[FFmpeg-devel] [PATCH] ALAC Encoder

[Ramiro Polla]

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?

Ramiro Polla