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Description: 通信系统 ITU-T G.729 Annex -D 语音压缩标准附录D-Coding of speech at 8 kbit/s using Conjugate-
Structure Algebraic-Code-Excited Linear-Prediction
(CS-ACELP)Annex D: 6.4 kbit/s CS-ACELP speech coding algorithm ITU-T Recommendation G.729 – Annex D
(Previously CCITT Recommendation)
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Author: yangsher |
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Description: 通信系统 ITU－T G.729 Annex －E 语音压缩标准附录 E-Coding of speech at 8 kbit/s using Conjugate-Structure Algebraic-Code-Excited Linear-Prediction(CS-ACELP)Annex E: 11.8 kbit/s CS-ACELP speech codingalgorithm
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Author: yangsher |
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Description: Implementation of a speech codec based on coding of speech at 8 kbit/s
using conjugate-structure algebraic-code-excited linear-prediction (CS-ACELP)
- We took .wav files that is sampled at 8000 Hz using 16-bit linear PCM. The encoding process is done every 10ms frame or 80 samples. For the preprocessing stage, the samples are high passed with cut-off frequency of 140 Hz and scaled down by 2. A total of 240 samples are buffer for windowing and autocorrelation computation. The autocorrelation coefficients are used to calculate the LP filter coefficients using the Levinson-Durbin algorithm. The LP filter coefficients are converted to Line Spectral Pair (LSP) coefficients. LSP coefficients are converted back to the LP filter coefficients, which is just the reverse process of the conversion from LP to LSP. This module is exactly what the decoder will need in order to convert the LSP coefficients to LP coefficients. We decided not to implement the LSF quantization module because we did not have the codebook information when we designed our system.
The open-loop pitch delay is calculated first for each frame. Then the closed-loop pitch
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Author: coco |
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Description: G.729 语音编码算法叫做对结构代数码激励线性预测语音压缩编码，它的核
心原理是线性预测和矢量量化。这种编码标准已在1995年H月工TU一TSG15全会上
通过，并在1996年6月的工TU一TSG15末次会议上通过了G.729的附件A“减少复杂
度的skbit/5cs一ACELP语音编解码器”，正式成为国际电信标准
-ITU一TG72915one
ofthestandardsthatcanProcesssPeechsignalefficiently
.
Thisstandard15basedon
Linea-rPrediction一AnalysisandVecto-rQuantization
.
ThesPeechsignalencodedby
thisstandardcanbetransmittedattherateofskbi/ts，whilethequaliytofthesPeech15
good
.
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Author: 王婧 |
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Description:
该方案旨在简化，如播放，暂停播放声音基本波，停止和恢复。
自从我在www.mathworks.com通知少人有解决 如何停止播放 ，所以我要做到这一点，它张贴在这里波的声音。
希望你们喜欢的节目：对
编码器实现了G.723.1的多脉冲模式（在6.3 kb / s的运行）。解码器可以解码的5.3 kb / s的，在6.3 kb的多脉冲/秒，连续传输，数据包丢失隐藏）编码器（所有的ACELP方式。
该编码器和解码器之间的标准接口是位文件（同位文件由ITU - T参考代码产生兼容）。第二个选项之间传递的编码器和解码器使用Matlab的数据信息文件。此选项允许测试通过传递量化或unquantized之间的编码器和解码器的信息。
有报道[1]描述了G.723.1编码与Matlab实现与分布包括在内。 ITU - T建议G.723.1的[2]在线提供。该建议含-This program intended to simplify the playing basic of wave sound such as play, pause, stop and resume.
Since i notice in www.mathworks.com less people have the solution to how to stop wave sound from playing , so i try to do this and post it here.
Hope you guys enjoy the program :P
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Author: zaaa |
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Description: 附件代码为正交匹配追踪的源程序，用matlab编写，程序简单实用。-Annex code orthogonal matching pursuit of the source, written with matlab, the program is simple and practical.
