There have been some problems with FATE audio testing. First off, the qt-ima4-stereo test spec was testing against the wrong file for the past year. Stereo IMA ADPCM decoding could have broken in QuickTime and we might have never been alerted. Sloppy.
More seriously, I found out that many of my existing, bitexact audio tests have not been constructed properly. This is due to the fact that these 2 commands:
ffmpeg -i file.ext file.wav
ffmpeg -i file.ext -f wav - > file.wav
do not yield equivalent sets of bytes inside file.wav. Part of the reason is that, after writing out all the audio samples, the muxer needs to rewind to the header so that it can write the data payload length. When writing data to stdout, the program does not have the option to rewind the output stream. However, I don’t understand the entire discrepancy. Using the file qt-ima4-mono with the above command lines:
The file that is routed through stdout is notably smaller (9728 bytes smaller). I was going to write this off as the stdout file failing to be flushed. However, the behavior is consistent across all machines and platforms.
My proposed solution is to update all of the audio tests to use this raw format target:
ffmpeg -i file.ext -f s16le -
Since the output is equivalent to:
ffmpeg -i file.ext -f s16le file.s16le
Moving right along, there is the much bigger task of testing perceptual audio decoders. Working down the FATE Test Coverage list, these perceptual audio codecs will get the naive, one-off wave reference treatment in lieu of a proper conformance suite: ATRAC3, RealAudio Cooker, DCA (DTS), IMC, Nellymoser, Qcelp, QDesign, RealAudio 28.8, Truespeech, Vorbis, and WMA v1.
Then there is the matter of MPEG audio codecs for which we have access to extensive conformance suites. Thanks to Kostya and Benjamin for furnishing pointers to precise information discussing how to verify if your MP1/2/3 or AAC audio decoder is up to snuff. This page at Underbit describes exactly how the spec describes conformance for MPEG 1, layers 1, 2, and 3, and also evaluates the conformance of various implementations. The comparison ostensibly predates FFmpeg. This Mp4-tech mailing list post shows the way regarding AAC conformance.
So I need to automate the MP1/2/3 and AAC test entries. I estimate the automated process will work something like this:
- Decode encoded file
- Run comparison of decoded wave against original wave
- For MP1/2/3, this seems to entail converting both the FFmpeg output and the original wave output floating point numbers to a normalized range of -1.0..1.0, computing the root mean square of the difference signal, and verifying that the RMS is less than 1 / (32768 * sqrt(12))
- For AAC, well, I’m still researching the precise criteria
- If the decoded wave is within tolerance, add a new test
The part where I get a bit fuzzy is: what should the test spec be? Should I generate a reference wave and test future decoded waves against it using my one-off wave reference method? Or, should I just go ahead and compute the RMS of the difference signal? I assume that if I use the nifty numpy library for the task, it couldn’t possibly make any measurable difference in the performance of FATE testing vs. using the one-off wave reference method (computing absolute value of the difference signal and checking that no discrete points exceed 1).
One trade-off is that I would need to store the full 24-bit reference waves in order to properly compute RMS, which is 50% more data than I would need with the one-off method. And I’m still not sure how to process the 24-bit data in any event.