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: 63725 |
Author:徐勇 |
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Description: 采用TMS320VC5402实现一个16阶低通滤波器，音频接口采用的是AD50，汇编语言编写-TMS320VC5402 used to achieve a 16 band low-pass filter, Audio interface is AD50, the compilation of language Platform: |
Size: 25344 |
Author:潘枫 |
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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 |
Hits:
Description: 采用TMS320VC5402实现一个16阶低通滤波器，音频接口采用的是AD50，汇编语言编写-TMS320VC5402 used to achieve a 16 band low-pass filter, Audio interface is AD50, the compilation of language Platform: |
Size: 30720 |
Author:潘枫 |
Hits:
Description: 大功率音频放大器的设计，基于低通滤波器，H桥PWM技术-High-power audio amplifier design, based on the low-pass filter, H bridge PWM technology Platform: |
Size: 3111936 |
Author:Yang hong |
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Description: 一个音频滤波器的C++类，可以实现低通，高通滤波，并可以指定频率，带宽和增益值-An audio filter C++ Category, you can realize low-pass, high pass filter, and can specify the frequency, bandwidth and gain value Platform: |
Size: 2048 |
Author:cxcxcx |
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Description: 一个用C语言程序实现读写语音文件的语音信号低通滤波例子-A C language program used to read and write audio files of voice signal low-pass filter example Platform: |
Size: 1024 |
Author:董飞 |
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Description: matlab的滤波器，有菜单选择，高通，带通和低通，并且内有一个音频文件-matlab filter, there is the menu selection, high-pass, band-pass and low pass, and inside there is an audio file Platform: |
Size: 135168 |
Author:zengxubin |
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Description: 1）设计Butterworth型音频抗混叠滤波器；
2）参数：
下通带频率300Hz；上通带频率3400Hz；
下阻带频率280Hz；上阻带频率3600Hz；
通带最大衰减0.3dB；
阻带最小衰减40dB；
3）采用一低通滤波器和一高通滤波器级联；
4）分别确定LPF和HPF的性能指标；
5）求出两滤波器的系统函数和频率响应，并画出其幅频特性曲线；
6）求整个滤波器的系统函数和频率响应，并画出其幅频特性曲线。-1) the design of Butterworth-type audio anti-aliasing filter 2) parameters: passband frequency of 300Hz on pass-band frequency 3400Hz under the stop-band frequency of 280Hz on stop-band frequency 3600Hz passband maximum attenuation of 0.3dB minimum stopband attenuation of 40dB 3) the use of a low-pass filter and a high-pass filter cascade 4) separately to determine the performance of LPF and HPF target 5) to derive two-filter system function and frequency response, and draw its increase frequency characteristic curve 6) for the entire filter system function and frequency response, and draw its amplitude-frequency characteristic curve. Platform: |
Size: 15360 |
Author:hsf |
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Description: 先采集一单声道音频信号（.wav）并用WAVREAD文件采样读取，并对其进行频谱分析。分别用窗函数法和双线性变换法设计低通、高通、带通三种FIR滤波器和IIR滤波器。用M文件使信号通过滤波器并对输出信号进行时域和频域分析。-First acquisition of a mono audio signal (. Wav) files with sampling WAVREAD read, and their spectral analysis. Window function, respectively and the design of bilinear transform low-pass, high pass, band-pass FIR filters and IIR three filters. M document with the signal through the filter and the output signal in time domain and frequency domain analysis. Platform: |
Size: 197632 |
Author:宋立泉 |
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Description: An Introduction To Compressive Sampling-Conventional approaches to sampling signals or images follow Shannon’s celebrated theorem: the sampling rate must be at least twice the maximum frequency present in the signal (the so-called Nyquist rate). In fact, this
principle underlies nearly all signal acquisition protocols used in consumer
audio and visual electronics, medical imaging devices, radio receivers, and
so on. (For some signals, such as images that are not naturally bandlimited, the sampling
rate is dictated not by the Shannon theorem but by the desired temporal or spatial
resolution. However, it is common in such systems to use an antialiasing low-pass filter
to bandlimit the signal before sampling, and so the Shannon theorem plays an implicit
role.) In the field of data conversion, for example, standard analog-to-digital converter
(ADC) technology implements the usual quantized Shannon representation: the signal is
uniformly sampled at or above the Nyquist rate Platform: |
Size: 1144832 |
Author:yjsdqq |
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Description: 音频信号处理，主要完成对于音频信号的滤波，内含自己编写的低通、高通和带组数字滤波器源码-Audio signal processing, mainly to complete the filtering for audio signals, including the preparation of their own low-pass, high-pass digital filter and source code with group Platform: |
Size: 827392 |
Author:mimi |
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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: 一个GUI，低通滤波器，可以滤除音频信号的噪声部分，参数可以自己设定。基本原理采用多种不同滤波器进行比较-A GUI, low-pass filter, to filter out the noise part of the audio signal, the parameter can be set. The basic principles of using a variety of different filters are compared Platform: |
Size: 3072 |
Author:孙涛 |
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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 low pass filter