lavc: export Dirac parsing API used by the ogg demuxer as public

[ffmpeg] / libavformat / aacdec.c

diff --git a/libavformat/aacdec.c b/libavformat/aacdec.c
index 8737e56409871ce17c8a0952f7d1988f412b55ae..566c3e78b916c9a5a4f15e780c966787fd5672ff 100644 (file)
--- a/libavformat/aacdec.c
+++ b/libavformat/aacdec.c
@@ -3,30 +3,29 @@
  * Copyright (c) 2008 Michael Niedermayer <michaelni@gmx.at>
  * Copyright (c) 2009 Robert Swain ( rob opendot cl )
  *
- * This file is part of FFmpeg.
+ * This file is part of Libav.
  *
- * FFmpeg is free software; you can redistribute it and/or
+ * Libav 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,
+ * Libav 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
+ * License along with Libav; 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 "internal.h"
+#include "rawdec.h"
 #include "id3v1.h"
 
-
 static int adts_aac_probe(AVProbeData *p)
 {
     int max_frames = 0, first_frames = 0;
@@ -36,61 +35,73 @@ static int adts_aac_probe(AVProbeData *p)
     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) {
+    for (; buf < end; buf = buf2 + 1) {
         buf2 = buf;
 
-        for(frames = 0; buf2 < end; frames++) {
+        for (frames = 0; buf2 < end; frames++) {
             uint32_t header = AV_RB16(buf2);
-            if((header&0xFFF6) != 0xFFF0)
+            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) & 0x8FFF;
-            if(fsize < 7)
+            }
+            fsize = (AV_RB32(buf2 + 3) >> 13) & 0x1FFF;
+            if (fsize < 7)
                 break;
             buf2 += fsize;
         }
         max_frames = FFMAX(max_frames, frames);
-        if(buf == buf0)
-            first_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;
+
+    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,
-                                AVFormatParameters *ap)
+static int adts_aac_read_header(AVFormatContext *s)
 {
     AVStream *st;
 
-    st = av_new_stream(s, 0);
+    st = avformat_new_stream(s, NULL);
     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;
+    st->codec->codec_id   = s->iformat->raw_codec_id;
+    st->need_parsing      = AVSTREAM_PARSE_FULL;
 
     ff_id3v1_read(s);
-    ff_id3v2_read(s, ID3v2_DEFAULT_MAGIC);
+
+    // LCM of all possible ADTS sample rates
+    avpriv_set_pts_info(st, 64, 1, 28224000);
 
     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,
+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,
 };