[FFmpeg-devel] Review request - ra288.{c,h} ra144.{c,h}

Michael Niedermayer michaelni
Tue Sep 16 22:35:52 CEST 2008


On Tue, Sep 16, 2008 at 08:23:19PM +0200, Vitor Sessak wrote:
> Michael Niedermayer wrote:
> > On Mon, Sep 15, 2008 at 07:49:41PM +0200, Vitor Sessak wrote:
> >> Michael Niedermayer wrote:
> >>> On Sun, Sep 14, 2008 at 11:29:08PM +0200, Vitor Sessak wrote:
> >>>> Michael Niedermayer wrote:
> >>>>> On Sun, Sep 14, 2008 at 08:17:18PM +0200, Vitor Sessak wrote:
> >>>>>> Michael Niedermayer wrote:
> >>>>>>> On Sun, Sep 14, 2008 at 05:55:16PM +0200, Vitor Sessak wrote:
> >>>>> [...]
> >>>>>>>>>>>>> [...]
> >>>>>>>>>>>>>> static int ra288_decode_frame(AVCodecContext * avctx, void *data,
> >>>>>>>>>>>>>>                               int *data_size, const uint8_t * buf,
> >>>>>>>>>>>>>>                               int buf_size)
> >>>>>>>>>>>>>> {
> >>>>>>>>>>>>>>     int16_t *out = data;
> >>>>>>>>>>>>>>     int i, j;
> >>>>>>>>>>>>>>     RA288Context *ractx = avctx->priv_data;
> >>>>>>>>>>>>>>     GetBitContext gb;
> >>>>>>>>>>>>>>
> >>>>>>>>>>>>>>     if (buf_size < avctx->block_align) {
> >>>>>>>>>>>>>>         av_log(avctx, AV_LOG_ERROR,
> >>>>>>>>>>>>>>                "Error! Input buffer is too small [%d<%d]\n",
> >>>>>>>>>>>>>>                buf_size, avctx->block_align);
> >>>>>>>>>>>>>>         return 0;
> >>>>>>>>>>>>>>     }
> >>>>>>>>>>>>>>
> >>>>>>>>>>>>>>     if (*data_size < 32*5*2)
> >>>>>>>>>>>>>>         return -1;
> >>>>>>>>>>>>>>
> >>>>>>>>>>>>>>     init_get_bits(&gb, buf, avctx->block_align * 8);
> >>>>>>>>>>>>>>
> >>>>>>>>>>>>>>     for (i=0; i < 32; i++) {
> >>>>>>>>>>>>>>         float gain = amptable[get_bits(&gb, 3)];
> >>>>>>>>>>>>>>         int cb_coef = get_bits(&gb, 6 + (i&1));
> >>>>>>>>>>>>>>
> >>>>>>>>>>>>>>         decode(ractx, gain, cb_coef);
> >>>>>>>>>>>>>>
> >>>>>>>>>>>>>>         for (j=0; j < 5; j++)
> >>>>>>>>>>>>>>             *(out++) = 8 * ractx->sp_block[36 + j];
> >>>>>>>>>>>>> if float output works already, then this could output floats, if not then
> >>>>>>>>>>>>> this could use lrintf()
> >>>>>>>>>>>> I've tried the float output (with the attached patch) and it didn't work. 
> >>>>>>>>>>> ok
> >>>>>>>>>>>
> >>>>>>>>>>>
> >>>>>>>>>>>> Using lrint() changes slightly the output (PSNR about 99), is it expected?
> >>>>>>>>>>> yes, it does round differently (=more correctly)
> >>>>>>>>>> Too correct maybe. PSNR to binary decoder with SVN:
> >>>>>>>>>>
> >>>>>>>>>> stddev:    0.15 PSNR:112.70 bytes:   990720/  1013760
> >>>>>>>>>> stddev:    0.04 PSNR:122.74 bytes:   368640/   368640
> >>>>>>>>>> stddev:    0.07 PSNR:118.84 bytes:   460800/   458752
> >>>>>>>>>> stddev:    0.31 PSNR:106.24 bytes:  6451200/  6451200
> >>>>>>>>>>
> >>>>>>>>>> Using lrint()
> >>>>>>>>>>
> >>>>>>>>>> stddev:    0.70 PSNR: 99.33 bytes:   990720/  1013760
> >>>>>>>>>> stddev:    0.70 PSNR: 99.35 bytes:   368640/   368640
> >>>>>>>>>> stddev:    0.70 PSNR: 99.35 bytes:   460800/   458752
> >>>>>>>>>> stddev:    0.75 PSNR: 98.76 bytes:  6451200/  6451200
> >>>>>>>>> yes, the rounding is more accurate, and differs by +-1 50% of the time from
> >>>>>>>>> the binary decoder, sqrt(0.5) ~ 0.7
> >>>>>>>>>
> >>>>>>>>> If you want a proof that it is better, you should compare the original
> >>>>>>>>> pcm that is
> >>>>>>>>>
> >>>>>>>>> X -> encoder -> binary decoder -> Y
> >>>>>>>>>              -> FF decoder ->Z
> >>>>>>>>>
> >>>>>>>>> and look at how the X-Y and X-Z change relative to each other.
> >>>>>>>>>
> >>>>>>>>> Also you would see a similar PSNR change relative to the binary decoder if
> >>>>>>>>> you would output floats.
