[FFmpeg-devel] Review request - ra288.{c,h} ra144.{c,h}
Michael Niedermayer
michaelni
Sun Sep 14 19:12:47 CEST 2008
On Sun, Sep 14, 2008 at 05:55:16PM +0200, Vitor Sessak wrote:
> Michael Niedermayer wrote:
> > On Sat, Sep 13, 2008 at 09:48:46PM +0200, Vitor Sessak wrote:
> >> Michael Niedermayer wrote:
> >>> On Sat, Sep 13, 2008 at 07:07:26PM +0200, Vitor Sessak wrote:
> >>>> Michael Niedermayer wrote:
> >>>>> On Fri, Sep 05, 2008 at 12:23:58AM +0200, Vitor Sessak wrote:
> >>>>>> Vitor Sessak wrote:
> >>> [...]
> >>>>> [...]
> >>>>>> static void colmult(float *tgt, const float *m1, const float *m2, int n)
> >>>>>> {
> >>>>>> while (n--)
> >>>>>> *tgt++ = *m1++ * *m2++;
> >>>>>> }
> >>>>> such function is commonly called apply_window() in other codecs
> >>>>>> static void decode(RA288Context *ractx, float gain, int cb_coef)
> >>>>>> {
> >>>>>> int i, j;
> >>>>>> double sumsum;
> >>>>>> float sum, buffer[5];
> >>>>>> float *block = ractx->sp_block + 36; // Current block
> >>>>>>
> >>>>>> memmove(ractx->sp_block, ractx->sp_block + 5,
> >>>>>> 36*sizeof(*ractx->sp_block));
> >>>>>>
> >>>>>> for (i=0; i < 5; i++) {
> >>>>>> block[i] = 0.;
> >>>>>> for (j=0; j < 36; j++)
> >>>>>> block[i] -= block[i-1-j]*ractx->sp_lpc[j];
> >>>>>> }
> >>>>>>
> >>>>>> /* block 46 of G.728 spec */
> >>>>>> sum = 32.;
> >>>>>> for (i=0; i < 10; i++)
> >>>>>> sum -= ractx->gain_block[9-i] * ractx->gain_lpc[i];
> >>>>>>
> >>>>>> /* block 47 of G.728 spec */
> >>>>>> sum = av_clipf(sum, 0, 60);
> >>>>>>
> >>>>>> /* block 48 of G.728 spec */
> >>>>>> sumsum = exp(sum * 0.1151292546497) * gain; /* pow(10.0,sum/20)*gain
> >>>>>> */
> >>>>>>
> >>>>>> for (i=0; i < 5; i++)
> >>>>>> buffer[i] = codetable[cb_coef][i] * sumsum;
> >>>>>>
> >>>>>> sum = scalar_product_float(buffer, buffer, 5) / 5;
> >>>>>>
> >>>>>> sum = FFMAX(sum, 1);
> >>>>>>
> >>>>>> /* shift and store */
> >>>>>> memmove(ractx->gain_block, ractx->gain_block + 1,
> >>>>>> 9 * sizeof(*ractx->gain_block));
> >>>>>>
> >>>>>> ractx->gain_block[9] = 10 * log10(sum) - 32;
> >>>>>>
> >>>>>> for (i=1; i < 5; i++)
> >>>>>> for (j=i-1; j >= 0; j--)
> >>>>>> buffer[i] -= ractx->sp_lpc[i-j-1] * buffer[j];
> >>>>>>
> >>>>>> /* output */
> >>>>>> for (i=0; i < 5; i++)
> >>>>>> block[i] = av_clipf(block[i] + buffer[i], -4095, 4095);
> >>>>> can the buffer values be stored in block and sp_lpc applied over both
> >>>>> in one pass instead of this 2 pass and add-clip thing?
> >>>> I can't apply sp_lpc to buffer+block, so I need two buffers...
> >>> What i was thinking about was:
> >>>
> >>> /* block 46 of G.728 spec */
> >>> sum = 32.;
> >>> for (i=0; i < 10; i++)
> >>> sum -= gain_block[9-i] * ractx->gain_lpc[i];
> >>>
> >>> /* block 47 of G.728 spec */
> >>> sum = av_clipf(sum, 0, 60);
> >>>
> >>> /* block 48 of G.728 spec */
> >>> sumsum = exp(sum * 0.1151292546497) * gain; /* pow(10.0,sum/20)*gain */
> >>>
> >>> for (i=0; i < 5; i++)
> >>> buffer[i] = codetable[cb_coef][i] * sumsum * (1./2048.);
> >>>
> >>> sum = scalar_product_float(buffer, buffer, 5) / 5;
> >>>
> >>> sum = FFMAX(sum, 1);
> >>>
> >>> /* shift and store */
> >>> memmove(gain_block, gain_block + 1, 9 * sizeof(*gain_block));
> >>>
> >>> gain_block[9] = 10 * log10(sum) - 32;
> >>>
> >>> for (i=0; i < 5; i++) {
> >>> block[i] = buffer[i];
> >> Here you are overwriting the value of block[i] (while previous code used
> >> this value).
> >
> > previous code did:
> > for (i=0; i < 5; i++) {
> > block[i] = 0.;
> >
> > so that certainly was not useing it
>
> Ok, the problem is the following. Now we have (moving down a few loops):
>
> for (i=1; i < 5; i++)
> for (j=i-1; j >= 0; j--)
> buffer[i] -= ractx->sp_lpc[i-j-1] * buffer[j];
>
> for (i=0; i < 5; i++) {
> block[i] = 0.;
> for (j=0; j < 36; j++)
> block[i] -= block[i-1-j]*ractx->sp_lpc[j];
> }
>
> /* output */
> for (i=0; i < 5; i++)
> block[i] = av_clipf(block[i] + buffer[i], -4095, 4095);
>
> And I cannot change block[i] = 0. to block[i] = buffer[i] because
> block[0] will be read for evaluating block[1] in the second loop.
no and yes
the second loop applies a LPC filter to block assuming the future samples
are 0
the first loop applies the same LPC filter to buffer assuming the previous
samples are 0
the third loop combines them so that they match as if a LPC filter had
been applied to their sum i think.
>
> >>>>> [...]
> >>>>>> 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
[...]
--
Michael GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC787040B0FAB
Asymptotically faster algorithms should always be preferred if you have
asymptotical amounts of data
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