公开(公告)号：US09401156B2

公开(公告)日：2016-07-26

申请号：US12215649

申请日：2008-06-27

申请人： Huan-Yu Su ,  Yang Gao

发明人： Huan-Yu Su ,  Yang Gao

IPC分类号： G10L19/09 ,  G10L19/12 ,  G10L19/18 ,  G10L19/20 ,  G10L25/90 ,  G10L19/02 ,  G10L19/00

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 using an adaptive tilt compensation by a speech decoder. The method comprises receiving a bit stream including a plurality of parameters representative of a speech signal; identifying an adaptive code vector and a fixed code vector using the plurality of parameters; scaling the adaptive code vector and the fixed code vector to generate a scaled adaptive code vector and a scaled fixed code vector; summing the scaled adaptive code vector and the scaled fixed code vector to generate a synthesized output; calculating a first reflection coefficient based on the plurality of parameters representative of the speech signal; multiplying the first reflection coefficient by a factor to generate a tilt factor; and applying the tilt factor to the synthesized output based on an encoding bit rate.

摘要翻译： 提供了一种通过语音解码器使用自适应倾斜补偿的方法。 该方法包括：接收包括表示语音信号的多个参数的比特流; 使用所述多个参数来识别自适应码矢量和固定码矢量; 缩放自适应码矢量和固定码矢量以生成缩放的自适应码矢量和缩放的固定码矢量; 对经缩放的自适应码矢量和缩放的固定码矢量求和以产生合成输出; 基于表示所述语音信号的多个参数来计算第一反射系数; 将第一反射系数乘以因子以产生倾斜因子; 以及基于编码比特率将所述倾斜因子应用于所述合成输出。

公开(公告)号：US20130267270A1

公开(公告)日：2013-10-10

申请号：US13468978

申请日：2012-05-10

申请人： Marco Y.C. Cheng ,  Huan-Yu Su ,  James W. Johnston

发明人： Marco Y.C. Cheng ,  Huan-Yu Su ,  James W. Johnston

IPC分类号： H04W48/08 ,  H04W88/04

CPC分类号： H04W84/005 ,  H04W28/06 ,  H04W52/0219 ,  H04W52/0245 ,  H04W76/12 ,  H04W88/04 ,  H04W92/18 ,  Y02B70/00 ,  Y02D70/122 ,  Y02D70/124 ,  Y02D70/1262 ,  Y02D70/1264 ,  Y02D70/142 ,  Y02D70/144 ,  Y02D70/162 ,  Y02D70/166

摘要： Provided is a system for wireless communications including several base stations supporting a wide area wireless network and several mobile user equipment (UE) devices. Each mobile UE device may be configured to transmit a request to establish a local wireless connection with one or more of the UE devices. The mobile UE device may receive a response containing connectivity information from each of the mobile UE devices and then select one of the mobile UE devices based on the connectivity information received from each of the mobile UE devices. The mobile UE device may then establish a local wireless connection with the selected mobile UE device. The mobile UE device may then communicate with one of the base stations in the wide area wireless network through the selected mobile UE device, utilizing the local wireless connection.

摘要翻译： 提供了一种用于无线通信的系统，包括支持广域无线网络的几个基站和几个移动用户设备（UE）设备。 每个移动UE设备可以被配置为发送与一个或多个UE设备建立本地无线连接的请求。 移动UE设备可以从每个移动UE设备接收包含连接性信息的响应，然后基于从每个移动UE设备接收到的连接性信息来选择一个移动UE设备。 然后，移动UE设备可以建立与所选移动UE设备的本地无线连接。 移动UE设备然后可以利用本地无线连接通过所选移动UE设备与广域无线网络中的一个基站进行通信。

公开(公告)号：US07876892B1

公开(公告)日：2011-01-25

申请号：US11318375

申请日：2005-12-22

申请人： Huan-Yu Su ,  Jonathan Peace

发明人： Huan-Yu Su ,  Jonathan Peace

IPC分类号： H04M9/08

CPC分类号： H04B3/234

摘要： There is provided a method for use by an echo canceller to cancel a far echo at a variable delay and a near echo at a fixed delay. The method comprises constructing an echo signal model based on an incoming signal, determining a variable echo delay for a far echo caused by a far echo source, determining a fixed echo delay for a near echo caused by a near echo source, subtracting the echo signal model from an outgoing signal at a window placed around the variable echo delay to cancel far echo, e.g. when the echo canceller determines existence of the far echo, and subtracting the echo signal model from the outgoing signal at a window placed around the fixed echo delay to cancel near echo, e.g. regardless of existence of the near echo, wherein the fixed echo delay is smaller than the variable echo delay.

