搜索资源列表
sox-12.17.1.tar
- Sox是最为著名的Open Source声音文件格式转换工具。已经被广泛移植到Dos、windows、OS2、Sun、Next、Unix、Linux等多个操作系统平台。Sox项目是由Lance Norskog创立的，后来被众多的开发者逐步完善，现在已经能够支持等多种声音文件格式和声音处理效果。基本上常见的声音格式都能够支持。更加有用的是，Sox能够进行声音滤波、采样频率转换，这对那些从事声讯平台开发或维护的朋友非常有用。当然，Sox里面也包括一些DSP算法，有兴趣的朋友可以下载回去研究。Sox可
Lab540-AudioFIR
- 基于TI DSP TMS320DM642的音频FIR滤波处理程序代码-based on TI's TMS320DM642 DSP audio processing FIR filter code
wavdest
- Writes an audio stream to a WAV file. It takes a single audio stream as input, and its output pin must be connected to the File Writer filter. This sample filter is based on the CTransformFilter class.
PWM_VOICE
- 用LPC213X系列的PWM来产生语音。语音数据来源于PC机转换后得到的。最大可以是16位的。 由于是PWM所以输出端口需要低通滤波才能够得到比较干净的声音。 可以经过修改使用LPC2132以上的ARM的D/A来产生。 This project outputs audio on PWM0 using wave samples that are stored in the on-chip Flash ROM. -LPC213X series used to generate th
DSP_dg
- 这是本人用Matlab实现音频噪声滤除的试验，要求分析并选用适当滤波器-using Matlab audio noise filtering tests, requirements analysis and the choice of appropriate filter
DirectShowBOOK04
- 本书以DirectX SDK 9.0版为蓝本，内容几乎涵盖了在Windows平台上使用DirectShow进行C++编程的方方面面。全书共分4个部分。第1部分详细介绍了DirectShow的基础知识。第2部分重点讨论了Filter的开发，以及DirectShow应用程序的开发，包括目前非常流行的音视频采集、数码摄像机的支持、非线性编辑等应用。第3部分深入分析了DirectShow SDK提供的部分典型源代码。第4部分结合作者个人的一些开发实践，通过案例和开放源代码分析，进一步介绍DirectSh
filter.mixing_auds.v13
- 这是一个DirectShow过滤器，已经编译并且运行良好，它可以把最多16路输入音频信号混合为1路输出音频信号；（对于不同的采样率，采用线性重采样算法；其内部混合运算达到24Bits精度；支持立体声、5.1环绕、7.1环绕输出）-This is a DirectShow filter, has been compiled and run well, it can be up to 16 input audio signal to an output mixed audio signal (fo
filter.parser_ssif.vs8
- 这是一个DirectShow过滤器，它可以解析3D蓝光影碟里面的SSIF文件，并且将各个视频流、音频流、字幕流分别输出；已经编译并且运行良好，运行效率极高；-This is a DirectShow filter, which can be resolved inside the 3D Blu-ray SSIF files, and the various video streams, audio streams, subtitle streams are output has been co
filter.resamp_cdda.vs8
- 这是一个DirectShow过滤器，它专门用于CD音频重采样（把44100Hz采样率的音频信号 转化为 48000Hz的）；采用了高精度的奈奎斯特重采样算法，品质极佳；已经编译并且运行良好；-This is a DirectShow filter, it specifically for CD audio resampling (the audio signal into 44100Hz to 48000Hz sampling rate) and the use of a high-preci
Audio_Filter
- Audio filter for Matlab How is it working? 1) Take the current samples and some previous samples (which previously had been stored) for multiplied by a factor defined. 2) Take the output values in past moments and multiply by other fac
filter
- Apply FIR filter to 44 kHz raw 16-bit PCM linear audio then downsample to 11 kHz.
filter(LMS)
- LMS自适应滤波去噪算法，里面一个去噪功能模块及一个声音处理模块。可去除音频中的噪声-LMS adaptive filtering denoising algorithm, which is a denoising function module and a voice processing module. Noise in audio
FILTER
- 对于原始音频信号进行高通、低通、带通、带阻滤波（含GUI文件），可以手动设置噪声信号，并对比染噪前和滤波后的信号区别。-For the original audio signal high-pass, low pass, band pass, band stop filter (with GUI files), you can manually set the noise signal, and compares the difference signal noise before transf
audio-equalizer-algorithm
- 硕士论文。主要包括：1、数字滤波器的设计原理及其matlab实现 2、数字均衡器的不同设计算法-Master thesis. Mainly include：1、Design principle of digital filter and its matlab implementation 2、Different design algorithms for digital equalizer
Sub-band-filter
- MP3语音编码方面的自带滤波原理，详细讲述了相关原理与概念-Filtering principle comes with MP3 audio coding, and a detailed account of the relevant principles and concepts
filter.CompressDTS
- 把PCM音频流保存为DTS文件的dshow过滤器；-Save the PCM audio stream to DTS file dshow filter
filter-of-the-audio-signal
- 简单的语音信号的采样及滤波,以音乐片段为例-the sampling and filtering of the audio signal，such as music
gyhhku
- 用LabVIEW实现对输入音频滤波达到降噪效果后通过滤波处理，回复原正弦信号，欢迎参阅。。(LabVIEW to achieve the input audio filter to achieve noise reduction effect, through the filtering process, reply to the original sinusoidal signal, welcome to see..)
Informe 5
- An equilizer using digital filter in matlab
ffmpeg
- FFmpeg is the leading multimedia framework, able to decode, encode, transcode, mux, demux, stream, filter and play pretty much anything that humans and machines have created. It supports the most obscure ancient formats up to the cutting edge. No mat