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authorTim Redfern <tim@eclectronics.org>2013-08-26 15:10:18 +0100
committerTim Redfern <tim@eclectronics.org>2013-08-26 15:10:18 +0100
commit150c9823e71a161e97003849cf8b2f55b21520bd (patch)
tree3559c840cf403d1386708b2591d58f928c7b160d /ffmpeg1/libavformat/aacdec.c
parentb4b1e2630c95d5e6014463f7608d59dc2322a3b8 (diff)
adding ffmpeg specific version
Diffstat (limited to 'ffmpeg1/libavformat/aacdec.c')
-rw-r--r--ffmpeg1/libavformat/aacdec.c94
1 files changed, 94 insertions, 0 deletions
diff --git a/ffmpeg1/libavformat/aacdec.c b/ffmpeg1/libavformat/aacdec.c
new file mode 100644
index 0000000..7c17dd0
--- /dev/null
+++ b/ffmpeg1/libavformat/aacdec.c
@@ -0,0 +1,94 @@
+/*
+ * 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 "internal.h"
+#include "rawdec.h"
+#include "id3v1.h"
+
+
+static int adts_aac_probe(AVProbeData *p)
+{
+ int max_frames = 0, first_frames = 0;
+ int fsize, frames;
+ const uint8_t *buf0 = p->buf;
+ const uint8_t *buf2;
+ const uint8_t *buf;
+ const uint8_t *end = buf0 + p->buf_size - 7;
+
+ 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) & 0x1FFF;
+ if(fsize < 7)
+ break;
+ fsize = FFMIN(fsize, end - buf2);
+ 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)
+{
+ AVStream *st;
+
+ st = avformat_new_stream(s, NULL);
+ if (!st)
+ return AVERROR(ENOMEM);
+
+ st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
+ st->codec->codec_id = s->iformat->raw_codec_id;
+ st->need_parsing = AVSTREAM_PARSE_FULL_RAW;
+
+ ff_id3v1_read(s);
+
+ //LCM of all possible ADTS sample rates
+ avpriv_set_pts_info(st, 64, 1, 28224000);
+
+ return 0;
+}
+
+AVInputFormat ff_aac_demuxer = {
+ .name = "aac",
+ .long_name = NULL_IF_CONFIG_SMALL("raw ADTS AAC (Advanced Audio Coding)"),
+ .read_probe = adts_aac_probe,
+ .read_header = adts_aac_read_header,
+ .read_packet = ff_raw_read_partial_packet,
+ .flags = AVFMT_GENERIC_INDEX,
+ .extensions = "aac",
+ .raw_codec_id = AV_CODEC_ID_AAC,
+};