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audioaggregator: Sync pad values before aggregating

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We need to sync the pad values before taking the aggregator and pad locks
otherwise the element will just deadlock if there's any property changes
scheduled using GstController since that involves taking the aggregator and pad
locks.

Also add a test for this.

https://bugzilla.gnome.org/show_bug.cgi?id=749574
This commit is contained in:
Nirbheek Chauhan 2015-05-19 16:08:08 +05:30 committed by Tim-Philipp Müller
parent a1cdede940
commit 74d7944cbb
2 changed files with 103 additions and 10 deletions
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37
gst/audiomixer/gstaudioaggregator.c
View file

@ -739,7 +739,6 @@ gst_audio_aggregator_fill_buffer (GstAudioAggregator * aagg,
GstClockTime start_time, end_time;
gboolean discont = FALSE;
guint64 start_offset, end_offset;
GstClockTime timestamp, stream_time = GST_CLOCK_TIME_NONE;
gint rate, bpf;
GstAggregatorPad *aggpad = GST_AGGREGATOR_PAD (pad);
@ -762,15 +761,6 @@ gst_audio_aggregator_fill_buffer (GstAudioAggregator * aagg,
goto done;
}
timestamp = GST_BUFFER_PTS (inbuf);
stream_time = gst_segment_to_stream_time (&aggpad->segment, GST_FORMAT_TIME,
timestamp);
/* sync object properties on stream time */
/* TODO: Ideally we would want to do that on every sample */
if (GST_CLOCK_TIME_IS_VALID (stream_time))
gst_object_sync_values (GST_OBJECT (pad), stream_time);
start_time = GST_BUFFER_PTS (inbuf);
end_time =
start_time + gst_util_uint64_scale_ceil (pad->priv->size, GST_SECOND,
@ -964,6 +954,29 @@ gst_audio_aggregator_create_output_buffer (GstAudioAggregator * aagg,
return outbuf;
}
static gboolean
sync_pad_values (GstAudioAggregator * aagg, GstAudioAggregatorPad * pad)
{
GstAggregatorPad *bpad = GST_AGGREGATOR_PAD (pad);
GstClockTime timestamp, stream_time;
if (pad->priv->buffer == NULL)
return TRUE;
timestamp = GST_BUFFER_PTS (pad->priv->buffer);
GST_OBJECT_LOCK (bpad);
stream_time = gst_segment_to_stream_time (&bpad->segment, GST_FORMAT_TIME,
timestamp);
GST_OBJECT_UNLOCK (bpad);
/* sync object properties on stream time */
/* TODO: Ideally we would want to do that on every sample */
if (GST_CLOCK_TIME_IS_VALID (stream_time))
gst_object_sync_values (GST_OBJECT (pad), stream_time);
return TRUE;
}
static GstFlowReturn
gst_audio_aggregator_aggregate (GstAggregator * agg, gboolean timeout)
{
@ -1011,6 +1024,10 @@ gst_audio_aggregator_aggregate (GstAggregator * agg, gboolean timeout)
element = GST_ELEMENT (agg);
aagg = GST_AUDIO_AGGREGATOR (agg);
/* Sync pad properties to the stream time */
gst_aggregator_iterate_sinkpads (agg,
(GstAggregatorPadForeachFunc) sync_pad_values, NULL);
GST_AUDIO_AGGREGATOR_LOCK (aagg);
GST_OBJECT_LOCK (agg);

76
tests/check/elements/audiomixer.c
View file

@ -35,6 +35,8 @@
#include <gst/check/gstconsistencychecker.h>
#include <gst/audio/audio.h>
#include <gst/base/gstbasesrc.h>
#include <gst/controller/gstdirectcontrolbinding.h>
#include <gst/controller/gstinterpolationcontrolsource.h>
static GMainLoop *main_loop;
@ -1835,6 +1837,79 @@ GST_START_TEST (test_segment_base_handling)
GST_END_TEST;
static void
set_pad_volume_fade (GstPad * pad, GstClockTime start, gdouble start_value,
GstClockTime end, gdouble end_value)
{
GstControlSource *cs;
GstTimedValueControlSource *tvcs;
cs = gst_interpolation_control_source_new ();
fail_unless (gst_object_add_control_binding (GST_OBJECT_CAST (pad),
gst_direct_control_binding_new_absolute (GST_OBJECT_CAST (pad),
"volume", cs)));
/* set volume interpolation mode */
g_object_set (cs, "mode", GST_INTERPOLATION_MODE_LINEAR, NULL);
tvcs = (GstTimedValueControlSource *) cs;
fail_unless (gst_timed_value_control_source_set (tvcs, start, start_value));
fail_unless (gst_timed_value_control_source_set (tvcs, end, end_value));
gst_object_unref (cs);
}
GST_START_TEST (test_sinkpad_property_controller)
{
GstBus *bus;
GstMessage *msg;
GstElement *pipeline, *sink, *mix, *src1;
GstPad *srcpad, *sinkpad;
GError *error = NULL;
gchar *debug;
pipeline = gst_pipeline_new ("pipeline");
mix = gst_element_factory_make ("audiomixer", "audiomixer");
sink = gst_element_factory_make ("fakesink", "sink");
src1 = gst_element_factory_make ("audiotestsrc", "src1");
g_object_set (src1, "num-buffers", 100, NULL);
gst_bin_add_many (GST_BIN (pipeline), src1, mix, sink, NULL);
fail_unless (gst_element_link (mix, sink));
srcpad = gst_element_get_static_pad (src1, "src");
sinkpad = gst_element_get_request_pad (mix, "sink_0");
fail_unless (gst_pad_link (srcpad, sinkpad) == GST_PAD_LINK_OK);
set_pad_volume_fade (sinkpad, 0, 0, 1.0, 2.0);
gst_object_unref (sinkpad);
gst_object_unref (srcpad);
gst_element_set_state (pipeline, GST_STATE_PLAYING);
bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline));
msg = gst_bus_timed_pop_filtered (bus, GST_CLOCK_TIME_NONE,
GST_MESSAGE_EOS | GST_MESSAGE_ERROR);
switch (GST_MESSAGE_TYPE (msg)) {
case GST_MESSAGE_ERROR:
gst_message_parse_error (msg, &error, &debug);
g_printerr ("ERROR from element %s: %s\n",
GST_OBJECT_NAME (msg->src), error->message);
g_printerr ("Debug info: %s\n", debug);
g_error_free (error);
g_free (debug);
break;
case GST_MESSAGE_EOS:
break;
default:
g_assert_not_reached ();
}
gst_message_unref (msg);
g_object_unref (bus);
gst_element_set_state (pipeline, GST_STATE_NULL);
gst_object_unref (pipeline);
}
GST_END_TEST;
static Suite *
audiomixer_suite (void)
{
@ -1859,6 +1934,7 @@ audiomixer_suite (void)
tcase_add_test (tc_chain, test_sync_discont);
tcase_add_test (tc_chain, test_sync_unaligned);
tcase_add_test (tc_chain, test_segment_base_handling);
tcase_add_test (tc_chain, test_sinkpad_property_controller);
/* Use a longer timeout */
#ifdef HAVE_VALGRIND