ffmpeg-2.8.5

git-svn-id: svn://kolibrios.org@6147 a494cfbc-eb01-0410-851d-a64ba20cac60

contrib/sdk/sources/ffmpeg/ffmpeg-2.8/libavformat/aacdec.c

@@ -0,0 +1,115 @@
+/*
+ * 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"
+#include "apetag.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) {
+                if (buf != buf0) {
+                    // Found something that isn't an ADTS header, starting
+                    // from a position other than the start of the buffer.
+                    // Discard the count we've accumulated so far since it
+                    // probably was a false positive.
+                    frames = 0;
+                }
+                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_EXTENSION + 1;
+    else if (max_frames > 100)
+        return AVPROBE_SCORE_EXTENSION;
+    else if (max_frames >= 3)
+        return AVPROBE_SCORE_EXTENSION / 2;
+    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);
+    if (s->pb->seekable &&
+        !av_dict_get(s->metadata, "", NULL, AV_DICT_IGNORE_SUFFIX)) {
+        int64_t cur = avio_tell(s->pb);
+        ff_ape_parse_tag(s);
+        avio_seek(s->pb, cur, SEEK_SET);
+    }
+
+    // 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",
+    .mime_type    = "audio/aac,audio/aacp,audio/x-aac",
+    .raw_codec_id = AV_CODEC_ID_AAC,
+};