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Topic: Forcing The Sample Rate (Read 2401 times) previous topic - next topic
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Forcing The Sample Rate

I'm having an issue in trans-coding an AAC stream to MP2.  The AAC stream has been demuxed from an AVI file, in this somehow it's header was dropped or screwed up, Foobar is reading the AAC stream as being 44.1khz when it's actually 48khz, this is of course resulting in dramatic synch issues as  Foobar is working with the file at 9% slower speed than it should be.

Is there any way to force the sample rate at which Foobar reads a file, thusly correcting this problem?

Forcing The Sample Rate

Reply #1 –
I made a simple DSP you can use to do that.

Forcing The Sample Rate

Reply #2 –
Quote
I made a simple DSP you can use to do that.

Maybe there should be a little explanation on the pref page of this plugin, other than "Change samplerate", so that uninformed people won't confuse it with one of the (output) resamplers (SSRC or PPHS) ....?
"ONLY THOSE WHO ATTEMPT THE IMPOSSIBLE WILL ACHIEVE THE ABSURD"
        - Oceania Association of Autonomous Astronauts

Forcing The Sample Rate

Reply #3 –
Ah, perfect.  Exactly what I was needing, thank you.  Without this, I was going to be converting th AAC to a WAV, then using Besweet to encode the WAV, forcing the sample rate to 48000 and encoding it to MP2.  This makes life simpler, thanks.