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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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7eb98ba4f3
When checking the behaviour of live seeking on audiomixer or adder we don't *really* need real audio devices. audiotestsrc in live mode is enough to test the behaviour of those elements. Also avoids people repeatedly wasting hours trying to figure out whether that failing behaviour is due to their code or not.
1993 lines
62 KiB
C
1993 lines
62 KiB
C
/* GStreamer
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*
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* unit test for audiomixer
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*
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* Copyright (C) 2005 Thomas Vander Stichele <thomas at apestaart dot org>
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* Copyright (C) 2013 Sebastian Dröge <sebastian@centricular.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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#ifdef HAVE_CONFIG_H
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# include <config.h>
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#endif
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#ifdef HAVE_VALGRIND
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# include <valgrind/valgrind.h>
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#endif
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#include <gst/check/gstcheck.h>
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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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/* fixtures */
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static void
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test_setup (void)
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{
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main_loop = g_main_loop_new (NULL, FALSE);
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}
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static void
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test_teardown (void)
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{
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g_main_loop_unref (main_loop);
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main_loop = NULL;
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}
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/* some test helpers */
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static GstElement *
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setup_pipeline (GstElement * audiomixer, gint num_srcs, GstElement * capsfilter)
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{
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GstElement *pipeline, *src, *sink;
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gint i;
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pipeline = gst_pipeline_new ("pipeline");
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if (!audiomixer) {
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audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
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}
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sink = gst_element_factory_make ("fakesink", "sink");
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gst_bin_add_many (GST_BIN (pipeline), audiomixer, sink, NULL);
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if (capsfilter) {
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gst_bin_add (GST_BIN (pipeline), capsfilter);
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gst_element_link_many (audiomixer, capsfilter, sink, NULL);
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} else {
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gst_element_link (audiomixer, sink);
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}
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for (i = 0; i < num_srcs; i++) {
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src = gst_element_factory_make ("audiotestsrc", NULL);
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g_object_set (src, "wave", 4, NULL); /* silence */
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gst_bin_add (GST_BIN (pipeline), src);
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gst_element_link (src, audiomixer);
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}
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return pipeline;
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}
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static GstCaps *
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get_element_sink_pad_caps (GstElement * pipeline, const gchar * element_name)
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{
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GstElement *sink;
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GstCaps *caps;
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GstPad *pad;
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sink = gst_bin_get_by_name (GST_BIN (pipeline), "sink");
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pad = gst_element_get_static_pad (sink, "sink");
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caps = gst_pad_get_current_caps (pad);
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gst_object_unref (pad);
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gst_object_unref (sink);
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return caps;
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}
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static void
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set_state_and_wait (GstElement * pipeline, GstState state)
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{
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GstStateChangeReturn state_res;
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/* prepare paused/playing */
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state_res = gst_element_set_state (pipeline, state);
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ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
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/* wait for preroll */
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state_res = gst_element_get_state (pipeline, NULL, NULL, GST_CLOCK_TIME_NONE);
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ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
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}
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static gboolean
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set_playing (GstElement * element)
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{
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GstStateChangeReturn state_res;
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state_res = gst_element_set_state (element, GST_STATE_PLAYING);
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ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
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return FALSE;
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}
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static void
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play_and_wait (GstElement * pipeline)
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{
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GstStateChangeReturn state_res;
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g_idle_add ((GSourceFunc) set_playing, pipeline);
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GST_INFO ("running main loop");
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g_main_loop_run (main_loop);
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state_res = gst_element_set_state (pipeline, GST_STATE_NULL);
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ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
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}
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static void
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message_received (GstBus * bus, GstMessage * message, GstPipeline * bin)
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{
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GST_INFO ("bus message from \"%" GST_PTR_FORMAT "\": %" GST_PTR_FORMAT,
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GST_MESSAGE_SRC (message), message);
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switch (message->type) {
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case GST_MESSAGE_EOS:
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g_main_loop_quit (main_loop);
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break;
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case GST_MESSAGE_WARNING:{
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GError *gerror;
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gchar *debug;
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gst_message_parse_warning (message, &gerror, &debug);
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gst_object_default_error (GST_MESSAGE_SRC (message), gerror, debug);
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g_error_free (gerror);
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g_free (debug);
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break;
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}
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case GST_MESSAGE_ERROR:{
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GError *gerror;
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gchar *debug;
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gst_message_parse_error (message, &gerror, &debug);
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gst_object_default_error (GST_MESSAGE_SRC (message), gerror, debug);
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g_error_free (gerror);
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g_free (debug);
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g_main_loop_quit (main_loop);
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break;
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}
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default:
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break;
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}
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}
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static GstBuffer *
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new_buffer (gsize num_bytes, gint data, GstClockTime ts, GstClockTime dur,
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GstBufferFlags flags)
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{
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GstMapInfo map;
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GstBuffer *buffer = gst_buffer_new_and_alloc (num_bytes);
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gst_buffer_map (buffer, &map, GST_MAP_WRITE);
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memset (map.data, data, map.size);
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gst_buffer_unmap (buffer, &map);
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GST_BUFFER_TIMESTAMP (buffer) = ts;
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GST_BUFFER_DURATION (buffer) = dur;
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if (flags)
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GST_BUFFER_FLAG_SET (buffer, flags);
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GST_DEBUG ("created buffer %p", buffer);
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return buffer;
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}
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/* make sure downstream gets a CAPS event before buffers are sent */
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GST_START_TEST (test_caps)
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{
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GstElement *pipeline;
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GstCaps *caps;
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/* build pipeline */
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pipeline = setup_pipeline (NULL, 1, NULL);
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/* prepare playing */
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set_state_and_wait (pipeline, GST_STATE_PAUSED);
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/* check caps on fakesink */
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caps = get_element_sink_pad_caps (pipeline, "sink");
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fail_unless (caps != NULL);
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gst_caps_unref (caps);
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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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/* check that caps set on the property are honoured */
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GST_START_TEST (test_filter_caps)
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{
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GstElement *pipeline, *audiomixer, *capsfilter;
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GstCaps *filter_caps, *caps;
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filter_caps = gst_caps_new_simple ("audio/x-raw",
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"format", G_TYPE_STRING, GST_AUDIO_NE (F32),
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"layout", G_TYPE_STRING, "interleaved",
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"rate", G_TYPE_INT, 44100, "channels", G_TYPE_INT, 1,
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"channel-mask", GST_TYPE_BITMASK, (guint64) 0x04, NULL);
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capsfilter = gst_element_factory_make ("capsfilter", NULL);
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/* build pipeline */
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audiomixer = gst_element_factory_make ("audiomixer", NULL);
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g_object_set (capsfilter, "caps", filter_caps, NULL);
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pipeline = setup_pipeline (audiomixer, 1, capsfilter);