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Author: miaoguijun |
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Description: ITU-T G.729 Source codeITU-T G.729 Annex B ANSI-C Source CodeVersion 1.4 Last modified: November, 2000 */TITLE-----Fixed-point description of Recommendation G.729 with ANNEX BCoding of Speech at 8 kbit/s using Conjugate-StructureAlgebraic-Code-Excited Linear-Prediction (CS-ACELP) withVoice Activity Decision(VAD), Discontinuous Transmission(DTX), and Comfort Noise Generation(CNG).SOFTWARE AND INTELLECTUAL PROPERTY----------------------------------This software package is provided as part of ITU-T Recommendation G.729B.Copyright (c) 1996, AT&T, France Telecom, Lucent Technologies, NTT, Rockwell International, Universite de Sherbrooke.All rights reserved.The copy of the source C code, version 1.4, is given under Copyright of the authors, only for the purpose of establishing the specification of a codec.VERSION-------This is version 1.4.-ITU-T G.729 Source codeITU-T G.729 Annex B ANSI-C Source CodeVersion 1.4 Last modified: November, 2000*/TITLE-----Fixed-point description of Recommendation G.729 with ANNEX BCoding of Speech at 8 kbit/s using Conjugate-StructureAlgebraic-Code-Excited Linear-Prediction (CS-ACELP) withVoice Activity Decision(VAD), Discontinuous Transmission(DTX), and Comfort Noise Generation(CNG).SOFTWARE AND INTELLECTUAL PROPERTY----------------------------------This software package is provided as part of ITU-T Recommendation G.729B.Copyright (c) 1996, AT&T, France Telecom, Lucent Technologies, NTT, Rockwell International, Universite de Sherbrooke.All rights reserved.The copy of the source C code, version 1.4, is given under Copyright of the authors, only for the purpose of establishing the specification of a codec.VERSION-------This is version 1.4.
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Author: 丁一 |
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Description: Compressed audio format developed by Ericsson used by many 3G cell phones for voice recordings such as MMS messages incorporates the Algebraic Code Excited Linear Prediction (ACELP) algorithm, which is designed to efficiently compress human speech audio.-Compressed audio format developed by Ericsson used by many 3G cell phones for voice recordings such as MMS messages incorporates the Algebraic Code Excited Linear Prediction (ACELP) algorithm, which is designed to efficiently compress human speech audio.
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Author: ha |
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Description: Performs the main encoder routine Fixed-point C simulation of AMR WB ACELP coding algorithm with 20 msspeech frames for wideband speech signals.
-Performs the main encoder routine Fixed-point C simulation of AMR WB ACELP coding algorithm with 20 msspeech frames for wideband speech signals.
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Author: gokingma |
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Description: kalman滤波器经典入门matlab程序
[Simulink_kalman.rar] - 实现kalman滤波算法，通过跟踪，估计物体运动轨迹。-,matlab,matlab例程/matlab，做了仿真
[matlab.rar] - 收集的matlab，simulink的全部基本命令，方便大家查阅！
[selfadptivesignalprocess.rar] - 自适应信号处理算法的实现，非常有价值
[pcm.rar] - 语音编码方案的选取对移动通信系统的通话质量、信道容量等有重要影响。本文讨论了TD-SCDMA系统中AMR语音编码的自适应机制，同时分析了AMR中代数码本线性预测(ACELP)算法及实现过程。该方案可以在一块TMS320C5510上实现。
[kalmanFilterProgram.rar] - 目标跟踪问题的应用背景是雷达数据处理，即雷达在搜索到目标并记录目标的位置数据, 对测量到的目标位置数据（称为点迹）进行处理，自动形成航迹，并对目标在下一时刻的位置进行预测。-在matlab simulink 中实现kalman滤波器设计，可以生动的演示kalman滤波在运动估计中的预测作用-Matlab simulink in achieving kalman filter design can be a vivid kalman滤波器经典入门matlab程序
[Simulink_kalman.rar]- 实现kalman滤波算法，通过跟踪，估计物体运动轨迹。-,matlab,matlab例程/matlab，做了仿真
[matlab.rar]- 收集的matlab，simulink的全部基本命令，方便大家查阅！
[selfadptivesignalprocess.rar]- 自适应信号处理算法的实现，非常有价值
[pcm.rar]- 语音编码方案的选取对移动通信系统的通话质量、信道容量等有重要影响。本文讨论了TD-SCDMA系统中AMR语音编码的自适应机制，同时分析了AMR中代数码本线性预测(ACELP)算法及实现过程。该方案可以在一块TMS320C5510上实现。
[kalmanFilterProgram.rar]- 目标跟踪问题的应用背景是雷达数据处理，即雷达在搜索到目标并记录目标的位置数据, 对测量到的目标位置数据（称为点迹）进行处理，自动形成航迹，并对目标在下一时刻的位置进行预测。demonstration of kalman filtering in the motion estimation of the prediction
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Author: 王子 |
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Description: ACELP important algorithms for speech compression.-ACELP important algorithms for speech compression.
