[FFmpeg-devel] Review request - ra288.{c,h} ra144.{c,h}
Michael Niedermayer
michaelni
Mon Sep 15 21:41:19 CEST 2008
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.
[...]
--
Michael GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC787040B0FAB
it is not once nor twice but times without number that the same ideas make
their appearance in the world. -- Aristotle
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