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/* prepare playing */
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set_state_and_wait (pipeline, GST_STATE_PAUSED);
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/* check caps on fakesink */
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caps = get_element_sink_pad_caps (pipeline, "sink");
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fail_unless (caps != NULL);
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GST_INFO_OBJECT (pipeline, "received caps: %" GST_PTR_FORMAT, caps);
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fail_unless (gst_caps_is_equal_fixed (caps, filter_caps));
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gst_caps_unref (caps);
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gst_element_set_state (pipeline, GST_STATE_NULL);
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gst_object_unref (pipeline);
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gst_caps_unref (filter_caps);
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}
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GST_END_TEST;
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static GstFormat format = GST_FORMAT_UNDEFINED;
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static gint64 position = -1;
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static void
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test_event_message_received (GstBus * bus, GstMessage * message,
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GstPipeline * bin)
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{
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GST_INFO ("bus message from \"%" GST_PTR_FORMAT "\": %" GST_PTR_FORMAT,
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GST_MESSAGE_SRC (message), message);
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switch (message->type) {
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case GST_MESSAGE_SEGMENT_DONE:
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gst_message_parse_segment_done (message, &format, &position);
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GST_INFO ("received segment_done : %" G_GINT64_FORMAT, position);
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g_main_loop_quit (main_loop);
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break;
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default:
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g_assert_not_reached ();
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break;
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}
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}
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GST_START_TEST (test_event)
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{
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GstElement *bin, *src1, *src2, *audiomixer, *sink;
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GstBus *bus;
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GstEvent *seek_event;
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gboolean res;
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GstPad *srcpad, *sinkpad;
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GstStreamConsistency *chk_1, *chk_2, *chk_3;
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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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src1 = gst_element_factory_make ("audiotestsrc", "src1");
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g_object_set (src1, "wave", 4, NULL); /* silence */
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src2 = gst_element_factory_make ("audiotestsrc", "src2");
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g_object_set (src2, "wave", 4, NULL); /* silence */
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audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
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sink = gst_element_factory_make ("fakesink", "sink");
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gst_bin_add_many (GST_BIN (bin), src1, src2, audiomixer, sink, NULL);
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res = gst_element_link (src1, audiomixer);
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fail_unless (res == TRUE, NULL);
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res = gst_element_link (src2, audiomixer);
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fail_unless (res == TRUE, NULL);
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res = gst_element_link (audiomixer, sink);
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fail_unless (res == TRUE, NULL);
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srcpad = gst_element_get_static_pad (audiomixer, "src");
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chk_3 = gst_consistency_checker_new (srcpad);
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gst_object_unref (srcpad);
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/* create consistency checkers for the pads */
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srcpad = gst_element_get_static_pad (src1, "src");
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chk_1 = gst_consistency_checker_new (srcpad);
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sinkpad = gst_pad_get_peer (srcpad);
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gst_consistency_checker_add_pad (chk_3, sinkpad);
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gst_object_unref (sinkpad);
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gst_object_unref (srcpad);
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srcpad = gst_element_get_static_pad (src2, "src");
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chk_2 = gst_consistency_checker_new (srcpad);
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sinkpad = gst_pad_get_peer (srcpad);
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gst_consistency_checker_add_pad (chk_3, sinkpad);
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gst_object_unref (sinkpad);
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gst_object_unref (srcpad);
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seek_event = gst_event_new_seek (1.0, GST_FORMAT_TIME,
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GST_SEEK_FLAG_SEGMENT | GST_SEEK_FLAG_FLUSH,
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GST_SEEK_TYPE_SET, (GstClockTime) 0,
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GST_SEEK_TYPE_SET, (GstClockTime) 2 * GST_SECOND);
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format = GST_FORMAT_UNDEFINED;
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position = -1;
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g_signal_connect (bus, "message::segment-done",
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(GCallback) test_event_message_received, bin);
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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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GST_INFO ("starting test");
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/* prepare playing */
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set_state_and_wait (bin, GST_STATE_PAUSED);
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res = gst_element_send_event (bin, seek_event);
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fail_unless (res == TRUE, NULL);
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/* run pipeline */
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play_and_wait (bin);
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ck_assert_int_eq (position, 2 * GST_SECOND);
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/* cleanup */
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gst_consistency_checker_free (chk_1);
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gst_consistency_checker_free (chk_2);
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gst_consistency_checker_free (chk_3);
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gst_bus_remove_signal_watch (bus);
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gst_object_unref (bus);
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gst_object_unref (bin);
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}
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GST_END_TEST;
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static guint play_count = 0;
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static GstEvent *play_seek_event = NULL;
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static void
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test_play_twice_message_received (GstBus * bus, GstMessage * message,
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GstElement * bin)
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{
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gboolean res;
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GstStateChangeReturn state_res;
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GST_INFO ("bus message from \"%" GST_PTR_FORMAT "\": %" GST_PTR_FORMAT,
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GST_MESSAGE_SRC (message), message);
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switch (message->type) {
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case GST_MESSAGE_SEGMENT_DONE:
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play_count++;
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if (play_count == 1) {
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state_res = gst_element_set_state (bin, GST_STATE_READY);
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ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
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/* prepare playing again */
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set_state_and_wait (bin, GST_STATE_PAUSED);
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gst_event_set_seqnum (play_seek_event, gst_util_seqnum_next ());
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res = gst_element_send_event (bin, gst_event_ref (play_seek_event));
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fail_unless (res == TRUE, NULL);
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state_res = gst_element_set_state (bin, GST_STATE_PLAYING);
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ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
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} else {
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g_main_loop_quit (main_loop);
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}
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break;
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default:
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g_assert_not_reached ();
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break;
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}
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}
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GST_START_TEST (test_play_twice)
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{
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GstElement *bin, *audiomixer;
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GstBus *bus;
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gboolean res;
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GstPad *srcpad;
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GstStreamConsistency *consist;
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GST_INFO ("preparing test");
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/* build pipeline */
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audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
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bin = setup_pipeline (audiomixer, 2, NULL);
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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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srcpad = gst_element_get_static_pad (audiomixer, "src");
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consist = gst_consistency_checker_new (srcpad);
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gst_object_unref (srcpad);
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play_seek_event = gst_event_new_seek (1.0, GST_FORMAT_TIME,
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GST_SEEK_FLAG_SEGMENT | GST_SEEK_FLAG_FLUSH,
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GST_SEEK_TYPE_SET, (GstClockTime) 0,
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GST_SEEK_TYPE_SET, (GstClockTime) 2 * GST_SECOND);
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play_count = 0;
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g_signal_connect (bus, "message::segment-done",
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(GCallback) test_play_twice_message_received, bin);
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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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GST_INFO ("starting test");
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/* prepare playing */
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set_state_and_wait (bin, GST_STATE_PAUSED);
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gst_event_set_seqnum (play_seek_event, gst_util_seqnum_next ());
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res = gst_element_send_event (bin, gst_event_ref (play_seek_event));
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fail_unless (res == TRUE, NULL);
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GST_INFO ("seeked");
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/* run pipeline */
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play_and_wait (bin);
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ck_assert_int_eq (play_count, 2);
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/* cleanup */
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gst_consistency_checker_free (consist);
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gst_event_unref (play_seek_event);
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gst_bus_remove_signal_watch (bus);
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gst_object_unref (bus);
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gst_object_unref (bin);
