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audioaggregator: Sync pad values before aggregating
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
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d92375eaae
commit
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3 changed files with 105 additions and 12 deletions
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@ -739,7 +739,6 @@ gst_audio_aggregator_fill_buffer (GstAudioAggregator * aagg,
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GstClockTime start_time, end_time;
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gboolean discont = FALSE;
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guint64 start_offset, end_offset;
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GstClockTime timestamp, stream_time = GST_CLOCK_TIME_NONE;
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gint rate, bpf;
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GstAggregatorPad *aggpad = GST_AGGREGATOR_PAD (pad);
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@ -762,15 +761,6 @@ gst_audio_aggregator_fill_buffer (GstAudioAggregator * aagg,
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goto done;
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}
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timestamp = GST_BUFFER_PTS (inbuf);
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stream_time = gst_segment_to_stream_time (&aggpad->segment, GST_FORMAT_TIME,
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timestamp);
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/* sync object properties on stream time */
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/* TODO: Ideally we would want to do that on every sample */
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if (GST_CLOCK_TIME_IS_VALID (stream_time))
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gst_object_sync_values (GST_OBJECT (pad), stream_time);
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start_time = GST_BUFFER_PTS (inbuf);
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end_time =
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start_time + gst_util_uint64_scale_ceil (pad->priv->size, GST_SECOND,
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@ -964,6 +954,29 @@ gst_audio_aggregator_create_output_buffer (GstAudioAggregator * aagg,
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return outbuf;
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}
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static gboolean
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sync_pad_values (GstAudioAggregator * aagg, GstAudioAggregatorPad * pad)
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{
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GstAggregatorPad *bpad = GST_AGGREGATOR_PAD (pad);
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GstClockTime timestamp, stream_time;
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if (pad->priv->buffer == NULL)
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return TRUE;
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timestamp = GST_BUFFER_PTS (pad->priv->buffer);
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GST_OBJECT_LOCK (bpad);
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stream_time = gst_segment_to_stream_time (&bpad->segment, GST_FORMAT_TIME,
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timestamp);
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GST_OBJECT_UNLOCK (bpad);
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/* sync object properties on stream time */
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/* TODO: Ideally we would want to do that on every sample */
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if (GST_CLOCK_TIME_IS_VALID (stream_time))
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gst_object_sync_values (GST_OBJECT (pad), stream_time);
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return TRUE;
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}
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static GstFlowReturn
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gst_audio_aggregator_aggregate (GstAggregator * agg, gboolean timeout)
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{
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@ -1011,6 +1024,10 @@ gst_audio_aggregator_aggregate (GstAggregator * agg, gboolean timeout)
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element = GST_ELEMENT (agg);
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aagg = GST_AUDIO_AGGREGATOR (agg);
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/* Sync pad properties to the stream time */
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gst_aggregator_iterate_sinkpads (agg,
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(GstAggregatorPadForeachFunc) sync_pad_values, NULL);
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GST_AUDIO_AGGREGATOR_LOCK (aagg);
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GST_OBJECT_LOCK (agg);
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@ -296,8 +296,8 @@ AM_CFLAGS = $(GST_CFLAGS) $(GST_CHECK_CFLAGS) $(GST_OPTION_CFLAGS) \
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-UG_DISABLE_ASSERT -UG_DISABLE_CAST_CHECKS
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LDADD = $(GST_CHECK_LIBS)
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elements_audiomixer_LDADD = $(GST_BASE_LIBS) -lgstbase-@GST_API_VERSION@ $(LDADD)
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elements_audiomixer_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(AM_CFLAGS)
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elements_audiomixer_LDADD = $(GST_BASE_LIBS) $(GST_CONTROLLER_LIBS) -lgstbase-@GST_API_VERSION@ $(LDADD)
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elements_audiomixer_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(GST_CONTROLLER_CFLAGS) $(AM_CFLAGS)
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elements_audiointerleave_LDADD = $(GST_BASE_LIBS) -lgstbase-@GST_API_VERSION@ -lgstaudio-@GST_API_VERSION@ $(LDADD)
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elements_audiointerleave_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(AM_CFLAGS)
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@ -35,6 +35,8 @@
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#include <gst/check/gstconsistencychecker.h>
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#include <gst/audio/audio.h>
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#include <gst/base/gstbasesrc.h>
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#include <gst/controller/gstdirectcontrolbinding.h>
