This makes using the generic indexing code with mp3 easier at a later
point.
---
libavformat/avformat.h | 8 ++++++++
libavformat/utils.c | 4 ++++
2 files changed, 12 insertions(+)
diff --git a/libavformat/avformat.h b/libavformat/avformat.h
index 514e646..93cfb20 100644
--- a/libavformat/avformat.h
+++ b/libavformat/avformat.h
@@ -1083,6 +1083,14 @@ typedef struct AVStream {
int skip_samples;
/**
+ * The number of samples that should be skipped from the start of the
stream
+ * (i.e. the timestamp of the first sample that should be actually output).
+ * Intended for use with formats such as mp3 with ad-hoc gapless audio
+ * support. It is assumed that negative timestamps do not exist.
+ */
+ int64_t start_skip_samples;
+
+ /**
* If not 0, the first audio sample that should be discarded from the
stream.
* This is broken by design (needs global sample count), but can't be
* avoided for broken by design formats such as mp3 with ad-hoc gapless
diff --git a/libavformat/utils.c b/libavformat/utils.c
index 950b3c6..2f6122d 100644
--- a/libavformat/utils.c
+++ b/libavformat/utils.c
@@ -1416,6 +1416,8 @@ static int read_frame_internal(AVFormatContext *s,
AVPacket *pkt)
sample < st->last_discard_sample)
discard_padding = FFMIN(end_sample - st->first_discard_sample,
duration);
}
+ if (st->start_skip_samples && (pkt->pts == 0 || pkt->pts ==
RELATIVE_TS_BASE))
+ st->skip_samples = st->start_skip_samples;
if (st->skip_samples || discard_padding) {
uint8_t *p = av_packet_new_side_data(pkt,
AV_PKT_DATA_SKIP_SAMPLES, 10);
if (p) {
@@ -1645,6 +1647,8 @@ void ff_read_frame_flush(AVFormatContext *s)
if (s->internal->inject_global_side_data)
st->inject_global_side_data = 1;
+
+ st->skip_samples = 0;
}
}
--
2.1.4
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