wav

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] # number of frames s = w.readframes(f) w.close() # Edit s = numpy.fromstring(s, numpy.int16) * 5 / 10 # half

I'm trying to process audio data. I'm working with Java. I've extracted the audio data to an array. Now I should pass N data samples to a function that calculates the Discrete Fourier Transform (or Fast Fourier Transform, which is more efficient). I've read documentation but I'm getting more and more confused. What I'm trying to calculate is the magnitude spectrum (|X(k)|). Can anyone help me? Thanks Robert Harvey Richard G. Baldwin has a number of very good articles on Fast Fourier Transform algorithms in Java on the Developer.com website. In particular, the following articles should prove to

I have to downsample a wav file from 44100Hz to 16000Hz without using any external python libraries, so preferably wave and/or audioop . I tried just changing the wav files framerate to 16000 by using setframerate function but that just slows down the entire recording. How can I just downsample the audio file to 16kHz and maintain the same length of the audio? Thank you very much in advance You can use Librosa's load() function, import librosa y, s = librosa.load('test.wav', sr=8000) # Downsample 44.1kHz to 8kHz The extra effort to install Librosa is probably worth the peace of mind. Pro-tip:

package soundTest; import java.applet.*; import java.net.*; import javax.swing.*; import javax.sound.sampled.*; import java.io.*; import java.util.*; public class SoundTest { public static void main(String[] args) throws Exception { try { AudioClip clip1 = Applet.newAudioClip(new URL(new File("E0.wav").getAbsolutePath())); clip1.play(); } catch (MalformedURLException murle) { System.out.println(murle); } URL url = new URL( "http://www.mediafire.com/listen/tok9j9s1hnogj1y/downloads/E0.wav"); Clip clip = AudioSystem.getClip(); AudioInputStream ais = AudioSystem.getAudioInputStream(url); clip

I'm writing an app where I record audio and upload the audio file over the web. In order to speed up the upload I want to start uploading before I've finished recording. The file I'm creating is a WAV file. My plan was to use multiple data chunks. So instead of the normal encoding (RIFF, fmt , data) I’m using (RIFF, fmt , data, data, ..., data). The first issue is that the RIFF header wants the total length of the whole file, but that is of course not known when streaming the audio (I’m now using an arbitrary number). The other problem is that I'm not sure if it's valid since Audacity doesn't