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fb2k's compensation for decoder delay is inconsistent, 529 samples sometimes go missing. Bug or feature?
post Nov 2 2010, 09:49
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As I understand it, the first non-VBR-header frame of an MP3 file should decode to 529 (the usual decoder delay) + 1152 samples, and every frame thereafter should decode to exactly 1152 samples. The player should discard those first 529, and then it should discard however many samples the VBR header said was the encoder delay. When the last frames are reached, the player should discard however many samples the VBR header said was padding. It should be as simple as that, right?

foobar2000 1.1 seems to behave inconsistently in this regard. timcupery apparently noticed the problem in 0.9 a while back but didn't get any responses, perhaps because he asked about several things at the same time.

Here's a quick test to demonstrate.

flac.exe -c -d -s Boing_Boom_Tschak.flac | lame.exe -V0 -Boing_Boom_Tschak.mp3
The resulting MP3 has 1155 frames + the VBR header frame w/LAME tag. The LAME tag specifies encoder delay 576 and padding 991. This checks out fine: 1155 frames × 1152 samples/frame = 1330560 samples. 1330560 - 576 - 991 = 1328993, same as the original FLAC.

Now load up the MP3 in foobar2000 1.1 and convert it to WAV. Output should be 1328993 samples and should align with the FLAC in a wave editor. No problem so far.

Now in foobar2000, edit the MP3's gapless playback information. Change encoder delay to 0 and length-of-track to 1330560. The playlist item's properties window should now show delay & padding of 0. Convert to WAV again. We would expect the output to be 1330560 samples. However, it's now 1330031. In a wave editor, it's apparent that fb2k removed 529 samples from the beginning, apparently overcompensating for decoder delay by a factor of 2!

It's not just happening in conversion, but in ordinary playback as well. Any MP3 having a delay and padding both set to 0 loses its first 529 samples. Because of this, seams are audible in what should be a gapless series of MP3s created by mp3DirectCut after splitting a CBR, no-bit-reservoir MP3 on frame boundaries.

Back to the Kraftwerk sample—some experimentation shows that after using fb2k to edit the gapless playback information, leaving the encoder delay at 0 and setting the padding to any value between 0 and 529 (length-of-track 1330560 to 1330031), results in the same 1330031-sample output. But setting the padding to 530 (length-of-track 1330030) results in the requested 1330030 samples. In a wave editor, one can observe 576 samples of encoder delay, followed by 1328993 samples of audio, followed by not 530 but 461 samples of padding, as that's what's required to reach 1330030. I notice 1330030 is 530 shy of a frame boundary, probably not a coincidence.

I haven't even experimented with encoder delay values yet, and I'm finding it rather painful to wrap my head around this behavior. I don't understand why decoder delay even enters into the equation. I should be able to set encoder delay & padding to whatever I want and get those samples trimmed from the beginning and end, respectively, and never be exposed to decoder delay. What's going on here? Bug? Feature?

This post has been edited by mjb2006: Nov 2 2010, 09:50
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post Nov 2 2010, 19:08
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Where do you arrive at the conclusion that 529 is "overcompensating for decoder delay by a factor of 2?" 529 is not an even number. It is exactly the decoder delay for layer 3 files, in this case.
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