wav

How to convert .wav to .mp3 file in php

怎甘沉沦 提交于 2019-11-29 14:36:05
I am currently developing a website using the CodeIgniter framework and in one of the Process I need to download a .wav music file and convert it to .mp3 file. All the coding needs to be done only in PHP and I cannot use Ruby or java to do so. And I am currently hosting on a shared env i.e. Dreamhost. So please let me know how to work out this problem. Use FFMPEG . There is a PHP extension for it , but I've found better success just calling exec() . Most hosts I have used (even shared hosting) will install it if it isn't already available. Edit: Dreamhost provides this for you. See this: http:

Why are an integers bytes stored backwards? Does this apply to headers only?

五迷三道 提交于 2019-11-29 12:14:19
问题 I'm currently trying to decipher WAV files. From headers to the PCM data. I've found a PDF (http://www.tdt.com/T2Support/technical_notes/tn0132.pdf) detailing the anatomy of a WAV file, and I've been able to extract and make sense of the appropriate header data using Ghex2. But my questions are: Why are the integers bytes stored backwards? I.e. dec. 20 is stored as 0x14000000 instead of 0x00000014. Are the integers of the PCM data also stored backwards? 回答1: WAV files are little-endian (least

How to split a wav file into smaller chunks using Java?

那年仲夏 提交于 2019-11-29 07:19:34
I have a very huge WAV file, about 100MB in size. I would like to use Java to read this wav file and split it into smaller chunks for every 2 seconds of audio. Is it possible to do this in Java? Could you please suggest me an API with which I can achieve it? Thanks in advance, Snehal You can use AudioInputStream and its AudioFileFormat member (which contains an AudioFormat instance) to know what to write (format, sample rate), you can use AudioSystem to write it. Based on the sample rate of the format you can find out how many bytes of audio are 2 seconds, and go on a loop of reading that many

Playing a sound from a wave form stored in an array

时光毁灭记忆、已成空白 提交于 2019-11-29 07:17:27
I'm currently experimenting with generating sounds in Python, and I'm curious how I can take a n array representing a waveform (with a sample rate of 44100 hz), and play it. I'm looking for pure Python here, rather than relying on a library that supports more than just .wav format. You should use a library. Writing it all in pure python could be many thousands of lines of code, to interface with the audio hardware! With a library, e.g. audiere, it will be as simple as this: import audiere ds = audiere.open_device() os = ds.open_array(input_array, 44100) os.play() There's also pyglet, pygame,

What is a channel in a .wav file format?Do all channels play simultaneaously when a wav file is played?

南楼画角 提交于 2019-11-29 03:50:54
I read about.wav file format by googling,all I could figure was that Frames are made of samples(of some defined bit depth) and a wav stereo file has a multiple of something called channels.... The confusion is whether a channel is made up of frames? Do all channels play along when I play some audio file? If a channel is made up of frames,are all channels equal in length(bit wise)? Please answer if someone can,I have to display each channel separately when playing a wav file in waveform In each frame in wav there are channels. If you have stereo sound, then each frame contains two samples (left

Change the volume of a wav file in python

﹥>﹥吖頭↗ 提交于 2019-11-29 03:46:42
问题 I have a 2 seconds 16bit single channel 8khz wav file and I need to change its volume. It should be quite straightforward, because changing the volume is the same as changing the amplitude of the signal, and I just need to attenuate it, that is to multiply it for a number between 0 and 1. But it doesn't work: the new sound is lower but VERY full of noise. What am I doing wrong? Here is my code: import wave, numpy, struct # Open w = wave.open("input.wav","rb") p = w.getparams() f = p[3] #

Using pyDub to chop up a long audio file

本秂侑毒 提交于 2019-11-29 03:25:39
问题 I'd like to use pyDub to take a long WAV file of individual words (and silence in between) as input, then strip out all the silence, and output the remaining chunks is individual WAV files. The filenames can just be sequential numbers, like 001.wav, 002.wav, 003.wav, etc. The "Yet another Example?" example on the Github page does something very similar, but rather than outputting separate files, it combines the silence-stripped segments back together into one file: from pydub import

Filtering a wav file using python

孤人 提交于 2019-11-29 01:00:30
问题 So i recently successfully built a system which will record, plot, and playback an audio wav file entirely with python. Now, I'm trying to put some filtering and audio mixing in between the when i record and when i start plotting and outputting the file to the speakers. However, i have no idea where to start. Right now I'm to read in a the intial wav file, apply a low pass filter, and then re-pack the newly filtered data into a new wav file. Here is the code i used to plot the initial data

Python: write a wav file into numpy float array

点点圈 提交于 2019-11-29 00:08:36
ifile = wave.open("input.wav") how can I write this file into a numpy float array now? Joran Beasley >>> from scipy.io.wavfile import read >>> a = read("adios.wav") >>> numpy.array(a[1],dtype=float) array([ 128., 128., 128., ..., 128., 128., 128.]) typically it would be bytes which are then ints... here we just convert it to float type you can read about read here https://docs.scipy.org/doc/scipy/reference/tutorial/io.html#module-scipy.io.wavfile 来源: https://stackoverflow.com/questions/16778878/python-write-a-wav-file-into-numpy-float-array

How to edit raw PCM audio data without an audio library?

三世轮回 提交于 2019-11-28 23:55:36
I'm interested in precisely extracting portions of a PCM WAV file, down to the sample level. Most audio modules seem to rely on platform-specific audio libraries. I want to make this cross platform and speed is not an issue, are there any native python audio modules that can do this? If not, I'll have to interpret the PCM binary. While I'm sure I can dig up the PCM specs fairly easily, and raw formats are easy enough to walk, I've never actually dealt with binary data in Python before. Are there any good resources that explain how to do this? Specifically relating to audio would just be icing.