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

[Vitor Sessak]

Michael Niedermayer wrote:
> 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?

Original     wav:  967706 bytes
FFmpeg   decoder:  990764 bytes   1 second  of silence in the end
Original decoder: 1013804 bytes   3 seconds of silence in the end

Anyway, nothing of that explains the PSNR discrepancy...

-Vitor