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1. 发明申请

US20050137863A1 Method and apparatus for speech coding 有权
标题翻译：用于语音编码的方法和装置
公开(公告)号：US20050137863A1
公开(公告)日：2005-06-23
申请号：US10964861
申请日：2004-10-14
申请人： Mark Jasiuk , Tenkasi Ramabadran , Udar Mittal , James Ashley , Michael McLaughlin
发明人： Mark Jasiuk , Tenkasi Ramabadran , Udar Mittal , James Ashley , Michael McLaughlin
IPC分类号： G10L19/08 , G10L19/04
CPC分类号： G10L19/09
摘要： A method and apparatus for prediction in a speech-coding system is provided herein. The method of a 1st order long-term predictor (LTP) filter, using a sub-sample resolution delay, is extended to a multi-tap LTP filter, or, viewed from another vantage point, the conventional integer-sample resolution multi-tap LTP filter is extended to use sub-sample resolution delay. This novel formulation of a multi-tap LTP filter offers a number of advantages over the prior-art LTP filter configurations. Particularly, defining the lag with sub-sample resolution makes it possible to explicitly model the delay values that have a fractional component, within the limits of resolution of the over-sampling factor used by the interpolation filter. The coefficients of such a multi-tap LTP filter are thus largely freed from modeling the effect of delays that have a fractional component. Consequently their main function is to maximize the prediction gain of the LTP filter via modeling the degree of periodicity that is present and by imposing spectral shaping.
摘要翻译：本发明提供了语音编码系统中的预测方法和装置。使用子样本分辨率延迟的1＆lt; ST＆gt;阶长期预测器（LTP）滤波器的方法被扩展到多抽头LTP滤波器，或者从另一个有利位置观察，常规的整数抽样分辨率多抽头LTP滤波器被扩展为使用子样本分辨率延迟。多抽头LTP滤波器的这种新颖配方提供了优于现有技术的LTP滤波器配置的许多优点。特别地，使用子样本分辨率定义滞后使得可以在内插滤波器使用的过采样因子的分辨率的限度内明确地建模具有分数分量的延迟值。因此，这种多抽头LTP滤波器的系数在很大程度上不会对具有分数分量的延迟的影响进行建模。因此，它们的主要功能是通过建模存在的周期程度和施加频谱整形来最大化LTP滤波器的预测增益。

2. 发明申请

US20050096903A1 Method and apparatus for performing harmonic noise weighting in digital speech coders 有权
标题翻译：在数字语音编码器中执行谐波噪声加权的方法和装置
公开(公告)号：US20050096903A1
公开(公告)日：2005-05-05
申请号：US10965462
申请日：2004-10-14
申请人： Udar Mittal , James Ashley
发明人： Udar Mittal , James Ashley
IPC分类号： G10L21/02 , G10L11/04
CPC分类号： G10L21/0364 , G10L21/0264
摘要： To address the need for choosing values of harmonic noise weighting (HNW) coefficient (εp) so that the amount of harmonic noise weighting cam be optimized, a method and apparatus for performing harmonic noise weighting in digital speech coders is provided herein. During operation, received speech is analyzed to determine a pitch period. HNW coefficients are then chosen based on the pitch period, and a perceptual noise weighting filter (C(z)) is determined based on the harmonic-noise weighting (HNW) coefficients (εp).
摘要翻译：为了满足选择谐波噪声加权（HNW）系数（epsilon）的值的需要，以便谐波噪声加权凸轮的数量被优化，在数字语音中执行谐波噪声加权的方法和装置本文提供了编码器。在操作期间，分析接收到的语音以确定音调周期。然后基于音调周期来选择HNW系数，并且基于谐波噪声加权（HNW）系数（ε-p P）来确定感知噪声加权滤波器（C（z））。

