[FFmpeg-cvslog] aadec: improve seeking in mp3 content

Karsten Otto git at videolan.org
Sat Jul 14 23:17:56 EEST 2018


ffmpeg | branch: master | Karsten Otto <ottoka at posteo.de> | Sat Jul 14 11:20:46 2018 +0200| [c5329d64b1264ef1431732aad6f5b08d0c4b55f4] | committer: Michael Niedermayer

aadec: improve seeking in mp3 content

MP3 frames may not be aligned to aa chunk boundaries. When seeking,
calculate the expected frame offset in the target chunk. Adjust the
timestamp and truncate the next packet accordingly.

This solution works for the majority of tested audio material. For
some rare encodings with mp3 padding or embedded id3 tags, it will
mispredict the correct offset, and at worst skip an extra frame.

Signed-off-by: Michael Niedermayer <michael at niedermayer.cc>

> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=c5329d64b1264ef1431732aad6f5b08d0c4b55f4
---

 libavformat/aadec.c | 19 ++++++++++++++++---
 1 file changed, 16 insertions(+), 3 deletions(-)

diff --git a/libavformat/aadec.c b/libavformat/aadec.c
index 17ad20686b..d83f283ffe 100644
--- a/libavformat/aadec.c
+++ b/libavformat/aadec.c
@@ -37,6 +37,7 @@
 #define TEA_BLOCK_SIZE 8
 #define CHAPTER_HEADER_SIZE 8
 #define TIMEPREC 1000
+#define MP3_FRAME_SIZE 104
 
 typedef struct AADemuxContext {
     AVClass *class;
@@ -50,6 +51,7 @@ typedef struct AADemuxContext {
     int64_t current_chapter_size;
     int64_t content_start;
     int64_t content_end;
+    int seek_offset;
 } AADemuxContext;
 
 static int get_second_size(char *codec_name)
@@ -177,6 +179,7 @@ static int aa_read_header(AVFormatContext *s)
         st->codecpar->sample_rate = 22050;
         st->need_parsing = AVSTREAM_PARSE_FULL_RAW;
         avpriv_set_pts_info(st, 64, 8, 32000 * TIMEPREC);
+        // encoded audio frame is MP3_FRAME_SIZE bytes (+1 with padding, unlikely)
     } else if (!strcmp(codec_name, "acelp85")) {
         st->codecpar->codec_id = AV_CODEC_ID_SIPR;
         st->codecpar->block_align = 19;
@@ -228,6 +231,7 @@ static int aa_read_header(AVFormatContext *s)
     ff_update_cur_dts(s, st, 0);
     avio_seek(pb, start, SEEK_SET);
     c->current_chapter_size = 0;
+    c->seek_offset = 0;
 
     return 0;
 }
@@ -293,12 +297,16 @@ static int aa_read_packet(AVFormatContext *s, AVPacket *pkt)
     if (c->current_chapter_size <= 0)
         c->current_chapter_size = 0;
 
-    ret = av_new_packet(pkt, written);
+    if (c->seek_offset > written)
+        c->seek_offset = 0; // ignore wrong estimate
+
+    ret = av_new_packet(pkt, written - c->seek_offset);
     if (ret < 0)
         return ret;
-    memcpy(pkt->data, buf, written);
+    memcpy(pkt->data, buf + c->seek_offset, written - c->seek_offset);
     pkt->pos = pos;
 
+    c->seek_offset = 0;
     return 0;
 }
 
@@ -342,7 +350,12 @@ static int aa_read_seek(AVFormatContext *s,
     c->current_chapter_size = chapter_size - chapter_pos;
     c->chapter_idx = 1 + chapter_idx;
 
-    ff_update_cur_dts(s, s->streams[0], ch->start + chapter_pos * TIMEPREC);
+    // for unaligned frames, estimate offset of first frame in block (assume no padding)
+    if (s->streams[0]->codecpar->codec_id == AV_CODEC_ID_MP3) {
+        c->seek_offset = (MP3_FRAME_SIZE - chapter_pos % MP3_FRAME_SIZE) % MP3_FRAME_SIZE;
+    }
+
+    ff_update_cur_dts(s, s->streams[0], ch->start + (chapter_pos + c->seek_offset) * TIMEPREC);
 
     return 1;
 }



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