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Description: 基于数字语音的FIR音频滤波，是数字信号处理的必须掌握的算法。-FIR digital speech-based audio filtering, digital signal processing algorithms to master.
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Size: 564224 |
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Description: 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
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Size: 48128 |
Author: dy |
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Description: CCS FIR高通音频滤波器，用于DSP实验-CCS FIR high-pass audio filter for DSP experimental
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Size: 29696 |
Author: dy |
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Description: 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
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Size: 297984 |
Author: 南宫崔浩 |
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Description: 详细介绍了模拟和数字滤波器的原理和设计，并给出了想要的C语言实现代码。对音频进行研究的开发人员有很大的借鉴意义
-Introduces the principle and design of analog and digital filter, and gives the desired C language code. There is great significance to study the audio developers
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Size: 2075648 |
Author: 刘红军 |
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Description: 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
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Size: 155648 |
Author: marrymiffy |
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Description: 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 (for different sampling rates, the use of linear resampling algorithm its internal mixing precision arithmetic reach 24Bits support for stereo, 5.1 surround, 7.1 surround output)
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Size: 1527808 |
Author: 夏永明 |
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Description: 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 compiled and run good, high operating efficiency
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Size: 709632 |
Author: 夏永明 |
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Description: 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-precision Nyquist resampling algorithms, excellent quality has been compiled and run good
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Size: 1206272 |
Author: 夏永明 |
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Description: 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 factors.
3) Finally, all the results of all these multiplications are added, giving an outlet to the current moment.
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Size: 6144 |
Author: tomquick |
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Description: 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.
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Size: 4892672 |
Author: 小林 |
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Description: Apply FIR filter to 44 kHz raw 16-bit PCM linear audio then downsample to 11 kHz.
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Size: 4096 |
Author: hgqdhd |
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Description: 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
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Size: 2048 |
Author: lonsonOS |
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Description: 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 transfection and filtered.
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Size: 867328 |
Author: 兆林 |
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Description: 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
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Size: 4733952 |
Author: 七夜雨 |
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Description: 本文的重点是软件实现的DSP算法音频效果，如回声、颤音、合唱等。本研究的主要贡献在于实现一个软件应用程序能对音频信号进行实时处理。-This paper is focused on the software
implementation of DSP oriented algorithms for different
audio effects, like Echo, 3-Tap Echo, Vibrato, Tremolo
and Chorus. The main contribution of this research
consists in a software application which is able to
process in real time the audio signals received the
electric guitar using PC sound card. In addition, the
application also includes a digital filter class used for
implementation of a low pass filter as well as other filter
types, a tone generator and the options for storing the
sound samples in a .wav file. The application is written
in C# language which uses DirectSound library in order
to process the sound samples on a low level.
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Size: 394240 |
Author: 张 |
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Description: 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
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Size: 10348544 |
Author: 胖熊 |
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Description: 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
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Size: 2448384 |
Author: 夏永明 |
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Description: 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.
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Size: 25600 |
Author: When |
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Description: 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
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Size: 8359936 |
Author: DanielWang |
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