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}
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GST_END_TEST;
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GST_START_TEST (test_play_twice_then_add_and_play_again)
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{
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GstElement *bin, *src, *audiomixer;
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GstBus *bus;
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gboolean res;
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GstStateChangeReturn state_res;
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gint i;
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GstPad *srcpad;
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GstStreamConsistency *consist;
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GST_INFO ("preparing test");
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/* build pipeline */
|
|
audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
|
|
bin = setup_pipeline (audiomixer, 2, NULL);
|
|
bus = gst_element_get_bus (bin);
|
|
gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH);
|
|
|
|
srcpad = gst_element_get_static_pad (audiomixer, "src");
|
|
consist = gst_consistency_checker_new (srcpad);
|
|
gst_object_unref (srcpad);
|
|
|
|
play_seek_event = gst_event_new_seek (1.0, GST_FORMAT_TIME,
|
|
GST_SEEK_FLAG_SEGMENT | GST_SEEK_FLAG_FLUSH,
|
|
GST_SEEK_TYPE_SET, (GstClockTime) 0,
|
|
GST_SEEK_TYPE_SET, (GstClockTime) 2 * GST_SECOND);
|
|
|
|
g_signal_connect (bus, "message::segment-done",
|
|
(GCallback) test_play_twice_message_received, bin);
|
|
g_signal_connect (bus, "message::error", (GCallback) message_received, bin);
|
|
g_signal_connect (bus, "message::warning", (GCallback) message_received, bin);
|
|
g_signal_connect (bus, "message::eos", (GCallback) message_received, bin);
|
|
|
|
/* run it twice */
|
|
for (i = 0; i < 2; i++) {
|
|
play_count = 0;
|
|
|
|
GST_INFO ("starting test-loop %d", i);
|
|
|
|
/* prepare playing */
|
|
set_state_and_wait (bin, GST_STATE_PAUSED);
|
|
|
|
gst_event_set_seqnum (play_seek_event, gst_util_seqnum_next ());
|
|
res = gst_element_send_event (bin, gst_event_ref (play_seek_event));
|
|
fail_unless (res == TRUE, NULL);
|
|
|
|
GST_INFO ("seeked");
|
|
|
|
/* run pipeline */
|
|
play_and_wait (bin);
|
|
|
|
ck_assert_int_eq (play_count, 2);
|
|
|
|
/* plug another source */
|
|
if (i == 0) {
|
|
src = gst_element_factory_make ("audiotestsrc", NULL);
|
|
g_object_set (src, "wave", 4, NULL); /* silence */
|
|
gst_bin_add (GST_BIN (bin), src);
|
|
|
|
res = gst_element_link (src, audiomixer);
|
|
fail_unless (res == TRUE, NULL);
|
|
}
|
|
|
|
gst_consistency_checker_reset (consist);
|
|
}
|
|
|
|
state_res = gst_element_set_state (bin, GST_STATE_NULL);
|
|
ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
|
|
|
|
/* cleanup */
|
|
gst_event_unref (play_seek_event);
|
|
gst_consistency_checker_free (consist);
|
|
gst_bus_remove_signal_watch (bus);
|
|
gst_object_unref (bus);
|
|
gst_object_unref (bin);
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
/* test failing seeks on live-sources */
|
|
GST_START_TEST (test_live_seeking)
|
|
{
|
|
GstElement *bin, *src1 = NULL, *cf, *src2, *audiomixer, *sink;
|
|
GstCaps *caps;
|
|
GstBus *bus;
|
|
gboolean res;
|
|
GstPad *srcpad;
|
|
GstPad *sinkpad;
|
|
gint i;
|
|
GstStreamConsistency *consist;
|
|
|
|
GST_INFO ("preparing test");
|
|
play_seek_event = NULL;
|
|
|
|
/* build pipeline */
|
|
bin = gst_pipeline_new ("pipeline");
|
|
bus = gst_element_get_bus (bin);
|
|
gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH);
|
|
|
|
src1 = gst_element_factory_make ("audiotestsrc", "src1");
|
|
g_object_set (src1, "wave", 4, "is-live", TRUE, NULL); /* silence */
|
|
|
|
audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
|
|
cf = gst_element_factory_make ("capsfilter", "capsfilter");
|
|
sink = gst_element_factory_make ("fakesink", "sink");
|
|
|
|
gst_bin_add_many (GST_BIN (bin), src1, cf, audiomixer, sink, NULL);
|
|
res = gst_element_link_many (src1, cf, audiomixer, sink, NULL);
|
|
fail_unless (res == TRUE, NULL);
|
|
|
|
/* get the caps for the livesrc, we'll reuse this for the non-live source */
|
|
set_state_and_wait (bin, GST_STATE_PLAYING);
|
|
|
|
sinkpad = gst_element_get_static_pad (sink, "sink");
|
|
fail_unless (sinkpad != NULL);
|
|
caps = gst_pad_get_current_caps (sinkpad);
|
|
fail_unless (caps != NULL);
|
|
gst_object_unref (sinkpad);
|
|
|
|
gst_element_set_state (bin, GST_STATE_NULL);
|
|
|
|
g_object_set (cf, "caps", caps, NULL);
|
|
|
|
src2 = gst_element_factory_make ("audiotestsrc", "src2");
|
|
g_object_set (src2, "wave", 4, NULL); /* silence */
|
|
gst_bin_add (GST_BIN (bin), src2);
|
|
|
|
res = gst_element_link_filtered (src2, audiomixer, caps);
|
|
fail_unless (res == TRUE, NULL);
|
|
|
|
gst_caps_unref (caps);
|
|
|
|
play_seek_event = gst_event_new_seek (1.0, GST_FORMAT_TIME,
|
|
GST_SEEK_FLAG_FLUSH,
|
|
GST_SEEK_TYPE_SET, (GstClockTime) 0,
|
|
GST_SEEK_TYPE_SET, (GstClockTime) 2 * GST_SECOND);
|
|
|
|
g_signal_connect (bus, "message::error", (GCallback) message_received, bin);
|
|
g_signal_connect (bus, "message::warning", (GCallback) message_received, bin);
|
|
g_signal_connect (bus, "message::eos", (GCallback) message_received, bin);
|
|
|
|
srcpad = gst_element_get_static_pad (audiomixer, "src");
|
|
consist = gst_consistency_checker_new (srcpad);
|
|
gst_object_unref (srcpad);
|
|
|
|
GST_INFO ("starting test");
|
|
|
|
/* run it twice */
|
|
for (i = 0; i < 2; i++) {
|
|
|
|
GST_INFO ("starting test-loop %d", i);
|
|
|
|
/* prepare playing */
|
|
set_state_and_wait (bin, GST_STATE_PAUSED);
|
|
|
|
gst_event_set_seqnum (play_seek_event, gst_util_seqnum_next ());
|
|
res = gst_element_send_event (bin, gst_event_ref (play_seek_event));
|
|
fail_unless (res == TRUE, NULL);
|
|
|
|
GST_INFO ("seeked");
|
|
|
|
/* run pipeline */
|
|
play_and_wait (bin);
|
|
|
|
gst_consistency_checker_reset (consist);
|
|
}
|
|
|
|
/* cleanup */
|
|
GST_INFO ("cleaning up");
|
|
gst_consistency_checker_free (consist);
|
|
if (play_seek_event)
|
|
gst_event_unref (play_seek_event);
|
|
gst_bus_remove_signal_watch (bus);
|
|
gst_object_unref (bus);
|
|
gst_object_unref (bin);
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
/* check if adding pads work as expected */
|
|
GST_START_TEST (test_add_pad)
|
|
{
|
|
GstElement *bin, *src1, *src2, *audiomixer, *sink;
|
|
GstBus *bus;
|
|
GstPad *srcpad;
|
|
gboolean res;
|
|
GstStateChangeReturn state_res;
|
|
|
|
GST_INFO ("preparing test");
|
|
|
|
/* build pipeline */
|
|
bin = gst_pipeline_new ("pipeline");
|
|
bus = gst_element_get_bus (bin);
|
|
gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH);
|
|
|
|
src1 = gst_element_factory_make ("audiotestsrc", "src1");
|
|
g_object_set (src1, "num-buffers", 4, "wave", /* silence */ 4, NULL);
|
|
src2 = gst_element_factory_make ("audiotestsrc", "src2");
|
|
/* one buffer less, we connect with 1 buffer of delay */
|
|
g_object_set (src2, "num-buffers", 3, "wave", /* silence */ 4, NULL);
|
|
audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
|
|
sink = gst_element_factory_make ("fakesink", "sink");
|
|
gst_bin_add_many (GST_BIN (bin), src1, audiomixer, sink, NULL);
|
|
|
|
res = gst_element_link (src1, audiomixer);
|
|
fail_unless (res == TRUE, NULL);
|
|
res = gst_element_link (audiomixer, sink);
|
|
fail_unless (res == TRUE, NULL);
|
|
|
|
srcpad = gst_element_get_static_pad (audiomixer, "src");
|
|
gst_object_unref (srcpad);
|
|
|
|
g_signal_connect (bus, "message::segment-done", (GCallback) message_received,
|
|
bin);
|
|
g_signal_connect (bus, "message::error", (GCallback) message_received, bin);
|
|
g_signal_connect (bus, "message::warning", (GCallback) message_received, bin);
|
|
g_signal_connect (bus, "message::eos", (GCallback) message_received, bin);
|
|
|
|
GST_INFO ("starting test");
|
|
|
|
/* prepare playing */
|
|
set_state_and_wait (bin, GST_STATE_PAUSED);
|
|
|
|
/* add other element */
|
|
gst_bin_add_many (GST_BIN (bin), src2, NULL);
|
|
|
|
/* now link the second element */
|
|
res = gst_element_link (src2, audiomixer);
|
|
fail_unless (res == TRUE, NULL);
|
|
|
|
/* set to PAUSED as well */
|
|
state_res = gst_element_set_state (src2, GST_STATE_PAUSED);
|
|
ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
|
|
|
|
/* now play all */
|
|
play_and_wait (bin);
|
|
|
|
/* cleanup */
|
|
gst_bus_remove_signal_watch (bus);
|
|
gst_object_unref (bus);
|
|
gst_object_unref (bin);
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
/* check if removing pads work as expected */
|
|
GST_START_TEST (test_remove_pad)
|
|
{
|
|
GstElement *bin, *src, *audiomixer, *sink;
|
|
GstBus *bus;
|
|
GstPad *pad, *srcpad;
|
|
gboolean res;
|
|
GstStateChangeReturn state_res;
|
|
|
|
GST_INFO ("preparing test");
|
|
|
|
/* build pipeline */
|
|
bin = gst_pipeline_new ("pipeline");
|
|
bus = gst_element_get_bus (bin);
|
|
gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH);
|
|
|
|
src = gst_element_factory_make ("audiotestsrc", "src");
|
|
g_object_set (src, "num-buffers", 4, "wave", 4, NULL);
|
|
audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
|
|
sink = gst_element_factory_make ("fakesink", "sink");
|
|
gst_bin_add_many (GST_BIN (bin), src, audiomixer, sink, NULL);
|
|
|
|
res = gst_element_link (src, audiomixer);
|
|
fail_unless (res == TRUE, NULL);
|
|
res = gst_element_link (audiomixer, sink);
|
|
fail_unless (res == TRUE, NULL);
|
|
|
|
/* create an unconnected sinkpad in audiomixer */
|
|
pad = gst_element_get_request_pad (audiomixer, "sink_%u");
|
|
fail_if (pad == NULL, NULL);
|
|
|
|
srcpad = gst_element_get_static_pad (audiomixer, "src");
|
|
gst_object_unref (srcpad);
|
|
|
|
g_signal_connect (bus, "message::segment-done", (GCallback) message_received,
|
|
bin);
|
|
g_signal_connect (bus, "message::error", (GCallback) message_received, bin);
|
|
g_signal_connect (bus, "message::warning", (GCallback) message_received, bin);
|
|
g_signal_connect (bus, "message::eos", (GCallback) message_received, bin);
|
|
|
|
GST_INFO ("starting test");
|
|
|
|
/* prepare playing, this will not preroll as audiomixer is waiting
|
|
* on the unconnected sinkpad. */
|
|
state_res = gst_element_set_state (bin, GST_STATE_PAUSED);
|
|
ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
|
|
|
|
/* wait for completion for one second, will return ASYNC */
|
|
state_res = gst_element_get_state (GST_ELEMENT (bin), NULL, NULL, GST_SECOND);
|
|
ck_assert_int_eq (state_res, GST_STATE_CHANGE_ASYNC);
|
|
|
|
/* get rid of the pad now, audiomixer should stop waiting on it and
|
|
* continue the preroll */
|
|
gst_element_release_request_pad (audiomixer, pad);
|
|
gst_object_unref (pad);
|
|
|
|
/* wait for completion, should work now */
|
|
state_res =
|
|
gst_element_get_state (GST_ELEMENT (bin), NULL, NULL,
|
|
GST_CLOCK_TIME_NONE);
|
|
ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
|
|
|
|
/* now play all */
|
|
play_and_wait (bin);
|
|
|
|
/* cleanup */
|
|
gst_bus_remove_signal_watch (bus);
|
|
gst_object_unref (G_OBJECT (bus));
|
|
gst_object_unref (G_OBJECT (bin));
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
|
|
static GstBuffer *handoff_buffer = NULL;
|
|
|
|
static void
|
|
handoff_buffer_cb (GstElement * fakesink, GstBuffer * buffer, GstPad * pad,
|
|
gpointer user_data)
|
|
{
|
|
GST_DEBUG ("got buffer -- SIZE: %" G_GSIZE_FORMAT
|
|
" -- %p PTS is %" GST_TIME_FORMAT " END is %" GST_TIME_FORMAT,
|
|
gst_buffer_get_size (buffer), buffer,
|
|
GST_TIME_ARGS (GST_BUFFER_PTS (buffer)),
|
|
GST_TIME_ARGS (GST_BUFFER_PTS (buffer) + GST_BUFFER_DURATION (buffer)));
|
|
|
|
gst_buffer_replace (&handoff_buffer, buffer);
|
|
}
|
|
|
|
/* check if clipping works as expected */
|
|
GST_START_TEST (test_clip)
|
|
{
|
|
GstSegment segment;
|
|
GstElement *bin, *audiomixer, *sink;
|
|
GstBus *bus;
|
|
GstPad *sinkpad;
|
|
gboolean res;
|
|
GstStateChangeReturn state_res;
|
|
GstFlowReturn ret;
|
|
GstEvent *event;
|
|
GstBuffer *buffer;
|
|
GstCaps *caps;
|
|
GstQuery *drain = gst_query_new_drain ();
|
|
|
|
GST_INFO ("preparing test");
|
|
|
|
/* build pipeline */
|
|
bin = gst_pipeline_new ("pipeline");
|
|
bus = gst_element_get_bus (bin);
|
|
gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH);
|
|
|
|
g_signal_connect (bus, "message::error", (GCallback) message_received, bin);
|
|
g_signal_connect (bus, "message::warning", (GCallback) message_received, bin);
|
|
g_signal_connect (bus, "message::eos", (GCallback) message_received, bin);
|
|
|
|
/* just an audiomixer and a fakesink */
|
|
audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
|
|
g_object_set (audiomixer, "output-buffer-duration", 50 * GST_MSECOND, NULL);
|
|
sink = gst_element_factory_make ("fakesink", "sink");
|
|
g_object_set (sink, "signal-handoffs", TRUE, NULL);
|
|
g_signal_connect (sink, "handoff", (GCallback) handoff_buffer_cb, NULL);
|
|
gst_bin_add_many (GST_BIN (bin), audiomixer, sink, NULL);
|
|
|
|
res = gst_element_link (audiomixer, sink);
|
|
fail_unless (res == TRUE, NULL);
|
|
|
|
/* set to playing */
|
|
state_res = gst_element_set_state (bin, GST_STATE_PLAYING);
|
|
ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
|
|
|
|
/* create an unconnected sinkpad in audiomixer, should also automatically activate
|
|
* the pad */
|
|
sinkpad = gst_element_get_request_pad (audiomixer, "sink_%u");
|
|
fail_if (sinkpad == NULL, NULL);
|
|
|
|
gst_pad_send_event (sinkpad, gst_event_new_stream_start ("test"));
|
|
|
|
caps = gst_caps_new_simple ("audio/x-raw",
|
|
"format", G_TYPE_STRING, GST_AUDIO_NE (S16),
|
|
"layout", G_TYPE_STRING, "interleaved",
|
|
"rate", G_TYPE_INT, 44100, "channels", G_TYPE_INT, 2, NULL);
|
|
|
|
gst_pad_set_caps (sinkpad, caps);
|
|
gst_caps_unref (caps);
|
|
|
|
/* send segment to audiomixer */
|
|
gst_segment_init (&segment, GST_FORMAT_TIME);
|
|
segment.start = GST_SECOND;
|
|
segment.stop = 2 * GST_SECOND;
|
|
segment.time = 0;
|
|
event = gst_event_new_segment (&segment);
|
|
gst_pad_send_event (sinkpad, event);
|
|
|
|
/* should be clipped and ok */
|
|
buffer = new_buffer (44100, 0, 0, 250 * GST_MSECOND, 0);
|
|
GST_DEBUG ("pushing buffer %p END is %" GST_TIME_FORMAT,
|
|
buffer,
|
|
GST_TIME_ARGS (GST_BUFFER_PTS (buffer) + GST_BUFFER_DURATION (buffer)));
|
|
ret = gst_pad_chain (sinkpad, buffer);
|
|
ck_assert_int_eq (ret, GST_FLOW_OK);
|
|
/* The aggregation is done in a dedicated thread, so we can't
|
|
* know when it is actually going to happen, so we use a DRAIN query
|
|
* to wait for it to complete.