3. 发明授权

US08890723B2 Encoder that optimizes bit allocation for information sub-parts 有权
标题翻译：用于优化信息子部分的位分配的编码器
公开(公告)号：US08890723B2
公开(公告)日：2014-11-18
申请号：US13481608
申请日：2012-05-25
申请人： James Ashley , Udar Mittal
发明人： James Ashley , Udar Mittal
IPC分类号： H03M7/30 , H03M7/40
CPC分类号： H03M7/4006 , H03M7/3082
摘要： A digital information encoder including a divider configured to divide a block of information into a plurality of sub-parts, an initial bit allocator configured to perform an initial allocation of bits to a KTH sub-part of said plurality of sub-parts, a processor configured to compute an estimated number of bits for encoding said KTH sub-part, and a bit allocation adjuster configured to obtain an adjusted bit allocation for said KTH sub-part by adjusting said initial allocation of bits to said KTH sub-part based, at least in part, on said estimated number of bits, wherein the encoder encodes said KTH sub-part using said adjusted bit allocation for said KTH sub-part.
摘要翻译：一种数字信息编码器，包括：分配器，用于将信息块划分成多个子部分;初始位分配器，被配置为对所述多个子部件的KTH子部分执行位的初始分配;处理器被配置为计算用于对所述KTH子部分进行编码的估计位数，以及比特分配调整器，被配置为通过基于所述KTH子部分的位的初始分配来获得所述KTH子部分的调整比特分配，至少部分地基于所述估计的比特数，其中所述编码器使用所述调整的所述KTH子部分的比特分配对所述KTH子部分进行编码。

4. 发明申请

US20070027680A1 Method and apparatus for coding an information signal using pitch delay contour adjustment 有权
标题翻译：使用音调延迟轮廓调整对信息信号进行编码的方法和装置
公开(公告)号：US20070027680A1
公开(公告)日：2007-02-01
申请号：US11190680
申请日：2005-07-27
申请人： James Ashley , Udar Mittal
发明人： James Ashley , Udar Mittal
IPC分类号： G10L11/04
CPC分类号： G10L19/09 , G10L21/00 , G10L21/06
摘要： In a speech encoder/decoder a pitch delay contour endpoint modifier is employed to shift the endpoints of a pitch delay interpolation curve up or down. Parficularly, the endpoints of the pitch delay interpolation curve are shifted based on a variation and/or a standard deviation in pitch delay.
摘要翻译：在语音编码器/解码器中，使用音调延迟轮廓端点修改器来向上或向下移动音调延迟内插曲线的端点。特别地，音调延迟插值曲线的端点基于音调延迟的变化和/或标准偏差而移位。

5. 发明申请

US20070109977A1 Method and apparatus for improving listener differentiation of talkers during a conference call 审中-公开
标题翻译：在电话会议期间改善讲话者听众差异的方法和装置
公开(公告)号：US20070109977A1
公开(公告)日：2007-05-17
申请号：US11273670
申请日：2005-11-14
申请人： Udar Mittal , James Ashley
发明人： Udar Mittal , James Ashley
IPC分类号： H04L12/16
CPC分类号： G10L21/038
摘要： A method and apparatus for improving listener differentiation of talkers during a conference call is provided herein. Particularly, during a teleconference a node (101) will extend the bandwidth of received signals (e.g., speech). Each caller within the conference call will then have their voice projected by the node (101) to a particular spot in three-dimensional space.
摘要翻译：本文提供了一种用于在电话会议期间改善讲话者的听众区分的方法和装置。特别地，在电话会议期间，节点（101）将扩展接收信号（例如语音）的带宽。电话会议中的每个呼叫者然后将其节点（101）投射到三维空间中的特定点的声音。

