Hello Ned,
I know a fair amount about certain audio codecs and audio file container
formats, especially MP3 (the codec and the container and ID3 metadata)
and AAC (codec) and M4A/MP4 (the container and ill-defined metadata
standard).
As I have discussed in at least one other email to this list recently
[1], FFmpeg probably reads metadata from the container files and does
not calculate it by actually analyzing the compressed audio. There are
also tons of reasons for this metadata to be incorrect in the container
file. The AAC/M4A ecosystem is much worse about standards compliance
than the MP3/ID3 ecosystem because there really are no standards at all
for AAC/M4A metadata, and the audio compression is far more flexible and
configurable than MP3 audio compression. The de facto popular standard
is basically "whatever works with iTunes", because iTunes runs on
several operating systems and Apple is a major source for AAC-format
audio files, because that is the favorite format on the iTunes Store.
What you ideally want is a software library that can decode the audio
data itself and deduce the duration. I can suggest Python libraries that
do this for MP3 audio streams, but sadly I am less familiar with AAC
audio streams. MediaInfo (software package) is something that might do
it, because it supports a lot of things, but I don't honestly know.
If knowing the duration in seconds is very important, in the context of
music, be very careful just reading the metadata in the file. Sometimes
the metadata was created by looking at the file size and dividing by the
bitrate. Sometimes this was done by a batch metadata editor after an
album cover was embedded in the file, so the "duration" gets an extra 4
seconds because of the JPEG of the album cover, for example. In other
cases, the target bitrate for a variable bitrate (VBR) AAC stream will
be used in the division operation, which means that the deduced duration
is off by however much the audio deviated from the target bitrate. If
the file is pure silence, the duration will be way off because that
compresses very small.
[1] Here is the email where I talk about AAC stream metadata,
specifically bitrate:
https://ffmpeg.org/pipermail/ffmpeg-user/2018-March/039037.html
Subject line was:
Re: [FFmpeg-user] Why _COPYing_ a variable encoded audio (AAC) channel
will produce a variably encoded one?
Date: Thu Mar 1 02:20:15 EET 2018.
Zak
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