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authorwm4 <wm4@nowhere>2013-07-11 19:15:09 +0200
committerwm4 <wm4@nowhere>2013-07-11 19:15:09 +0200
commit23e303859aa93572f00b17e3b2bc0a552ad7c348 (patch)
tree155f4f429170b770e54bdd630886830168e220bc /core
parent7a4f9cc4d2509a0d9afa68ea53ed97b72525d001 (diff)
downloadmpv-23e303859aa93572f00b17e3b2bc0a552ad7c348.tar.bz2
mpv-23e303859aa93572f00b17e3b2bc0a552ad7c348.tar.xz
mplayer: fix incorrect audio sync after format changes
This is not directly related to the handling of format changes itself, but playing audio normally after the change. This was broken: the output byte rate was not recalculated, so audio-video sync was simply broken. Fix this by calculating the byte rate on the fly, instead of storing it in sh_audio. Format changes are relatively common (switches between stereo and 5.1 in TV recordings), so this fixes a somewhat critical bug.
Diffstat (limited to 'core')
-rw-r--r--core/mplayer.c8
1 files changed, 6 insertions, 2 deletions
diff --git a/core/mplayer.c b/core/mplayer.c
index 87bef0e3c5..7b3cf65026 100644
--- a/core/mplayer.c
+++ b/core/mplayer.c
@@ -1668,6 +1668,10 @@ static double written_audio_pts(struct MPContext *mpctx)
sh_audio_t *sh_audio = mpctx->sh_audio;
if (!sh_audio)
return MP_NOPTS_VALUE;
+
+ double bps = sh_audio->channels.num * sh_audio->samplerate *
+ sh_audio->samplesize;
+
// first calculate the end pts of audio that has been output by decoder
double a_pts = sh_audio->pts;
if (a_pts == MP_NOPTS_VALUE)
@@ -1676,13 +1680,13 @@ static double written_audio_pts(struct MPContext *mpctx)
// sh_audio->pts is the timestamp of the latest input packet with
// known pts that the decoder has decoded. sh_audio->pts_bytes is
// the amount of bytes the decoder has written after that timestamp.
- a_pts += sh_audio->pts_bytes / (double) sh_audio->o_bps;
+ a_pts += sh_audio->pts_bytes / bps;
// Now a_pts hopefully holds the pts for end of audio from decoder.
// Subtract data in buffers between decoder and audio out.
// Decoded but not filtered
- a_pts -= sh_audio->a_buffer_len / (double)sh_audio->o_bps;
+ a_pts -= sh_audio->a_buffer_len / bps;
// Data buffered in audio filters, measured in bytes of "missing" output
double buffered_output = af_calc_delay(sh_audio->afilter);