--- a/gst_plugins_base/tsrc/check/elements/audioresample/src/audioresample.c Fri Mar 19 09:35:09 2010 +0200
+++ b/gst_plugins_base/tsrc/check/elements/audioresample/src/audioresample.c Fri Apr 16 15:15:52 2010 +0300
@@ -1,6 +1,6 @@
/* GStreamer
*
- * unit test for audioresample
+ * unit test for audioresample, based on the audioresample unit test
*
* Copyright (C) <2005> Thomas Vander Stichele <thomas at apestaart dot org>
* Copyright (C) <2006> Tim-Philipp Müller <tim at centricular net>
@@ -20,27 +20,37 @@
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
-
-
#include <gst/gst_global.h>
#include <unistd.h>
+
#include <gst/check/gstcheck.h>
-
-
+#include <gst/audio/audio.h>
#define LOG_FILE "c:\\logs\\audioresample_logs.txt"
#include "std_log_result.h"
#define LOG_FILENAME_LINE __FILE__, __LINE__
+//char* xmlfile = "gstsystemclock";
+
void create_xml(int result)
{
+
if(result)
+ {
assert_failed = 1;
-
+ }
+
testResultXml(xmlfile);
close_log_file();
+
+ if(result)
+ {
+ exit (-1);
+ }
+
}
+
#include "libgstreamer_wsd_solution.h"
@@ -79,16 +89,19 @@
#endif
-
-
-
/* For ease of programming we use globals to keep refs for our floating
* src and sink pads we create; otherwise we always have to do get_pad,
* get_peer, and then remove references in every test function */
static GstPad *mysrcpad, *mysinkpad;
+#define RESAMPLE_CAPS_FLOAT \
+ "audio/x-raw-float, " \
+ "channels = (int) [ 1, MAX ], " \
+ "rate = (int) [ 1, MAX ], " \
+ "endianness = (int) BYTE_ORDER, " \
+ "width = (int) { 32, 64 }"
-#define RESAMPLE_CAPS_TEMPLATE_STRING \
+#define RESAMPLE_CAPS_INT \
"audio/x-raw-int, " \
"channels = (int) [ 1, MAX ], " \
"rate = (int) [ 1, MAX ], " \
@@ -97,6 +110,10 @@
"depth = (int) 16, " \
"signed = (bool) TRUE"
+#define RESAMPLE_CAPS_TEMPLATE_STRING \
+ RESAMPLE_CAPS_FLOAT " ; " \
+ RESAMPLE_CAPS_INT
+
static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
@@ -109,20 +126,25 @@
);
static GstElement *
-setup_audioresample (int channels, int inrate, int outrate)
+setup_audioresample (int channels, int inrate, int outrate, int width,
+ gboolean fp)
{
GstElement *audioresample;
GstCaps *caps;
GstStructure *structure;
- GstPad *pad;
GST_DEBUG ("setup_audioresample");
audioresample = gst_check_setup_element ("audioresample");
- caps = gst_caps_from_string (RESAMPLE_CAPS_TEMPLATE_STRING);
+ if (fp)
+ caps = gst_caps_from_string (RESAMPLE_CAPS_FLOAT);
+ else
+ caps = gst_caps_from_string (RESAMPLE_CAPS_INT);
structure = gst_caps_get_structure (caps, 0);
gst_structure_set (structure, "channels", G_TYPE_INT, channels,
- "rate", G_TYPE_INT, inrate, NULL);
+ "rate", G_TYPE_INT, inrate, "width", G_TYPE_INT, width, NULL);
+ if (!fp)
+ gst_structure_set (structure, "depth", G_TYPE_INT, width, NULL);
fail_unless (gst_caps_is_fixed (caps));
fail_unless (gst_element_set_state (audioresample,
@@ -130,27 +152,30 @@
"could not set to paused");
mysrcpad = gst_check_setup_src_pad (audioresample, &srctemplate, caps);
- pad = gst_pad_get_peer (mysrcpad);
- gst_pad_set_caps (pad, caps);
- gst_object_unref (GST_OBJECT (pad));
+ gst_pad_set_caps (mysrcpad, caps);
gst_caps_unref (caps);
- gst_pad_set_active (mysrcpad, TRUE);