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Author: haitham |
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Description: tetra etsi acelp codec complete with makefile to build on linux with g-tetra etsi acelp codec complete with makefile to build on linux with gcc
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Author: Thomas |
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Description: G.729的全名Code Excited Linear Prediction Model (CELP) and Conjugate-Structure Algebraic CELP (CS-ACELP). 共轭结构-代数码激励线性预测编码.
G.729家族一共包括三个版本G.729, G.729A,G.729B,G.729AB:
G.729 是G.729的第一个版本,
G.729A 相对于G.729简化了算法.
G.729B 给G.729增加了静音压缩的算法.
G.729Ab 给G.729A增加了静音压缩的算法.
G.729属于合成分析(abs)的语音编码方法,这种方法在中低比特率的情况下可得到较好的语音质量.
编码工作于10ms的语音帧上,8K采样率,具有80个样点,,对每一帧提取CELP模型的有关参量(LP滤波器系数,自适应码书和固定码书的指针和增益),这些参量被编码和传送.在解码器中,这些参量被用来恢复激励信号和合成滤波器的参量.语音信号由激励信号通过LP合成滤波器重构得到.
注：8000Hz采样率，16位样本，压缩率为16:1，80个16位样本经过编码后为10个8位编码串；我用G.729a试过，语音还可以，音乐就不行了。另：商业应用G.729每个许可是10美元，一个许可就是一路通话；不知道现在还是不是。
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Author: dmwang1982@163.com |
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Description: G.729的全名Code Excited Linear Prediction Model (CELP) and Conjugate-Structure Algebraic CELP (CS-ACELP). 共轭结构-代数码激励线性预测编码. -Code Excited Linear Prediction Model (CELP) and Conjugate-Structure Algebraic CELP (CS-ACELP
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Author: 曹旭辉 |
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Description: 支持高精度视频数据，在高动态范围场景提供更多图像细节
支持先进编码工具，在获得更好图像质量的同时获得更高编码效率
支持感兴趣区域（ROI）变质量编码，在网络带宽或存储空间有限的情况下，提供更符
合监控需要的高质量视频编码
支持可伸缩视频编码（SVC），满足不同传输网络带宽和数据存储环境的需求
u 支持代数码书激励线性预测（ACELP）和变换音频编码（TAC）切换的双核音频编码，
保证对语音和环境（背景）声音均有较好的编码效果
支持声音识别特征参数编码，避免编码失真对语音识别和声纹识别的影响
支持监控专用信息（绝对时间、智能分析结果、报警信息），便于音视频内容的有效
管理和综合利用
支持加密和认证，保证监控数据的保密性、真实性和完整性(Support high precision video data to provide more image details in a high dynamic range scene
Support advanced coding tools to obtain better coding efficiency while obtaining better image quality
Support region of interest (ROI) variable quality coding, providing more characters in the case of limited network bandwidth or storage space.
High quality video coding for monitoring requirements
Support scalable video coding (SVC) to meet the needs of different transmission network bandwidth and data storage environment
U supports dual core audio coding, which enables digital books to stimulate linear prediction (ACELP) and switched audio coding (TAC) switching.)
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