[FFmpeg-devel] [PATCH] ALAC Encoder

Michael Niedermayer michaelni
Mon Aug 18 16:01:22 CEST 2008


On Mon, Aug 18, 2008 at 09:38:53AM -0300, Ramiro Polla wrote:
> Hi,
> 
> On Sun, Aug 17, 2008 at 12:55 PM, Michael Niedermayer <michaelni at gmx.at> wrote:
> > On Sun, Aug 17, 2008 at 11:17:27AM -0300, Ramiro Polla wrote:
> >> On Sun, Aug 17, 2008 at 10:15 AM, Michael Niedermayer <michaelni at gmx.at> wrote:
> >> > On Sun, Aug 17, 2008 at 09:09:00AM +0530, Jai Menon wrote:
> >> >> Hi,
> >> >>
> >> >> On Sunday 17 Aug 2008 8:05:14 am Michael Niedermayer wrote:
> >> >> > On Sun, Aug 17, 2008 at 04:14:43AM +0530, Jai Menon wrote:
> >> > [...]
> >> >> > > +static void alac_stereo_decorrelation(AlacEncodeContext *s)
> >> >> > > +{
> >> >> > > +    int32_t *left = s->sample_buf[0], *right = s->sample_buf[1];
> >> >> > > +    int32_t tmp;
> >> >> > > +    int i;
> >> >> > > +
> >> >> > > +    for(i=0; i<s->avctx->frame_size; i++) {
> >> >> > > +        tmp = left[i];
> >> >> > > +        left[i] = (tmp + right[i]) >> 1;
> >> >> > > +        right[i] = tmp - right[i];
> >> >> > > +    }
> >> >> > >
> >> >> > > +    s->interlacing_leftweight = 1;
> >> >> > > +    s->interlacing_shift = 1;
> >> >> >
> >> >> > i do not belive this is optimal
> >> >> >
> >> >>
> >> >> It may not be optimal in the sense that I do not adaptively select the
> >> >> decorrelation scheme, but this is just the first iteration which aims at
> >> >> getting a basic encoder into svn. And it is better than doing no
> >> >> deorrelation. I did initially try out an adaptive approach but the difference
> >> >> in compression wasn't that great. I'm looking into how this can be done in a
> >> >> better manner. Till then, I was hoping if we could go with this.
> >> >
> >> > see the pca.c/h i posted in a reply to ramiro a few days ago
> >> > it might be worth a try ...
> >>
> >> Speaking of that... I haven't finished integrating it in MLP (I'm
> >> working on some other stuff atm), but it seems to be what I need.
> >
> >> Could you get it cleaned up and committed like you suggested?
> >
> > done
> 
> I can get it working in my tests, but not in MLP =(
> 
> I take something like this (I'll name it here s[channels][samples]):
>             [samples]
> [channel 0] 0 1 2 3 4 5 6 7 8
> [channel 1] 0 1 2 3 4 5 6 7 8
> [noise   0] 0 1 2 3 4 5 6 7 8
> [noise   1] 0 1 2 3 4 5 6 7 8
> 
> Pass it through the pca, <num_channels> samples at a time
> pca_add(s[0][0], s[1][0], s[2][0], s[3][0])
> pca_add(s[0][1], s[1][1], s[2][1], s[3][1])
> ...
> pca_add(s[0][x], s[1][x], s[2][x], s[3][x])
> 
> Solve the pca and get the eigenvectors (I'll name it here as e[][])
> 
> Multiply them both to a new buffer (s2[channels][samples])
> s2[0][0] = s[0][0] * e[0][0] + s[1][0] * e[0][1] +
>            s[2][0] * e[0][2] + s[3][0] * e[0][3]
> s2[0][1] = s[0][1] * e[0][0] + s[1][1] * e[0][1] +
>            s[2][1] * e[0][2] + s[3][1] * e[0][3]
> ...
> s2[0][x] = s[0][x] * e[0][0] + s[1][x] * e[0][1] +
>            s[2][x] * e[0][2] + s[3][x] * e[0][3]
> 
> s2[1][0] = s[0][0] * e[1][0] + s[1][0] * e[1][1] +
>            s[2][0] * e[1][2] + s[3][0] * e[1][3]
> s2[1][1] = s[0][1] * e[1][0] + s[1][1] * e[1][1] +
>            s[2][1] * e[1][2] + s[3][1] * e[1][3]
> ...
> s2[1][x] = s[0][x] * e[1][0] + s[1][x] * e[1][1] + s[2][x] * e[1][2] +
> s[3][x] * e[1][3]
> 
> In this case I'd get some values in s2[0][x], and 0 for s2[i>0][x],
> since all channels are equal.
> 
> I then multiply again by the eigenvectors (with something transposed
> in the middle, I forgot which), and get s[][] again.
> 
> But MLP doesn't have the intermediate s2 buffer. It overwrites s[][]
> directly. From my stereo samples, it usually only infers s[1][] out of
> s[0,2,3][] (2, 3 being noise channels). So there's only one lossless
> matrix for channel 1. It works like:
> s[1][0] = s[0][0] * e[1][0] + s[1][0] * e[1][1] +
>           s[2][0] * e[1][2] + s[3][0] * e[1][3]
> s[1][1] = s[0][1] * e[1][0] + s[1][1] * e[1][1] +
>           s[2][1] * e[1][2] + s[3][1] * e[1][3]
> ...
> s[1][x] = s[0][x] * e[1][0] + s[1][x] * e[1][1] +
>           s[2][x] * e[1][2] + s[3][x] * e[1][3]
> 
> If there were more matrices, it would always use the previously
> overwritten values.
> 
> Can I still achieve this with PCA?

yes

there are many things that can be tried ...
first is to simply just decorrelate only one channel
and leave the others as is, this should be easy and similar to the
other encoder

s[r] = s[0]e[0]/e[r] + s[1]e[1]/e[r] + ...+ s[n]e[n]/e[r]

e here is the eigenvector with the smallest eigenvalue
r is choosen so that e[r] is largest

second, lets assume E is our eigenvector matrix with which we want to
transform the channels, S is a vector of the samples of all channels at a
single time
what we want is 
E*S

and when we decorrelated one channel (as for example in "first" above)
then we have applied a linear transform to S that is S1=L*S (L being a matrix
with one row of the e[x]/e[r] coeffs and otherwise the identify matrix)
this would make our original goal of E*S look like E*L^-1*S1, now 
we can multiply the left side out to get E1*S1 with E1=(E*L^-1)
and then repeat "first" with the next eigenvector.

note1 L^-1 is just the identify matrix with one row replaced by e[x]/e[r]
and signs fliped on all but one element of that row if i didnt make an
error, so this should all be rather trivial and fast 1-2 line loops in C
theres not even any need to ever store L or L^-1 as matrixes, its just
convenient to think of them that way ...

note2 i do not know which order is the best when more than 1 channel is
decorrelated like this, but it does matter, different order leads to
different scalings for the channels due to preserving of losslessness



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

Dictatorship naturally arises out of democracy, and the most aggravated
form of tyranny and slavery out of the most extreme liberty. -- Plato
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