搜索资源列表
firDitong
- CCS FIR低通音频滤波器 以优化运行速度 用于DSP实验-CCS FIR low-pass audio filter to optimize speed for DSP experiments
firGaotong
- CCS FIR高通音频滤波器，用于DSP实验-CCS FIR high-pass audio filter for DSP experimental
Adaptive-filter-code
- 自适应滤波器代码，利用自适应滤波器对音频号进行滤波，从而实现对音频中噪声滤除，使音频信号更清晰-Adaptive filter code, to filter the audio signal by using the adaptive filter, so as to realize the audio filtering noise in the audio signal, so that more clearly
tree-structure-filter-bank
- MATLAB仿真设计树形结构滤波器组 并将原来语音文件与滤波接收后的问价进行对比 能实现功能-MATLAB Simulation design tree structure filter bank and the original audio file and filter asking price after receiving comparison function can be realized
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
Audio-processing-system
- 这是一篇基于FPGA的音频处理系统的论文，讲述了音频信号处理的相关知识和基于FPGA的数字滤波器设计等。-This is an FPGA-based audio processing system of the paper, about the knowledge of audio signal processing and FPGA-based digital filter design.
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
audio
- 基本的音频处理程序，包含ola、resample、filter等功能。-Basic audio processing procedures, it can do ola, resample, 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