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/* GStreamer unit tests for flvmux
*
* Copyright (C) 2009 Tim-Philipp Müller <tim centricular net>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#include <gst/check/gstcheck.h>
#include <gst/gst.h>
static GstBusSyncReply
error_cb (GstBus * bus, GstMessage * msg, gpointer user_data)
{
if (GST_MESSAGE_TYPE (msg) == GST_MESSAGE_ERROR) {
GError *err = NULL;
gchar *dbg = NULL;
gst_message_parse_error (msg, &err, &dbg);
g_error ("ERROR: %s\n%s\n", err->message, dbg);
}
return GST_BUS_PASS;
}
static void
handoff_cb (GstElement * element, GstBuffer * buf, GstPad * pad,
gint * p_counter)
{
*p_counter += 1;
GST_LOG ("counter = %d", *p_counter);
fail_unless (GST_BUFFER_CAPS (buf) != NULL);
}
static void
mux_pcm_audio (guint num_buffers, guint repeat)
{
GstElement *src, *sink, *flvmux, *conv, *pipeline;
gint counter;
GST_LOG ("num_buffers = %u", num_buffers);
pipeline = gst_pipeline_new ("pipeline");
fail_unless (pipeline != NULL, "Failed to create pipeline!");
/* kids, don't use a sync handler for this at home, really; we do because
* we just want to abort and nothing else */
gst_bus_set_sync_handler (GST_ELEMENT_BUS (pipeline), error_cb, NULL);
src = gst_element_factory_make ("audiotestsrc", "audiotestsrc");
fail_unless (src != NULL, "Failed to create 'audiotestsrc' element!");
g_object_set (src, "num-buffers", num_buffers, NULL);
conv = gst_element_factory_make ("audioconvert", "audioconvert");
fail_unless (conv != NULL, "Failed to create 'audioconvert' element!");
flvmux = gst_element_factory_make ("flvmux", "flvmux");
fail_unless (flvmux != NULL, "Failed to create 'flvmux' element!");
sink = gst_element_factory_make ("fakesink", "fakesink");
fail_unless (sink != NULL, "Failed to create 'fakesink' element!");
g_object_set (sink, "signal-handoffs", TRUE, NULL);
g_signal_connect (sink, "handoff", G_CALLBACK (handoff_cb), &counter);
gst_bin_add_many (GST_BIN (pipeline), src, conv, flvmux, sink, NULL);
fail_unless (gst_element_link (src, conv));
fail_unless (gst_element_link (flvmux, sink));
do {
GstStateChangeReturn state_ret;
GstMessage *msg;
GstPad *sinkpad, *srcpad;
GST_LOG ("repeat=%d", repeat);
/* now link the elements */
sinkpad = gst_element_get_request_pad (flvmux, "audio");
fail_unless (sinkpad != NULL, "Could not get audio request pad");
srcpad = gst_element_get_static_pad (conv, "src");
fail_unless (srcpad != NULL, "Could not get audioconvert's source pad");
fail_unless_equals_int (gst_pad_link (srcpad, sinkpad), GST_PAD_LINK_OK);
gst_object_unref (srcpad);
gst_object_unref (sinkpad);
counter = 0;
state_ret = gst_element_set_state (pipeline, GST_STATE_PAUSED);
fail_unless (state_ret != GST_STATE_CHANGE_FAILURE);
if (state_ret == GST_STATE_CHANGE_ASYNC) {
GST_LOG ("waiting for pipeline to reach PAUSED state");
state_ret = gst_element_get_state (pipeline, NULL, NULL, -1);
fail_unless_equals_int (state_ret, GST_STATE_CHANGE_SUCCESS);
}
GST_LOG ("PAUSED, let's do the rest of it");
state_ret = gst_element_set_state (pipeline, GST_STATE_PLAYING);
fail_unless (state_ret != GST_STATE_CHANGE_FAILURE);
msg = gst_bus_poll (GST_ELEMENT_BUS (pipeline), GST_MESSAGE_EOS, -1);
fail_unless (msg != NULL, "Expected EOS message on bus!");
GST_LOG ("EOS");
gst_message_unref (msg);
/* should have some output */
fail_unless (counter > 2);
fail_unless_equals_int (gst_element_set_state (pipeline, GST_STATE_NULL),
GST_STATE_CHANGE_SUCCESS);
/* repeat = test re-usability */
--repeat;
} while (repeat > 0);
gst_object_unref (pipeline);
}
GST_START_TEST (test_index_writing)
{
guint bufs;
/* note: there's a magic 128 value in flvmux when doing index writing */
for (bufs = 1; bufs < 500; bufs += 33) {
mux_pcm_audio (bufs, 2);
}
gst_task_cleanup_all ();
}
GST_END_TEST;
static Suite *
flvmux_suite (void)
{
Suite *s = suite_create ("flvmux");
TCase *tc_chain = tcase_create ("general");
suite_add_tcase (s, tc_chain);
tcase_add_test (tc_chain, test_index_writing);
return s;
}
GST_CHECK_MAIN (flvmux)
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