Description: 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可以用于任何用途。但是发布源代码时必须包括版权声明，发布二进制代码必须声明作者。-Sox is the most famous Open Source voice file format conversion tools. Has been widely transplant Dos, windows, OS2, Sun, Next, Unix, Linux and other operating system platforms. Sox was founded by Lance Norskog, later many developers have gradually improved. can now support multiple audio files formats and audio processing effects. Basically common formats can support. More useful is that the Sox to a voice filter, the sampling frequency conversion, This voice of those engaged in the development or maintenance of platforms friends very useful. Of course, the Sox also including some inside DSP Algorithm, interested friends can be downloaded to go back and study. Sox can be used for any other purpose. However, the distribution of source code must include the copyright statement released bina Platform: |
Size: 312208 |
Author:李增军 |
Hits:
Description: 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可以用于任何用途。但是发布源代码时必须包括版权声明，发布二进制代码必须声明作者。-Sox is the most famous Open Source voice file format conversion tools. Has been widely transplant Dos, windows, OS2, Sun, Next, Unix, Linux and other operating system platforms. Sox was founded by Lance Norskog, later many developers have gradually improved. can now support multiple audio files formats and audio processing effects. Basically common formats can support. More useful is that the Sox to a voice filter, the sampling frequency conversion, This voice of those engaged in the development or maintenance of platforms friends very useful. Of course, the Sox also including some inside DSP Algorithm, interested friends can be downloaded to go back and study. Sox can be used for any other purpose. However, the distribution of source code must include the copyright statement released bina Platform: |
Size: 312320 |
Author:李增军 |
Hits:
Description: 语音降噪。从Codec AD50采集话筒语音，通过DSP TMS320vc5402处理，在送到AD50输出降噪后语音，涉及加汉宁窗，切比雪夫滤波器，快速傅立叶变换和反FFT，有声无声判断谱分解，谱合成等功能-Voice noise. Codec AD50 collected from the microphone voice, through the DSP TMS320vc5402 treatment, in AD50 to the output noise after the voice, involve Hanning windows, Chebyshev filter, Fast Fourier Transform and anti-FFT, audio silent judge spectral decomposition, spectral synthesis and other functions Platform: |
Size: 44032 |
Author:黄胜华 |
Hits:
Description: 利用matlab实现语音输入、8bit PCM音频截短、滤波输出。用于将大于8bit的PCM语音截短为8bit的PCM语音。-Matlab realize the use of voice input, 8bit PCM audio truncated, filter output. Will be greater than for the PCM voice 8bit to 8bit truncated the PCM voice. Platform: |
Size: 1497088 |
Author:hexd |
Hits:
Description: 录制一段个人的语音信号，并对录制的信号进行采样；画出采样后语音信号的时域波形和频谱图；给定滤波器的性能指标，采用双线性变换法和窗函数法设计滤波器，并画出滤波器的频率响应；然后用自己设计的滤波器对采集的信号进行滤波，画出滤波后信号的时域波形和频谱，并对滤波前后的信号进行对比，分析信号的变化；回放语音信号；最后，设计一个信号处理系统界面。-Record a personal voice signal, and recording the signal is sampled draw sampled speech signal time-domain waveform and frequency spectrum for a given filter performance, the bilinear transformation and design of filter window functions and draw the frequency response filter then filter of their own design collection of the signal filtering, to draw the filtered signal in time domain waveforms and frequency spectrum, and filtering the signal before and after comparison, analysis of signal changes playback audio signals Finally, a signal processing system interface design. Platform: |
Size: 6144 |
Author:liudan |
Hits:
Description: ISD4004系列工作电压3V,单片录放时间8至16分钟,音质好,适用于移动电话及其他便携式电子产品中。芯片采用CMOS技术,内含振荡器、防混淆滤波器、平滑滤波器、音频放大器、自动静噪及高密度多电平闪烁存贮陈列。芯片设计是基于所有操作必须由微控制器控制,操作命令可通过串行通信接口(SPI或Microwire)送入。芯片采用多电平直接模拟量存储技术, 每个采样值直接存贮在片内闪烁存贮器中,因此能够非常真实、自然地再现语音、音乐、音调和效果声,避免了一般固体录音电路因量化和压缩造成的量化噪声和"金属声"。采样频率可为 4.0,5.3,6.4,8.0kHz,频率越低,录放时间越长,而音质则有所下降,片内信息存于闪烁存贮器中,可在断电情况下保存100年(典型值),反复录音10万次。-ISD4004 series voltage 3V, single playback time of 8-16 minutes, good sound quality for mobile phones and other portable electronic products. Chip using CMOS technology, contains the oscillator, anti-aliasing filter, smoothing filter, audio amplifier, automatic squelch and high-density multi-level flash storage display. Chip design is based on all operations must be controlled by a micro-controller, the operation command through the serial communication interface (SPI or Microwire) into. Chip multi-level direct analog storage technology, the value of each sample directly stored in the flash memory chip, and therefore can be very real, natural reproduction of voice, music, tone and effect sound, solid recording circuit to avoid the general quantization and compression caused by quantization noise and the "metallic sound." Sampling frequency can 4.0,5.3,6.4,8.0 kHz, lower frequency, the longer the recorder, while the sound quality is down, piece of information stored in the flash memory Platform: |
