waveform

I was wondering if anyone has any suggestions on how to go about rendering a waveform of an audio file. I wold like to enable the user to set an in and out point of an audio track and I need to have a waveform so you can see where to put the points. Are there any libraries available for this or does it need to be a completely custom solution? Is it even called a waveform ? Maybe there's a better word for it so I can do some more searching. TIA ! Nick Haddad I'm reposting my answer from this question , since it applies here as well: When displaying an audio waveform, you will want to do some

I'm writing an audio waveform editor in Cocoa with a wide range of zoom options. At its widest, it shows a waveform for an entire song (~10 million samples in view). At its narrowest, it shows a pixel accurate representation of the sound wave (~1 thousand samples in a view). I want to be able to smoothly transition between these zoom levels. Some commercial editors like Ableton Live seem to do this in a very inexpensive fashion. My current implementation satisfies my desired zoom range, but is inefficient and choppy. The design is largely inspired by this excellent article on drawing waveforms

one of the default live wallpapers that came with my phone was a wallpaper that displayed the wave form of music playing in the background in real time. I was wondering how one could go about doing this. It seems I should somehow access the streaming data of the background music and compute the samples in real time, but I have no idea how to get the streaming data. I think I can display the wave form by following this tutorial http://codeidol.com/java/swing/Audio/Build-an-Audio-Waveform-Display/ , but I'm not exactly sure how to do it. Help would be greatly appreciated/ Christopher Souvey

I am trying to implement a low pass filter in Java. My requirement is very simple,I have to eliminate signals beyond a particular frequency (Single dimension). Looks like Butterworth filter would suit my need. Now the important thing is that CPU time should be as low as possible. There would be close to a million sample the filter would have to process and our users don't like waiting too long. Are there any readymade implementation of Butterworth filters which has optimal algorithms for filtering. I have a page describing a very simple, very low-CPU low-pass filter that is also able to be

I'm writing a c program to generate a sinusoidal wave that slowly ramps up frequency from f1 to f2 for a giving time interval. I have written this c program to ramp the frequency from 0 to 10 Hz but the problem is that the frequency changes after completion of 360 degrees. If I try to change the frequency between 0 and 360 degree that the transition is not smooth and it is abrupt. This is the equation the sin that I have used y = Amplitude*sin(freq*phase) int main(int argc, char *argv[]) { double y, freq,phase; int count; // for convenience of plotting in matlab so all the waves are spread on

I reading song from iPod library using assetUrl (in code it named audioUrl) I can play it many ways, I can cut it, I can make some precessing with this but... I really don't understand what I gonna do with this CMSampleBufferRef to get data for drawing waveform! I need info about peak values, how I can get it this (maybe another) way? AVAssetTrack * songTrack = [audioUrl.tracks objectAtIndex:0]; AVAssetReaderTrackOutput * output = [[AVAssetReaderTrackOutput alloc] initWithTrack:songTrack outputSettings:nil]; [reader addOutput:output]; [output release]; NSMutableData * fullSongData = [