Description: karaoke系統開發library,使用DC-DSP Filter對Audio做採集和特效,相當重量及的Sample-karaoke system development library, using DC-Audio DSP Filter to do Acquisition and special effects, a considerable weight and the Sample Platform: |
Size: 245130 |
Author:starlife |
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Description: 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 the PWM voice. Voice data from the PC after the conversion. It is the largest of 16. Because there is PWM output ports need low-pass filter can be relatively clean voice. It has been modified over the use LPC2132 ARM D/A to produce. This project outputs audio on PWM0 using wave samples that are stored in the on-chip Flash ROM. Platform: |
Size: 63488 |
Author:徐勇 |
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Description: This article describes the process of storing the audio data of a movie file (.mpeg, .mpg, .avi and .dat) on the hard disk in a WAV file using DirectShow filters.
-This article describes the process of storing the audio data of a movie file (. Mpeg,. Mpg. Avi and. Dat) on the hard disk in a WAV file using DirectShow filters. Platform: |
Size: 108544 |
Author:secondman_Jon |
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Description: karaoke系統開發library,使用DC-DSP Filter對Audio做採集和特效,相當重量及的Sample-karaoke system development library, using DC-Audio DSP Filter to do Acquisition and special effects, a considerable weight and the Sample Platform: |
Size: 244736 |
Author:starlife |
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Description: * 书 名：《DirectShow开发指南》
* 作 者： 陆其明（著） 金邦飞（审校）
* 内容提要：
本书以DirectX SDK 9.0版为蓝本，涉及的内容几乎涵盖了在Windows平台上使
用DirectShow进行C++编码的方方面面。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如音视频采集、数码摄像机的支持、非线性
编辑等应用。第3部深入分析了DirectShow SDK提供的部分典型源代码例子。第
4部分结合作者个人的一些开发实践，通过案例和开放源码分析，进一步介绍
DirectShow的实务应用。-* Title : "DirectShow Development Guide"* Author : Lu Qiming (a) JIN Bang fly (Revision)* Synopsis : The book is DirectX 9.0 SDK version, modeled The contents cover almost the Windows platform using DirectShow coding for the C side aspect. The book is divided into four parts. Part 1 introduced DirectShow basic knowledge. Part 2 focused on the development of Filter, and DirectShow application development, including the current very popular, such as audio and video collection, digital cameras, nonlinear editing application. No. 3-depth analysis of the DirectShow SDK source code for some typical examples. Part 4 of the individual authors combine some development practice through case and open-source analysis, further information on the practical application of DirectShow. Platform: |
Size: 398336 |
Author:long |
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Description: Really Great DirectShow Project by
"Jesse Stone" from Taiwan. It Include
1.AC3 Filter
2.Mpeg1 Splitter for Video and Audio
two pins
3.Directshow BaseClass for debug using
4.Standard mpge1 file that you
correctly try it in the GraphEdit
5.I am sure the project is easy for
using and I am sure it can work 100％,Really Great DirectShow Project by Platform: |
Size: 8269824 |
Author:小誠 |
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Description: 电话通信中我们常会遇到回音,如何消除?本程序用自适应滤波器消除回音的方法来处理.(使用于音频信号处理)-Telephone communication, we often encounter echo, how to eliminate? This procedure using adaptive filter method to eliminate the echo treatment. (For use in audio signal processing) Platform: |
Size: 4096 |
Author: |
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Description: * 内容提要：
本书以DirectX SDK 9.0版为蓝本，涉及的内容几乎涵盖了在Windows平台上使用DirectShow进行C++编码的方方面面。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如音视频采集、数码摄像机的支持、非线性编辑等应用。第3部深入分析了DirectShow SDK提供的部分典型源代码例子。第4部分结合作者个人的一些开发实践，通过案例和开放源码分析，进一步介绍DirectShow的实务应用。-* Summary: The book version of DirectX SDK 9.0 is based on, involving almost covered in the Windows platform using DirectShow for C++ All aspects of coding. Book is divided into four parts. Part 1 details the basic knowledge of DirectShow. Part 2 focused on the development of Filter, as well as the DirectShow application development, including the very popular collection, such as audio and video, digital camera support, non-linear editing applications. No. 3-depth analysis of some of the DirectShow SDK source code for a typical example. Part 4 the author combines some of the development of personal practice, through case analysis and open source, and further introduce the practical application of DirectShow. Platform: |