|
|
*/
|
|
gst_pad_query (sinkpad, drain);
|
|
fail_unless (handoff_buffer == NULL);
|
|
|
|
/* should be partially clipped */
|
|
buffer = new_buffer (44100, 0, 900 * GST_MSECOND, 250 * GST_MSECOND,
|
|
GST_BUFFER_FLAG_DISCONT);
|
|
GST_DEBUG ("pushing buffer %p START %" GST_TIME_FORMAT " -- DURATION is %"
|
|
GST_TIME_FORMAT, buffer, GST_TIME_ARGS (GST_BUFFER_PTS (buffer)),
|
|
GST_TIME_ARGS (GST_BUFFER_DURATION (buffer)));
|
|
ret = gst_pad_chain (sinkpad, buffer);
|
|
ck_assert_int_eq (ret, GST_FLOW_OK);
|
|
gst_pad_query (sinkpad, drain);
|
|
|
|
fail_unless (handoff_buffer != NULL);
|
|
ck_assert_int_eq (GST_BUFFER_PTS (handoff_buffer) +
|
|
GST_BUFFER_DURATION (handoff_buffer), 150 * GST_MSECOND);
|
|
gst_buffer_replace (&handoff_buffer, NULL);
|
|
|
|
/* should not be clipped */
|
|
buffer = new_buffer (44100, 0, 1150 * GST_MSECOND, 250 * GST_MSECOND, 0);
|
|
GST_DEBUG ("pushing buffer %p END is %" GST_TIME_FORMAT,
|
|
buffer,
|
|
GST_TIME_ARGS (GST_BUFFER_PTS (buffer) + GST_BUFFER_DURATION (buffer)));
|
|
ret = gst_pad_chain (sinkpad, buffer);
|
|
ck_assert_int_eq (ret, GST_FLOW_OK);
|
|
gst_pad_query (sinkpad, drain);
|
|
fail_unless (handoff_buffer != NULL);
|
|
ck_assert_int_eq (GST_BUFFER_PTS (handoff_buffer) +
|
|
GST_BUFFER_DURATION (handoff_buffer), 400 * GST_MSECOND);
|
|
gst_buffer_replace (&handoff_buffer, NULL);
|
|
fail_unless (handoff_buffer == NULL);
|
|
|
|
/* should be clipped and ok */
|
|
buffer = new_buffer (44100, 0, 2 * GST_SECOND, 250 * GST_MSECOND,
|
|
GST_BUFFER_FLAG_DISCONT);
|
|
GST_DEBUG ("pushing buffer %p PTS is %" GST_TIME_FORMAT
|
|
" END is %" GST_TIME_FORMAT,
|
|
buffer,
|
|
GST_TIME_ARGS (GST_BUFFER_PTS (buffer)),
|
|
GST_TIME_ARGS (GST_BUFFER_PTS (buffer) + GST_BUFFER_DURATION (buffer)));
|
|
ret = gst_pad_chain (sinkpad, buffer);
|
|
ck_assert_int_eq (ret, GST_FLOW_OK);
|
|
gst_pad_query (sinkpad, drain);
|
|
fail_unless (handoff_buffer == NULL);
|
|
|
|
gst_element_release_request_pad (audiomixer, sinkpad);
|
|
gst_object_unref (sinkpad);
|
|
gst_element_set_state (bin, GST_STATE_NULL);
|
|
gst_bus_remove_signal_watch (bus);
|
|
gst_object_unref (bus);
|
|
gst_object_unref (bin);
|
|
gst_query_unref (drain);
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
GST_START_TEST (test_duration_is_max)
|
|
{
|
|
GstElement *bin, *src[3], *audiomixer, *sink;
|
|
GstStateChangeReturn state_res;
|
|
GstFormat format = GST_FORMAT_TIME;
|
|
gboolean res;
|
|
gint64 duration;
|
|
|
|
GST_INFO ("preparing test");
|
|
|
|
/* build pipeline */
|
|
bin = gst_pipeline_new ("pipeline");
|
|
|
|
/* 3 sources, an audiomixer and a fakesink */
|
|
src[0] = gst_element_factory_make ("audiotestsrc", NULL);
|
|
src[1] = gst_element_factory_make ("audiotestsrc", NULL);
|
|
src[2] = gst_element_factory_make ("audiotestsrc", NULL);
|
|
audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
|
|
sink = gst_element_factory_make ("fakesink", "sink");
|
|
gst_bin_add_many (GST_BIN (bin), src[0], src[1], src[2], audiomixer, sink,
|
|
NULL);
|
|
|
|
gst_element_link (src[0], audiomixer);
|
|
gst_element_link (src[1], audiomixer);
|
|
gst_element_link (src[2], audiomixer);
|
|
gst_element_link (audiomixer, sink);
|
|
|
|
/* irks, duration is reset on basesrc */
|
|
state_res = gst_element_set_state (bin, GST_STATE_PAUSED);
|
|
fail_unless (state_res != GST_STATE_CHANGE_FAILURE, NULL);
|
|
|
|
/* set durations on src */
|
|
GST_BASE_SRC (src[0])->segment.duration = 1000;
|
|
GST_BASE_SRC (src[1])->segment.duration = 3000;
|
|
GST_BASE_SRC (src[2])->segment.duration = 2000;
|
|
|
|
/* set to playing */
|
|
set_state_and_wait (bin, GST_STATE_PLAYING);
|
|
|
|
res = gst_element_query_duration (GST_ELEMENT (bin), format, &duration);
|
|
fail_unless (res, NULL);
|
|
|
|
ck_assert_int_eq (duration, 3000);
|
|
|
|
gst_element_set_state (bin, GST_STATE_NULL);
|
|
gst_object_unref (bin);
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
GST_START_TEST (test_duration_unknown_overrides)
|
|
{
|
|
GstElement *bin, *src[3], *audiomixer, *sink;
|
|
GstStateChangeReturn state_res;
|
|
GstFormat format = GST_FORMAT_TIME;
|
|
gboolean res;
|
|
gint64 duration;
|
|
|
|
GST_INFO ("preparing test");
|
|
|
|
/* build pipeline */
|
|
bin = gst_pipeline_new ("pipeline");
|
|
|
|
/* 3 sources, an audiomixer and a fakesink */
|
|
src[0] = gst_element_factory_make ("audiotestsrc", NULL);
|
|
src[1] = gst_element_factory_make ("audiotestsrc", NULL);
|
|
src[2] = gst_element_factory_make ("audiotestsrc", NULL);
|
|
audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
|
|
sink = gst_element_factory_make ("fakesink", "sink");
|
|
gst_bin_add_many (GST_BIN (bin), src[0], src[1], src[2], audiomixer, sink,
|
|
NULL);
|
|
|
|
gst_element_link (src[0], audiomixer);
|
|
gst_element_link (src[1], audiomixer);
|
|
gst_element_link (src[2], audiomixer);
|
|
gst_element_link (audiomixer, sink);
|
|
|
|
/* irks, duration is reset on basesrc */
|
|
state_res = gst_element_set_state (bin, GST_STATE_PAUSED);
|
|
fail_unless (state_res != GST_STATE_CHANGE_FAILURE, NULL);
|
|
|
|
/* set durations on src */
|
|
GST_BASE_SRC (src[0])->segment.duration = GST_CLOCK_TIME_NONE;
|
|
GST_BASE_SRC (src[1])->segment.duration = 3000;
|
|
GST_BASE_SRC (src[2])->segment.duration = 2000;
|
|
|
|
/* set to playing */
|
|
set_state_and_wait (bin, GST_STATE_PLAYING);
|
|
|
|
res = gst_element_query_duration (GST_ELEMENT (bin), format, &duration);
|
|
fail_unless (res, NULL);
|
|
|
|
ck_assert_int_eq (duration, GST_CLOCK_TIME_NONE);
|
|
|
|
gst_element_set_state (bin, GST_STATE_NULL);
|
|
gst_object_unref (bin);
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
|
|
static gboolean looped = FALSE;
|
|
|
|
static void
|
|
loop_segment_done (GstBus * bus, GstMessage * message, GstElement * bin)
|
|
{
|
|
GST_INFO ("bus message from \"%" GST_PTR_FORMAT "\": %" GST_PTR_FORMAT,
|
|
GST_MESSAGE_SRC (message), message);
|
|
|
|
if (looped) {
|
|
g_main_loop_quit (main_loop);
|
|
} else {
|
|
GstEvent *seek_event;
|
|
gboolean res;
|
|
|
|
seek_event = gst_event_new_seek (1.0, GST_FORMAT_TIME,
|
|
GST_SEEK_FLAG_SEGMENT,
|
|
GST_SEEK_TYPE_SET, (GstClockTime) 0,
|
|
GST_SEEK_TYPE_SET, (GstClockTime) 1 * GST_SECOND);
|
|
|
|
res = gst_element_send_event (bin, seek_event);
|
|
fail_unless (res == TRUE, NULL);
|
|
looped = TRUE;
|
|
}
|
|
}
|
|
|
|
GST_START_TEST (test_loop)
|
|
{
|
|
GstElement *bin;
|
|
GstBus *bus;
|
|
GstEvent *seek_event;
|
|
gboolean res;
|
|
|
|
GST_INFO ("preparing test");
|
|
|
|
/* build pipeline */
|
|
bin = setup_pipeline (NULL, 2, NULL);
|
|
bus = gst_element_get_bus (bin);
|
|
gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH);
|
|
|
|
seek_event = gst_event_new_seek (1.0, GST_FORMAT_TIME,
|
|
GST_SEEK_FLAG_SEGMENT | GST_SEEK_FLAG_FLUSH,
|
|
GST_SEEK_TYPE_SET, (GstClockTime) 0,
|
|
GST_SEEK_TYPE_SET, (GstClockTime) 1 * GST_SECOND);
|
|
|
|
g_signal_connect (bus, "message::segment-done",
|
|
(GCallback) loop_segment_done, bin);
|
|
g_signal_connect (bus, "message::error", (GCallback) message_received, bin);
|
|
g_signal_connect (bus, "message::warning", (GCallback) message_received, bin);
|
|
g_signal_connect (bus, "message::eos", (GCallback) message_received, bin);
|
|
|
|
GST_INFO ("starting test");
|
|
|
|
/* prepare playing */
|
|
set_state_and_wait (bin, GST_STATE_PAUSED);
|
|
|
|
res = gst_element_send_event (bin, seek_event);
|
|
fail_unless (res == TRUE, NULL);
|
|
|
|
/* run pipeline */
|
|
play_and_wait (bin);
|
|
|
|
fail_unless (looped);
|
|
|
|
/* cleanup */
|
|
gst_bus_remove_signal_watch (bus);
|
|
gst_object_unref (bus);
|
|
gst_object_unref (bin);
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
GST_START_TEST (test_flush_start_flush_stop)
|
|
{
|
|
