move ADTS AAC demuxer to its own file

Originally committed as revision 24967 to svn://svn.ffmpeg.org/ffmpeg/trunk

1 files changed, 96 insertions, 0 deletions

diff --git a/libavformat/aacdec.c b/libavformat/aacdec.c
new file mode 100644
index 0000000000..8737e56409
--- /dev/null
+++ b/libavformat/aacdec.c
@@ -0,0 +1,96 @@
+/*
+ * raw ADTS AAC demuxer
+ * Copyright (c) 2008 Michael Niedermayer <michaelni@gmx.at>
+ * Copyright (c) 2009 Robert Swain ( rob opendot cl )
+ *
+ * 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
+ */
+
+#include "libavutil/intreadwrite.h"
+#include "avformat.h"
+#include "raw.h"
+#include "id3v2.h"
+#include "id3v1.h"
+
+
+static int adts_aac_probe(AVProbeData *p)
+{
+    int max_frames = 0, first_frames = 0;
+    int fsize, frames;
+    uint8_t *buf0 = p->buf;
+    uint8_t *buf2;
+    uint8_t *buf;
+    uint8_t *end = buf0 + p->buf_size - 7;
+
+    if (ff_id3v2_match(buf0, ID3v2_DEFAULT_MAGIC)) {
+        buf0 += ff_id3v2_tag_len(buf0);
+    }
+    buf = buf0;
+
+    for(; buf < end; buf= buf2+1) {
+        buf2 = buf;
+
+        for(frames = 0; buf2 < end; frames++) {
+            uint32_t header = AV_RB16(buf2);
+            if((header&0xFFF6) != 0xFFF0)
+                break;
+            fsize = (AV_RB32(buf2+3)>>13) & 0x8FFF;
+            if(fsize < 7)
+                break;
+            buf2 += fsize;
+        }
+        max_frames = FFMAX(max_frames, frames);
+        if(buf == buf0)
+            first_frames= frames;
+    }
+    if   (first_frames>=3) return AVPROBE_SCORE_MAX/2+1;
+    else if(max_frames>500)return AVPROBE_SCORE_MAX/2;
+    else if(max_frames>=3) return AVPROBE_SCORE_MAX/4;
+    else if(max_frames>=1) return 1;
+    else                   return 0;
+}
+
+static int adts_aac_read_header(AVFormatContext *s,
+                                AVFormatParameters *ap)
+{
+    AVStream *st;
+
+    st = av_new_stream(s, 0);
+    if (!st)
+        return AVERROR(ENOMEM);
+
+    st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
+    st->codec->codec_id = s->iformat->value;
+    st->need_parsing = AVSTREAM_PARSE_FULL;
+
+    ff_id3v1_read(s);
+    ff_id3v2_read(s, ID3v2_DEFAULT_MAGIC);
+
+    return 0;
+}
+
+AVInputFormat aac_demuxer = {
+    "aac",
+    NULL_IF_CONFIG_SMALL("raw ADTS AAC"),
+    0,
+    adts_aac_probe,
+    adts_aac_read_header,
+    ff_raw_read_partial_packet,
+    .flags= AVFMT_GENERIC_INDEX,
+    .extensions = "aac",
+    .value = CODEC_ID_AAC,
+};