> >>>>>>>> I've already tried comparing PSNR to the original input when I was 
> >>>>>>>> looking for a way to test float codecs in FATE.
> >>>>>>>>
> >>>>>>>> vitor at vitor$ ffmpeg -i luckynightmono2.ra -ac 1 -ar 8000 test.wav
> >>>>>>>> vitor at vitor$ ffmpeg -i luckynight.wav -ac 1 -ar 8000 test2.wav
> >>>>>>>> vitor at vitor$ tiny_psnr test.wav test2.wav 2 0 44
> >>>>>>>> stddev: 5981.39 PSNR: 20.78 bytes:   990720/   967662
> >>>>>>>> vitor at vitor$ tiny_psnr test.wav test2.wav 2 2 44
> >>>>>>>> stddev: 5982.77 PSNR: 20.78 bytes:   990718/   967662
> >>>>>>>> vitor at vitor$ tiny_psnr test.wav test2.wav 2 100 44
> >>>>>>>> stddev: 6012.76 PSNR: 20.74 bytes:   990620/   967662
> >>>>>>>>
> >>>>>>>> And by looking at results, if I change the "skip bytes" parameter I 
> >>>>>>>> don't get much change in PSNR. For me, this is a signal that the value I 
> >>>>>>>> got is meaningless (since it don't change a lot if I compare it with 
> >>>>>>>> different data). I asked about it in IRC and people told me that PSNR 
> >>>>>>>> didn't worked very well to LPC vocoders. Sample in 
> >>>>>>>> http://samples.mplayerhq.hu/real/AC-28_8/ .
> >>>>>>> considering that the claimed encoder input has
> >>>>>>> 10668716 bytes of 44.1khz at stereo
> >>>>>>> and that /2/44100*8000 is ~967684
> >>>>>>> and the ra288 decoder output has 990764 bytes i cant help but wonder
> >>>>>>> why, but of course this is incompareable. PSNR or otherwise
> >>>>>> Yes, the files have different sizes. That's why I started poking with 
> >>>>>> "skip bytes" and tried to cut the files. But I didn't succeeded in 
> >>>>>> making they match whatever I did.
> >>>>> how has the .ra file been generated?
> >>>>> what happens with a 2x as long input file? does the size difference
> >>>>> stay constant or grow?
> >>>>>
> >>>>> what does the binary decoder produce for it? is that also too big?
> >>>> Original     wav:  967706 bytes
> >>>> FFmpeg   decoder:  990764 bytes
> >>>> Original decoder: 1013804 bytes
> >>>>
> >>>> Go figure...
> >>> the decoder outputs 3 seconds more than what is in the claimed original.
> >>> How does it sound? is the audio stretched to the bigger length are there
> >>> 3 seconds of distortion or silence somewhere?
> >> Original     wav:  967706 bytes
> >> FFmpeg   decoder:  990764 bytes   1 second  of silence in the end
> >> Original decoder: 1013804 bytes   3 seconds of silence in the end
> >>
> >> Anyway, nothing of that explains the PSNR discrepancy...
> > 
> > ok, so lets forget about the PSNR, and rather try a simpler test for
> > the accuracy, just try to cast a float to an int and try lrintf()
> > and print the differens, or sum or squared differences, it should be
> > obvious which is more accurate.
> 
> The problem is not just when using PSNR, but I also fail to see any 
> similarity between the two files in a hex editor.

PEBCAK

looking with gnuplot at the files shows clearly that they are near identical
but drift, creating one with -ar 8018 and using
tiny_psnr luckymonora.wav luckymono2.wav 2 34 44
shows:
stddev: 3474.96 PSNR: 25.50 bytes:   990686/   969838

which cuts the difference to half of what you had, but thats not how
one compares files.
We need a  8khz wav and the corresponding ra288, a 44khz that was "somehow,
we dont know" resampled to 8khz is not useable.

[...]
-- 
Michael     GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC787040B0FAB

Freedom in capitalist society always remains about the same as it was in
ancient Greek republics: Freedom for slave owners. -- Vladimir Lenin
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