GstPadTemplate *sink_template;
|
|
GstPad *tmppad, *srcpad1, *sinkpad1, *sinkpad2, *audiomixer_src;
|
|
GstElement *pipeline, *src1, *src2, *audiomixer, *sink;
|
|
|
|
GST_INFO ("preparing test");
|
|
|
|
/* build pipeline */
|
|
pipeline = gst_pipeline_new ("pipeline");
|
|
src1 = gst_element_factory_make ("audiotestsrc", "src1");
|
|
g_object_set (src1, "wave", 4, NULL); /* silence */
|
|
src2 = gst_element_factory_make ("audiotestsrc", "src2");
|
|
g_object_set (src2, "wave", 4, NULL); /* silence */
|
|
audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
|
|
sink = gst_element_factory_make ("fakesink", "sink");
|
|
gst_bin_add_many (GST_BIN (pipeline), src1, src2, audiomixer, sink, NULL);
|
|
|
|
sink_template =
|
|
gst_element_class_get_pad_template (GST_ELEMENT_GET_CLASS (audiomixer),
|
|
"sink_%u");
|
|
fail_unless (GST_IS_PAD_TEMPLATE (sink_template));
|
|
sinkpad1 = gst_element_request_pad (audiomixer, sink_template, NULL, NULL);
|
|
srcpad1 = gst_element_get_static_pad (src1, "src");
|
|
gst_pad_link (srcpad1, sinkpad1);
|
|
|
|
sinkpad2 = gst_element_request_pad (audiomixer, sink_template, NULL, NULL);
|
|
tmppad = gst_element_get_static_pad (src2, "src");
|
|
gst_pad_link (tmppad, sinkpad2);
|
|
gst_object_unref (tmppad);
|
|
|
|
gst_element_link (audiomixer, sink);
|
|
|
|
/* prepare playing */
|
|
set_state_and_wait (pipeline, GST_STATE_PLAYING);
|
|
|
|
audiomixer_src = gst_element_get_static_pad (audiomixer, "src");
|
|
fail_if (GST_PAD_IS_FLUSHING (audiomixer_src));
|
|
gst_pad_send_event (sinkpad1, gst_event_new_flush_start ());
|
|
fail_if (GST_PAD_IS_FLUSHING (audiomixer_src));
|
|
fail_unless (GST_PAD_IS_FLUSHING (sinkpad1));
|
|
/* Hold the streamlock to make sure the flush stop is not between
|
|
the attempted push of a segment event and of the following buffer. */
|
|
GST_PAD_STREAM_LOCK (srcpad1);
|
|
gst_pad_send_event (sinkpad1, gst_event_new_flush_stop (TRUE));
|
|
GST_PAD_STREAM_UNLOCK (srcpad1);
|
|
fail_if (GST_PAD_IS_FLUSHING (audiomixer_src));
|
|
fail_if (GST_PAD_IS_FLUSHING (sinkpad1));
|
|
gst_object_unref (audiomixer_src);
|
|
|
|
gst_element_release_request_pad (audiomixer, sinkpad1);
|
|
gst_object_unref (sinkpad1);
|
|
gst_element_release_request_pad (audiomixer, sinkpad2);
|
|
gst_object_unref (sinkpad2);
|
|
gst_object_unref (srcpad1);
|
|
|
|
/* cleanup */
|
|
gst_element_set_state (pipeline, GST_STATE_NULL);
|
|
gst_object_unref (pipeline);
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
static void
|
|
handoff_buffer_collect_cb (GstElement * fakesink, GstBuffer * buffer,
|
|
GstPad * pad, gpointer user_data)
|
|
{
|
|
GList **received_buffers = user_data;
|
|
|
|
GST_DEBUG ("got buffer %p", buffer);
|
|
*received_buffers =
|
|
g_list_append (*received_buffers, gst_buffer_ref (buffer));
|
|
}
|
|
|
|
typedef void (*SendBuffersFunction) (GstPad * pad1, GstPad * pad2);
|
|
typedef void (*CheckBuffersFunction) (GList * buffers);
|
|
|
|
static void
|
|
run_sync_test (SendBuffersFunction send_buffers,
|
|
CheckBuffersFunction check_buffers)
|
|
{
|
|
GstSegment segment;
|
|
GstElement *bin, *audiomixer, *queue1, *queue2, *sink;
|
|
GstBus *bus;
|
|
GstPad *sinkpad1, *sinkpad2;
|
|
GstPad *queue1_sinkpad, *queue2_sinkpad;
|
|
GstPad *pad;
|
|
gboolean res;
|
|
GstStateChangeReturn state_res;
|
|
GstEvent *event;
|
|
GstCaps *caps;
|
|
GList *received_buffers = NULL;
|
|
|
|
GST_INFO ("preparing test");
|
|
|
|
/* build pipeline */
|
|
bin = gst_pipeline_new ("pipeline");
|
|
bus = gst_element_get_bus (bin);
|
|
gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH);
|
|
|
|
g_signal_connect (bus, "message::error", (GCallback) message_received, bin);
|
|
g_signal_connect (bus, "message::warning", (GCallback) message_received, bin);
|
|
g_signal_connect (bus, "message::eos", (GCallback) message_received, bin);
|
|
|
|
/* just an audiomixer and a fakesink */
|
|
queue1 = gst_element_factory_make ("queue", "queue1");
|
|
queue2 = gst_element_factory_make ("queue", "queue2");
|
|
audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
|
|
g_object_set (audiomixer, "output-buffer-duration", 500 * GST_MSECOND, NULL);
|
|
sink = gst_element_factory_make ("fakesink", "sink");
|
|
g_object_set (sink, "signal-handoffs", TRUE, NULL);
|
|
g_signal_connect (sink, "handoff", (GCallback) handoff_buffer_collect_cb,
|
|
&received_buffers);
|
|
gst_bin_add_many (GST_BIN (bin), queue1, queue2, audiomixer, sink, NULL);
|
|
|
|
res = gst_element_link (audiomixer, sink);
|
|
fail_unless (res == TRUE, NULL);
|
|
|
|
/* set to paused */
|
|
state_res = gst_element_set_state (bin, GST_STATE_PAUSED);
|
|
ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
|
|
|
|
/* create an unconnected sinkpad in audiomixer, should also automatically activate
|
|
* the pad */
|
|
sinkpad1 = gst_element_get_request_pad (audiomixer, "sink_%u");
|
|
fail_if (sinkpad1 == NULL, NULL);
|
|
|
|
queue1_sinkpad = gst_element_get_static_pad (queue1, "sink");
|
|
pad = gst_element_get_static_pad (queue1, "src");
|
|
fail_unless (gst_pad_link (pad, sinkpad1) == GST_PAD_LINK_OK);
|
|
gst_object_unref (pad);
|
|
|
|
sinkpad2 = gst_element_get_request_pad (audiomixer, "sink_%u");
|
|
fail_if (sinkpad2 == NULL, NULL);
|
|
|
|
queue2_sinkpad = gst_element_get_static_pad (queue2, "sink");
|
|
pad = gst_element_get_static_pad (queue2, "src");
|
|
fail_unless (gst_pad_link (pad, sinkpad2) == GST_PAD_LINK_OK);
|
|
gst_object_unref (pad);
|
|
|
|
gst_pad_send_event (queue1_sinkpad, gst_event_new_stream_start ("test"));
|
|
gst_pad_send_event (queue2_sinkpad, gst_event_new_stream_start ("test"));
|
|
|
|
caps = gst_caps_new_simple ("audio/x-raw",
|
|
"format", G_TYPE_STRING, GST_AUDIO_NE (S16),
|
|
"layout", G_TYPE_STRING, "interleaved",
|
|
"rate", G_TYPE_INT, 1000, "channels", G_TYPE_INT, 1, NULL);
|
|
|
|
gst_pad_set_caps (queue1_sinkpad, caps);
|
|
gst_pad_set_caps (queue2_sinkpad, caps);
|
|
gst_caps_unref (caps);
|
|
|
|
/* send segment to audiomixer */
|
|
gst_segment_init (&segment, GST_FORMAT_TIME);
|
|
event = gst_event_new_segment (&segment);
|
|
gst_pad_send_event (queue1_sinkpad, gst_event_ref (event));
|
|
gst_pad_send_event (queue2_sinkpad, event);
|
|
|
|
/* Push buffers */
|
|
send_buffers (queue1_sinkpad, queue2_sinkpad);
|
|
|
|
/* Set PLAYING */
|
|
g_idle_add ((GSourceFunc) set_playing, bin);
|
|
|
|
/* Collect buffers and messages */
|
|
g_main_loop_run (main_loop);
|
|
|
|
/* Here we get once we got EOS, for errors we failed */
|
|
|
|
check_buffers (received_buffers);
|
|
|
|
g_list_free_full (received_buffers, (GDestroyNotify) gst_buffer_unref);
|
|
|
|
gst_element_release_request_pad (audiomixer, sinkpad1);
|
|
gst_object_unref (sinkpad1);
|
|
gst_object_unref (queue1_sinkpad);
|
|
gst_element_release_request_pad (audiomixer, sinkpad2);
|
|
gst_object_unref (sinkpad2);
|
|
gst_object_unref (queue2_sinkpad);
|
|
gst_element_set_state (bin, GST_STATE_NULL);
|
|
gst_bus_remove_signal_watch (bus);
|
|
gst_object_unref (bus);
|
|
gst_object_unref (bin);
|
|
}
|
|
|
|
static void
|
|
send_buffers_sync (GstPad * pad1, GstPad * pad2)
|
|
{
|
|
GstBuffer *buffer;
|
|
GstFlowReturn ret;
|
|
|
|
buffer = new_buffer (2000, 1, 1 * GST_SECOND, 1 * GST_SECOND, 0);
|
|
ret = gst_pad_chain (pad1, buffer);
|
|
ck_assert_int_eq (ret, GST_FLOW_OK);
|
|
|
|
buffer = new_buffer (2000, 1, 2 * GST_SECOND, 1 * GST_SECOND, 0);
|
|
ret = gst_pad_chain (pad1, buffer);
|
|
ck_assert_int_eq (ret, GST_FLOW_OK);
|
|
|
|
gst_pad_send_event (pad1, gst_event_new_eos ());
|
|
|
|
buffer = new_buffer (2000, 2, 2 * GST_SECOND, 1 * GST_SECOND, 0);
|
|
ret = gst_pad_chain (pad2, buffer);
|
|
ck_assert_int_eq (ret, GST_FLOW_OK);
|
|
|
|
buffer = new_buffer (2000, 2, 3 * GST_SECOND, 1 * GST_SECOND, 0);
