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159 lines
4.7 KiB
C
159 lines
4.7 KiB
C
/* GStreamer unit tests for flvmux
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*
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* Copyright (C) 2009 Tim-Philipp Müller <tim centricular net>
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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., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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#include <gst/check/gstcheck.h>
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#include <gst/gst.h>
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static GstBusSyncReply
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error_cb (GstBus * bus, GstMessage * msg, gpointer user_data)
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{
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if (GST_MESSAGE_TYPE (msg) == GST_MESSAGE_ERROR) {
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GError *err = NULL;
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gchar *dbg = NULL;
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gst_message_parse_error (msg, &err, &dbg);
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g_error ("ERROR: %s\n%s\n", err->message, dbg);
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}
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return GST_BUS_PASS;
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}
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static void
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handoff_cb (GstElement * element, GstBuffer * buf, GstPad * pad,
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gint * p_counter)
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{
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*p_counter += 1;
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GST_LOG ("counter = %d", *p_counter);
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}
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static void
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mux_pcm_audio (guint num_buffers, guint repeat)
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{
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GstElement *src, *sink, *flvmux, *conv, *pipeline;
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GstPad *sinkpad, *srcpad;
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gint counter;
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GST_LOG ("num_buffers = %u", num_buffers);
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pipeline = gst_pipeline_new ("pipeline");
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fail_unless (pipeline != NULL, "Failed to create pipeline!");
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/* kids, don't use a sync handler for this at home, really; we do because
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* we just want to abort and nothing else */
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gst_bus_set_sync_handler (GST_ELEMENT_BUS (pipeline), error_cb, NULL);
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src = gst_element_factory_make ("audiotestsrc", "audiotestsrc");
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fail_unless (src != NULL, "Failed to create 'audiotestsrc' element!");
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g_object_set (src, "num-buffers", num_buffers, NULL);
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conv = gst_element_factory_make ("audioconvert", "audioconvert");
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fail_unless (conv != NULL, "Failed to create 'audioconvert' element!");
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flvmux = gst_element_factory_make ("flvmux", "flvmux");
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fail_unless (flvmux != NULL, "Failed to create 'flvmux' element!");
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sink = gst_element_factory_make ("fakesink", "fakesink");
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fail_unless (sink != NULL, "Failed to create 'fakesink' element!");
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g_object_set (sink, "signal-handoffs", TRUE, NULL);
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g_signal_connect (sink, "handoff", G_CALLBACK (handoff_cb), &counter);
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gst_bin_add_many (GST_BIN (pipeline), src, conv, flvmux, sink, NULL);
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fail_unless (gst_element_link (src, conv));
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fail_unless (gst_element_link (flvmux, sink));
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/* now link the elements */
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sinkpad = gst_element_get_request_pad (flvmux, "audio");
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fail_unless (sinkpad != NULL, "Could not get audio request pad");
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srcpad = gst_element_get_static_pad (conv, "src");
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fail_unless (srcpad != NULL, "Could not get audioconvert's source pad");
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fail_unless_equals_int (gst_pad_link (srcpad, sinkpad), GST_PAD_LINK_OK);
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gst_object_unref (srcpad);
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gst_object_unref (sinkpad);
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do {
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GstStateChangeReturn state_ret;
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GstMessage *msg;
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GST_LOG ("repeat=%d", repeat);
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counter = 0;
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state_ret = gst_element_set_state (pipeline, GST_STATE_PAUSED);
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fail_unless (state_ret != GST_STATE_CHANGE_FAILURE);
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if (state_ret == GST_STATE_CHANGE_ASYNC) {
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GST_LOG ("waiting for pipeline to reach PAUSED state");
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state_ret = gst_element_get_state (pipeline, NULL, NULL, -1);
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fail_unless_equals_int (state_ret, GST_STATE_CHANGE_SUCCESS);
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}
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GST_LOG ("PAUSED, let's do the rest of it");
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state_ret = gst_element_set_state (pipeline, GST_STATE_PLAYING);
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fail_unless (state_ret != GST_STATE_CHANGE_FAILURE);
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msg = gst_bus_poll (GST_ELEMENT_BUS (pipeline), GST_MESSAGE_EOS, -1);
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fail_unless (msg != NULL, "Expected EOS message on bus!");
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GST_LOG ("EOS");
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gst_message_unref (msg);
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/* should have some output */
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fail_unless (counter > 2);
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fail_unless_equals_int (gst_element_set_state (pipeline, GST_STATE_NULL),
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GST_STATE_CHANGE_SUCCESS);
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/* repeat = test re-usability */
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--repeat;
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} while (repeat > 0);
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gst_object_unref (pipeline);
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}
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GST_START_TEST (test_index_writing)
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{
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/* note: there's a magic 128 value in flvmux when doing index writing */
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if ((__i__ % 33) == 1)
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mux_pcm_audio (__i__, 2);
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}
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GST_END_TEST;
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static Suite *
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flvmux_suite (void)
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{
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Suite *s = suite_create ("flvmux");
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TCase *tc_chain = tcase_create ("general");
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suite_add_tcase (s, tc_chain);
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tcase_add_loop_test (tc_chain, test_index_writing, 1, 499);
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return s;
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}
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GST_CHECK_MAIN (flvmux)
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