Ok.
Full Processing is on.
I'm using the Lame MP3 encoder for the audio compression, and it's not working.
I'm getting an ACM Failure on Compress.
The source stream is:
44.1KHz, Mono
The compression is:
44.1Khz, Mono, 84kbps (might be 86, but whatever)
It's not working. It works for the first 20 seconds of the AVI and then poops out.
I'm going to make some assumptions.
If PCM Audio = 48KHz, then the compression sample rate must be the same.
So, if I had a 48KHz stereo 16 bit source, I extract the audio, open up Adobe Audition (Cool Edit) and convert to 44.1Khz, Mono, 16bit.
This should work.
Anything I'm doing wrong??
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For Video distributed on-line, are there any restrictions to the audio stream?
Conventions?
I'm capturing a source:
1. 48KHz 16bit Stereo
2. Extracting WAV
3. Converting WAV to 44.1 16bit Mono
4. Using it as the "new source" ala Virtual Dub
5. Trying to compress using Lame MP3 44.1Khz, 16bit ~90kbps CBR
Keeps generating the ACM error referenced above.
What characteristics much match between source and destination in the audio stream?
Sample Rate? Appears so.
Bit rate? Doesn't apply between source/target, does it?
Mono vs. Stereo? Appears so.
Ugh.
So far, the only thing that works consistently is:
48Khz, 16bit Stereo at around 128kbps Layer 3 (Lame)
I don't want/need that, but nothing else works. Even if I convert the source audio first.
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