wav

I'm trying to understand the header of a WAV file. I've opened an example file and got this: 5249 4646 e857 1400 5741 5645 666d 7420 1000 0000 0100 0200 44ac 0000 10b1 0200 0400 1000 I've been reading this data representation tutorial . I understand that 52 is one byte and represents the ASCII letter R . I understand up to the 1000 0000 . Why does that represent decimal 16? The tutorial says that the value at that position is always 0x10 . How does 1000 0000 equate to 0x10 . Also, when reading the file, will a program know whether to expect a number or ASCII? Presumably it'll check against a

I would like to generate a sine tone in php. But constructing my wav I need to give the values in bytes. I don't know how to do that: Here is the code I have: $freqOfTone = 440; $sampleRate = 44100; $samplesCount = 80000; $amplitude = 0.25 * 32768; $w = 2 * pi() * $freqOfTone / $sampleRate; //$dataArray = new $text = "RIFF" ."80036" ."WAVE" ."fmt " ."16" ."1" ."1" ."44100" ."44100" ."1" ."8" ."data" ."80000"; for ($n = 0; $n < $samplesCount; $n++) { $text .= (int)($amplitude * sin($n * $w)); } $myfile = fopen("sine.wav", "w") or die("Unable to open file!"); fwrite($myfile, $text); fclose(

I have no choice but to play wav files directly in the browser (serverside encoding to mp3 isn't an option, unfortunately.) What's the best way to do this? I'd really like to take advantage of the HTML 5 audio tag but my target audience includes many, many teens using IE6. As far as I'm aware flash isn't an option, but speedy playback really is critical. Thanks. Nowadays, the best way is probably just to use the HTML5 <audio> tag. In the past, you might have done it like this: Background: <embed src="bgsound.wav" hidden="true" autostart="true" loop="1"> On Click: <a href="success.wav">Play

Say I have a program that reads a .WAV or .AIFF file, and the file's audio is encoded as floating-point sample-values. Is it correct for my program to assume that any well-formed (floating-point-based) .WAV or .AIFF file will contain sample values only in the range [-1.0f,+1.0f]? I couldn't find anything in the WAV or AIFF specifications that addresses this point. And if that is not a valid assumption, how can one know what the full dynamic range of the audio in the file was intended to be? (I could read the entire file and find out what the file's actual minimum and maximum sample values are,

I need to extract the samples of a single channel from a wav file that will contain up to 12 (11.1 format) channels. I know that within a normal stereo file samples are interleaved, first left, and then right, like so, [1st L] [1st R] [2nd L] [2nd R]... So, to read the left channel I'd do this, for (var i = 0; i < myByteArray.Length; i += (bitDepth / 8) * 2) { // Get bytes and convert to actual samples. } And to get the right channel I'd simply do for (var i = (bitDepth / 8)... . But, what order is used for files with more than 2 channels? Microsoft have created a standard that covers up to 18