[FFmpeg-devel] [PATCH] atrac1 decoder and aea demuxer

[Benjamin Larsson]

compn wrote:
> On Tue, 01 Sep 2009 22:08:00 +0200, Benjamin Larsson wrote:
>> +static float            sf_tab[64];
>> +DECLARE_ALIGNED_16(static float,   qmf_window[48]);
>> +DECLARE_ALIGNED_16(static float, short_window[ 32]);
>> +DECLARE_ALIGNED_16(static float,   mid_window[128]);
>> +DECLARE_ALIGNED_16(static float,  long_window[256]);
> 
> missing a space or needs a space to prettyprint ^^

Fixed.

> 
>> +static int at1_parse_block_size_mode(GetBitContext* gb, int bsm[AT1_QMF_BANDS])
>> +{
>> +    int bsm_tmp;
>> +
>> +    /* low band */
>> +    bsm_tmp = get_bits(gb, 2);
>> +    if (bsm_tmp&1)
>> +        return -1;
>> +    bsm[IDX_LOW_BAND] = 2 - bsm_tmp;
>> +
>> +    /* middle band */
>> +    bsm_tmp = get_bits(gb, 2);
>> +    if (bsm_tmp&1)
>> +        return -1;
>> +    bsm[IDX_MID_BAND] = 2 - bsm_tmp;
>> +
>> +    /* high band */
>> +    bsm_tmp = get_bits(gb, 2);
>> +    if (bsm_tmp != 0 && bsm_tmp != 3)
>> +        return -1;
>> +    bsm[IDX_HIGH_BAND] = 3 - bsm_tmp;
>> +
>> +    skip_bits(gb, 2);
>> +    return 0;
>> +}
> 
> can some of this can be factored? low and mid look the same.

Well I think this is more clear this way.


> 
>> +    /* First packet starts at 0x800 */
>> +    url_fskip(s->pb, 264);
>> +    st->codec->channels = get_byte(s->pb);
>> +    url_fskip(s->pb, 1783);
>> +
>> +
>> +    st->codec->codec_type     = CODEC_TYPE_AUDIO;
>> +    st->codec->codec_id       = CODEC_ID_ATRAC1;
>> +    st->codec->sample_rate    = 44100;
>> +    st->codec->bit_rate       = 292000;
> 
> only 44100 is supported?

Yes.

> 
>> +
>> +    if (!((st->codec->channels == 1) || (st->codec->channels == 2)))
>> +        return -1;
> 
> no error message ?
> 
> ignore me if i ask stupid questions, just a quick review.
> 
> -compn


MvH
Benjamin Larsson