mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-22 00:06:36 +00:00
pulsesrc: improve clock handling
Post the notify outside of the pa_lock to avoid a deadlock caused by basesrc calling get_time with the object lock. Reset the clock on connect. Post clock-lost and clock-provide messages. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=673977 Conflicts: ext/pulse/pulsesrc.c
This commit is contained in:
parent
3bcae19398
commit
456c8e8205
1 changed files with 21 additions and 6 deletions
|
@ -295,13 +295,12 @@ gst_pulsesrc_init (GstPulseSrc * pulsesrc)
|
|||
GST_AUDIO_BASE_SRC_SLAVE_SKEW);
|
||||
|
||||
/* override with a custom clock */
|
||||
if (GST_AUDIO_BASE_SRC (pulsesrc)->clock) {
|
||||
if (GST_AUDIO_BASE_SRC (pulsesrc)->clock)
|
||||
gst_object_unref (GST_AUDIO_BASE_SRC (pulsesrc)->clock);
|
||||
}
|
||||
|
||||
GST_AUDIO_BASE_SRC (pulsesrc)->clock =
|
||||
gst_audio_clock_new ("GstPulseSrcClock",
|
||||
(GstAudioClockGetTimeFunc) gst_pulsesrc_get_time, pulsesrc);
|
||||
(GstAudioClockGetTimeFunc) gst_pulsesrc_get_time, pulsesrc, NULL);
|
||||
}
|
||||
|
||||
static void
|
||||
|
@ -996,14 +995,14 @@ gst_pulsesrc_read (GstAudioSrc * asrc, gpointer data, guint length)
|
|||
GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (asrc);
|
||||
size_t sum = 0;
|
||||
|
||||
pa_threaded_mainloop_lock (pulsesrc->mainloop);
|
||||
pulsesrc->in_read = TRUE;
|
||||
|
||||
if (g_atomic_int_compare_and_exchange (&pulsesrc->notify, 1, 0)) {
|
||||
g_object_notify (G_OBJECT (pulsesrc), "volume");
|
||||
g_object_notify (G_OBJECT (pulsesrc), "mute");
|
||||
}
|
||||
|
||||
pa_threaded_mainloop_lock (pulsesrc->mainloop);
|
||||
pulsesrc->in_read = TRUE;
|
||||
|
||||
if (pulsesrc->paused)
|
||||
goto was_paused;
|
||||
|
||||
|
@ -1333,6 +1332,7 @@ gst_pulsesrc_prepare (GstAudioSrc * asrc, GstAudioRingBufferSpec * spec)
|
|||
GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (asrc);
|
||||
pa_stream_flags_t flags;
|
||||
pa_operation *o;
|
||||
GstAudioClock *clock;
|
||||
|
||||
pa_threaded_mainloop_lock (pulsesrc->mainloop);
|
||||
|
||||
|
@ -1382,6 +1382,10 @@ gst_pulsesrc_prepare (GstAudioSrc * asrc, GstAudioRingBufferSpec * spec)
|
|||
goto connect_failed;
|
||||
}
|
||||
|
||||
/* our clock will now start from 0 again */
|
||||
clock = GST_AUDIO_CLOCK (GST_AUDIO_BASE_SRC (pulsesrc)->clock);
|
||||
gst_audio_clock_reset (clock, 0);
|
||||
|
||||
pulsesrc->corked = TRUE;
|
||||
|
||||
for (;;) {
|
||||
|
@ -1617,6 +1621,11 @@ gst_pulsesrc_change_state (GstElement * element, GstStateChange transition)
|
|||
goto mainloop_start_failed;
|
||||
}
|
||||
break;
|
||||
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
||||
gst_element_post_message (element,
|
||||
gst_message_new_clock_provide (GST_OBJECT_CAST (element),
|
||||
GST_AUDIO_BASE_SRC (this)->clock, TRUE));
|
||||
break;
|
||||
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
|
||||
/* uncork and start recording */
|
||||
gst_pulsesrc_play (this);
|
||||
|
@ -1650,6 +1659,12 @@ gst_pulsesrc_change_state (GstElement * element, GstStateChange transition)
|
|||
this->mainloop = NULL;
|
||||
}
|
||||
break;
|
||||
case GST_STATE_CHANGE_PAUSED_TO_READY:
|
||||
/* format_lost is reset in release() in baseaudiosink */
|
||||
gst_element_post_message (element,
|
||||
gst_message_new_clock_lost (GST_OBJECT_CAST (element),
|
||||
GST_AUDIO_BASE_SRC (this)->clock));
|
||||
break;
|
||||
default:
|
||||
break;
|
||||
}
|
||||
|
|
Loading…
Reference in a new issue