- caps = gst_caps_from_string (RESAMPLE_CAPS_TEMPLATE_STRING);
+ if (fp)
+ caps = gst_caps_from_string (RESAMPLE_CAPS_FLOAT);
+ else
+ caps = gst_caps_from_string (RESAMPLE_CAPS_INT);
structure = gst_caps_get_structure (caps, 0);
gst_structure_set (structure, "channels", G_TYPE_INT, channels,
- "rate", G_TYPE_INT, outrate, NULL);
+ "rate", G_TYPE_INT, outrate, "width", G_TYPE_INT, width, NULL);
+ if (!fp)
+ gst_structure_set (structure, "depth", G_TYPE_INT, width, NULL);
fail_unless (gst_caps_is_fixed (caps));
mysinkpad = gst_check_setup_sink_pad (audioresample, &sinktemplate, caps);
/* this installs a getcaps func that will always return the caps we set
* later */
+ gst_pad_set_caps (mysinkpad, caps);
gst_pad_use_fixed_caps (mysinkpad);
- pad = gst_pad_get_peer (mysinkpad);
- gst_pad_set_caps (pad, caps);
- gst_object_unref (GST_OBJECT (pad));
+
+ gst_pad_set_active (mysinkpad, TRUE);
+ gst_pad_set_active (mysrcpad, TRUE);
+
gst_caps_unref (caps);
- gst_pad_set_active (mysinkpad, TRUE);
return audioresample;
}
@@ -183,8 +208,11 @@
buffer = GST_BUFFER (l->data);
ASSERT_BUFFER_REFCOUNT (buffer, "buffer", 1);
GST_DEBUG ("buffer timestamp %" G_GUINT64_FORMAT ", duration %"
- G_GUINT64_FORMAT, GST_BUFFER_TIMESTAMP (buffer),
- GST_BUFFER_DURATION (buffer));
+ G_GUINT64_FORMAT " offset %" G_GUINT64_FORMAT " offset_end %"
+ G_GUINT64_FORMAT,
+ GST_BUFFER_TIMESTAMP (buffer),
+ GST_BUFFER_DURATION (buffer),
+ GST_BUFFER_OFFSET (buffer), GST_BUFFER_OFFSET_END (buffer));
fail_unless_equals_uint64 (timestamp, GST_BUFFER_TIMESTAMP (buffer));
fail_unless_equals_uint64 (offset, GST_BUFFER_OFFSET (buffer));
@@ -207,12 +235,12 @@
GstElement *audioresample;
GstBuffer *inbuffer, *outbuffer;
GstCaps *caps;
+ guint64 offset = 0;
int i, j;
gint16 *p;
-
- audioresample = setup_audioresample (2, inrate, outrate);
+ audioresample = setup_audioresample (2, inrate, outrate, 16, FALSE);
caps = gst_pad_get_negotiated_caps (mysrcpad);
fail_unless (gst_caps_is_fixed (caps));
@@ -223,10 +251,11 @@
for (j = 1; j <= numbuffers; ++j) {
inbuffer = gst_buffer_new_and_alloc (samples * 4);
- GST_BUFFER_DURATION (inbuffer) = samples * GST_SECOND / inrate;
+ GST_BUFFER_DURATION (inbuffer) = GST_FRAMES_TO_CLOCK_TIME (samples, inrate);
GST_BUFFER_TIMESTAMP (inbuffer) = GST_BUFFER_DURATION (inbuffer) * (j - 1);
- GST_BUFFER_OFFSET (inbuffer) = 0;
- GST_BUFFER_OFFSET_END (inbuffer) = samples;
+ GST_BUFFER_OFFSET (inbuffer) = offset;
+ offset += samples;
+ GST_BUFFER_OFFSET_END (inbuffer) = offset;
gst_buffer_set_caps (inbuffer, caps);
@@ -264,9 +293,9 @@
*/
void test_perfect_stream()
{
- xmlfile = "test_perfect_stream";
+ /* integral scalings */
+ xmlfile = "test_perfect_stream";
std_log(LOG_FILENAME_LINE, "Test Started test_perfect_stream");
- /* integral scalings */
test_perfect_stream_instance (48000, 24000, 500, 20);
test_perfect_stream_instance (48000, 12000, 500, 20);
test_perfect_stream_instance (12000, 24000, 500, 20);
@@ -299,7 +328,10 @@
int i, j;
gint16 *p;
- audioresample = setup_audioresample (2, inrate, outrate);
+ GST_DEBUG ("inrate:%d outrate:%d samples:%d numbuffers:%d",
+ inrate, outrate, samples, numbuffers);
+
