Looking at that bug report, it seems like it's because someone's doing
dodgy sums based on 44kHz...
At the end of the day, you still have to decode MP3's into WAV type
audio to play it back. The codec will output 11, 22, 44 etc. based on
the audio sample rate the file was encoded with...
What's probably happening is that the playback engine (after the codec
?) is probably running at 44.1kHz, and positioning info maybe based on
that, not taking into account the sample rate differences (bit of a
schoolboy error that one...)
It's quite amusing from an abstract viewpoint, but I feel for the poor
people trying to work with various sample rates :(
C'mon Adobe, make some noise (that works).
Kerry Thompson wrote:
Duncan Reid wrote:
this sounds similar to the sample rate bug, this one drove me absolutely
bat
shit.
https://bugs.adobe.com/jira/browse/FP-33
I've seen that bug report, and never quite understood it. It talks about 22
kHz mp3's, 11 kHz, and 48 kHz (?). Talking about kHz in mp3 files doesn't
make sense, so I've never quite understood the bug. Let me explain--forgive
me if this gets technical, but I've been involved in mp3 since the early
90s, when it came out of MIT labs, and before it was called mp3.
When you create an mp3 file, you take some source audio and compress it,
squeezing out sounds that can't be heard by the human ear. The lower the bit
rate, the more sounds are squeezed out, including sounds you _can_ hear,
until you get telephone-quality audio at the low end.
But this has nothing to do with kHz, unless you are talking about the
sampling rate of the source file, typically 44.1, 22, or 11 kHz. The
sampling rate of the source file has a lot to do with the quality of the
audio, much as the bit rate does in an mp3.
A young person's ear can hear sounds up to about 20 kHz. For technical
reasons I won't go into, digital audio sampling can record sounds up to half
the sampling rate. That is, a sampling rate of 44.1 kHz (CD-quality audio)
can reproduce sounds up to 22.05 kHz--a little beyond the hearing of a
youngster (our ability to hear the high frequencies drops off as we age). A
sampling rate of 22.05 kHz can reproduce sounds up to 11.025 kHz, which is
good enough for ambient music. 11.025 kHz sampling rate is good enough for
most speech--some of the sibilants we produce may go higher than 5 kHz, but
speech is still reasonable quality at that rate.
Now, you take an aiff, wave, or Red Book file at one of those sample rates,
and run it through an mp3 compressor. Even at the highest bit rates, you
will lose some sounds. It might just be the breathing of the audience, the
quiet turning of a page by a violinist, or some of the highest overtones of
the cymbal or snare, but, even with 44.1 source at the highest bit rate, you
lose some sound.
Mp3 is a lossy compression, so talking about kHz in the context of mp3 audio
is meaningless. So, I have never quite understood that bug.
Cordially,
Kerry Thompson
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