|
|
ret = gst_pad_chain (pad2, buffer);
|
|
ck_assert_int_eq (ret, GST_FLOW_OK);
|
|
|
|
gst_pad_send_event (pad2, gst_event_new_eos ());
|
|
}
|
|
|
|
static void
|
|
check_buffers_sync (GList * received_buffers)
|
|
{
|
|
GstBuffer *buffer;
|
|
GList *l;
|
|
gint i;
|
|
GstMapInfo map;
|
|
|
|
/* Should have 8 * 0.5s buffers */
|
|
fail_unless_equals_int (g_list_length (received_buffers), 8);
|
|
for (i = 0, l = received_buffers; l; l = l->next, i++) {
|
|
buffer = l->data;
|
|
|
|
gst_buffer_map (buffer, &map, GST_MAP_READ);
|
|
|
|
if (i == 0 && GST_BUFFER_TIMESTAMP (buffer) == 0) {
|
|
fail_unless (map.data[0] == 0);
|
|
fail_unless (map.data[map.size - 1] == 0);
|
|
} else if (i == 1 && GST_BUFFER_TIMESTAMP (buffer) == 500 * GST_MSECOND) {
|
|
fail_unless (map.data[0] == 0);
|
|
fail_unless (map.data[map.size - 1] == 0);
|
|
} else if (i == 2 && GST_BUFFER_TIMESTAMP (buffer) == 1000 * GST_MSECOND) {
|
|
fail_unless (map.data[0] == 1);
|
|
fail_unless (map.data[map.size - 1] == 1);
|
|
} else if (i == 3 && GST_BUFFER_TIMESTAMP (buffer) == 1500 * GST_MSECOND) {
|
|
fail_unless (map.data[0] == 1);
|
|
fail_unless (map.data[map.size - 1] == 1);
|
|
} else if (i == 4 && GST_BUFFER_TIMESTAMP (buffer) == 2000 * GST_MSECOND) {
|
|
fail_unless (map.data[0] == 3);
|
|
fail_unless (map.data[map.size - 1] == 3);
|
|
} else if (i == 5 && GST_BUFFER_TIMESTAMP (buffer) == 2500 * GST_MSECOND) {
|
|
fail_unless (map.data[0] == 3);
|
|
fail_unless (map.data[map.size - 1] == 3);
|
|
} else if (i == 6 && GST_BUFFER_TIMESTAMP (buffer) == 3000 * GST_MSECOND) {
|
|
fail_unless (map.data[0] == 2);
|
|
fail_unless (map.data[map.size - 1] == 2);
|
|
} else if (i == 7 && GST_BUFFER_TIMESTAMP (buffer) == 3500 * GST_MSECOND) {
|
|
fail_unless (map.data[0] == 2);
|
|
fail_unless (map.data[map.size - 1] == 2);
|
|
} else {
|
|
g_assert_not_reached ();
|
|
}
|
|
|
|
gst_buffer_unmap (buffer, &map);
|
|
|
|
}
|
|
}
|
|
|
|
GST_START_TEST (test_sync)
|
|
{
|
|
run_sync_test (send_buffers_sync, check_buffers_sync);
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
static void
|
|
send_buffers_sync_discont (GstPad * pad1, GstPad * pad2)
|
|
{
|
|
GstBuffer *buffer;
|
|
GstFlowReturn ret;
|
|
|
|
buffer = new_buffer (2000, 1, 1 * GST_SECOND, 1 * GST_SECOND, 0);
|
|
ret = gst_pad_chain (pad1, buffer);
|
|
ck_assert_int_eq (ret, GST_FLOW_OK);
|
|
|
|
buffer = new_buffer (2000, 1, 3 * GST_SECOND, 1 * GST_SECOND,
|
|
GST_BUFFER_FLAG_DISCONT);
|
|
ret = gst_pad_chain (pad1, buffer);
|
|
ck_assert_int_eq (ret, GST_FLOW_OK);
|
|
|
|
gst_pad_send_event (pad1, gst_event_new_eos ());
|
|
|
|
buffer = new_buffer (2000, 2, 2 * GST_SECOND, 1 * GST_SECOND, 0);
|
|
ret = gst_pad_chain (pad2, buffer);
|
|
ck_assert_int_eq (ret, GST_FLOW_OK);
|
|
|
|
buffer = new_buffer (2000, 2, 3 * GST_SECOND, 1 * GST_SECOND, 0);
|
|
ret = gst_pad_chain (pad2, buffer);
|
|
ck_assert_int_eq (ret, GST_FLOW_OK);
|
|
|
|
gst_pad_send_event (pad2, gst_event_new_eos ());
|
|
}
|
|
|
|
static void
|
|
check_buffers_sync_discont (GList * received_buffers)
|
|
{
|
|
GstBuffer *buffer;
|
|
GList *l;
|
|
gint i;
|
|
GstMapInfo map;
|
|
|
|
/* Should have 8 * 0.5s buffers */
|
|
fail_unless_equals_int (g_list_length (received_buffers), 8);
|
|
for (i = 0, l = received_buffers; l; l = l->next, i++) {
|
|
buffer = l->data;
|
|
|
|
gst_buffer_map (buffer, &map, GST_MAP_READ);
|
|
|
|
if (i == 0 && GST_BUFFER_TIMESTAMP (buffer) == 0) {
|
|
fail_unless (map.data[0] == 0);
|
|
fail_unless (map.data[map.size - 1] == 0);
|
|
} else if (i == 1 && GST_BUFFER_TIMESTAMP (buffer) == 500 * GST_MSECOND) {
|
|
fail_unless (map.data[0] == 0);
|
|
fail_unless (map.data[map.size - 1] == 0);
|
|
} else if (i == 2 && GST_BUFFER_TIMESTAMP (buffer) == 1000 * GST_MSECOND) {
|
|
fail_unless (map.data[0] == 1);
|
|
fail_unless (map.data[map.size - 1] == 1);
|
|
} else if (i == 3 && GST_BUFFER_TIMESTAMP (buffer) == 1500 * GST_MSECOND) {
|
|
fail_unless (map.data[0] == 1);
|
|
fail_unless (map.data[map.size - 1] == 1);
|
|
} else if (i == 4 && GST_BUFFER_TIMESTAMP (buffer) == 2000 * GST_MSECOND) {
|
|
fail_unless (map.data[0] == 2);
|
|
fail_unless (map.data[map.size - 1] == 2);
|
|
} else if (i == 5 && GST_BUFFER_TIMESTAMP (buffer) == 2500 * GST_MSECOND) {
|
|
fail_unless (map.data[0] == 2);
|
|
fail_unless (map.data[map.size - 1] == 2);
|
|
} else if (i == 6 && GST_BUFFER_TIMESTAMP (buffer) == 3000 * GST_MSECOND) {
|
|
fail_unless (map.data[0] == 3);
|
|
fail_unless (map.data[map.size - 1] == 3);
|
|
} else if (i == 7 && GST_BUFFER_TIMESTAMP (buffer) == 3500 * GST_MSECOND) {
|
|
fail_unless (map.data[0] == 3);
|
|
fail_unless (map.data[map.size - 1] == 3);
|
|
} else {
|
|
g_assert_not_reached ();
|
|
}
|
|
|
|
gst_buffer_unmap (buffer, &map);
|
|
|
|
}
|
|
}
|
|
|
|
GST_START_TEST (test_sync_discont)
|
|
{
|
|
run_sync_test (send_buffers_sync_discont, check_buffers_sync_discont);
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
static void
|
|
send_buffers_sync_unaligned (GstPad * pad1, GstPad * pad2)
|
|
{
|
|
GstBuffer *buffer;
|
|
GstFlowReturn ret;
|
|
|
|
buffer = new_buffer (2000, 1, 750 * GST_MSECOND, 1 * GST_SECOND, 0);
|
|
ret = gst_pad_chain (pad1, buffer);
|
|
ck_assert_int_eq (ret, GST_FLOW_OK);
|
|
|
|
buffer = new_buffer (2000, 1, 1750 * GST_MSECOND, 1 * GST_SECOND, 0);
|
|
ret = gst_pad_chain (pad1, buffer);
|
|
ck_assert_int_eq (ret, GST_FLOW_OK);
|
|
|
|
gst_pad_send_event (pad1, gst_event_new_eos ());
|
|
|
|
buffer = new_buffer (2000, 2, 1750 * GST_MSECOND, 1 * GST_SECOND, 0);
|
|
ret = gst_pad_chain (pad2, buffer);
|
|
ck_assert_int_eq (ret, GST_FLOW_OK);
|
|
|
|
buffer = new_buffer (2000, 2, 2750 * GST_MSECOND, 1 * GST_SECOND, 0);
|
|
ret = gst_pad_chain (pad2, buffer);
|
|
ck_assert_int_eq (ret, GST_FLOW_OK);
|
|
|
|
gst_pad_send_event (pad2, gst_event_new_eos ());
|
|
}
|
|
|
|
static void
|
|
check_buffers_sync_unaligned (GList * received_buffers)
|
|
{
|
|
GstBuffer *buffer;
|
|
GList *l;
|
|
gint i;
|
|
GstMapInfo map;
|
|
|
|
/* Should have 8 * 0.5s buffers */
|
|
fail_unless_equals_int (g_list_length (received_buffers), 8);
|
|
for (i = 0, l = received_buffers; l; l = l->next, i++) {
|
|
buffer = l->data;
|
|
|
|
gst_buffer_map (buffer, &map, GST_MAP_READ);
|
|
|
|
if (i == 0 && GST_BUFFER_TIMESTAMP (buffer) == 0) {
|
|
fail_unless (map.data[0] == 0);
|
|
fail_unless (map.data[map.size - 1] == 0);
|
|
} else if (i == 1 && GST_BUFFER_TIMESTAMP (buffer) == 500 * GST_MSECOND) {
|
|
fail_unless (map.data[0] == 0);
|
|
fail_unless (map.data[499] == 0);
|
|
fail_unless (map.data[500] == 1);
|
|
fail_unless (map.data[map.size - 1] == 1);
|
|
} else if (i == 2 && GST_BUFFER_TIMESTAMP (buffer) == 1000 * GST_MSECOND) {
|
|
fail_unless (map.data[0] == 1);
|
|
fail_unless (map.data[map.size - 1] == 1);
|
|
} else if (i == 3 && GST_BUFFER_TIMESTAMP (buffer) == 1500 * GST_MSECOND) {
|
|
fail_unless (map.data[0] == 1);
|
|
fail_unless (map.data[499] == 1);
|
|
fail_unless (map.data[500] == 3);
|
|
fail_unless (map.data[map.size - 1] == 3);
|
|
} else if (i == 4 && GST_BUFFER_TIMESTAMP (buffer) == 2000 * GST_MSECOND) {
|
|
fail_unless (map.data[0] == 3);
|
|
fail_unless (map.data[499] == 3);
|
|
fail_unless (map.data[500] == 3);
|
|
fail_unless (map.data[map.size - 1] == 3);
|
|
} else if (i == 5 && GST_BUFFER_TIMESTAMP (buffer) == 2500 * GST_MSECOND) {
|
|
fail_unless (map.data[0] == 3);
|
|
fail_unless (map.data[499] == 3);
|
|
fail_unless (map.data[500] == 2);
|
|
fail_unless (map.data[map.size - 1] == 2);
|
|
} else if (i == 6 && GST_BUFFER_TIMESTAMP (buffer) == 3000 * GST_MSECOND) {
|
|
fail_unless (map.data[0] == 2);
|
|
fail_unless (map.data[499] == 2);
|
|
fail_unless (map.data[500] == 2);
|
|
