oggdec: simplify start time calculation code.

[ffmpeg] / libavformat / mp3dec.c

diff --git a/libavformat/mp3dec.c b/libavformat/mp3dec.c
index 61405f71b73fc47f6c1b4211179a0d9530bfcbf2..aa936c0e98cdf0d15927ea744424c982929feef2 100644 (file)
--- a/libavformat/mp3dec.c
+++ b/libavformat/mp3dec.c
@@ -31,6 +31,8 @@
 
 typedef struct {
     int64_t filesize;
+    int start_pad;
+    int end_pad;
 } MP3Context;
 
 /* mp3 read */
@@ -82,6 +84,7 @@ static int mp3_read_probe(AVProbeData *p)
  */
 static int mp3_parse_vbr_tags(AVFormatContext *s, AVStream *st, int64_t base)
 {
+    MP3Context *mp3 = s->priv_data;
     uint32_t v, spf;
     unsigned frames = 0; /* Total number of frames in file */
     unsigned size = 0; /* Total number of bytes in the stream */
@@ -107,6 +110,20 @@ static int mp3_parse_vbr_tags(AVFormatContext *s, AVStream *st, int64_t base)
             frames = avio_rb32(s->pb);
         if(v & 0x2)
             size = avio_rb32(s->pb);
+        if(v & 4)
+            avio_skip(s->pb, 100);
+        if(v & 8)
+            avio_skip(s->pb, 4);
+
+        v = avio_rb32(s->pb);
+        if(v == MKBETAG('L', 'A', 'M', 'E') || v == MKBETAG('L', 'a', 'v', 'f')) {
+            avio_skip(s->pb, 21-4);
+            v= avio_rb24(s->pb);
+            mp3->start_pad = v>>12;
+            mp3->  end_pad = v&4095;
+            st->skip_samples = mp3->start_pad + 528 + 1;
+            av_log(s, AV_LOG_DEBUG, "pad %d %d\n", mp3->start_pad, mp3->  end_pad);
+        }
     }
 
     /* Check for VBRI tag (always 32 bytes after end of mpegaudio header) */
@@ -150,7 +167,7 @@ static int mp3_read_header(AVFormatContext *s)
 
     st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
     st->codec->codec_id = CODEC_ID_MP3;
-    st->need_parsing = AVSTREAM_PARSE_FULL;
+    st->need_parsing = AVSTREAM_PARSE_FULL_RAW;
     st->start_time = 0;
 
     // lcm of all mp3 sample rates
@@ -179,7 +196,6 @@ static int mp3_read_packet(AVFormatContext *s, AVPacket *pkt)
     MP3Context *mp3 = s->priv_data;
     int ret, size;
     int64_t pos;
-    //    AVStream *st = s->streams[0];
 
     size= MP3_PACKET_SIZE;
     pos = avio_tell(s->pb);
@@ -187,15 +203,15 @@ static int mp3_read_packet(AVFormatContext *s, AVPacket *pkt)
         size= FFMIN(size, mp3->filesize - pos);
 
     ret= av_get_packet(s->pb, pkt, size);
-
-    pkt->flags &= ~AV_PKT_FLAG_CORRUPT;
-    pkt->stream_index = 0;
     if (ret <= 0) {
         if(ret<0)
             return ret;
         return AVERROR_EOF;
     }
 
+    pkt->flags &= ~AV_PKT_FLAG_CORRUPT;
+    pkt->stream_index = 0;
+
     if (ret >= ID3v1_TAG_SIZE &&
         memcmp(&pkt->data[ret - ID3v1_TAG_SIZE], "TAG", 3) == 0)
         ret -= ID3v1_TAG_SIZE;
@@ -206,13 +222,24 @@ static int mp3_read_packet(AVFormatContext *s, AVPacket *pkt)
     return ret;
 }
 
+static int read_seek(AVFormatContext *s, int stream_index, int64_t timestamp, int flags)
+{
+    MP3Context *mp3 = s->priv_data;
+    AVStream *st = s->streams[stream_index];
+
+    st->skip_samples = timestamp <= 0 ? mp3->start_pad + 528 + 1 : 0;
+
+    return -1;
+}
+
 AVInputFormat ff_mp3_demuxer = {
     .name           = "mp3",
-    .long_name      = NULL_IF_CONFIG_SMALL("MPEG audio layer 2/3"),
+    .long_name      = NULL_IF_CONFIG_SMALL("MP2/3 (MPEG audio layer 2/3)"),
     .priv_data_size = sizeof(MP3Context),
     .read_probe     = mp3_read_probe,
     .read_header    = mp3_read_header,
     .read_packet    = mp3_read_packet,
+    .read_seek      = read_seek,
     .flags          = AVFMT_GENERIC_INDEX,
     .extensions     = "mp2,mp3,m2a", /* XXX: use probe */
 };