摘要翻译： 提供了一种由回波消除器使用的方法，以可变延迟和固定延迟的近似回波消除远端回波。 该方法包括基于输入信号构建回波信号模型，确定由远回波源引起的远回波的可变回波延迟，确定由近回波源引起的近距离回波的固定回波延迟，减去回波信号 模型从出站信号的窗口放置在可变回波延迟周围以取消远端回波，例如 当回波消除器确定远回波的存在，并且在围绕固定回波延迟的窗口处从输出信号中减去回波信号模型以消除近似回波，例如， 不管近回波是否存在，其中固定回波延迟小于可变回波延迟。

公开(公告)号：US20080294429A1

公开(公告)日：2008-11-27

申请号：US12215649

申请日：2008-06-27

申请人： Huan-Yu Su ,  Yang Gao

发明人： Huan-Yu Su ,  Yang Gao

IPC分类号： G10L19/12

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 using an adaptive tilt compensation by a speech decoder. The method comprises receiving a bit stream including a plurality of parameters representative of a speech signal; identifying an adaptive code vector and a fixed code vector using the plurality of parameters; scaling the adaptive code vector and the fixed code vector to generate a scaled adaptive code vector and a scaled fixed code vector; summing the scaled adaptive code vector and the scaled fixed code vector to generate a synthesized output; calculating a first reflection coefficient based on the plurality of parameters representative of the speech signal; multiplying the first reflection coefficient by a factor to generate a tilt factor; and applying the tilt factor to the synthesized output based on an encoding bit rate.

摘要翻译： 提供了一种通过语音解码器使用自适应倾斜补偿的方法。 该方法包括：接收包括表示语音信号的多个参数的比特流; 使用所述多个参数来识别自适应码矢量和固定码矢量; 缩放自适应码矢量和固定码矢量以生成缩放的自适应码矢量和缩放的固定码矢量; 对经缩放的自适应码矢量和缩放的固定码矢量求和以产生合成输出; 基于表示所述语音信号的多个参数来计算第一反射系数; 将第一反射系数乘以因子以产生倾斜因子; 以及基于编码比特率将所述倾斜因子应用于所述合成输出。

公开(公告)号：US20080288246A1

公开(公告)日：2008-11-20

申请号：US12220480

申请日：2008-07-23

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 using a processing circuitry for selecting a preferential pitch lag value from a plurality of pitch lag values, including a first pitch lag value and a second pitch lag value, for coding an input speech signal. The method comprises determining a first timing relationship between a previous pitch lag value and at least one of the plurality of pitch lag values; determining a second timing relationship between the first pitch lag value and the second pitch lag value; favoring one of the first pitch lag value and the second pitch lag value based on the first timing relationship and the second timing relationship to select one of the first pitch lag value and the second pitch lag value as the preferential pitch lag value; and converting the input speech signal into an encoded speech using the preferential pitch lag value.

摘要翻译： 提供了一种使用处理电路的方法，用于从包括第一音调滞后值和第二音调滞后值的多个音调滞后值中选择用于编码输入语音信号的优先音调滞后值。 该方法包括确定先前的音调滞后值与多个音调滞后值中的至少一个之间的第一定时关系; 确定所述第一音调滞后值和所述第二音调滞后值之间的第二定时关系; 基于第一定时关系和第二定时关系，优选第一音调滞后值和第二音调滞后值中的一个，以选择第一音调滞后值和第二音调滞后值之一作为优先音调滞后值; 以及使用优先音调滞后值将输入语音信号转换为编码语音。