Size: 2048 |
Author:小老鼠 |
Hits:
Description: 语音文件中有单频噪声，设计高频滤波器和低频滤波器进行语音滤波-Audio files with a single frequency noise, high frequency filters and low-frequency filter design for voice filtering Platform: |
Size: 491520 |
Author:郑倩 |
Hits:
Description: 题目要求：
1.录制一段自己的语音信号，并对录制的信号进行采样；
2.画出采样后的语音信号的时域波形和频谱图；
3.给定滤波器的性能指标，采用窗函数法和双线性变换法设计滤波器，
并划出滤波器的频域响应；
4.用该滤波器对采集的信号进行滤波，画出滤波后信号的时域波形和频谱，
并对滤波前后的信号进行对比，分析信号的变化；
5.回放语音信号；
6.设计一个信号处理系统界面。-Subject to: 1. Recorded his voice for some signal, and recording the signal is sampled 2. Draw the sampled speech signal time-domain waveform and frequency spectrum 3. For a given filter performance, using window function bilinear transformation design method and filter, and set aside the filter frequency response 4. with the filter on the collection of signal filtering, to draw the filtered signal in time domain waveforms and frequency spectrum, and before and after filtering The signal contrast, analysis of signal changes 5. playback of audio signals 6. to design a signal processing system interface. Platform: |
Size: 2048 |
Author:pengtianwei |
Hits:
Description: 1.录制一段自己的语音信号,对录制的信号进行采样 画出采样后语音信号的时域波形和频谱图 给定滤波器的性能指标,采matlab设计数字滤波器,并画出滤波器的频率响应 然后用自己设计的滤波器对采集的信号进行滤波,画出滤波后信号的时域波形和频谱,并对滤波前后的信号进行对比,分析信号的变化 回放语音信号。2.-1. Recorded his voice for some signal, the signal is sampled on the record draw sampled speech signal time-domain waveform and frequency spectrum performance for a given filter, mining matlab digital filter design, and draw the filter frequency response then use their own filters designed to filter the collected signals, to draw the filtered time domain waveform and frequency spectrum, and comparing the signal before and after filtering, analysis of signal changes playback of audio signals. 2. Platform: |
Size: 51200 |
Author:liujia |
Hits:
Description: ISD4004 系列工作电压3V,单片录放时间8 至16 分钟,音质好,适用于移动电话及其他便携式电子产品中。芯片采用CMOS 技术,内含振荡器、防混淆滤波器、平滑滤波器、音频放大器、自动静噪及高密度多电平闪烁存贮陈列。芯片设计是基于所有操作必须由微控制器控制,操作命令可通过串行通信接口(SPI 或Microwire)送入。-The ISD4004 series voltage is 3V, the monolithic recording time to 16 minutes, good sound quality for mobile phones and other portable electronic products. Chip using CMOS technology, contains the oscillator, anti-aliasing filter, smoothing filter, audio amplifier, automatic squelch and high-density multi-level flash storage display. The chip design is based on all operations must be controlled by the microcontroller, operation command can be fed via a serial communications interface (SPI or Microwire). Platform: |
Size: 27648 |
Author:范阳阳 |
Hits:
Description:
采用winner的语音降噪实现代码，适用于语音和音频系统中，可直接调用。-Speech noise reduction using Winner filter implementation code, for voice and audio systems, can be called directly. Platform: |
Size: 1024 |
Author:梁小涛 |
Hits:
Description: Linear-Predictive-Coder
MATLAB Implementation of LPC algorithm for speech signal
# Why LPC?
In communication systems it is often necessary to transmit audio(speech) signal in compressed or encoded form because of bandwidth limitation of the channel. In this regard, ‘Linear predictive coding(LPC)’ is an effctive method of speech coding at a low bit-rate.
# Features
** Analysis/Encoding phase,Synthesis/Decoding phase.
**Human voice modelled with all-pole filter.
** LPC parameters(filter coefficients, pitch, gain etc) extraction at the decoding phase.
** Non-overlapping frames of 30 milliseconds in duration
# How To Run
** Make sure MATLAB(latest version) is installed
** Put both files(LPC.m with .mp3 file) in the same folder
** Open LPC.m file and run it.
## Comments
Different audio (.mp3) files can be coded/decoded by changing the input file name in the code. Platform: |
Size: 2048 |
Author:japaoli |
Hits:
Search - AUDIO FILTER FOR VOICE