+ audioresample = setup_audioresample (2, inrate, outrate, 16, FALSE);
caps = gst_pad_get_negotiated_caps (mysrcpad);
fail_unless (gst_caps_is_fixed (caps));
@@ -329,6 +361,11 @@
++p;
}
+ GST_DEBUG ("Sending Buffer time:%" G_GUINT64_FORMAT " duration:%"
+ G_GINT64_FORMAT " discont:%d offset:%" G_GUINT64_FORMAT " offset_end:%"
+ G_GUINT64_FORMAT, GST_BUFFER_TIMESTAMP (inbuffer),
+ GST_BUFFER_DURATION (inbuffer), GST_BUFFER_IS_DISCONT (inbuffer),
+ GST_BUFFER_OFFSET (inbuffer), GST_BUFFER_OFFSET_END (inbuffer));
/* pushing gives away my reference ... */
fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
@@ -336,9 +373,14 @@
outbuffer = g_list_nth_data (buffers, g_list_length (buffers) - 1);
fail_if (outbuffer == NULL);
fail_unless_equals_uint64 (ints, GST_BUFFER_TIMESTAMP (outbuffer));
+ GST_DEBUG ("Got Buffer time:%" G_GUINT64_FORMAT " duration:%"
+ G_GINT64_FORMAT " discont:%d offset:%" G_GUINT64_FORMAT " offset_end:%"
+ G_GUINT64_FORMAT, GST_BUFFER_TIMESTAMP (outbuffer),
+ GST_BUFFER_DURATION (outbuffer), GST_BUFFER_IS_DISCONT (outbuffer),
+ GST_BUFFER_OFFSET (outbuffer), GST_BUFFER_OFFSET_END (outbuffer));
if (j > 1) {
fail_unless (GST_BUFFER_IS_DISCONT (outbuffer),
- "expected discont buffer");
+ "expected discont for buffer #%d", j);
}
}
@@ -349,10 +391,9 @@
void test_discont_stream()
{
- xmlfile = "test_discont_stream";
+ /* integral scalings */
+ xmlfile = "test_discont_stream";
std_log(LOG_FILENAME_LINE, "Test Started test_discont_stream");
-
- /* integral scalings */
test_discont_stream_instance (48000, 24000, 500, 20);
test_discont_stream_instance (48000, 12000, 500, 20);
test_discont_stream_instance (12000, 24000, 500, 20);
@@ -364,7 +405,6 @@
/* wacky scalings */
test_discont_stream_instance (12345, 54321, 500, 20);
-
test_discont_stream_instance (101, 99, 500, 20);
std_log(LOG_FILENAME_LINE, "Test Successful");
@@ -382,9 +422,8 @@
GstBuffer *inbuffer;
GstCaps *caps;
xmlfile = "test_reuse";
- std_log(LOG_FILENAME_LINE, "Test Started test_reuse");
-
- audioresample = setup_audioresample (1, 9343, 48000);
+std_log(LOG_FILENAME_LINE, "Test Started test_reuse");
+ audioresample = setup_audioresample (1, 9343, 48000, 16, FALSE);
caps = gst_pad_get_negotiated_caps (mysrcpad);
fail_unless (gst_caps_is_fixed (caps));
@@ -436,6 +475,7 @@
cleanup_audioresample (audioresample);
gst_caps_unref (caps);
+
std_log(LOG_FILENAME_LINE, "Test Successful");
create_xml(0);
}
@@ -448,10 +488,10 @@
GstCaps *caps;
guint i;
xmlfile = "test_shutdown";
- std_log(LOG_FILENAME_LINE, "Test Started test_shutdown");
-
+std_log(LOG_FILENAME_LINE, "Test Started test_shutdown");
/* create pipeline, force audioresample to actually resample */
pipeline = gst_pipeline_new (NULL);
+
src = gst_check_setup_element ("audiotestsrc");
cf1 = gst_check_setup_element ("capsfilter");
ar = gst_check_setup_element ("audioresample");
@@ -486,31 +526,341 @@
}
gst_object_unref (pipeline);
+
+ std_log(LOG_FILENAME_LINE, "Test Successful");
+ create_xml(0);
+}
+
+
+
+static GstFlowReturn
+live_switch_alloc_only_48000 (GstPad * pad, guint64 offset,
+ guint size, GstCaps * caps, GstBuffer ** buf)
+{
+ GstStructure *structure;
+ gint rate;
+ gint channels;
+ GstCaps *desired;
+