fail_unless (map.data[map.size - 1] == 2);
|
|
} else if (i == 7 && GST_BUFFER_TIMESTAMP (buffer) == 3500 * GST_MSECOND) {
|
|
fail_unless (map.size == 500);
|
|
fail_unless (GST_BUFFER_DURATION (buffer) == 250 * GST_MSECOND);
|
|
fail_unless (map.data[0] == 2);
|
|
fail_unless (map.data[499] == 2);
|
|
} else {
|
|
g_assert_not_reached ();
|
|
}
|
|
|
|
gst_buffer_unmap (buffer, &map);
|
|
|
|
}
|
|
}
|
|
|
|
GST_START_TEST (test_sync_unaligned)
|
|
{
|
|
run_sync_test (send_buffers_sync_unaligned, check_buffers_sync_unaligned);
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
GST_START_TEST (test_segment_base_handling)
|
|
{
|
|
GstElement *pipeline, *sink, *mix, *src1, *src2;
|
|
GstPad *srcpad, *sinkpad;
|
|
GstClockTime end_time;
|
|
GstSample *last_sample = NULL;
|
|
GstSample *sample;
|
|
GstBuffer *buf;
|
|
GstCaps *caps;
|
|
|
|
caps = gst_caps_new_simple ("audio/x-raw", "rate", G_TYPE_INT, 44100,
|
|
"channels", G_TYPE_INT, 2, NULL);
|
|
|
|
pipeline = gst_pipeline_new ("pipeline");
|
|
mix = gst_element_factory_make ("audiomixer", "audiomixer");
|
|
sink = gst_element_factory_make ("appsink", "sink");
|
|
g_object_set (sink, "caps", caps, "sync", FALSE, NULL);
|
|
gst_caps_unref (caps);
|
|
/* 50 buffers of 1/10 sec = 5 sec */
|
|
src1 = gst_element_factory_make ("audiotestsrc", "src1");
|
|
g_object_set (src1, "samplesperbuffer", 4410, "num-buffers", 50, NULL);
|
|
src2 = gst_element_factory_make ("audiotestsrc", "src2");
|
|
g_object_set (src2, "samplesperbuffer", 4410, "num-buffers", 50, NULL);
|
|
gst_bin_add_many (GST_BIN (pipeline), src1, src2, 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_1");
|
|
fail_unless (gst_pad_link (srcpad, sinkpad) == GST_PAD_LINK_OK);
|
|
gst_object_unref (sinkpad);
|
|
gst_object_unref (srcpad);
|
|
|
|
srcpad = gst_element_get_static_pad (src2, "src");
|
|
sinkpad = gst_element_get_request_pad (mix, "sink_2");
|
|
fail_unless (gst_pad_link (srcpad, sinkpad) == GST_PAD_LINK_OK);
|
|
/* set a pad offset of another 5 seconds */
|
|
gst_pad_set_offset (sinkpad, 5 * GST_SECOND);
|
|
gst_object_unref (sinkpad);
|
|
gst_object_unref (srcpad);
|
|
|
|
gst_element_set_state (pipeline, GST_STATE_PLAYING);
|
|
|
|
do {
|
|
g_signal_emit_by_name (sink, "pull-sample", &sample);
|
|
if (sample == NULL)
|
|
break;
|
|
if (last_sample)
|
|
gst_sample_unref (last_sample);
|
|
last_sample = sample;
|
|
} while (TRUE);
|
|
|
|
buf = gst_sample_get_buffer (last_sample);
|
|
end_time = GST_BUFFER_TIMESTAMP (buf) + GST_BUFFER_DURATION (buf);
|
|
fail_unless_equals_int64 (end_time, 10 * GST_SECOND);
|
|
gst_sample_unref (last_sample);
|
|
|
|
gst_element_set_state (pipeline, GST_STATE_NULL);
|
|
gst_object_unref (pipeline);
|
|
}
|
|
|
|
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 void
|
|
change_src_caps (GstElement * fakesink, GstBuffer * buffer, GstPad * pad,
|
|
GstElement * capsfilter)
|
|
{
|
|
GstCaps *caps = gst_caps_new_simple ("audio/x-raw",
|
|
"format", G_TYPE_STRING, GST_AUDIO_NE (S32),
|
|
"layout", G_TYPE_STRING, "interleaved",
|
|
"rate", G_TYPE_INT, 10, "channels", G_TYPE_INT, 1, NULL);
|
|
|
|
g_object_set (capsfilter, "caps", caps, NULL);
|
|
gst_caps_unref (caps);
|
|
g_signal_connect (fakesink, "handoff", (GCallback) handoff_buffer_cb, NULL);
|
|
g_signal_handlers_disconnect_by_func (fakesink, change_src_caps, capsfilter);
|
|
}
|
|
|
|
/* In this test, we create an input buffer with a duration of 2 seconds,
|
|
* and require the audiomixer to output 1 second long buffers.
|
|
* The input buffer will thus be mixed twice, and the audiomixer will
|
|
* output two buffers.
|
|
*
|
|
* After audiomixer has output a first buffer, we change its output format
|
|
* from S8 to S32. As our sample rate stays the same at 10 fps, and we use
|
|
* mono, the first buffer should be 10 bytes long, and the second 40.
|
|
*
|
|
* The input buffer is made up of 15 0-valued bytes, and 5 1-valued bytes.
|
|
* We verify that the second buffer contains 5 0-valued integers, and
|
|
* 5 1 << 24 valued integers.
|
|
*/
|
|
GST_START_TEST (test_change_output_caps)
|
|
{
|
|
GstSegment segment;
|
|
GstElement *bin, *audiomixer, *capsfilter, *sink;
|
|
GstBus *bus;
|
|
GstPad *sinkpad;
|
|
gboolean res;
|
|
GstStateChangeReturn state_res;
|
|
GstFlowReturn ret;
|
|
GstEvent *event;
|
|
GstBuffer *buffer;
|
|
GstCaps *caps;
|
|
GstQuery *drain = gst_query_new_drain ();
|
|
GstMapInfo inmap;
|
|
GstMapInfo outmap;
|
|
gsize i;
|
|
|
|
bin = gst_pipeline_new ("pipeline");
|
|
bus = gst_element_get_bus (bin);
|
|
gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH);
|
|
|
|
g_signal_connect (bus, "message::error", (GCallback) message_received, bin);
|
|
g_signal_connect (bus, "message::warning", (GCallback) message_received, bin);
|
|
g_signal_connect (bus, "message::eos", (GCallback) message_received, bin);
|
|
|
|
audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
|
|
g_object_set (audiomixer, "output-buffer-duration", GST_SECOND, NULL);
|
|
capsfilter = gst_element_factory_make ("capsfilter", NULL);
|
|
sink = gst_element_factory_make ("fakesink", "sink");
|
|
g_object_set (sink, "signal-handoffs", TRUE, NULL);
|
|
g_signal_connect (sink, "handoff", (GCallback) change_src_caps, capsfilter);
|
|
gst_bin_add_many (GST_BIN (bin), audiomixer, capsfilter, sink, NULL);
|
|
|
|
res = gst_element_link_many (audiomixer, capsfilter, sink, NULL);
|
|
fail_unless (res == TRUE, NULL);
|
|
|
|
state_res = gst_element_set_state (bin, GST_STATE_PLAYING);
|
|
ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
|
|
|
|
sinkpad = gst_element_get_request_pad (audiomixer, "sink_%u");
|
|
fail_if (sinkpad == NULL, NULL);
|
|
|
|
gst_pad_send_event (sinkpad, gst_event_new_stream_start ("test"));
|
|
|
|
caps = gst_caps_new_simple ("audio/x-raw",
|
|
"format", G_TYPE_STRING, "S8",
|
|
"layout", G_TYPE_STRING, "interleaved",
|
|
"rate", G_TYPE_INT, 10, "channels", G_TYPE_INT, 1, NULL);
|
|
|
|
gst_pad_set_caps (sinkpad, caps);
|
|
g_object_set (capsfilter, "caps", caps, NULL);
|
|
gst_caps_unref (caps);
|
|
|
|
gst_segment_init (&segment, GST_FORMAT_TIME);
|
|
segment.start = 0;
|
|
segment.stop = 2 * GST_SECOND;
|
|
segment.time = 0;
|
|
event = gst_event_new_segment (&segment);
|
|
gst_pad_send_event (sinkpad, event);
|
|
|
|
gst_buffer_replace (&handoff_buffer, NULL);
|
|
|
|
buffer = new_buffer (20, 0, 0, 2 * GST_SECOND, 0);
|
|
gst_buffer_map (buffer, &inmap, GST_MAP_WRITE);
|
|
memset (inmap.data + 15, 1, 5);
|
|
gst_buffer_unmap (buffer, &inmap);
|
|
ret = gst_pad_chain (sinkpad, buffer);
|
|
ck_assert_int_eq (ret, GST_FLOW_OK);
|
|
gst_pad_query (sinkpad, drain);
|
|
fail_unless (handoff_buffer != NULL);
|
|
fail_unless_equals_int (gst_buffer_get_size (handoff_buffer), 40);
|
|
|
|
gst_buffer_map (handoff_buffer, &outmap, GST_MAP_READ);
|
|
for (i = 0; i < 10; i++) {
|
|
guint32 sample;
|
|
|
|
#if G_BYTE_ORDER == G_LITTLE_ENDIAN
|
|
sample = GUINT32_FROM_LE (((guint32 *) outmap.data)[i]);
|
|
#else
|
|
sample = GUINT32_FROM_BE (((guint32 *) outmap.data)[i]);
|
|
#endif
|
|
|
|
if (i < 5) {
|
|
fail_unless_equals_int (sample, 0);
|
|
} else {
|
|
fail_unless_equals_int (sample, 1 << 24);
|
|
}
|
|
}
|
|
gst_buffer_unmap (handoff_buffer, &outmap);
|
|
gst_clear_buffer (&handoff_buffer);
|
|
|
|
gst_element_release_request_pad (audiomixer, sinkpad);
|
|
gst_object_unref (sinkpad);
|
|
gst_element_set_state (bin, GST_STATE_NULL);
|
|
gst_bus_remove_signal_watch (bus);
|
|
gst_object_unref (bus);
|
|
gst_object_unref (bin);
|
|
gst_query_unref (drain);
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
/* In this test, we create two input buffers with a duration of 1 second,
|
|
* and require the audiomixer to output 1.5 second long buffers.
