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audio: interpolate audio timestamps

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Deal with jittering Matroska crap timestamps. This reuses the mechanism
that is needed for frames without PTS, and adds a heuristic to it. If
the interpolated timestamp is less than 1ms away from the real one, it
might be due to Matroska timestamp rounding (or other file formats with
such rounding, or files remuxed from Matroska).

While there actually isn't much of a need to do this (audio PTS
jittering by such a low amount doesn't negatively influence much), it
helps with identifying jitter from other sources.
This commit is contained in:
wm4 2015-11-08 18:05:16 +01:00
parent d91434756b
commit 0ff3ffb2be
1 changed files with 12 additions and 2 deletions
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14
audio/decode/dec_audio.c
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@ -172,8 +172,18 @@ static int decode_new_frame(struct dec_audio *da)
if (da->waiting) {
if (da->waiting->pts != MP_NOPTS_VALUE) {
da->pts = da->waiting->pts;
da->pts_offset = 0;
if (da->pts != MP_NOPTS_VALUE) {
da->pts += da->pts_offset / (double)da->waiting->rate;
da->pts_offset = 0;
}
// Keep the interpolated timestamp if it doesn't deviate more
// than 1 ms from the real one. (MKV rounded timestamps.)
if (da->pts == MP_NOPTS_VALUE || da->pts_offset != 0 ||
fabs(da->pts - da->waiting->pts) > 0.001)
{
da->pts = da->waiting->pts;
da->pts_offset = 0;
}
}
da->pts_offset += da->waiting->samples;
da->decode_format = *da->waiting;