+ structure = gst_caps_get_structure (caps, 0);
+ fail_unless (gst_structure_get_int (structure, "rate", &rate));
+ fail_unless (gst_structure_get_int (structure, "channels", &channels));
+
+ if (rate < 48000)
+ return GST_FLOW_NOT_NEGOTIATED;
+
+ desired = gst_caps_copy (caps);
+ gst_caps_set_simple (desired, "rate", G_TYPE_INT, 48000, NULL);
+
+ *buf = gst_buffer_new_and_alloc (channels * 48000);
+ gst_buffer_set_caps (*buf, desired);
+ gst_caps_unref (desired);
+
+ return GST_FLOW_OK;
+}
+
+static GstCaps *
+live_switch_get_sink_caps (GstPad * pad)
+{
+ GstCaps *result;
+
+ result = gst_caps_copy (GST_PAD_CAPS (pad));
+
+ gst_caps_set_simple (result,
+ "rate", GST_TYPE_INT_RANGE, 48000, G_MAXINT, NULL);
+
+ return result;
+}
+
+static void
+live_switch_push (int rate, GstCaps * caps)
+{
+ GstBuffer *inbuffer;
+ GstCaps *desired;
+ GList *l;
+
+ desired = gst_caps_copy (caps);
+ gst_caps_set_simple (desired, "rate", G_TYPE_INT, rate, NULL);
+ gst_pad_set_caps (mysrcpad, desired);
+
+ fail_unless (gst_pad_alloc_buffer_and_set_caps (mysrcpad,
+ GST_BUFFER_OFFSET_NONE, rate * 4, desired, &inbuffer) == GST_FLOW_OK);
+
+ /* When the basetransform hits the non-configured case it always
+ * returns a buffer with exactly the same caps as we requested so the actual
+ * renegotiation (if needed) will be done in the _chain*/
+ fail_unless (inbuffer != NULL);
+ GST_DEBUG ("desired: %" GST_PTR_FORMAT ".... got: %" GST_PTR_FORMAT,
+ desired, GST_BUFFER_CAPS (inbuffer));
+ fail_unless (gst_caps_is_equal (desired, GST_BUFFER_CAPS (inbuffer)));
+
+ memset (GST_BUFFER_DATA (inbuffer), 0, GST_BUFFER_SIZE (inbuffer));
+ GST_BUFFER_DURATION (inbuffer) = GST_SECOND;
+ GST_BUFFER_TIMESTAMP (inbuffer) = 0;
+ GST_BUFFER_OFFSET (inbuffer) = 0;
+
+ /* pushing gives away my reference ... */
+ fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
+
+ /* ... but it ends up being collected on the global buffer list */
+ fail_unless_equals_int (g_list_length (buffers), 1);
+
+ for (l = buffers; l; l = l->next) {
+ GstBuffer *buffer = GST_BUFFER (l->data);
+
+ gst_buffer_unref (buffer);
+ }
+
+ g_list_free (buffers);
+ buffers = NULL;
+
+ gst_caps_unref (desired);
+}
+
+void test_live_switch()
+{
+ GstElement *audioresample;
+ GstEvent *newseg;
+ GstCaps *caps;
+ xmlfile = "test_live_switch";
+std_log(LOG_FILENAME_LINE, "Test Started test_live_switch");
+ audioresample = setup_audioresample (4, 48000, 48000, 16, FALSE);
+
+ /* Let the sinkpad act like something that can only handle things of
+ * rate 48000- and can only allocate buffers for that rate, but if someone
+ * tries to get a buffer with a rate higher then 48000 tries to renegotiate
+ * */
+ gst_pad_set_bufferalloc_function (mysinkpad, live_switch_alloc_only_48000);
+ gst_pad_set_getcaps_function (mysinkpad, live_switch_get_sink_caps);
+
+ gst_pad_use_fixed_caps (mysrcpad);
+
+ caps = gst_pad_get_negotiated_caps (mysrcpad);
+ fail_unless (gst_caps_is_fixed (caps));
+
+ fail_unless (gst_element_set_state (audioresample,
+ GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
+ "could not set to playing");
+
+ newseg = gst_event_new_new_segment (FALSE, 1.0, GST_FORMAT_TIME, 0, -1, 0);
