Very Interesting, thanks for the explanation Kerry...

I'll admit I know very little about the this subject, I just pump out the
files in a format that sounds acceptable, for me an mp3 with a sample rate
of 22khz would exhibit the behavior within the bug report, FP9 and above (
have not tested in 10 yet ), whereas creating an mp3 file with a sample rate
of 44khz worked perfectly.  It's strange indeed.

Again knowing very little about this subject one of the comments in that bug
report noted by a Mr. John McGucken is that this also happens with Microsoft
Visual Studio, and as he put it, and this is only his best "guess":

" Microsoft Visual studio shows the exact same issue!
Therefore, I would guess that the Flash Player is built using components
from Visual studio (I.E. The MP3 play back and syncing). More concerning
this would indicate that the issue actually lies with Microsoft and the
Flash Player is exhibiting this abnormal as a result of this. "

Thanks again for the informative post.

Dunc



On Fri, Aug 8, 2008 at 12:10 PM, Kerry Thompson <[EMAIL PROTECTED]>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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