ALSA demuxer: use av_gettime and a timefilter.

The PTS for captured audio was measured using snd_pcm_htimestamp.

snd_pcm_htimestamp hangs when the input is a dsnoop plugin.

Furthermore, at some point, snd_pcm_htimestamp started returning monotonic
timestamps rather than wall clock timestamps, in most but not all
situations.
Monotonic timestamps are fine, but ffmpeg uses wall clock timestamps
everywhere else, and we have no API to inform the user which kind of
timestamps it is.

A separate snd_pcm_htimestamp is only slightly less accurate than
snd_pcm_htimestamp: the standard deviation for the difference between two
consecutive timestamps is (on my hardware):
- ~13 µs with snd_pcm_htimestamp;
- ~35 µs with av_gettime;
-  ~5 µs with av_gettime and a timefilter.

1 files changed, 1 insertions, 0 deletions

diff --git a/libavdevice/alsa-audio-common.c b/libavdevice/alsa-audio-common.c
index 8c5be3c864..6188721f48 100644
--- a/libavdevice/alsa-audio-common.c
+++ b/libavdevice/alsa-audio-common.c
@@ -316,6 +316,7 @@ av_cold int ff_alsa_close(AVFormatContext *s1)
     AlsaData *s = s1->priv_data;
 
     av_freep(&s->reorder_buf);
+    ff_timefilter_destroy(s->timefilter);
     snd_pcm_close(s->h);
     return 0;
 }