wav

I have a web page which decodes wave files for certain reasons. Chrome and Safari seem to work fine. Firefox occasionally is unable to decode the file and gives the error: "The buffer passed to decodeAudioData contains invalid content which cannot be decoded successfully." I have created a jsfiddle which illustrates the issue: var audioCtx = new (window.AudioContext || window.webkitAudioContext)(); var source; function getData() { source = audioCtx.createBufferSource(); request = new XMLHttpRequest(); request.open('GET', 'https://mpclubtest.s3.amazonaws.com/Malice_Bass.wav', true); request

Hi I need to downsample a wav audio file's sample rate from 44.1kHz to 8kHz. I have to do all the work manually with a byte array...it's for academic purposes. I am currently using 2 classes, Sink and Source, to pop and push arrays of bytes. Everything goes well until I reach the part where I need to downsample the data chunk using a linear interpolation. Since I'm downsampling from 44100 to 8000 Hz, how do I interpolate a byte array containing something like 128 000 000 bytes? Right now I'm popping 5, 6 or 7 bytes depending on i%2 == 0, i%2 == 1 and i%80 == 0 and push the average of these 5,

I am trying to extract amplitude array from an audio file(WAV file). I will be using this amplitude array to plot amplitude vs time graph for the given wav file. I am able to plot the graph myself but does not know how to extract the amplitude from given audio(wav) file in java? Here is a helper class that you can use. The getSampleInt() method is what you need to get the amplitude: File file = ...; WavFile wav = new WavFile(file); int amplitudeExample = wav.getSampleInt(140); // 140th amplitude value. for (int i = 0; i < wav.getFramesCount(); i++) { int amplitude = wav.getSampleInt(i); //

I am writing a C program for editing a Wav audio file. I have loaded all file datas in an array of unsigned integer values (UINT16_T). Now, i would like to reduce the volume of the file. I thought it was enough to decrease the value (of a certain percentage) of the single values. But if i do that, i obtain an audio file with noise (I think I understand is called "static" or "click noise") Why? Which is the right procedure? Thank You! This is the piece of code affected: FILE* fp; FILE* fp2; /*Size of my file*/ #define BUFFER_SIZE 28242852 /*Array with file data*/ unsigned char *buffer; /*Array

Since I've posted this question , I've been trying to write a valid WAV file myself from raw PCM data. I've managed to write the FLAC converter (tested and works), but it does not encode the WAV files I've been writing. I'm not sure what I'm doing wrong. I've been scouring the internet looking at other individuals source code and comparing it to my own, but I still can't get it to work. Here is the whittled down source code (sorry it's still a bit long, it takes a bit of code to record to a .wav on my own): // Compile with "g++ test.ccp -o test -lasound" // Use the newer ALSA API #define ALSA