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-rw-r--r--libavformat/loasdec.c87
1 files changed, 87 insertions, 0 deletions
diff --git a/libavformat/loasdec.c b/libavformat/loasdec.c
new file mode 100644
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+++ b/libavformat/loasdec.c
@@ -0,0 +1,87 @@
+/*
+ * LOAS AudioSyncStream demuxer
+ * Copyright (c) 2008 Michael Niedermayer <michaelni@gmx.at>
+ *
+ * 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 "libavutil/internal.h"
+#include "avformat.h"
+#include "internal.h"
+#include "rawdec.h"
+
+static int loas_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 - 3;
+ buf = buf0;
+
+ for(; buf < end; buf= buf2+1) {
+ buf2 = buf;
+
+ for(frames = 0; buf2 < end; frames++) {
+ uint32_t header = AV_RB24(buf2);
+ if((header >> 13) != 0x2B7)
+ break;
+ fsize = (header & 0x1FFF) + 3;
+ 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_EXTENSION+1;
+ else if(max_frames>100)return AVPROBE_SCORE_EXTENSION;
+ else if(max_frames>=3) return AVPROBE_SCORE_EXTENSION / 2;
+ else return 0;
+}
+
+static int loas_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;
+
+ //LCM of all possible AAC sample rates
+ avpriv_set_pts_info(st, 64, 1, 28224000);
+
+ return 0;
+}
+
+AVInputFormat ff_loas_demuxer = {
+ .name = "loas",
+ .long_name = NULL_IF_CONFIG_SMALL("LOAS AudioSyncStream"),
+ .read_probe = loas_probe,
+ .read_header = loas_read_header,
+ .read_packet = ff_raw_read_partial_packet,
+ .flags= AVFMT_GENERIC_INDEX,
+ .raw_codec_id = AV_CODEC_ID_AAC_LATM,
+};