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Description: 通过改变频率来改变音乐的音调，升调或降调都可以。另外还有移相（phaser.c)及反相(reverse.c)两个源程序文件-by changing the frequency to change the tone music, or transfer or demotion can. Another phase shifter (phaser.c) and RP (reverse.c) 2 source files
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Size: 12125 |
Author: 张承云 |
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Description: 语音合成程序！psalo频域基音同步叠加方法。它首先对原始语音信号进行短时频域变换，得到短时谱和短时谱包络。短时谱除以短时谱包络得到声源短时谱，对声源短时谱的实部和虚部分别进行线性插值，就可以达到改变语音信号基频的目的，然后再进行频域反变换，可得到变换后的短时语音信号。短时谱包络部分也可以独立改变，以达到改变音色的目的。-speech synthesis procedures! Psalo frequency domain pitch synchronous superposition method. It is first of the original speech signal for short-time frequency domain transform, to be short-term and short-term spectral envelope spectrum. Short-term spectrum divided by the short-time spectral envelope of the sound source to be short-term spectrum of the sound spectrum of short-term real and imaginary parts of linear interpolation, they could change the voice-frequency signals to the end, then we will proceed to frequency-domain transform, Transform available after the short speech signal. Short-term spectral envelope can be independent of changes to achieve the objective of changing colors.
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Size: 2553 |
Author: hzh |
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Description: 通过改变频率来改变音乐的音调，升调或降调都可以。另外还有移相（phaser.c)及反相(reverse.c)两个源程序文件-by changing the frequency to change the tone music, or transfer or demotion can. Another phase shifter (phaser.c) and RP (reverse.c) 2 source files
Platform: |
Size: 12288 |
Author: 张承云 |
Hits:
Description: 语音合成程序！psalo频域基音同步叠加方法。它首先对原始语音信号进行短时频域变换，得到短时谱和短时谱包络。短时谱除以短时谱包络得到声源短时谱，对声源短时谱的实部和虚部分别进行线性插值，就可以达到改变语音信号基频的目的，然后再进行频域反变换，可得到变换后的短时语音信号。短时谱包络部分也可以独立改变，以达到改变音色的目的。-speech synthesis procedures! Psalo frequency domain pitch synchronous superposition method. It is first of the original speech signal for short-time frequency domain transform, to be short-term and short-term spectral envelope spectrum. Short-term spectrum divided by the short-time spectral envelope of the sound source to be short-term spectrum of the sound spectrum of short-term real and imaginary parts of linear interpolation, they could change the voice-frequency signals to the end, then we will proceed to frequency-domain transform, Transform available after the short speech signal. Short-term spectral envelope can be independent of changes to achieve the objective of changing colors.
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Size: 2048 |
Author: hzh |
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Description: 用短时傅立叶变换方法实现歌曲变调功能，变调最小步长为1个半音-Using short-time Fourier transform method sandhi features songs, changing the minimum step size for a chromatic
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Size: 8192 |
Author: Jack |
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Description: 通过驱动实验板上的蜂鸣器发声，实现音乐的播放。
2.从TABLE中取出播放音乐的音调（1、2....7）通过改变该表中的值即可实现播放不同的音乐。
3.从TABLE_YP中取出播放音乐的音频，即该音调持续的时间。
4.调整DELAY延时的长短即可实现音乐播放的快慢-Through the drive plate experiment audible buzzer, realize music player. 2. TABLE removed from the pitch to play music (1,2 .... 7) by changing the values in the table to realize a different music player. 3. TABLE_YP removed from the audio playback of music, that is, the tone duration. 4. Adjust the length of delay DELAY music player can realize the processing speed of
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Size: 3072 |
Author: chuanji |
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Description: sample changing for sound pitch and tempo.
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Size: 66560 |
Author: carl simpson |
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Description: 语音合成程序源码！psalo频域基音同步叠加方法。它首先对原始语音信号进行短时频域变换，的到短时谱与短时谱包络。短时谱除以短时谱包络的到声源短时谱，对声源短时谱的实部与与虚部分别进行线性插值，就能达到改变变语音信号基频的目的，然后再进行频域反变换，可的到变换后的短时语音信号。短时谱包络部分也能独立改变，以达到改变音色的目的。
-Voice synthesis program source! psalo frequency domain pitch synchronous superposition method. It was first carried out on the original speech signal a short-time frequency-domain transform, to the short-time spectrum and short-time spectrum envelope. The short-time spectrum divided by the short-time spectrum envelope of the short time spectrum of the sound source, the short time spectrum of the real part of the sound source, and the imaginary parts of the linear interpolation, can achieve the purpose of changing the fundamental frequency of the alternating speech signal, and then then the inverse transform of the frequency domain, can be to the short-time speech signal after conversion. The short-time spectral envelope section can be varied independently, in order to achieve the purpose of changing the tone.
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Size: 2048 |
Author: moderate |
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Description: 语音信号进行分析、处理和设计。用MATLAB做变声器的原理是通过改变输入声音频率，进而改变声音的音色、音调，使输出声音在感官上与原声音不同。变声器是借助对声音音色和音调的双重复合改变，实现输出声音的改变。通过自己发声，共振峰频率的改变是基于重采样实现的-Speech signal analysis, processing and design. MATLAB do voice-changer works by changing the input frequency of the sound, and then change the sound of the tone, pitch, so that the output audio with the original sound in different senses. Voice Changer is the use of sound timbre and tone of the dual composite changes, realize the output sound changes. Through their own sound, formant frequency change is implemented based on resampling
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Size: 10240 |
Author: 刘安兰 |
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Description: 语音信号进行分析、处理和设计。用MATLAB做变声器的原理是通过改变输入声音频率，进而改变声音的音色、音调，使输出声音在感官上与原声音不同。变声器是借助对声音音色和音调的双重复合改变，实现输出声音的改变。通过自己发声，共振峰频率的改变是基于重采样实现的-Speech signal analysis, processing and design. MATLAB do voice-changer works by changing the input frequency of the sound, and then change the sound of the tone, pitch, so that the output audio with the original sound in different senses. Voice Changer is the use of sound timbre and tone of the dual composite changes, realize the output sound changes. Through their own sound, formant frequency change is implemented based on resampling
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Size: 7168 |
Author: 刘安兰 |
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Description: 使用电脑系统蜂鸣器的API模拟演奏音乐和图片超链接，
通过改变蜂鸣器工作时频率的变化来模拟音调的变化，
并附带图片超链接写法。-Using a computer simulation of the API system buzzer playing music and pictures hyperlinks,
By changing the frequency variation buzzer to simulate the change in pitch,
And comes with pictures hyperlink wording.
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Size: 5497856 |
Author: feng |
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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.
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Size: 2048 |
Author: japaoli |
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