公开(公告)号：US20070255561A1

公开(公告)日：2007-11-01

申请号：US11827915

申请日：2007-07-12

申请人： Huan-Yu Su ,  Yang Gao

发明人： Huan-Yu Su ,  Yang Gao

IPC分类号： G10L21/00

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

摘要： In accordance with one aspect of the invention, a selector supports the selection of a first encoding scheme or the second encoding scheme based upon the detection or absence of the triggering characteristic in the interval of the input speech signal. The first encoding scheme has a pitch pre-processing procedure for processing the input speech signal to form a revised speech signal biased toward an ideal voiced and stationary characteristic. The pre-processing procedure allows the encoder to fully capture the benefits of a bandwidth-efficient, long-term predictive procedure for a greater amount of speech components of an input speech signal than would otherwise be possible. In accordance with another aspect of the invention, the second encoding scheme entails a long-term prediction mode for encoding the pitch on a sub-frame by sub-frame basis. The long-term prediction mode is tailored to where the generally periodic component of the speech is generally not stationary or less than completely periodic and requires greater frequency of updates from the adaptive codebook to achieve a desired perceptual quality of the reproduced speech under a long-term predictive procedure.

摘要翻译： 根据本发明的一个方面，选择器基于输入语音信号的间隔中的触发特性的检测或不存在，支持选择第一编码方案或第二编码方案。 第一编码方案具有用于处理输入语音信号以形成偏向理想有声和静态特征的修正语音信号的音调预处理过程。 预处理过程允许编码器完全捕获带宽有效的长期预测程序的优点，用于输入语音信号的大量语音分量比否则可能的更多。 根据本发明的另一方面，第二编码方案需要一种长期预测模式，用于以子帧为基础对子帧上的音调进行编码。 长期预测模式被定制为语音的大致周期性分量通常不是静止的或小于完全周期性的，并且需要来自自适应码本的更高频率的更新以在长时间内实现再现语音的期望感知质量， 术语预测程序。

公开(公告)号：US06937979B2

公开(公告)日：2005-08-30

申请号：US09896682

申请日：2001-06-29

申请人： Yang Gao ,  Huan-Yu Su

发明人： Yang Gao ,  Huan-Yu Su

IPC分类号： G10L19/14 ,  G10L21/02 ,  G10L19/00

CPC分类号： G10L19/265 ,  G10L19/18 ,  G10L21/0364

摘要： In a coding procedure, a spectral content of a speech signal is estimated. A preferential coding algorithm or preferential value of at least one coding parameter is selected based on the estimated spectral content of the speech signal. The speech signal is coded in accordance with the selected coding algorithm or the selected coding parameter to control the operation of one or more of the following: a pre-processing filter, a post-processing filter, a coding control coefficient, a weighting filter, a synthesis filter, and a quantization table.

摘要翻译： 在编码过程中，估计语音信号的频谱内容。 基于所估计的语音信号的频谱内容来选择优选编码算法或至少一个编码参数的优先值。 语音信号根据所选择的编码算法或选择的编码参数进行编码，以控制以下一个或多个的操作：预处理滤波器，后处理滤波器，编码控制系数，加权滤波器， 合成滤波器和量化表。

公开(公告)号：US06757649B1

公开(公告)日：2004-06-29

申请号：US10409404

申请日：2003-04-08

申请人： Yang Gao ,  Adil Benyassine ,  Jes Thyssen ,  Eyal Shlomot ,  Huan-yu Su

发明人： Yang Gao ,  Adil Benyassine ,  Jes Thyssen ,  Eyal Shlomot ,  Huan-yu Su

IPC分类号： G10L1912

CPC分类号： G10L19/00 ,  G10L19/167 ,  G10L19/24 ,  H03G3/00

摘要： A speech compression system capable of encoding a speech signal into a bitstream for subsequent decoding to generate synthesized speech is disclosed. The speech compression system optimizes the bandwidth consumed by the bitstream by balancing the desired average bit rate with the perceptual quality of the reconstructed speech. The speech compression system comprises a full-rate codec, a half-rate codec, a quarter-rate codec and an eighth-rate codec. The codecs are selectively activated based on a rate selection. In addition, the full and half-rate codecs are selectively activated based on a type classification. Each codec is selectively activated to encode and decode the speech signals at different bit rates emphasizing different aspects of the speech signal to enhance overall quality of the synthesized speech.

