Initial commit of the MD studio demuxer, not hooked up yet

Originally committed as revision 19812 to svn://svn.ffmpeg.org/ffmpeg/trunk

1 files changed, 110 insertions, 0 deletions

diff --git a/libavformat/aea.c b/libavformat/aea.c
new file mode 100644
index 0000000000..f0e200d3fe
--- /dev/null
+++ b/libavformat/aea.c
@@ -0,0 +1,110 @@
+/*
+ * MD STUDIO audio demuxer
+ *
+ * Copyright (c) 2009 Benjamin Larsson
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file libavformat/aea.c
+ */
+
+#include "avformat.h"
+#include "raw.h"
+#include "libavutil/intreadwrite.h"
+
+#define AT1_SU_SIZE     212
+
+static int aea_read_probe(AVProbeData *p)
+{
+    if (p->buf_size <= 2048+212)
+        return 0;
+
+    /* Magic is '00 08 00 00' in Little Endian*/
+    if(AV_RL32(p->buf)==0x800) {
+        int bsm_s, bsm_e, inb_s, inb_e;
+        bsm_s = p->buf[2048];
+        inb_s = p->buf[2048+1];
+        inb_e = p->buf[2048+210];
+        bsm_e = p->buf[2048+211];
+
+
+        /* Check so that the redundant bsm bytes and info bytes are valid
+         * the block size mode bytes have to be the same
+         * the info bytes have to be the same
+         * the block size mode and info byte can't be the same
+         */
+        if (bsm_s == bsm_e && inb_s == inb_e && bsm_s != inb_s)
+            return AVPROBE_SCORE_MAX/2;
+    }
+    return 0;
+}
+
+static int aea_read_header(AVFormatContext *s,
+                           AVFormatParameters *ap)
+{
+    AVStream *st = av_new_stream(s, 0);
+    if (!st)
+        return AVERROR(ENOMEM);
+
+    /* Parse the amount of channels and skip to pos 2048(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;
+
+    if (st->codec->channels != 1 && st->codec->channels != 2) {
+        av_log(s,AV_LOG_ERROR,"Channels %d not supported!\n",st->codec->channels);
+        return -1;
+    }
+
+    st->codec->channel_layout = (st->codec->channels==1) ? CH_LAYOUT_MONO : CH_LAYOUT_STEREO;
+
+    st->codec->block_align = AT1_SU_SIZE * st->codec->channels;
+    return 0;
+}
+
+static int aea_read_packet(AVFormatContext *s, AVPacket *pkt)
+{
+    int ret = av_get_packet(s->pb, pkt, s->streams[0]->codec->block_align);
+
+    pkt->stream_index = 0;
+    if (ret <= 0)
+        return AVERROR(EIO);
+
+    return ret;
+}
+
+AVInputFormat aea_demuxer = {
+    "aea",
+    NULL_IF_CONFIG_SMALL("MD STUDIO audio"),
+    0,
+    aea_read_probe,
+    aea_read_header,
+    aea_read_packet,
+    0,
+    pcm_read_seek,
+    .flags= AVFMT_GENERIC_INDEX,
+    .extensions = "aea",
+};
+