[FFmpeg-devel] [PATCH] Some ra144.c simplifications
Vitor Sessak
vitor1001
Sat Jun 28 23:27:40 CEST 2008
Michael Niedermayer wrote:
> On Tue, Jun 24, 2008 at 11:35:12PM +0200, Vitor Sessak wrote:
>> Michael Niedermayer wrote:
>>> On Sat, Jun 21, 2008 at 07:53:09AM +0200, Vitor Sessak wrote:
>>>> Michael Niedermayer wrote:
>>>>> On Wed, Jun 04, 2008 at 08:18:10PM +0200, Vitor Sessak wrote:
>>>>>> Michael Niedermayer wrote:
>>>>>>> On Wed, May 28, 2008 at 09:23:02PM +0200, Vitor Sessak wrote:
>>>>>>>> Michael Niedermayer wrote:
>>>>>>>>> On Wed, May 28, 2008 at 06:56:45PM +0200, Vitor Sessak wrote:
>>>>>>>>>> Michael Niedermayer wrote:
>>>>>>>>>>> On Tue, May 27, 2008 at 09:16:09PM +0200, Vitor Sessak wrote:
>>>>>>>>>>>> Michael Niedermayer wrote:
>>>>>>>>>>>>> On Sun, May 25, 2008 at 07:11:52PM +0200, Vitor Sessak wrote:
>>>>>>>>>>>>>> Michael Niedermayer wrote:
>>>>>>>>>>>>>>> On Sun, May 25, 2008 at 06:05:15PM +0200, Vitor Sessak wrote:
>>>>>>>>>>>>>>> [...]
>>>>>>>>>>>>>>>>>> ok
>>>>>>>>>>>>>>>>> One more...
>>>>>>>>>>>>>>>> ... and some more cleanup:
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> ra144_vector_add_wav.diff: Make add_wav() receive a vector
>>>>>>>>>>>>>>>> instead of three integers
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> ra144_params_dec2.diff: Do not calculate anything based in l,
>>>>>>>>>>>>>>>> it is unrolled in the loop anyway
>>>>>>>>>>>>>>> ok
>>>>>>>>>>>>>> Now s/(unsigned) short/(u)int16_t.
>>>>>>>>>>>>> ok
>>>>>>>>>>>> Next one. dec2() interpolates the block coefficients from the
>>>>>>>>>>>> previous one and fall back to a block-dependent schema if the
>>>>>>>>>>>> interpolation results in an unstable filter...
>>>>>>>>>>> [...]
>>>>>>>>>>>> + // Interpolate block coefficients from the this frame forth
>>>>>>>>>>>> block and
>>>>>>>>>>>> + // last frame forth block
>>>>>>>>>>>> for (x=0; x<30; x++)
>>>>>>>>>>>> - decsp[x] = (a * inp[x] + b * inp2[x]) >> 2;
>>>>>>>>>>>> + decsp[x] = (a * ractx->lpc_coef[x] + b *
>>>>>>>>>>>> ractx->lpc_coef_old[x])>> 2;
>>>>>>>>>>> ff_acelp_weighted_vector_sum()
>>>>>>>>>> Ok, but to do that I need to use int16_t. So I propose to apply my
>>>>>>>>>> original patch and then the attached one.
>>>>>>>>> hmm, ok
>>>>>>>> Done. Now remove the dec1() function (that was memcpy + 1 line of
>>>>>>>> code). As a side effect, it removes the need of a memcpy (the dec1()
>>>>>>>> call at decode_frame()).
>>>>>>> ok
>>>>>> Now the first patch (ra144_rescale_energy.diff) split the energy
>>>>>> rescaling out of the rms() function. The next patch remove *lpc_refl from
>>>>>> the context, since the only thing needed from the last frame is the non
>>>>>> rescaled output of rms().
>>>>> ok
>>>> Now, I'm almost finished with this. Two things remains:
>>>>
>>>> 1- When decoding a ra144 encoded file, ffmpeg produces lots of "Multiple
>>>> frames in a packet from stream 0" (see
>>>> http://fate.multimedia.cx/index.php?test_result=1911120 for an example).
>>>> This is because the decoder receives a 1000 byte sample and decode only 20
>>>> bytes. The attached patch fix this (it decode all the 50 blocks).
>>> wrong solution, we need a AVParser that splits the 1000bytes in 20byte
>>> packets. A generic one that works based on block_align might be usefull
>>> for other cases as well ...
>> I'll see it later.
>>
>>>
>>>> 2- There are lots of unused table entries. Ok to remove them or do you
>>>> thing they can useful for anything (another codec?)?
>>> remove
>>>
>>>
>>>> Finally, if there is anything else you don't like about ra144.{c,h}, now is
>>>> the time to say if you want me to have a look at it...
>>> 1st pass review of ra144.c is below :)
>>> (yes you regret now that you asked, i know ...)
>> It was my masochistic side that made that question =)
>
> good, please remind me to do another review when you are finished
> with that one :)
>
> With your help ra144.c will soon be pretty nice and clean, next comes
> ra288.c i assume :)
Why not? By the way, can I just clean it and you flame me in -cvslogs?
>
>
> [...]
>>>> static void lpc_filter(const int16_t *lpc_coefs, const int16_t *adapt_coef,
>>>> void *out, int *statbuf, int len)
>>>> {
>>>> int x, i;
>>>> uint16_t work[50];
>>>> int16_t *ptr = work;
>>>>
>>>> memcpy(work, statbuf,20);
>>> 10*sizeof(int16_t)
>>>
>>>
>>>> memcpy(work + 10, adapt_coef, len * 2);
>>>>
>>>> for (i=0; i<len; i++) {
>>>> int sum = 0;
>>>> int new_val;
>>>>
>>>> for(x=0; x<10; x++)
>>>> sum += lpc_coefs[9-x] * ptr[x];
>>>>
>>>> sum >>= 12;
>>>>
>>>> new_val = ptr[10] - sum;
>>>>
>>>> if (new_val < -32768 || new_val > 32767) {
>>>> memset(out, 0, len * 2);
>>>> memset(statbuf, 0, 20);
>>>> return;
>>>> }
>>>>
>>>> ptr[10] = new_val;
>>>> ptr++;
>>>> }
>>>>
>>>> memcpy(out, work+10, len * 2);
>>>> memcpy(statbuf, work + 40, 20);
>>>> }
>>> duplicate of ff_acelp_lp_synthesis_filter)(
>> No, they are slightly different (ff_acelp_lp_synthesis_filter use the
>> last n input values to predict out[n]
>
> no it does not, it uses filter_length which you could set to 10
Ok, I did it (patch attached). But there were two things I was not happy
with. First, ff_acelp_lp_synthesis_filter completely ignores
filter_coeffs[0] and uses filter_coeffs[1..filter_length]. I have no
idea why it is like that, but I think the correct inner loop would be
> for(i=0; i<filter_length; i++)
> sum -= filter_coeffs[i] * out[n-i-1];
where now filter_length is 10 (and not 11!) for the 10th order filter.
Also, I had to change the initial value of sum in
ff_acelp_lp_synthesis_filter(). I don't know if I can just change it or
if I should add a "round" boolean to the parameters of
ff_acelp_lp_synthesis_filter...
One last thing, can I claim copyright on this file?
-Vitor
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