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 _______________________________________________ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-devel