19/09 ,  G10L2025/783

摘要： There are provided speech coding methods and systems for estimating a plurality of speech parameters of a speech signal for coding the speech signal using one of a plurality of speech coding algorithms, the plurality of speech parameters includes pitch information, the plurality of speech parameters is calculated using a plurality of thresholds. An example method includes estimating a background noise level in the speech signal to determine a signal to noise ratio (SNR) for the speech signal, adjusting one or more of the plurality of thresholds based on the SNR to generate one or more SNR adjusted thresholds, analyzing the speech signal to extract the pitch information using the one or more SNR adjusted thresholds, and repeating the estimating, the adjusting and the analyzing to code the speech signal using one the plurality of speech coding algorithms.

摘要翻译： 提供了语音编码方法和系统，用于使用多种语音编码算法中的一种来估计用于对语音信号进行编码的语音信号的多个语音参数，所述多个语音参数包括音调信息，所述多个语音参数被计算 使用多个阈值。 示例性方法包括估计语音信号中的背景噪声电平以确定语音信号的信噪比（SNR），基于SNR调整多个阈值中的一个或多个阈值以产生一个或多个SNR调整阈值， 分析语音信号以使用一个或多个SNR调整的阈值提取音调信息，并且使用多个语音编码算法中的一个重复对该语音信号的估计，调整和分析。

公开(公告)号：US06856954B1

公开(公告)日：2005-02-15

申请号：US09627375

申请日：2000-07-28

申请人： Huan-Yu Su

发明人： Huan-Yu Su

IPC分类号： G10L19/02 ,  H04L1/00 ,  H04L29/08 ,  G10L19/00

CPC分类号： H04L1/0014

摘要： A flexible variable rate vocoder and related method of operation. The vocoder selects a target average data rate responsive to at least one network parameter and at least one external parameter.

摘要翻译： 灵活的可变速率声码器及相关操作方法。 声码器响应于至少一个网络参数和至少一个外部参数来选择目标平均数据速率。