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#include <gst/controller/gstinterpolationcontrolsource.h>
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static GMainLoop *main_loop;
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@ -1835,6 +1837,79 @@ GST_START_TEST (test_segment_base_handling)
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GST_END_TEST;
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static void
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set_pad_volume_fade (GstPad * pad, GstClockTime start, gdouble start_value,
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GstClockTime end, gdouble end_value)
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{
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GstControlSource *cs;
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GstTimedValueControlSource *tvcs;
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cs = gst_interpolation_control_source_new ();
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fail_unless (gst_object_add_control_binding (GST_OBJECT_CAST (pad),
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gst_direct_control_binding_new_absolute (GST_OBJECT_CAST (pad),
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"volume", cs)));
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/* set volume interpolation mode */
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g_object_set (cs, "mode", GST_INTERPOLATION_MODE_LINEAR, NULL);
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tvcs = (GstTimedValueControlSource *) cs;
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fail_unless (gst_timed_value_control_source_set (tvcs, start, start_value));
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fail_unless (gst_timed_value_control_source_set (tvcs, end, end_value));
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gst_object_unref (cs);
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}
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GST_START_TEST (test_sinkpad_property_controller)
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{
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GstBus *bus;
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GstMessage *msg;
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GstElement *pipeline, *sink, *mix, *src1;
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GstPad *srcpad, *sinkpad;
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GError *error = NULL;
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gchar *debug;
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pipeline = gst_pipeline_new ("pipeline");
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mix = gst_element_factory_make ("audiomixer", "audiomixer");
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sink = gst_element_factory_make ("fakesink", "sink");
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src1 = gst_element_factory_make ("audiotestsrc", "src1");
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g_object_set (src1, "num-buffers", 100, NULL);
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gst_bin_add_many (GST_BIN (pipeline), src1, mix, sink, NULL);
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fail_unless (gst_element_link (mix, sink));
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srcpad = gst_element_get_static_pad (src1, "src");
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sinkpad = gst_element_get_request_pad (mix, "sink_0");
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fail_unless (gst_pad_link (srcpad, sinkpad) == GST_PAD_LINK_OK);
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set_pad_volume_fade (sinkpad, 0, 0, 1.0, 2.0);
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gst_object_unref (sinkpad);
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gst_object_unref (srcpad);
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gst_element_set_state (pipeline, GST_STATE_PLAYING);
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bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline));
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msg = gst_bus_timed_pop_filtered (bus, GST_CLOCK_TIME_NONE,
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GST_MESSAGE_EOS | GST_MESSAGE_ERROR);
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switch (GST_MESSAGE_TYPE (msg)) {
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case GST_MESSAGE_ERROR:
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gst_message_parse_error (msg, &error, &debug);
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g_printerr ("ERROR from element %s: %s\n",
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GST_OBJECT_NAME (msg->src), error->message);
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g_printerr ("Debug info: %s\n", debug);
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g_error_free (error);
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g_free (debug);
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break;
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case GST_MESSAGE_EOS:
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break;
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default:
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g_assert_not_reached ();
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}
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gst_message_unref (msg);
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g_object_unref (bus);
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gst_element_set_state (pipeline, GST_STATE_NULL);
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gst_object_unref (pipeline);
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}
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GST_END_TEST;
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static Suite *
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audiomixer_suite (void)
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{
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@ -1859,6 +1934,7 @@ audiomixer_suite (void)
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tcase_add_test (tc_chain, test_sync_discont);
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tcase_add_test (tc_chain, test_sync_unaligned);
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tcase_add_test (tc_chain, test_segment_base_handling);
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tcase_add_test (tc_chain, test_sinkpad_property_controller);
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/* Use a longer timeout */
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#ifdef HAVE_VALGRIND
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