mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2025-01-02 21:48:55 +00:00
Code Issues 1 Projects Releases Wiki Activity

pulsesrc: improve clock handling

Browse source 
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:
Wim Taymans 2012-06-07 15:03:02 +02:00
parent 3bcae19398
commit 456c8e8205
1 changed files with 21 additions and 6 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

27
ext/pulse/pulsesrc.c
View file

@ -295,13 +295,12 @@ gst_pulsesrc_init (GstPulseSrc * pulsesrc)
GST_AUDIO_BASE_SRC_SLAVE_SKEW); GST_AUDIO_BASE_SRC_SLAVE_SKEW);
/* override with a custom clock */ /* 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_object_unref (GST_AUDIO_BASE_SRC (pulsesrc)->clock);
}
GST_AUDIO_BASE_SRC (pulsesrc)->clock = GST_AUDIO_BASE_SRC (pulsesrc)->clock =
gst_audio_clock_new ("GstPulseSrcClock", gst_audio_clock_new ("GstPulseSrcClock",
(GstAudioClockGetTimeFunc) gst_pulsesrc_get_time, pulsesrc); (GstAudioClockGetTimeFunc) gst_pulsesrc_get_time, pulsesrc, NULL);
} }
static void static void
@ -996,14 +995,14 @@ gst_pulsesrc_read (GstAudioSrc * asrc, gpointer data, guint length)
GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (asrc); GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (asrc);
size_t sum = 0; 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)) { if (g_atomic_int_compare_and_exchange (&pulsesrc->notify, 1, 0)) {
g_object_notify (G_OBJECT (pulsesrc), "volume"); g_object_notify (G_OBJECT (pulsesrc), "volume");
g_object_notify (G_OBJECT (pulsesrc), "mute"); g_object_notify (G_OBJECT (pulsesrc), "mute");
} }
pa_threaded_mainloop_lock (pulsesrc->mainloop);
pulsesrc->in_read = TRUE;
if (pulsesrc->paused) if (pulsesrc->paused)
goto was_paused; goto was_paused;
@ -1333,6 +1332,7 @@ gst_pulsesrc_prepare (GstAudioSrc * asrc, GstAudioRingBufferSpec * spec)
GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (asrc); GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (asrc);
pa_stream_flags_t flags; pa_stream_flags_t flags;
pa_operation *o; pa_operation *o;
GstAudioClock *clock;
pa_threaded_mainloop_lock (pulsesrc->mainloop); pa_threaded_mainloop_lock (pulsesrc->mainloop);
@ -1382,6 +1382,10 @@ gst_pulsesrc_prepare (GstAudioSrc * asrc, GstAudioRingBufferSpec * spec)
goto connect_failed; 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; pulsesrc->corked = TRUE;
for (;;) { for (;;) {
@ -1617,6 +1621,11 @@ gst_pulsesrc_change_state (GstElement * element, GstStateChange transition)
goto mainloop_start_failed; goto mainloop_start_failed;
} }
break; 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: case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
/* uncork and start recording */ /* uncork and start recording */
gst_pulsesrc_play (this); gst_pulsesrc_play (this);
@ -1650,6 +1659,12 @@ gst_pulsesrc_change_state (GstElement * element, GstStateChange transition)
this->mainloop = NULL; this->mainloop = NULL;
} }
break; 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: default:
break; break;
} }