摘要翻译： 公开了能够将语音信号编码为比特流以进行后续解码以产生合成语音的语音压缩系统。 语音压缩系统通过将期望的平均比特率与重构语音的感知质量进行平衡来优化比特流消耗的带宽。 语音压缩系统包括全速率编解码器，半速率编解码器，四分之一速率编解码器和八速率编解码器。 基于速率选择来选择性地激活编解码器。 此外，基于类型分类，全速率和半速率编解码器被选择性地激活。 选择性地激活每个编解码器以以强调语音信号的不同方面的不同比特率对语音信号进行编码和解码，以增强合成语音的整体质量。

公开(公告)号：US06636829B1

公开(公告)日：2003-10-21

申请号：US09617191

申请日：2000-07-14

申请人： Adil Benyassine ,  Eyal Shlomot ,  Huan-Yu Su

发明人： Adil Benyassine ,  Eyal Shlomot ,  Huan-Yu Su

IPC分类号： G10L1900

CPC分类号： G10L19/08 ,  G10L19/005 ,  G10L19/07 ,  G10L19/083 ,  G10L25/90 ,  G10L2019/0012

摘要： An exemplary decoder comprises a receiver that receives parameters of a speech signal on a frame-by-frame basis, a control logic for decoding parameters and for resynthesizing the speech signal, the control logic including a minimum spacing indicative of a minimum difference required between LSFs of consecutive frames, a frame recovery logic that, when a lost frame detector detects a lost frame, sets the minimum spacing for the lost frame to a first value which is greater than the minimum spacing for the previously received frame, and/or uses pitch lag parameters of a plurality of previously received frames to extrapolate a pitch lag parameter for the lost frame, and/or sets gain parameter of a subframe of the lost frame in a first manner if the lost gain parameter is an adaptive codebook gain parameter and in a second manner if the lost gain parameter is a fixed codebook gain parameter.

摘要翻译： 示例性解码器包括接收器，其逐帧地接收语音信号的参数，用于解码参数并用于再合成语音信号的控制逻辑，所述控制逻辑包括指示LSF之间所需的最小差异的最小间隔 连续帧的帧恢复逻辑，当丢失帧检测器检测到丢失帧时，将丢失帧的最小间隔设置为大于先前接收帧的最小间隔的第一值，和/或使用间距 多个先前接收的帧的滞后参数，以推断丢失帧的音调滞后参数，和/或以丢失的增益参数为自适应码本增益参数，以第一种方式设置丢失帧的子帧的增益参数，并且 丢失增益参数是固定码本增益参数的第二种方式。

公开(公告)号：US06345248B1

公开(公告)日：2002-02-05

申请号：US09433002

申请日：1999-11-02

申请人： Huan-Yu Su ,  Tom Hong Li

发明人： Huan-Yu Su ,  Tom Hong Li

IPC分类号： G10L1912

CPC分类号： G10L19/08 ,  G10L19/06 ,  G10L2019/0011

摘要： A pitch lag coding device and method using interframe correlation inherent in pitch lag values to reduce coding bit requirements. A pitch lag value is extracted for a given speech frame, and then refined for each subframe. For every speech frame having N samples of speech, LPC analysis and vector quantization are performed for the whole coding frame. The LPC residual obtained for each frame is then processed such that pitch lag values for all subframes within the coding frame are analyzed concurrently. The remaining coding parameters, i.e., the codebook search, gain parameters, and excitation signal, are then analyzed sequentially according to their respective subframes.

摘要翻译： 音调滞后编码装置和方法，使用音调滞后值固有的帧间相关性来减少编码比特要求。 为给定的语音帧提取音调滞后值，然后针对每个子帧进行细化。 对于具有N个语音样本的每个语音帧，对于整个编码帧执行LPC分析和矢量量化。 然后对每个帧获得的LPC残差进行处理，使得同时分析编码帧内的所有子帧的音调滞后值。 然后根据其各自的子帧依次分析剩余的编码参数，即码本搜索，增益参数和激励信号。