Size: 41665536 |
Author:马敏 |
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Description: 用MATLAB实现了该算法，并对所实现方法进行了验证，在步骤（3）中则给出了求出滤波器类型及参数的方案，并进行验证，在步骤（4）中通过MATLAB的AWGN函数等相关知识实现了AWGN信道的仿真。
本文首先介绍数字水印和音频水印的相关概念和发展概况，接着介绍了完成仿真工作所需的基础知识，之后提出DA/AD变换的仿真实现，-Using MATLAB realization of the algorithm, and the realization method of authentication, in step (3) in the given filter type and parameters to derive the program and verify, in step (4) through MATLAB-AWGN function and other relevant knowledge to achieve the AWGN channel simulation. This paper first introduces the digital watermarking and audio watermarking-related concepts and developments, and then introduced a complete simulation of the basic knowledge required, after the DA/AD conversion of the simulation, Platform: |
Size: 3072 |
Author:byleimin |
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Description: After the run of the "MyEqualizer.m" file a graphical user interface appear. By pressing the "Load" we can load audio signal (type .wmv). After we can model the frequency behavior of the FIR filter that is used to filter that signal. After we can listen the filtred audio signal by pressing the "Play " button. In the GUI M-file I m using three importanat function: [ ]=coeff() that generate cell contain the coefficient of the equalizer, equalizer_plot() who plot the frequency behavior of equalizer and the equalizer_play who filter and play the filtered audio signal. Platform: |
Size: 1456128 |
Author:Nafiou |
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Description: 用Matlab实现的音频FIR滤波，提供音频文件实例，效果很好-Implemented using Matlab audio FIR filter, to provide examples of audio files with good results Platform: |
Size: 301056 |
Author:Howard |
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Description: 采用FFT和低通滤波器的两种语音除噪算法比较，MATLAB编程，附效果图-Using FFT and low pass filter audio noise canceling algorithm for the two compare, MATLAB programming, with a results map Platform: |
Size: 49152 |
Author:陈洪 |
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Description: Used on cellular phones to filter out
background noise. Measure the signal to noise ratio of an audio input in realtime. Pass speech but give silence where the input is noise (hiss, snap crackle and pop).
Includes a tutorial on how to use statistics (correlation coefficient) to calculate the signal to noise ratio.
Realtime implementation using Texas Instruments TMS320C5510 DSK and TMS320C6713 DSK. Code mostly in C, except where it would be too slow and assembly language is used. Platform: |
Size: 1053696 |
Author:ivan |
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Description: Digital filtering algorithms are most commonly implemented
using general purpose digital signal processing chips for audio
applications, or special purpose digital filtering chips and application-
specific integrated circuits (ASICs) for higher rates. This
paper describes an approach to the implementation of digital filter
algorithms based on field programmable gate arrays (FPGAs). Platform: |
Size: 318464 |
Author:JAVAD |
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Description: 音频信号的水印设计，以音频信号为载体，将秘密信号嵌入载体音频信号当中，
实现音频信号的水印。采用变换域小波变换（DWT）的技术，将音频信号进行小波变换，提取出低频分量，然后将水印嵌入到音频信号的低频当中。实验当中先在MATLAB 上面进行了仿真实验，并且通过加入噪声以及滤波等进行干扰以此检测了算法的透明性和健壮性，然后利用C 语言进行了测试，最后经
过一些修改，例如加入定时器以及进行中断的处理和加入缓冲在DSP 板子上实现了加入水印的音频声音的播放。-Audio signal watermark design to audio signal as the carrier, the carrier signal is embedded in the audio signal.
Realize audio signal watermark. The transform domain wavelet transform (DWT) technology, the audio signal wavelet transform to extract the low frequency component, and then the watermark is embedded in the audio signal of low frequency. Experiments on first in MATLAB, the simulation experiment and by adding noise and filter and so on interference to test the algorithm the transparency and robustness, and by using the C language were tested, and the
Some modification, such as join timer and interrupt handling and join buffer in DSP board to add watermark audio sound broadcast.
Platform: |
Size: 300032 |
Author:江晨城 |
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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 Platform: |
Size: 297984 |
Author:南宫崔浩 |
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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. Platform: |
Size: 394240 |
Author:张 |
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Search - audio using filter