[gstreamer-bugs] [Bug 360246] [audioconvert] Optionally apply dithering
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Sat Apr 14 15:06:32 PDT 2007
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GStreamer | gst-plugins-base | Ver: HEAD CVS
------- Comment #11 from David Schleef 2007-04-14 22:06 UTC -------
git clone git://people.freedesktop.org/~ds/cog
The code in gstdither.c quantizes int16_t audio into values that can be
directly shifted into int8_t. I.e., the low 8 bits of each output sample is 0.
I tried integrating this code into audioconvert.c and got frustrated -- I'd
want to rewrite audioconvert.c into four stages: unpack, mix, quantize, pack,
with better control over intermediate buffers.
Since audioconvert uses int32_t as the unpacked format, the code needs to be
adjusted to check for overflows and/or use int64_t as intermediates. Another
possibility is to shift the 32 bit values right by 8 bits (thus losing 8 LSBs)
before quantizing -- converting 32->24 doesn't need dithering anyway.
There's also code for converting float to int16_t, but it creates the float
samples by converting from int16_t.
The random number generator (for dithering) is from liboil. Liboil has good
random number generation, but it's used very poorly here. Ideally,
audioconvert should have a random number generator context that it uses only to
generate dithering, and reset it when other element state gets reset
(READY->PAUSED). Liboil doesn't have contextual RNG, whereas glib does. So
I'd recommend using glib's RNG until liboil is fixed.
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