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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2025-01-11 09:55:36 +00:00
adder: use collectpads clipping function
Install a clipping function in the collectpads and use the audio clipping helper function to perform clipping to the segment boundaries. Fixes #590265
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66ae01eced
commit
59ace1b9ee
3 changed files with 132 additions and 3 deletions
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@ -1,10 +1,12 @@
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plugin_LTLIBRARIES = libgstadder.la
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libgstadder_la_SOURCES = gstadder.c
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libgstadder_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_CFLAGS)
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libgstadder_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(GST_CFLAGS)
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#$(LIBOIL_CFLAGS)
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libgstadder_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
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libgstadder_la_LIBADD = $(GST_BASE_LIBS) $(GST_LIBS)
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libgstadder_la_LIBADD = \
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$(top_builddir)/gst-libs/gst/audio/libgstaudio-@GST_MAJORMINOR@.la \
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$(GST_BASE_LIBS) $(GST_LIBS)
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#$(LIBOIL_LIBS)
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libgstadder_la_LIBTOOLFLAGS = --tag=disable-static
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@ -136,6 +136,8 @@ static void gst_adder_release_pad (GstElement * element, GstPad * pad);
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static GstStateChangeReturn gst_adder_change_state (GstElement * element,
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GstStateChange transition);
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static GstBuffer *gst_adder_do_clip (GstCollectPads * pads,
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GstCollectData * data, GstBuffer * buffer, gpointer user_data);
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static GstFlowReturn gst_adder_collected (GstCollectPads * pads,
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gpointer user_data);
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@ -780,7 +782,7 @@ gst_adder_sink_event (GstPad * pad, GstEvent * event)
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case GST_EVENT_FLUSH_STOP:
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/* we received a flush-stop. The collect_event function will push the
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* event past our element. We simply forward all flush-stop events, even
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* when no flush-stop was pendingk, this is required because collectpads
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* when no flush-stop was pending, this is required because collectpads
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* does not provide an API to handle-but-not-forward the flush-stop.
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* We unset the pending flush-stop flag so that we don't send anymore
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* flush-stop from the collect function later.
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@ -882,6 +884,8 @@ gst_adder_init (GstAdder * adder)
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adder->collect = gst_collect_pads_new ();
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gst_collect_pads_set_function (adder->collect,
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GST_DEBUG_FUNCPTR (gst_adder_collected), adder);
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gst_collect_pads_set_clip_function (adder->collect,
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GST_DEBUG_FUNCPTR (gst_adder_do_clip), adder);
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}
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static void
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@ -1023,6 +1027,18 @@ gst_adder_release_pad (GstElement * element, GstPad * pad)
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gst_element_remove_pad (element, pad);
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}
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static GstBuffer *
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gst_adder_do_clip (GstCollectPads * pads, GstCollectData * data,
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GstBuffer * buffer, gpointer user_data)
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{
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GstAdder *adder = GST_ADDER (user_data);
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buffer = gst_audio_buffer_clip (buffer, &data->segment, adder->rate,
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adder->bps);
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return buffer;
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}
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static GstFlowReturn
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gst_adder_collected (GstCollectPads * pads, gpointer user_data)
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{
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@ -732,6 +732,116 @@ GST_START_TEST (test_remove_pad)
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GST_END_TEST;
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static GstBuffer *handoff_buffer = NULL;
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static void
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handoff_buffer_cb (GstElement * fakesink, GstBuffer * buffer, GstPad * pad,
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gpointer user_data)
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{
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GST_DEBUG ("got buffer %p", buffer);
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gst_buffer_replace (&handoff_buffer, buffer);
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}
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/* check if clipping works as expected */
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GST_START_TEST (test_clip)
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{
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GstElement *bin, *adder, *sink;
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GstBus *bus;
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GstPad *sinkpad;
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gboolean res;
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GstFlowReturn ret;
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GstEvent *event;
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GstBuffer *buffer;
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GstCaps *caps;
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GST_INFO ("preparing test");
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/* build pipeline */
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bin = gst_pipeline_new ("pipeline");
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bus = gst_element_get_bus (bin);