|
|
*
|
|
* After we have input two buffers, we change the output format
|
|
* from S8 to S32, then push a last buffer.
|
|
*
|
|
* This makes audioaggregator convert its "half-mixed" current_buffer,
|
|
* we can then ensure that the second output buffer is as expected.
|
|
*/
|
|
GST_START_TEST (test_change_output_caps_mid_output_buffer)
|
|
{
|
|
GstSegment segment;
|
|
GstElement *bin, *audiomixer, *capsfilter, *sink;
|
|
GstBus *bus;
|
|
GstPad *sinkpad;
|
|
gboolean res;
|
|
GstStateChangeReturn state_res;
|
|
GstFlowReturn ret;
|
|
GstEvent *event;
|
|
GstBuffer *buffer;
|
|
GstCaps *caps;
|
|
GstQuery *drain;
|
|
GstMapInfo inmap;
|
|
GstMapInfo outmap;
|
|
guint i;
|
|
|
|
bin = gst_pipeline_new ("pipeline");
|
|
bus = gst_element_get_bus (bin);
|
|
gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH);
|
|
|
|
g_signal_connect (bus, "message::error", (GCallback) message_received, bin);
|
|
g_signal_connect (bus, "message::warning", (GCallback) message_received, bin);
|
|
g_signal_connect (bus, "message::eos", (GCallback) message_received, bin);
|
|
|
|
audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
|
|
g_object_set (audiomixer, "output-buffer-duration", 1500 * GST_MSECOND, NULL);
|
|
capsfilter = gst_element_factory_make ("capsfilter", NULL);
|
|
sink = gst_element_factory_make ("fakesink", "sink");
|
|
gst_bin_add_many (GST_BIN (bin), audiomixer, capsfilter, sink, NULL);
|
|
|
|
res = gst_element_link_many (audiomixer, capsfilter, sink, NULL);
|
|
fail_unless (res == TRUE, NULL);
|
|
|
|
state_res = gst_element_set_state (bin, GST_STATE_PLAYING);
|
|
ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
|
|
|
|
sinkpad = gst_element_get_request_pad (audiomixer, "sink_%u");
|
|
fail_if (sinkpad == NULL, NULL);
|
|
|
|
gst_pad_send_event (sinkpad, gst_event_new_stream_start ("test"));
|
|
|
|
caps = gst_caps_new_simple ("audio/x-raw",
|
|
"format", G_TYPE_STRING, "S8",
|
|
"layout", G_TYPE_STRING, "interleaved",
|
|
"rate", G_TYPE_INT, 10, "channels", G_TYPE_INT, 1, NULL);
|
|
|
|
gst_pad_set_caps (sinkpad, caps);
|
|
g_object_set (capsfilter, "caps", caps, NULL);
|
|
gst_caps_unref (caps);
|
|
|
|
gst_segment_init (&segment, GST_FORMAT_TIME);
|
|
segment.start = 0;
|
|
segment.stop = 3 * GST_SECOND;
|
|
segment.time = 0;
|
|
event = gst_event_new_segment (&segment);
|
|
gst_pad_send_event (sinkpad, event);
|
|
|
|
buffer = new_buffer (10, 0, 0, 1 * GST_SECOND, 0);
|
|
ret = gst_pad_chain (sinkpad, buffer);
|
|
ck_assert_int_eq (ret, GST_FLOW_OK);
|
|
|
|
buffer = new_buffer (10, 0, 1 * GST_SECOND, 1 * GST_SECOND, 0);
|
|
gst_buffer_map (buffer, &inmap, GST_MAP_WRITE);
|
|
memset (inmap.data, 1, 10);
|
|
gst_buffer_unmap (buffer, &inmap);
|
|
ret = gst_pad_chain (sinkpad, buffer);
|
|
ck_assert_int_eq (ret, GST_FLOW_OK);
|
|
|
|
drain = gst_query_new_drain ();
|
|
gst_pad_query (sinkpad, drain);
|
|
gst_query_unref (drain);
|
|
|
|
caps = gst_caps_new_simple ("audio/x-raw",
|
|
"format", G_TYPE_STRING, GST_AUDIO_NE (S32),
|
|
"layout", G_TYPE_STRING, "interleaved",
|
|
"rate", G_TYPE_INT, 10, "channels", G_TYPE_INT, 1, NULL);
|
|
g_object_set (capsfilter, "caps", caps, NULL);
|
|
gst_caps_unref (caps);
|
|
|
|
gst_buffer_replace (&handoff_buffer, NULL);
|
|
g_object_set (sink, "signal-handoffs", TRUE, NULL);
|
|
g_signal_connect (sink, "handoff", (GCallback) handoff_buffer_cb, NULL);
|
|
|
|
buffer = new_buffer (10, 0, 2 * GST_SECOND, 1 * GST_SECOND, 0);
|
|
gst_buffer_map (buffer, &inmap, GST_MAP_WRITE);
|
|
memset (inmap.data, 0, 10);
|
|
gst_buffer_unmap (buffer, &inmap);
|
|
ret = gst_pad_chain (sinkpad, buffer);
|
|
ck_assert_int_eq (ret, GST_FLOW_OK);
|
|
|
|
drain = gst_query_new_drain ();
|
|
gst_pad_query (sinkpad, drain);
|
|
gst_query_unref (drain);
|
|
|
|
fail_unless (handoff_buffer);
|
|
fail_unless_equals_int (gst_buffer_get_size (handoff_buffer), 60);
|
|
|
|
gst_buffer_map (handoff_buffer, &outmap, GST_MAP_READ);
|
|
for (i = 0; i < 15; i++) {
|
|
guint32 sample;
|
|
|
|
#if G_BYTE_ORDER == G_LITTLE_ENDIAN
|
|
sample = GUINT32_FROM_LE (((guint32 *) outmap.data)[i]);
|
|
#else
|
|
sample = GUINT32_FROM_BE (((guint32 *) outmap.data)[i]);
|
|
#endif
|
|
|
|
if (i < 5) {
|
|
fail_unless_equals_int (sample, 1 << 24);
|
|
} else {
|
|
fail_unless_equals_int (sample, 0);
|
|
}
|
|
}
|
|
|
|
gst_buffer_unmap (handoff_buffer, &outmap);
|
|
gst_clear_buffer (&handoff_buffer);
|
|
|
|
gst_element_release_request_pad (audiomixer, sinkpad);
|
|
gst_object_unref (sinkpad);
|
|
gst_element_set_state (bin, GST_STATE_NULL);
|
|
gst_bus_remove_signal_watch (bus);
|
|
gst_object_unref (bus);
|
|
gst_object_unref (bin);
|
|
}
|
|
|
|
GST_END_TEST;
|
|
static Suite *
|
|
audiomixer_suite (void)
|
|
{
|
|
Suite *s = suite_create ("audiomixer");
|
|
TCase *tc_chain = tcase_create ("general");
|
|
|
|
suite_add_tcase (s, tc_chain);
|
|
tcase_add_test (tc_chain, test_caps);
|
|
tcase_add_test (tc_chain, test_filter_caps);
|
|
tcase_add_test (tc_chain, test_event);
|
|
tcase_add_test (tc_chain, test_play_twice);
|
|
tcase_add_test (tc_chain, test_play_twice_then_add_and_play_again);
|
|
tcase_add_test (tc_chain, test_live_seeking);
|
|
tcase_add_test (tc_chain, test_add_pad);
|
|
tcase_add_test (tc_chain, test_remove_pad);
|
|
tcase_add_test (tc_chain, test_clip);
|
|
tcase_add_test (tc_chain, test_duration_is_max);
|
|
tcase_add_test (tc_chain, test_duration_unknown_overrides);
|
|
tcase_add_test (tc_chain, test_loop);
|
|
tcase_add_test (tc_chain, test_flush_start_flush_stop);
|
|
tcase_add_test (tc_chain, test_sync);
|
|
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);
|
|
tcase_add_checked_fixture (tc_chain, test_setup, test_teardown);
|
|
tcase_add_test (tc_chain, test_change_output_caps);
|
|
tcase_add_test (tc_chain, test_change_output_caps_mid_output_buffer);
|
|
|
|
/* Use a longer timeout */
|
|
#ifdef HAVE_VALGRIND
|
|
if (RUNNING_ON_VALGRIND) {
|
|
tcase_set_timeout (tc_chain, 5 * 60);
|
|
} else
|
|
#endif
|
|
{
|
|
/* this is shorter than the default 60 seconds?! (tpm) */
|
|
/* tcase_set_timeout (tc_chain, 6); */
|
|
}
|
|
|
|
return s;
|
|
}
|
|
|
|
GST_CHECK_MAIN (audiomixer);
|