--- 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 <thomas at apestaart dot org>
* Copyright (C) <2006> Tim-Philipp Müller <tim at centricular net>
@@ -20,37 +20,27 @@
* 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"
@@ -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);