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i noted that a lot of times when i run a mastered track through a vst plugin (with the plugin set not to do anything, for example a flat equalizer in ozone), the meter inside the plugin will go above 0dB (sometimes as high as 1dB), how is that possible, if that maximum amplitude encoded in the mp3 file is 0dB, how can the plugin detect more than that?

is the maximum amplitude not 0dB in an mp3 encoding? if not why do vst's "complain" when it reaches more then 0dB?

EDIT: forget plugins, just adding the mp3 to an empty track (no changes to volume pan etc..), gets reaper's built in meter to turn red (both the track's and master's)

  • If you're running it through a VST, it's likely loaded in some host. Are you, by any chance, doing this in Ableton Live with the default Complex warp mode turned on? – Warrior Bob Jul 7 '15 at 16:28
  • no, but i can reproduce this in reaper, mixcraft, and some dedicated vst host i found (different plugins, all show db level above 0dB) – user1333057 Jul 7 '15 at 16:30
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    See here pleasurizemusic.com/en/… – noumenal Jul 7 '15 at 20:33
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So many possible reasons, really. It's possible that there is some setting that is not obvious and is increasing the level of the audio inside the application that is making it exceed 0 dBFS. For instance, if it's a stereo file and the left and right channels are both panned to the center, you could be adding 3 - 6 dB to the summed audio.

Probably more likely is the complicated interplay between MP3 codecs, virtual summing amps, and intersample overages. The MP3 format has to be converted back to PCM to be processed by the software, which means the software is resynthesizing some frequencies. If the encoded level is high enough, the resynthesis could be adding enough to bring it over the top when decoded. The virtual summing amp inside the software could be very short on headroom or there could be another summing quirk like the stereo summing concern above. And even PCM audio that is limited to full scale can be reconstructed into audio that exceeds the full scale level when the full waveform is reconstructed.

When CDs were the primary delivery format (16-bit PCM, essentially), mastering engineers quickly discovered that if they didn't leave at least 0.3 dB of headroom between the peak sample(s) and full scale, some playback equipment would clip, even in the digital domain. I hard limit at -1.0 dBFS because the last thing I want is a digital over and when you start transcoding to MP3 or AAC or whatever, strange things can happen. The original soundtrack to Akira on CD actually has obvious overs right at the beginning of the first track, and there are a few CDs from the early 90s where you can hear them if you listen closely.

  • so if i understand correctly this stems from the decoding of a lossy format, aka this cant happen with lossless formats? what do you mean by "the left and right channels are both panned to the center", i didn't change panning in the DAW (left goes to the left, right goes to the right, both at 0dB). – user1333057 Jul 8 '15 at 10:03
  • PCM is lossless and it can happen with lossless files but it is less common. The sentence about panning starts with "if" and is just an example. – Todd Wilcox Jul 8 '15 at 10:14
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    Regarding lossless formats exceeding 0 dBFS, see: musictech.net/2012/09/10mm-no211-inter-sample-peaks – Todd Wilcox Jul 8 '15 at 12:06

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