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    Method and apparatus for high performance low bit-rate coding of unvoiced speech
    4.
    发明授权
    Method and apparatus for high performance low bit-rate coding of unvoiced speech 有权
    无声语音的高性能低比特率编码方法和装置

    公开(公告)号：US07191125B2

    公开(公告)日：2007-03-13

    申请号：US11066356

    申请日：2005-02-24

    Applicant: Pengjun Huang

    Inventor： Pengjun Huang

    IPC: G10L21/02

    CPC classification number: G10L19/12 ,  G10L19/083 ,  G10L19/18 ,  G10L25/93

    Abstract: A low-bit-rate coding technique for unvoiced segments of speech, without loss of quality compared to the conventional Code Excited Linear Prediction (CELP) method operating at a much higher bit rate. A set of gains are derived from a residual signal after whitening the speech signal by a linear prediction filter. These gains are then quantized and applied to a randomly generated sparse excitation. The excitation is filtered, and its spectral characteristics are analyzed and compared to the spectral characteristics of the original residual signal. Based on this analysis, a filter is chosen to shape the spectral characteristics of the excitation to achieve optimal performance.

    Abstract translation: 一种用于无声段语音的低比特率编码技术，与以更高比特率运行的常规码激励线性预测（CELP）方法相比，没有质量损失。 通过线性预测滤波器对语音信号进行白化后的残差信号得到一组增益。 然后将这些增益量化并应用于随机产生的稀疏激发。 对激发进行滤波，并对其光谱特征进行分析，并将其与原始残留信号的光谱特性进行比较。 基于该分析，选择滤波器来形成激发的光谱特征以实现最佳性能。

    Low-complexity encoding/decoding of quantized MDCT spectrum in scalable speech and audio codecs
    5.
    发明授权
    Low-complexity encoding/decoding of quantized MDCT spectrum in scalable speech and audio codecs 有权

    公开(公告)号：US08527265B2

    公开(公告)日：2013-09-03

    申请号：US12255604

    申请日：2008-10-21

    Applicant: Yuriy Reznik ,  Pengjun Huang

    Inventor： Yuriy Reznik ,  Pengjun Huang

    IPC: G10L19/00

    CPC classification number: G10L19/24 ,  G10L19/038

    Abstract: A scalable speech and audio codec is provided that implements combinatorial spectrum encoding. A residual signal is obtained from a Code Excited Linear Prediction (CELP)-based encoding layer, where the residual signal is a difference between an original audio signal and a reconstructed version of the original audio signal. The residual signal is transformed at a Discrete Cosine Transform (DCT)-type transform layer to obtain a corresponding transform spectrum having a plurality of spectral lines. The transform spectrum spectral lines are transformed using a combinatorial position coding technique. The combinatorial position coding technique includes generating a lexicographical index for a selected subset of spectral lines, where each lexicographic index represents one of a plurality of possible binary strings representing the positions of the selected subset of spectral lines. The lexicographical index represents non-zero spectral lines in a binary string in fewer bits than the length of the binary string.

    Method and apparatus for high performance low bit-rate coding of unvoiced speech
    7.
    发明授权
    Method and apparatus for high performance low bit-rate coding of unvoiced speech 有权
    无声语音的高性能低比特率编码方法和装置

    公开(公告)号：US07493256B2

    公开(公告)日：2009-02-17

    申请号：US11685748

    申请日：2007-03-13

    Applicant: Pengjun Huang

    Inventor： Pengjun Huang

    IPC: G10L19/12 ,  G10L19/08 ,  G10L19/04 ,  G10L11/06

    CPC classification number: G10L19/12 ,  G10L19/083 ,  G10L19/18 ,  G10L25/93

    Abstract: A low-bit-rate coding technique for unvoiced segments of speech, without loss of quality compared to the conventional Code Excited Linear Prediction (CELP) method operating at a much higher bit rate. A set of gains are derived from a residual signal after whitening the speech signal by a linear prediction filter. These gains are then quantized and applied to a randomly generated sparse excitation. The excitation is filtered, and its spectral characteristics are analyzed and compared to the spectral characteristics of the original residual signal. Based on this analysis, a filter is chosen to shape the spectral characteristics of the excitation to achieve optimal performance.

    Abstract translation: 用于无声音段的低比特率编码技术，与以更高比特率运行的常规码激励线性预测（CELP）方法相比，没有质量损失。 通过线性预测滤波器对语音信号进行白化后的残差信号得到一组增益。 然后将这些增益量化并应用于随机产生的稀疏激发。 对激发进行滤波，并对其光谱特征进行分析，并将其与原始残留信号的光谱特性进行比较。 基于该分析，选择滤波器来形成激发的光谱特征以实现最佳性能。

    METHOD AND APPARATUS FOR HIGH PERFORMANCE LOW BIT-RATE CODING OF UNVOICED SPEECH
    8.
    发明申请
    METHOD AND APPARATUS FOR HIGH PERFORMANCE LOW BIT-RATE CODING OF UNVOICED SPEECH 有权
    用于高性能低码率编码的方法和装置

    公开(公告)号：US20070192092A1

    公开(公告)日：2007-08-16

    申请号：US11685748

    申请日：2007-03-13

    Applicant: Pengjun Huang

    Inventor： Pengjun Huang

    IPC: G10L19/00

    CPC classification number: G10L19/12 ,  G10L19/083 ,  G10L19/18 ,  G10L25/93

    Abstract: A low-bit-rate coding technique for unvoiced segments of speech, without loss of quality compared to the conventional Code Excited Linear Prediction (CELP) method operating at a much higher bit rate. A set of gains are derived from a residual signal after whitening the speech signal by a linear prediction filter. These gains are then quantized and applied to a randomly generated sparse excitation. The excitation is filtered, and its spectral characteristics are analyzed and compared to the spectral characteristics of the original residual signal. Based on this analysis, a filter is chosen to shape the spectral characteristics of the excitation to achieve optimal performance.