+ fail_unless (gst_pad_push_event (mysrcpad, newseg) != FALSE);
+
+ /* downstream can provide the requested rate, a buffer alloc will be passed
+ * on */
+ live_switch_push (48000, caps);
+
+ /* Downstream can never accept this rate, buffer alloc isn't passed on */
+ live_switch_push (40000, caps);
+
+ /* Downstream can provide the requested rate but will re-negotiate */
+ live_switch_push (50000, caps);
+
+ cleanup_audioresample (audioresample);
+ gst_caps_unref (caps);
+
std_log(LOG_FILENAME_LINE, "Test Successful");
create_xml(0);
}
-/*
-audioresample_suite (void)
+
+
+#ifndef GST_DISABLE_PARSE
+
+static GMainLoop *loop;
+static gint messages = 0;
+
+static void
+element_message_cb (GstBus * bus, GstMessage * message, gpointer user_data)
+{
+ gchar *s;
+
+ s = gst_structure_to_string (gst_message_get_structure (message));
+ GST_DEBUG ("Received message: %s", s);
+ g_free (s);
+
+ messages++;
+}
+
+static void
+eos_message_cb (GstBus * bus, GstMessage * message, gpointer user_data)
+{
+ GST_DEBUG ("Received eos");
+ g_main_loop_quit (loop);
+}
+
+static void
+test_pipeline (gint width, gboolean fp, gint inrate, gint outrate, gint quality)
+{
+ GstElement *pipeline;
+ GstBus *bus;
+ GError *error = NULL;
+ gchar *pipe_str;
+
+ pipe_str =
+ g_strdup_printf
+ ("audiotestsrc num-buffers=10 ! audioconvert ! audio/x-raw-%s,rate=%d,width=%d,channels=2 ! audioresample quality=%d ! audio/x-raw-%s,rate=%d,width=%d ! identity check-imperfect-timestamp=TRUE ! fakesink",
+ (fp) ? "float" : "int", inrate, width, quality, (fp) ? "float" : "int",
+ outrate, width);
+
+ pipeline = gst_parse_launch (pipe_str, &error);
+ fail_unless (pipeline != NULL, "Error parsing pipeline: %s",
+ error ? error->message : "(invalid error)");
+ g_free (pipe_str);
+
+ bus = gst_element_get_bus (pipeline);
+ fail_if (bus == NULL);
+ gst_bus_add_signal_watch (bus);
+ g_signal_connect (bus, "message::element", (GCallback) element_message_cb,
+ NULL);
+ g_signal_connect (bus, "message::eos", (GCallback) eos_message_cb, NULL);
+
+ gst_element_set_state (pipeline, GST_STATE_PLAYING);
+
+ /* run until we receive EOS */
+ loop = g_main_loop_new (NULL, FALSE);
+
+ g_main_loop_run (loop);
+
+ g_main_loop_unref (loop);
+ loop = NULL;
+
+ gst_element_set_state (pipeline, GST_STATE_NULL);
+
+ fail_if (messages > 0, "Received imperfect timestamp messages");
+ gst_object_unref (pipeline);
+}
+
+void test_pipelines()
{
-test_perfect_stream();
-test_discont_stream();
-test_reuse();
-test_shutdown();
-}*/
+ gint quality;
+ xmlfile = "test_pipelines";
+std_log(LOG_FILENAME_LINE, "Test Started test_pipelines");
+ /* Test qualities 0, 5 and 10 */
+ for (quality = 0; quality < 11; quality += 5) {
+ test_pipeline (8, FALSE, 44100, 48000, quality);
+ test_pipeline (8, FALSE, 48000, 44100, quality);
+
+ test_pipeline (16, FALSE, 44100, 48000, quality);
+ test_pipeline (16, FALSE, 48000, 44100, quality);
+
+ test_pipeline (24, FALSE, 44100, 48000, quality);
+ test_pipeline (24, FALSE, 48000, 44100, quality);
+
+ test_pipeline (32, FALSE, 44100, 48000, quality);
+ test_pipeline (32, FALSE, 48000, 44100, quality);
+
+ test_pipeline (32, TRUE, 44100, 48000, quality);
