On Jan 3, 2007, at 3:28 PM, Josh Coalson wrote:
the FLAC parts look OK but I don't know how the audio converter
works. I think that's where the problem is. it is probably
converting float to 32-bit int full scale.
well, here's what I found out about the 32-bit integer conversion.
If the original 16-bit sample is 0x52F3, it goes through floating
point, then comes to the 32-bit integer as 0x52F30000
As far as I know that's the proper conversion, yes? Simple adding
some LSB to make it 32-bit.
but if [audioFile range] is 16 (i.e. 16 bits per sample), then
after conversion, the integer PCM samples in outputBuffers
should all be in the range [-32768,32767], that's the first
thing I'd check.
OH so wait...
What you're saying then is that the buffers always have to be filled
with 32-bit-SIZED-FIELDS, inside of which
exists a low-aligned integer sample value?
So the VALUES aren't 32-bit integer, but the SPACE FOR THE SAMPLE
VALUE is 32-bit.
Wow, that's very trippy. Honestly, I *never* would have guessed that
from the documentation.
That would explain the silence, too.
Ev
Technical Knowledge Officer
Head Programmer/Designer
Audiofile Engineering
http://www.audiofile-engineering.com/
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