6. 发明授权

US09043201B2 Method and apparatus for processing audio frames to transition between different codecs 有权
标题翻译：用于处理音频帧以在不同编解码器之间转换的方法和装置
公开(公告)号：US09043201B2
公开(公告)日：2015-05-26
申请号：US13342462
申请日：2012-01-03
申请人： Udar Mittal , James P. Ashley
发明人： Udar Mittal , James P. Ashley
IPC分类号： G10L21/00 , G10L19/02 , G10L19/12 , G10L19/18
CPC分类号： G10L19/0212 , G10L19/12 , G10L19/18
摘要： A method (700, 800) and apparatus (100, 200) processes audio frames to transition between different codecs. The method can include producing (720), using a first coding method, a first frame of coded output audio samples by coding a first audio frame in a sequence of frames. The method can include forming (730) an overlap-add portion of the first frame using the first coding method. The method can include generating (740) a combination first frame of coded audio samples based on combining the first frame of coded output audio samples with the overlap-add portion of the first frame. The method can include initializing (760) a state of a second coding method based on the combination first frame of coded audio samples. The method can include constructing (770) an output signal based on the initialized state of the second coding method.
摘要翻译：方法（700,800）和装置（100,200）处理音频帧以在不同编解码器之间转换。该方法可以包括：使用第一编码方法，通过对帧序列中的第一音频帧进行编码来产生编码的输出音频样本的第一帧（720）。该方法可以包括使用第一编码方法形成（730）第一帧的重叠部分。该方法可以包括：通过将编码的输出音频样本的第一帧与第一帧的重叠相加部分组合来产生（740）编码音频样本的组合第一帧。该方法可以包括基于编码音频样本的组合第一帧来初始化（760）第二编码方法的状态。该方法可以包括基于第二编码方法的初始化状态来构造（770）输出信号。

7. 发明申请

US20130173259A1 Method and Apparatus for Processing Audio Frames to Transition Between Different Codecs 有权
标题翻译：用于处理音频帧以转换不同编解码器的方法和装置
公开(公告)号：US20130173259A1
公开(公告)日：2013-07-04
申请号：US13342462
申请日：2012-01-03
申请人： Udar Mittal , James P. Ashley
发明人： Udar Mittal , James P. Ashley
IPC分类号： G10L19/00
CPC分类号： G10L19/0212 , G10L19/12 , G10L19/18
摘要： A method (700, 800) and apparatus (100, 200) processes audio frames to transition between different codecs. The method can include producing (720), using a first coding method, a first frame of coded output audio samples by coding a first audio frame in a sequence of frames. The method can include forming (730) an overlap-add portion of the first frame using the first coding method. The method can include generating (740) a combination first frame of coded audio samples based on combining the first frame of coded output audio samples with the overlap-add portion of the first frame. The method can include initializing (760) a state of a second coding method based on the combination first frame of coded audio samples. The method can include constructing (770) an output signal based on the initialized state of the second coding method.
摘要翻译：方法（700,800）和装置（100,200）处理音频帧以在不同编解码器之间转换。该方法可以包括：使用第一编码方法，通过对帧序列中的第一音频帧进行编码来产生编码的输出音频样本的第一帧（720）。该方法可以包括使用第一编码方法形成（730）第一帧的重叠部分。该方法可以包括：通过将编码的输出音频样本的第一帧与第一帧的重叠相加部分组合来产生（740）编码音频样本的组合第一帧。该方法可以包括基于编码音频样本的组合第一帧来初始化（760）第二编码方法的状态。该方法可以包括基于第二编码方法的初始化状态来构造（770）输出信号。

8. 发明授权

US08219408B2 Audio signal decoder and method for producing a scaled reconstructed audio signal 有权
标题翻译：音频信号解码器和用于产生缩放的重构音频信号的方法
公开(公告)号：US08219408B2
公开(公告)日：2012-07-10
申请号：US12345117
申请日：2008-12-29
申请人： James P. Ashley , Udar Mittal
发明人： James P. Ashley , Udar Mittal
IPC分类号： G10L19/00
CPC分类号： G10L19/24 , G10L19/008
摘要： During operation a multiple channel audio input signal is received and coded to generate a coded audio signal. A balance factor having balance factor components each associated with an audio signal of the multiple channel audio signal is generated. A gain value to be applied to the coded audio signal to generate an estimate of the multiple channel audio signal based on the balance factor and the multiple channel audio signal is determined, with the gain value configured to minimize a distortion value between the multiple channel audio signal and the estimate of the multiple channel audio signal. The representation of the gain value may be output for transmission and/or storage.
摘要翻译：在操作期间，接收并编码多声道音频输入信号以产生编码音频信号。产生具有各自与多声道音频信号的音频信号相关联的平衡因子分量的平衡因子。确定要应用于编码音频信号以产生基于平衡因子和多声道音频信号的多声道音频信号的估计的增益值，其中增益值被配置为最小化多声道音频之间的失真值信号和多声道音频信号的估计。可以输出增益值的表示用于传输和/或存储。

