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

[Michael Niedermayer]

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?


[...]

-- 
Michael     GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC787040B0FAB

Breaking DRM is a little like attempting to break through a door even
though the window is wide open and the only thing in the house is a bunch
of things you dont want and which you would get tomorrow for free anyway
-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: application/pgp-signature
Size: 189 bytes
Desc: Digital signature
URL: <http://lists.mplayerhq.hu/pipermail/ffmpeg-devel/attachments/20080915/a6459136/attachment.pgp>