After seeking, determine the offset of the next frame in the decrypted
buffer by scanning the first few bytes for a valid mp3 header.
This significantly improves the listening experience for audio content
with untypical encoding.
---
This is a refinement of an earlier patch iteration, according to lessons
learned and discussions on the mailing list: Scan a limited range to find
the first shifted frame only, check for a very specific bit pattern, and
add extra checks and guards for better code maintainability.
Unfortunately, this variant violates the architectural layering between
demuxer and codec. But I did some more testing with untypical encodings,
where the current estimation mechanism fails, and the resulting audio on
seek was just too horribly annoying.
I believe the better listening experience outweighs the architectural
uglyness, so this should be in the official code base. But if you think
otherwise, just let me know, and I will keep this a private patch.
libavformat/aadec.c | 45 ++++++++++++++++++++++++++++-----------------
1 file changed, 28 insertions(+), 17 deletions(-)
diff --git a/libavformat/aadec.c b/libavformat/aadec.c
index d83f283ffe..9b1495c218 100644
--- a/libavformat/aadec.c
+++ b/libavformat/aadec.c
@@ -37,7 +37,7 @@
#define TEA_BLOCK_SIZE 8
#define CHAPTER_HEADER_SIZE 8
#define TIMEPREC 1000
-#define MP3_FRAME_SIZE 104
+#define MP3_FRAME_SIZE 105
typedef struct AADemuxContext {
AVClass *class;
@@ -51,7 +51,7 @@ typedef struct AADemuxContext {
int64_t current_chapter_size;
int64_t content_start;
int64_t content_end;
- int seek_offset;
+ int did_seek;
} AADemuxContext;
static int get_second_size(char *codec_name)
@@ -179,7 +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)
+ // encoded audio frame is MP3_FRAME_SIZE bytes (-1 without padding)
} else if (!strcmp(codec_name, "acelp85")) {
st->codecpar->codec_id = AV_CODEC_ID_SIPR;
st->codecpar->block_align = 19;
@@ -231,7 +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;
+ c->did_seek = 0;
return 0;
}
@@ -244,7 +244,7 @@ static int aa_read_packet(AVFormatContext *s, AVPacket *pkt)
int trailing_bytes;
int blocks;
uint8_t buf[MAX_CODEC_SECOND_SIZE * 2];
- int written = 0;
+ int written = 0, offset = 0;
int ret;
AADemuxContext *c = s->priv_data;
uint64_t pos = avio_tell(s->pb);
@@ -297,16 +297,33 @@ static int aa_read_packet(AVFormatContext *s, AVPacket
*pkt)
if (c->current_chapter_size <= 0)
c->current_chapter_size = 0;
- if (c->seek_offset > written)
- c->seek_offset = 0; // ignore wrong estimate
+ if (c->did_seek) {
+ c->did_seek = 0;
+
+ if (s->streams[0]->codecpar->codec_id == AV_CODEC_ID_MP3 && written >=
MP3_FRAME_SIZE + 3) {
+ for (offset = 0; offset < MP3_FRAME_SIZE; ++offset) {
+ // find mp3 header: sync, v2, layer3, no crc, 32kbps, 22kHz,
mono
+ if ((buf[offset + 0] ) == 0xff &&
+ (buf[offset + 1] ) == 0xf3 &&
+ (buf[offset + 2] & 0xfc) == 0x40 &&
+ (buf[offset + 3] & 0xf0) == 0xc0)
+ break;
+ }
+ if (offset == MP3_FRAME_SIZE) offset = 0; // not found, just use
as is
+ }
+
+ ff_update_cur_dts(s, s->streams[0],
+ (pos + offset - c->content_start - CHAPTER_HEADER_SIZE *
(c->chapter_idx - 1))
+ * TIMEPREC);
+ }
- ret = av_new_packet(pkt, written - c->seek_offset);
+ if (offset > written) offset = 0;
+ ret = av_new_packet(pkt, written - offset);
if (ret < 0)
return ret;
- memcpy(pkt->data, buf + c->seek_offset, written - c->seek_offset);
+ memcpy(pkt->data, buf + offset, written - offset);
pkt->pos = pos;
- c->seek_offset = 0;
return 0;
}
@@ -349,13 +366,7 @@ static int aa_read_seek(AVFormatContext *s,
c->current_codec_second_size = c->codec_second_size;
c->current_chapter_size = chapter_size - chapter_pos;
c->chapter_idx = 1 + chapter_idx;
-
- // 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);
+ c->did_seek = 1;
return 1;
}
--
2.14.3 (Apple Git-98)
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