I've just come home from a studio with a freshly-mastered track, which I have on my computer as a stereo .AIFF file. I would like to submit this track to a particular compilation online, and the organizer has asked me to email him a .WAV file, so I need to convert it to .WAV format. However, I don't want to introduce any coloring or artifacts in the sound, and I'm not confident that I would be able to catch this with my ears in time, if it did happen.

I'd like the conversion to be sample-accurate, if this is possible. It seems like it should be. I've never really run into this since I switched over to OSX from Windows so I'm not familiar with the available software yet.

Is there a tool or process for OSX that is known to convert .AIFF to .WAV without changing a single sample? Which is to say, without affecting the resulting audio waveform in any way?

4 Answers 4


Audacity is free and can import and export to both AIFF and WAV. There are also various command line tools you can install and use, like ffmpeg, mplayer and sox. They are harder to install and use though.

  • Audacity's a great editor. Is there any processing that it does that I couldn't get around? For example, if I load foo.WAV into Audacity, and immediately export bar.WAV, would both files have the EXACT same audio data? This might be pedantic, but I just don't want to introduce any errors into my recording from format conversion if I don't have to. Thanks for your help :) Feb 22, 2011 at 21:51
  • 1
    @Warrior Bob: Of course it would have the same audio data. It's an editor, it only changes what you tell it to change. And you asked this in the question, I wouldn't have said Audacity unless it fulfilled your requirements. Feb 23, 2011 at 9:13
  • That makes sense to me - and thank you for spelling it out like that. I think the core of my question is really superstition ("is this program REALLY not changing anything or can I just not hear the difference easily?"), and pointing out that it's an editor and shouldn't be changing things addresses that concern. Seems obvious now that you say it, but I hadn't really thought of it that way. Feb 23, 2011 at 15:13

I don't understand what you mean by "changing a single sample". As you may know, samples are 16 or 24 bit long strings that describe the particular sample, but the file format won't comply to this as easy.

AIFF and WAV are both lossless formats, in Pulse Code Modulation (also referred as Linear Pulse Code Modulation or LPCM due to the quantization processing involved), but one is based on Big endian (AIFF uncompressed) and WAV on little endian, changing the order of the bit stream so that it will start from the most or least significant bit, respectively.

iTunes, as practical as it may seems, uses AIFF-C, a compressed version (although this one actually uses little endian). So it might be fine for your iPod but probably you don't want that.

You could just simply use ProTools, Logic or Cubase to do the conversion? select region - cmd+K (or is it option? can't remember when I'm not in Tools) and select WAV 24bit, done! (I'm assuming you are giving it to your mastering engineer in 24 bit!) I would also leave embedded information out, as some readers (very frustrating example happened to me today with a Alesis HD24 where it would read embedded info as "sample rate mismatch") won't like it.

I'm afraid I might cross the line here by saying that I doubt you'd catch any difference between these using either of these software, although you might if you are using the mixer (pan laws, etc. Some mixers even ad processing). - hence the recommendation of "region export" like protools, cubase and logic offer - not bouncing the audio. Simple import, export region as.

Edited to correct some mistakes and add some small data!

  • Thanks for the answer; hopefully I can clarify. By "not changing a single sample" I mean "the bitstream decoded from my .AIFF file should match the bitstream decoded from the .WAV file." Basically, I would like to convert from AIFF to WAV without changing any audio information in the conversion process, which I'm pretty sure is possible. Apologies if I was unclear! I've had trouble wording it precisely. Feb 22, 2011 at 21:48
  • Microsoft’ WAV format is a clone of Apple’s AIFF. Both are lossless PCM audio files. If you convert an AIFF to WAV, you are just changing the packaging, the PCM data inside will be the exact same samples. This is a very common procedure because almost all lossless PCM audio files are either WAV or AIFF and they need to be converted back and forth for various situations. Ideally there would be one “.pcm” format but there is not. Feb 28, 2016 at 9:37


The short answer is that both AIFF and WAV encode all of the audio information from the original file in a completely lossless format. Both formats encode in PCM, but their "containers" are different. A 16bit, 48kHz stereo AIFF will be changed to a 16bit 48kHz stereo WAV with out any changes other than the "container." As long as the sample rate and bit depth are the same for the final wave as the original AIFF, then the two files will be bit (i.e. sample) accurate by default.

The same goes for FLAC or apple lossless, which are both smaller than Wav files, but are still bit accurate.


Answers above are correct. A 16bit 44.1kHz AIFF and WAV file should be identical lossless files once they have been opened in an editor. If you really want to confirm this (although I think the answers above are sufficient) you could open both files in an editor, and reverse the phase of the left & right channels of one version. When you play both versions together, you should get silence, as they will be identical but opposite.

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