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该代码是双速率的语音压缩编码(G.723.1)的matlab代码。能在matlab6.5以上运行-Dual-rate voice compressed encoding(G.723.1) based on MatLab platform. It works on MatLab 6.5 or later versions.
Date : 2008-10-13 Size : 2.04mb User : 刘开文

G.723.1语音压缩编码的英文原文和中文译文。该协议是双速率语音压缩编码,提供5.3和6.3kbps的速率,占用很窄的带宽。-G.723.1 voice coding the original English and Chinese translations. The agreement is a two-bit-rate speech coding, and 5.3. Rate, which takes up a very narrow bandwidth.
Date : 2008-10-13 Size : 506.6kb User :

GSM 6.10语音压缩算法,点对点之间语音通信声音清晰,音质优良,延时小,使用简单-GSM 6.10 voice compression algorithm, point-to-point voice communications between loud and excellent acoustic quality, Delay small, simple to use
Date : 2008-10-13 Size : 131.65kb User : 孙健

这是G.727标准文件-This is the G.727 standard documents
Date : 2025-12-21 Size : 142kb User : 唐思珑

这是G.729标准-This is the standard G.729
Date : 2025-12-21 Size : 100kb User : 唐思珑

ITU-T G.723语音压缩编解码算法源码和测试语音,内有标准附件和使用说明。-ITU-T G.723 voice compression codec algorithm source code and test voice, within the standard accessories and use.
Date : 2025-12-21 Size : 1.62mb User : 阙劲峰

该代码是双速率的语音压缩编码(G.723.1)的matlab代码。能在matlab6.5以上运行-Dual-rate voice compressed encoding(G.723.1) based on MatLab platform. It works on MatLab 6.5 or later versions.
Date : 2025-12-21 Size : 2.04mb User : 刘开文

这是LPC-10压缩算法的源代码,愿共享之.也希望能赐与MELP方面的算法源码.-This is the LPC-10 compression algorithm source code, and is willing to Sharing. I hope to bestow MELP the algorithm source code.
Date : 2025-12-21 Size : 779kb User : 李文化

3GPP中GSM的AMR编解码c源码和协议,包含8种不同速率的编解码方式-3GPP GSM AMR codec c-source agreement and contains eight different rate codec approach
Date : 2025-12-21 Size : 868kb User : 肯尼斯

GSM 6.10语音压缩算法,点对点之间语音通信声音清晰,音质优良,延时小,使用简单-GSM 6.10 voice compression algorithm, point-to-point voice communications between loud and excellent acoustic quality, Delay small, simple to use
Date : 2025-12-21 Size : 132kb User : 孙健

主要研究了超声回波信号的小波分析算法和BP神经网络算法。在超声检测中,作为检测基本数据的脉冲反射回波信号受到电子噪声和结构噪声的干扰,-major study of the ultrasonic echo signal wavelet analysis algorithm and BP neural network algorithm. In ultrasonic testing, to detect the basic data pulse echo signals reflected by the structure of electronic noise and noise interference,
Date : 2025-12-21 Size : 967kb User :

26104-610.zip为欧洲G3的语音压缩标准gsmamr的C语言浮点算法的标准源程序。-26104- 610.zip for the European G3 voice compression standard C language gsmamr Floating Point Algorithm standard source.
Date : 2025-12-21 Size : 286kb User : 文翔云

用于MTK6217平台的MP4软件转换工具,Realese给大家分享,希望对您的产品设计和定义有帮助-MP4 for MTK6217 platform software conversion tools, Realese to the U.S. share, in the hope that your product design and the definition has to help
Date : 2025-12-21 Size : 6.85mb User : yangjun

最新的AMR源代码. the newest AMR source code-AMR latest source code. The newest AMR source code
Date : 2025-12-21 Size : 901kb User : jiangdongping

Speech Signal Processing Toolkit 3.0
Date : 2025-12-21 Size : 562kb User : mindong

G.729A编码解码算法英文文档,ITU-T-err
Date : 2025-12-21 Size : 48kb User : li

语音的多线程编程的开发和研究,对于多线程入门的是不可多得的好文章-Voice of the multi-threaded programming development and research, for multi-threaded entry is rare good article
Date : 2025-12-21 Size : 12kb User : 阿强

amr语音编解码库,浮点运算,用于3gp格式的语音压缩-amr voice codec library, floating-point operations for 3gp format voice compression
Date : 2025-12-21 Size : 286kb User : lyoyou

本课程设计的主要目的是在MATLAB环境下,使用窗口设计法设计一个滤波器,并对语音信号进行滤波去噪。开发平台为MATLAB7.0,设计方法为窗口设计法。用麦克风采集一段语音信号,绘制波形并观察其频谱,给定相应技术指标,用凯塞窗设计一个满足指标的FIR滤波器,对该语音信号进行滤波去噪处理,比较滤波前后的波形和频谱并进行分析。在滤波前后语音信号波形的变化中,由于我们录制的语音信号噪声不大,所以观察并不明显,但在频域波形中,我们可以明显的看到设计的滤波器对语音信号进行了滤波处理,将噪声进行了滤除。此次滤波基本达到了要求,完成了设计指标。-The main purpose of this course is designed in the MATLAB environment, the use of window design method to design a filter, and the voice signal filtering noise. Development platform for MATLAB7.0, design methods for the window design method. Collecting a voice signal with a microphone, and observe its spectrum waveform drawing, given the corresponding technical indicators, using a Kaiser window design FIR filter meet the target, the filter denoising speech signal processing, a waveform and spectrum before and after filtering and analyzed. Before and after the filter in the speech signal waveform changes, because we recorded voice signal to noise is low, they observed not obvious, but the waveform in the frequency domain, we can clearly see the design of a filter for the filtering of the speech signal processing , the noise were filtered out. The filtering basically reached the requirements to complete the design specifications.
Date : 2025-12-21 Size : 115kb User : 吴丹

mbuni源码, Mbuni是一个开源的多媒体消息服务网关。-Mbuni is a fully-fledged Free/Open Source Multimedia Messaging Service (MMS) gateway
Date : 2025-12-21 Size : 608kb User : 何光
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