diff -r 567bb019e3e3 -r 7e817e7e631c gst_plugins_base/tsrc/check/elements/audioresample/src/audioresample.c --- a/gst_plugins_base/tsrc/check/elements/audioresample/src/audioresample.c Tue Aug 31 15:30:33 2010 +0300 +++ b/gst_plugins_base/tsrc/check/elements/audioresample/src/audioresample.c Wed Sep 01 12:16:41 2010 +0100 @@ -1,6 +1,6 @@ /* GStreamer * - * unit test for audioresample, based on the audioresample unit test + * unit test for audioresample * * Copyright (C) <2005> Thomas Vander Stichele * Copyright (C) <2006> Tim-Philipp Müller @@ -20,37 +20,27 @@ * Free Software Foundation, Inc., 59 Temple Place - Suite 330, * Boston, MA 02111-1307, USA. */ + + #include #include - #include -#include + + #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" @@ -58,50 +48,47 @@ static GET_GLOBAL_VAR_FROM_TLS(threads_running,gstcheck,gboolean) #define _gst_check_threads_running (*GET_GSTREAMER_WSD_VAR_NAME(threads_running,gstcheck,g)()) #else -IMPORT_C extern gboolean _gst_check_threads_running; +extern gboolean _gst_check_threads_running; #endif #if EMULATOR static GET_GLOBAL_VAR_FROM_TLS(raised_critical,gstcheck,gboolean) #define _gst_check_raised_critical (*GET_GSTREAMER_WSD_VAR_NAME(raised_critical,gstcheck,g)()) #else -IMPORT_C extern gboolean _gst_check_raised_critical; +extern gboolean _gst_check_raised_critical; #endif //gboolean _gst_check_raised_warning = FALSE; #if EMULATOR static GET_GLOBAL_VAR_FROM_TLS(raised_warning,gstcheck,gboolean) #define _gst_check_raised_warning (*GET_GSTREAMER_WSD_VAR_NAME(raised_warning,gstcheck,g)()) #else -IMPORT_C extern gboolean _gst_check_raised_warning; +extern gboolean _gst_check_raised_warning; #endif //gboolean _gst_check_expecting_log = FALSE; #if EMULATOR static GET_GLOBAL_VAR_FROM_TLS(expecting_log,gstcheck,gboolean) #define _gst_check_expecting_log (*GET_GSTREAMER_WSD_VAR_NAME(expecting_log,gstcheck,g)()) #else -IMPORT_C extern gboolean _gst_check_expecting_log; +extern gboolean _gst_check_expecting_log; #endif #if EMULATOR GET_GLOBAL_VAR_FROM_TLS(buffers,gstcheck,GList*) #define buffers (*GET_GSTREAMER_WSD_VAR_NAME(buffers,gstcheck,g)()) #else -IMPORT_C extern GList *buffers; +extern GList *buffers; #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_INT \ +#define RESAMPLE_CAPS_TEMPLATE_STRING \ "audio/x-raw-int, " \ "channels = (int) [ 1, MAX ], " \ "rate = (int) [ 1, MAX ], " \ @@ -110,10 +97,6 @@ "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, @@ -126,25 +109,20 @@ ); static GstElement * -setup_audioresample (int channels, int inrate, int outrate, int width, - gboolean fp) +setup_audioresample (int channels, int inrate, int outrate) { GstElement *audioresample; GstCaps *caps; GstStructure *structure; + GstPad *pad; GST_DEBUG ("setup_audioresample"); audioresample = gst_check_setup_element ("audioresample"); - if (fp) - caps = gst_caps_from_string (RESAMPLE_CAPS_FLOAT); - else - caps = gst_caps_from_string (RESAMPLE_CAPS_INT); + caps = gst_caps_from_string (RESAMPLE_CAPS_TEMPLATE_STRING); structure = gst_caps_get_structure (caps, 0); gst_structure_set (structure, "channels", G_TYPE_INT, channels, - "rate", G_TYPE_INT, inrate, "width", G_TYPE_INT, width, NULL); - if (!fp) - gst_structure_set (structure, "depth", G_TYPE_INT, width, NULL); + "rate", G_TYPE_INT, inrate, NULL); fail_unless (gst_caps_is_fixed (caps)); fail_unless (gst_element_set_state (audioresample, @@ -152,30 +130,27 @@ "could not set to paused"); mysrcpad = gst_check_setup_src_pad (audioresample, &srctemplate, caps); - gst_pad_set_caps (mysrcpad, caps); + pad = gst_pad_get_peer (mysrcpad); + gst_pad_set_caps (pad, caps); + gst_object_unref (GST_OBJECT (pad)); gst_caps_unref (caps); + gst_pad_set_active (mysrcpad, TRUE); - if (fp) - caps = gst_caps_from_string (RESAMPLE_CAPS_FLOAT); - else - caps = gst_caps_from_string (RESAMPLE_CAPS_INT); + caps = gst_caps_from_string (RESAMPLE_CAPS_TEMPLATE_STRING); structure = gst_caps_get_structure (caps, 0); gst_structure_set (structure, "channels", G_TYPE_INT, channels, - "rate", G_TYPE_INT, outrate, "width", G_TYPE_INT, width, NULL); - if (!fp) - gst_structure_set (structure, "depth", G_TYPE_INT, width, NULL); + "rate", G_TYPE_INT, outrate, 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_caps_unref (caps); gst_pad_set_active (mysinkpad, TRUE); - gst_pad_set_active (mysrcpad, TRUE); - - gst_caps_unref (caps); return audioresample; } @@ -208,11 +183,8 @@ buffer = GST_BUFFER (l->data); ASSERT_BUFFER_REFCOUNT (buffer, "buffer", 1); GST_DEBUG ("buffer timestamp %" G_GUINT64_FORMAT ", duration %" - 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)); + G_GUINT64_FORMAT, GST_BUFFER_TIMESTAMP (buffer), + GST_BUFFER_DURATION (buffer)); fail_unless_equals_uint64 (timestamp, GST_BUFFER_TIMESTAMP (buffer)); fail_unless_equals_uint64 (offset, GST_BUFFER_OFFSET (buffer)); @@ -235,12 +207,12 @@ GstElement *audioresample; GstBuffer *inbuffer, *outbuffer; GstCaps *caps; - guint64 offset = 0; int i, j; gint16 *p; + - audioresample = setup_audioresample (2, inrate, outrate, 16, FALSE); + audioresample = setup_audioresample (2, inrate, outrate); caps = gst_pad_get_negotiated_caps (mysrcpad); fail_unless (gst_caps_is_fixed (caps)); @@ -251,11 +223,10 @@ for (j = 1; j <= numbuffers; ++j) { inbuffer = gst_buffer_new_and_alloc (samples * 4); - GST_BUFFER_DURATION (inbuffer) = GST_FRAMES_TO_CLOCK_TIME (samples, inrate); + GST_BUFFER_DURATION (inbuffer) = samples * GST_SECOND / inrate; GST_BUFFER_TIMESTAMP (inbuffer) = GST_BUFFER_DURATION (inbuffer) * (j - 1); - GST_BUFFER_OFFSET (inbuffer) = offset; - offset += samples; - GST_BUFFER_OFFSET_END (inbuffer) = offset; + GST_BUFFER_OFFSET (inbuffer) = 0; + GST_BUFFER_OFFSET_END (inbuffer) = samples; gst_buffer_set_caps (inbuffer, caps); @@ -293,9 +264,9 @@ */ void test_perfect_stream() { + xmlfile = "test_perfect_stream"; + std_log(LOG_FILENAME_LINE, "Test Started test_perfect_stream"); /* integral scalings */ - xmlfile = "test_perfect_stream"; - std_log(LOG_FILENAME_LINE, "Test Started test_perfect_stream"); test_perfect_stream_instance (48000, 24000, 500, 20); test_perfect_stream_instance (48000, 12000, 500, 20); test_perfect_stream_instance (12000, 24000, 500, 20); @@ -328,10 +299,7 @@ int i, j; gint16 *p; - GST_DEBUG ("inrate:%d outrate:%d samples:%d numbuffers:%d", - inrate, outrate, samples, numbuffers); - - audioresample = setup_audioresample (2, inrate, outrate, 16, FALSE); + audioresample = setup_audioresample (2, inrate, outrate); caps = gst_pad_get_negotiated_caps (mysrcpad); fail_unless (gst_caps_is_fixed (caps)); @@ -361,11 +329,6 @@ ++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); @@ -373,14 +336,9 @@ 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 for buffer #%d", j); + "expected discont buffer"); } } @@ -391,9 +349,10 @@ void test_discont_stream() { + xmlfile = "test_discont_stream"; + std_log(LOG_FILENAME_LINE, "Test Started test_discont_stream"); + /* integral scalings */ - xmlfile = "test_discont_stream"; - std_log(LOG_FILENAME_LINE, "Test Started test_discont_stream"); test_discont_stream_instance (48000, 24000, 500, 20); test_discont_stream_instance (48000, 12000, 500, 20); test_discont_stream_instance (12000, 24000, 500, 20); @@ -405,6 +364,7 @@ /* 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"); @@ -422,8 +382,9 @@ GstBuffer *inbuffer; GstCaps *caps; xmlfile = "test_reuse"; -std_log(LOG_FILENAME_LINE, "Test Started test_reuse"); - audioresample = setup_audioresample (1, 9343, 48000, 16, FALSE); + std_log(LOG_FILENAME_LINE, "Test Started test_reuse"); + + audioresample = setup_audioresample (1, 9343, 48000); caps = gst_pad_get_negotiated_caps (mysrcpad); fail_unless (gst_caps_is_fixed (caps)); @@ -475,7 +436,6 @@ cleanup_audioresample (audioresample); gst_caps_unref (caps); - std_log(LOG_FILENAME_LINE, "Test Successful"); create_xml(0); } @@ -488,10 +448,10 @@ 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"); @@ -526,341 +486,31 @@ } 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); } - - -#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() +/* +audioresample_suite (void) { - 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; -//} +test_perfect_stream(); +test_discont_stream(); +test_reuse(); +test_shutdown(); +}*/ void (*fn[]) (void) = { test_perfect_stream, test_discont_stream, test_reuse, -test_shutdown, -test_live_switch, -test_pipelines, -test_preference_passthrough +test_shutdown }; char *args[] = { "test_perfect_stream", "test_discont_stream", "test_reuse", -"test_shutdown", -"test_live_switch", -"test_pipelines", -"test_preference_passthrough" +"test_shutdown" }; GST_CHECK_MAIN (audioresample);