    Abstract translation: 一种用于无声段语音的低比特率编码技术，与以更高比特率运行的常规码激励线性预测（CELP）方法相比，没有质量损失。 通过线性预测滤波器对语音信号进行白化后的残差信号得到一组增益。 然后将这些增益量化并应用于随机产生的稀疏激发。 对激发进行滤波，并对其光谱特征进行分析，并将其与原始残留信号的光谱特性进行比较。 基于该分析，选择滤波器来形成激发的光谱特征以实现最佳性能。

    Method and apparatus for improved detection of rate errors in variable rate receivers
    9.
    发明申请
    Method and apparatus for improved detection of rate errors in variable rate receivers 有权
    用于改进可变速率接收机中速率误差检测的方法和装置

    公开(公告)号：US20050050407A1

    公开(公告)日：2005-03-03

    申请号：US10938445

    申请日：2004-09-09

    Applicant: Khaled El-Maleh ,  Eddie-Lun Choy ,  Arasanipalai Ananthapadmanabhan ,  Andrew DeJaco ,  Pengjun Huang

    Inventor： Khaled El-Maleh ,  Eddie-Lun Choy ,  Arasanipalai Ananthapadmanabhan ,  Andrew DeJaco ,  Pengjun Huang

    IPC: H04B7/26 ,  H04L1/00 ,  H04L1/08 ,  H04L29/08 ,  H04B7/216 ,  G06F11/00

    CPC classification number: H04L1/08 ,  H04L1/0046 ,  H04L1/201

    Abstract: A system and method for detection of rate determination algorithm errors in variable rate communications system receivers. The disclosed embodiments prevent rate determination algorithm errors from causing audible artifacts such as screeches or beeps. The disclosed system and method detects frames with incorrectly determined data rates and performs frame erasure processing and/or memory state clean up to prevent propagation of distortion across multiple frames. Frames with incorrectly determined data rates are detected by checking illegal rate transitions, reserved bits, validating unused filter type bit combinations and analyzing relationships between fixed code-book gains and linear prediction coefficient gains.

    Abstract translation: 一种用于在可变速率通信系统接收机中检测速率确定算法错误的系统和方法。 所公开的实施例防止速率确定算法错误引起可听见的伪影，例如吱吱声或嘟嘟声。 所公开的系统和方法检测具有错误确定的数据速率的帧，并执行帧擦除处理和/或存储器状态清理，以防止跨多个帧的失真传播。 通过检查非法速率转换，保留位，验证未使用的过滤器类型位组合以及分析固定代码簿增益和线性预测系数增益之间的关系来检测具有不正确确定的数据速率的帧。

    Frame erasure compensation method in a variable rate speech coder
    10.
    发明授权
    Frame erasure compensation method in a variable rate speech coder 有权
    可变速率语音编码器中的帧擦除补偿方法

    公开(公告)号：US06584438B1

    公开(公告)日：2003-06-24

    申请号：US09557283

    申请日：2000-04-24

    Applicant: Sharath Manjunath ,  Pengjun Huang ,  Eddie-Lun Tik Choy

    Inventor： Sharath Manjunath ,  Pengjun Huang ,  Eddie-Lun Tik Choy

    IPC: G10L1300

    CPC classification number: G10L21/02 ,  G10L19/005 ,  G10L19/097

    Abstract: A frame erasure compensation method in a variable-rate speech coder includes quantizing, with a first encoder, a pitch lag value for a current frame and a first delta pitch lag value equal to the difference between the pitch lag value for the current frame and the pitch lag value for the previous frame. A second, predictive encoder quantizes only a second delta pitch lag value for the previous frame (equal to the difference between the pitch lag value for the previous frame and the pitch lag value for the frame prior to that frame). If the frame prior to the previous frame is processed as a frame erasure, the pitch lag value for the previous frame is obtained by subtracting the first delta pitch lag value from the pitch lag value for the current frame. The pitch lag value for the erasure frame is then obtained by subtracting the second delta pitch lag value from the pitch lag value for the previous frame. Additionally, a waveform interpolation method may be used to smooth discontinuities caused by changes in the coder pitch memory.

    Abstract translation: 可变速率语音编码器中的帧擦除补偿方法包括：利用第一编码器量化当前帧的音调滞后值，以及等于当前帧的音调滞后值与第 前一帧的音调滞后值。 第二预测编码器仅量化前一帧的第二增量音调滞后值（等于先前帧的音调滞后值与该帧之前的帧的音调滞后值之间的差）。 如果先前帧之前的帧被作为帧擦除处理，则通过从当前帧的音调滞后值中减去第一增量音调滞后值来获得先前帧的音调滞后值。 然后通过从前一帧的音调滞后值减去第二增量音调滞后值来获得擦除帧的音调滞后值。 此外，可以使用波形插值方法来平滑由编码器音调存储器的变化引起的不连续性。

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