9. 发明授权

US08207875B2 Encoder that optimizes bit allocation for information sub-parts 有权
标题翻译：用于优化信息子部分的位分配的编码器
公开(公告)号：US08207875B2
公开(公告)日：2012-06-26
申请号：US12607439
申请日：2009-10-28
申请人： James P. Ashley , Udar Mittal
发明人： James P. Ashley , Udar Mittal
IPC分类号： H03M7/34
CPC分类号： H03M7/4006 , H03M7/3082
摘要： A encoder/decoder architecture (200, 300, 700) that uses an arithmetic encoder (220) to encode the MSB portions of the output of a Factorial Pulse Coder (212), that encodes the output of a first-level source encoder (210), e.g., MDCT. Sub-parts (e.g., frequency bands) of portions (e.g., frames) of the signal are suitably sorted in increasing order based on a measure related to signal energy (e.g., signal energy itself). Doing this in a system (100) that overlays Arithmetic Encoding on Factorial Pulse coding results in bits being re-allocated to bands with higher signal energy content, ultimately yielding higher signal quality and higher bit utilization efficiency.
摘要翻译：一种编码器/解码器架构（200,300,700），其使用算术编码器（220）对因子脉冲编码器（212）的输出的MSB部分进行编码，其对第一级源编码器（210）的输出进行编码），例如MDCT。基于与信号能量（例如，信号能量本身）相关的测量，以增加的顺序适当地分类信号的部分（例如，帧）的子部分（例如，频带）。在覆盖因子脉冲编码的算术编码的系统（100）中进行这一操作导致将比特重新分配给具有较高信号能量内容的频带，最终产生更高的信号质量和更高的比特利用效率。

10. 发明申请

US20120095757A1 AUDIO SIGNAL BANDWIDTH EXTENSION IN CELP-BASED SPEECH CODER 有权
标题翻译：基于CELP的语音编码器中的音频信号带宽扩展
公开(公告)号：US20120095757A1
公开(公告)日：2012-04-19
申请号：US13247129
申请日：2011-09-28
申请人： Jonathan A. Gibbs , James P. Ashley , Udar Mittal
发明人： Jonathan A. Gibbs , James P. Ashley , Udar Mittal
IPC分类号： G10L19/12
CPC分类号： G10L21/038
摘要： A method for decoding an audio signal having a bandwidth that extends beyond a bandwidth of a CELP excitation signal in an audio decoder including a CELP-based decoder element. The method includes obtaining a second excitation signal having an audio bandwidth extending beyond the audio bandwidth of the CELP excitation signal, obtaining a set of signals by filtering the second excitation signal with a set of bandpass filters, scaling the set of signals using a set of energy-based parameters, and obtaining a composite output signal by combining the scaled set of signals with a signal based on the audio signal decoded by the CELP-based decoder element.
摘要翻译：一种用于对包括基于CELP的解码器元件的音频解码器中具有超出CELP激励信号的带宽的带宽的音频信号进行解码的方法。该方法包括获得具有超出CELP激励信号的音频带宽的音频带宽的第二激励信号，通过用一组带通滤波器对第二激励信号进行滤波来获得一组信号，使用一组基于能量的参数，并且通过将缩放的信号组合与基于由基于CELP的解码器元素解码的音频信号的信号组合来获得复合输出信号。

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