upload sdk

git-svn-id: svn://kolibrios.org@4349 a494cfbc-eb01-0410-851d-a64ba20cac60
2013-12-15 08:09:20 +00:00
parent 6c6781f799
commit 754f9336f0
5801 changed files with 1688660 additions and 0 deletions
--- a/contrib/sdk/sources/ffmpeg/libavformat/loasdec.c
+++ b/contrib/sdk/sources/ffmpeg/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,
+};