[FFmpeg-devel] [PATCH] adpcm: Store trellis nodes in a heap structure
Sat Nov 13 00:09:10 CET 2010
On Fri, Nov 12, 2010 at 10:54:33AM -0800, Jason Garrett-Glaser wrote:
> On Fri, Nov 12, 2010 at 4:33 AM, Martin Storsj? <martin at martin.st> wrote:
> > On Fri, 12 Nov 2010, Michael Niedermayer wrote:
> >> On Fri, Nov 12, 2010 at 12:41:24PM +0200, Martin Storsj? wrote:
> >> > On Thu, 11 Nov 2010, Reimar D?ffinger wrote:
> >> >
> >> > > On Thu, Nov 11, 2010 at 01:08:41AM +0200, Martin Storsj? wrote:
> >> > > > > In that case, do you feel like finding some setting that with all
> >> > > > > patches is about the same speed as without patches and compare the
> >> > > > > quality? IMO that would possibly be the most interesting comparison.
> >> > > >
> >> > > > If reading the graphs at
> >> > > > http://albin.abo.fi/~mstorsjo/adpcm-graphs/music1/, I find the following
> >> > > > test runs quite similar:
> >> > > > Original code, -trellis 6: 26.7 seconds, stddev 87.67, PSNR 57.47
> >> > > > Fully patched, -trellis 8: 22.8 seconds, stddev 85.08, PSNR 57.73
> >> > > >
> >> > > > Thus, with all the patches, you get better quality at comparable run
> >> > > > times. Or just roughly similar quality at very much shorter run time. :-)
> >> > >
> >> > > My question was rather: how does the "maximum" quality change, i.e.
> >> > > at the highest reasonable setting.
> >> > > I'd expect it to rather improve, but I think you so far only tested really
> >> > > fast settings (22 seconds is not a long encoding time in any way,
> >> > > even the original 1 minute something is still "acceptable", I remember
> >> > > when MP3-encoding was done at I think 1/8th real-time...)
> >> >
> >> > Well, then I guess it's all up to how much patience you have when defining
> >> > the "maximum" quality. If we consider 1/8th to 1/10th of realtime as
> >> > "maximum", we get these numbers:
> >> > Original code, -trellis 8: 245.8 seconds, stddev 83.65, PSNR 57.88
> >> > Fully patched, -trellis 11: 189.7 seconds, stddev 83.26, PSNR 57.92
> >> >
> >> > However, if checking the runtime_psnr graphs at
> >> > http://albin.abo.fi/~mstorsjo/adpcm-graphs/, one notices that the original
> >> > (and patch #1 and patch #3) will get better PSNR/runtime if extending the
> >> > benchmark to even larger trellis sizes. For the music1 sample, this
> >> > happens around sometimes after ~1/15 of realtime, for the other samples it
> >> > happens even later than that.
> >> so id say #1-#3 are ok to commit while #4 needs more work
> > Applied #1-3. I'll think more about #4 later to see if I can make a better
> > compromise between quality and runtime.
> > I don't think I had any code in the G.722 trellis encoder similar to the
> > one I'm removing in #4, so I guess I can try to update the G.722 trellis
> > patch with these improvments.
> > Thanks for the persistence on making graphs - the log(time) vs PSNR graph
> > really is a valuable tool for making these comparisons.
> > // Martin
> > _______________________________________________
> > ffmpeg-devel mailing list
> > ffmpeg-devel at mplayerhq.hu
> > https://lists.mplayerhq.hu/mailman/listinfo/ffmpeg-devel
> FYI, here is why PSNR sucks
> http://x264.nl/developers/Dark_Shikari/2.wav (from a game that used ADPCM audio)
> http://x264.nl/developers/Dark_Shikari/1.wav (ffmpeg's ADPCM, with
> -trellis 16, source was the (lossless) game soundtrack CD)
> ffmpeg sounds vastly worse. I blame PSNR.
these 2 files are encoded with vastly different encoders
the better sounding one dynamically adapts the 2 predictor coefficients while
ffmpeg leaves them at very dubious 0, 64 values.
Thus what you compare here is not just PSNR vs advanced metric (is the other
even using a advanced metric?) but a
encoder that dynamically adapts LPC coeffs against one which uses fixed and
actually quite bad one in addition to any other differences
Michael GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC787040B0FAB
Asymptotically faster algorithms should always be preferred if you have
asymptotical amounts of data
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