+ test_pipeline (32, TRUE, 48000, 44100, quality);
+
+ test_pipeline (64, TRUE, 44100, 48000, quality);
+ test_pipeline (64, TRUE, 48000, 44100, quality);
+ }
+
+ std_log(LOG_FILENAME_LINE, "Test Successful");
+ create_xml(0);
+}
+
+
+
+void test_preference_passthrough()
+{
+ GstStateChangeReturn ret;
+ GstElement *pipeline, *src;
+ GstStructure *s;
+ GstMessage *msg;
+ GstCaps *caps;
+ GstPad *pad;
+ GstBus *bus;
+ GError *error = NULL;
+ gint rate = 0;
+
+ xmlfile = "test_preference_passthrough";
+std_log(LOG_FILENAME_LINE, "Test Started test_preference_passthrough");
+ pipeline = gst_parse_launch ("audiotestsrc num-buffers=1 name=src ! "
+ "audioresample ! "
+ "audio/x-raw-int,rate=8000,channels=1,width=16,depth=16,signed=(boolean)true,endianness=(int)BYTE_ORDER ! "
+ "fakesink can-activate-pull=0 ", &error);
+ fail_unless (pipeline != NULL, "Error parsing pipeline: %s",
+ error ? error->message : "(invalid error)");
+
+ ret = gst_element_set_state (pipeline, GST_STATE_PLAYING);
+ fail_unless_equals_int (ret, GST_STATE_CHANGE_ASYNC);
+
+ /* run until we receive EOS */
+ bus = gst_element_get_bus (pipeline);
+ fail_if (bus == NULL);
+ msg = gst_bus_timed_pop_filtered (bus, -1, GST_MESSAGE_EOS);
+ gst_message_unref (msg);
+ gst_object_unref (bus);
+
+ src = gst_bin_get_by_name (GST_BIN (pipeline), "src");
+ fail_unless (src != NULL);
+ pad = gst_element_get_static_pad (src, "src");
+ fail_unless (pad != NULL);
+ caps = gst_pad_get_negotiated_caps (pad);
+ GST_LOG ("negotiated audiotestsrc caps: %" GST_PTR_FORMAT, caps);
+ fail_unless (caps != NULL);
+ s = gst_caps_get_structure (caps, 0);
+ fail_unless (gst_structure_get_int (s, "rate", &rate));
+ /* there's no need to resample, audiotestsrc supports any rate, so make
+ * sure audioresample provided upstream with the right caps to negotiate
+ * this correctly */
+ fail_unless_equals_int (rate, 8000);
+ gst_caps_unref (caps);
+ gst_object_unref (pad);
+ gst_object_unref (src);
+
+ gst_element_set_state (pipeline, GST_STATE_NULL);
+ gst_object_unref (pipeline);
+
+ std_log(LOG_FILENAME_LINE, "Test Successful");
+ create_xml(0);
+}
+
+
+
+#endif
+
+//static Suite *
+//audioresample_suite (void)
+//{
+// Suite *s = suite_create ("audioresample");
+// TCase *tc_chain = tcase_create ("general");
+//
+// suite_add_tcase (s, tc_chain);
+// tcase_add_test (tc_chain, test_perfect_stream);
+// tcase_add_test (tc_chain, test_discont_stream);
+// tcase_add_test (tc_chain, test_reuse);
+// tcase_add_test (tc_chain, test_shutdown);
+// tcase_add_test (tc_chain, test_live_switch);
+//
+//#ifndef GST_DISABLE_PARSE
+// tcase_set_timeout (tc_chain, 360);
+// tcase_add_test (tc_chain, test_pipelines);
+// tcase_add_test (tc_chain, test_preference_passthrough);
+//#endif
+//
+// return s;
+//}
void (*fn[]) (void) = {
test_perfect_stream,
test_discont_stream,
test_reuse,
-test_shutdown
+test_shutdown,
+test_live_switch,
+test_pipelines,
+test_preference_passthrough
};
char *args[] = {
"test_perfect_stream",
"test_discont_stream",
"test_reuse",
-"test_shutdown"
+"test_shutdown",
+"test_live_switch",
+"test_pipelines",
+"test_preference_passthrough"
};
GST_CHECK_MAIN (audioresample);