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

[compn]

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 ^^

>+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.

>+    /* 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?

>+
>+    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