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gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH);
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g_signal_connect (bus, "message::error", (GCallback) message_received, bin);
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g_signal_connect (bus, "message::warning", (GCallback) message_received, bin);
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g_signal_connect (bus, "message::eos", (GCallback) message_received, bin);
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/* just an adder and a fakesink */
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adder = gst_element_factory_make ("adder", "adder");
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sink = gst_element_factory_make ("fakesink", "sink");
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g_object_set (sink, "signal-handoffs", TRUE, NULL);
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g_signal_connect (sink, "handoff", (GCallback) handoff_buffer_cb, NULL);
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gst_bin_add_many (GST_BIN (bin), adder, sink, NULL);
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res = gst_element_link (adder, sink);
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fail_unless (res == TRUE, NULL);
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/* set to playing */
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res = gst_element_set_state (bin, GST_STATE_PLAYING);
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fail_unless (res != GST_STATE_CHANGE_FAILURE, NULL);
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/* create an unconnected sinkpad in adder, should also automatically activate
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* the pad */
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sinkpad = gst_element_get_request_pad (adder, "sink%d");
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fail_if (sinkpad == NULL, NULL);
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/* send segment to adder */
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event = gst_event_new_new_segment (FALSE, 1.0, GST_FORMAT_TIME,
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GST_SECOND, 2 * GST_SECOND, 0);
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gst_pad_send_event (sinkpad, event);
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caps = gst_caps_new_simple ("audio/x-raw-int",
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"rate", G_TYPE_INT, 44100,
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"channels", G_TYPE_INT, 2,
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"endianness", G_TYPE_INT, G_BYTE_ORDER,
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"width", G_TYPE_INT, 16,
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"depth", G_TYPE_INT, 16, "signed", G_TYPE_BOOLEAN, TRUE, NULL);
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/* should be clipped and ok */
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buffer = gst_buffer_new_and_alloc (44100);
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GST_BUFFER_TIMESTAMP (buffer) = 0;
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GST_BUFFER_DURATION (buffer) = 250 * GST_MSECOND;
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gst_buffer_set_caps (buffer, caps);
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GST_DEBUG ("pushing buffer %p", buffer);
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ret = gst_pad_chain (sinkpad, buffer);
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fail_unless (ret == GST_FLOW_OK);
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fail_unless (handoff_buffer == NULL);
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/* should be partially clipped */
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buffer = gst_buffer_new_and_alloc (44100);
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GST_BUFFER_TIMESTAMP (buffer) = 900 * GST_MSECOND;
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GST_BUFFER_DURATION (buffer) = 250 * GST_MSECOND;
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gst_buffer_set_caps (buffer, caps);
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GST_DEBUG ("pushing buffer %p", buffer);
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ret = gst_pad_chain (sinkpad, buffer);
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fail_unless (ret == GST_FLOW_OK);
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fail_unless (handoff_buffer != NULL);
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gst_buffer_replace (&handoff_buffer, NULL);
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/* should not be clipped */
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buffer = gst_buffer_new_and_alloc (44100);
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GST_BUFFER_TIMESTAMP (buffer) = 1 * GST_SECOND;
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GST_BUFFER_DURATION (buffer) = 250 * GST_MSECOND;
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gst_buffer_set_caps (buffer, caps);
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GST_DEBUG ("pushing buffer %p", buffer);
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ret = gst_pad_chain (sinkpad, buffer);
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fail_unless (ret == GST_FLOW_OK);
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fail_unless (handoff_buffer != NULL);
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gst_buffer_replace (&handoff_buffer, NULL);
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/* should be clipped and ok */
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buffer = gst_buffer_new_and_alloc (44100);
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GST_BUFFER_TIMESTAMP (buffer) = 2 * GST_SECOND;
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GST_BUFFER_DURATION (buffer) = 250 * GST_MSECOND;
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gst_buffer_set_caps (buffer, caps);
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GST_DEBUG ("pushing buffer %p", buffer);
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ret = gst_pad_chain (sinkpad, buffer);
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fail_unless (ret == GST_FLOW_OK);
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fail_unless (handoff_buffer == NULL);
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}
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GST_END_TEST;
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static Suite *
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adder_suite (void)
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{
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@ -745,6 +855,7 @@ adder_suite (void)
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tcase_add_test (tc_chain, test_live_seeking);
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tcase_add_test (tc_chain, test_add_pad);
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tcase_add_test (tc_chain, test_remove_pad);
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tcase_add_test (tc_chain, test_clip);
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/* Use a longer timeout */
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#ifdef HAVE_VALGRIND
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