tests/check/: Add some basic unit tests for audiorate. Disabled at the moment since it doesn't pass yet (see bug #363...

Original commit message from CVS:
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/audiorate.c: (probe_cb), (got_buf),
(do_perfect_stream_test), (GST_START_TEST), (audiorate_suite):
Add some basic unit tests for audiorate. Disabled at the moment
since it doesn't pass yet (see bug #363119).

ChangeLog

@@ -1,3 +1,12 @@
+2006-10-21  Tim-Philipp Müller  <tim at centricular dot net>
+
+	* tests/check/Makefile.am:
+	* tests/check/elements/.cvsignore:
+	* tests/check/elements/audiorate.c: (probe_cb), (got_buf),
+	(do_perfect_stream_test), (GST_START_TEST), (audiorate_suite):
+	  Add some basic unit tests for audiorate. Disabled at the moment
+	  since it doesn't pass yet (see bug #363119).
+
 2006-10-20  Tim-Philipp Müller  <tim at centricular dot net>
 
 	* gst/subparse/gstsubparse.c: (subrip_fix_up_markup),

common

@@ -1 +1 @@
-Subproject commit efcacf2625da231fbee99b68e0f5db6816cf6fad
+Subproject commit ee0bb43e2b66781d04078e2210404da48f6c68f0

tests/check/Makefile.am

@@ -81,6 +81,7 @@ VALGRIND_TO_FIX = \
 
 # these tests don't even pass
 noinst_PROGRAMS = \
+	elements/audiorate \
 	elements/ffmpegcolorspace
 
 AM_CFLAGS = $(GST_CFLAGS) $(GST_CHECK_CFLAGS)
@@ -111,6 +112,9 @@ elements_audioconvert_LDADD = \
 	$(top_builddir)/gst-libs/gst/audio/libgstaudio-@GST_MAJORMINOR@.la \
 	$(LDADD)
 
+elements_audiorate_LDADD =  $(LDADD)
+elements_audiorate_CFLAGS = $(CFLAGS) $(AM_CFLAGS)
+
 elements_gdpdepay_LDADD = $(GST_GDP_LIBS) $(LDADD)
 elements_gdppay_LDADD = $(GST_GDP_LIBS) $(LDADD)
 

tests/check/elements/.gitignore

@@ -2,6 +2,7 @@
 adder
 alsa
 audioconvert
+audiorate
 audioresample
 audiotestsrc
 gdpdepay

tests/check/elements/audiorate.c

@@ -0,0 +1,219 @@
+/* GStreamer unit tests for audiorate
+ *
+ * Copyright (C) 2006 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.
+ */
+
+#ifdef HAVE_CONFIG_H
+# include <config.h>
+#endif
+
+#include <gst/check/gstcheck.h>
+
+static gboolean
+probe_cb (GstPad * pad, GstBuffer * buf, gdouble * drop_probability)
+{
+  if (g_random_double () < *drop_probability) {
+    GST_LOG ("dropping buffer");
+    return FALSE;               /* drop buffer */
+  }
+
+  return TRUE;                  /* don't drop buffer */
+}
+
+static void
+got_buf (GstElement * fakesink, GstBuffer * buf, GstPad * pad, GList ** p_bufs)
+{
+  *p_bufs = g_list_append (*p_bufs, gst_buffer_ref (buf));
+}
+
+static void
+do_perfect_stream_test (guint rate, guint width, gdouble drop_probability)
+{
+  GstElement *pipe, *src, *conv, *filter, *audiorate, *sink;
+  GstMessage *msg;
+  GstCaps *caps;
+  GstPad *srcpad;
+  GList *l, *bufs = NULL;
+  GstClockTime next_time = GST_CLOCK_TIME_NONE;
+  gint64 next_offset = GST_BUFFER_OFFSET_NONE;
+
+  caps = gst_caps_new_simple ("audio/x-raw-int", "rate", G_TYPE_INT,
+      rate, "width", G_TYPE_INT, width, NULL);
+
+  GST_INFO ("-------- drop=%.0f%% caps = %" GST_PTR_FORMAT " ---------- ",
+      drop_probability * 100.0, caps);
+
+  g_assert (drop_probability >= 0.0 && drop_probability <= 1.0);
+  g_assert (width > 0 && (width % 8) == 0);
+
+  pipe = gst_pipeline_new ("pipeline");
+  fail_unless (pipe != NULL);
+
+  src = gst_element_factory_make ("audiotestsrc", "audiotestsrc");
+  fail_unless (src != NULL);
+
+  g_object_set (src, "num-buffers", 500, NULL);
+
+  conv = gst_element_factory_make ("audioconvert", "audioconvert");
+  fail_unless (conv != NULL);
+
+  filter = gst_element_factory_make ("capsfilter", "capsfilter");
+  fail_unless (filter != NULL);
+
+  g_object_set (filter, "caps", caps, NULL);
+
+  srcpad = gst_element_get_pad (filter, "src");
+  fail_unless (srcpad != NULL);
+  gst_pad_add_buffer_probe (srcpad, G_CALLBACK (probe_cb), &drop_probability);
+  gst_object_unref (srcpad);
+
+  audiorate = gst_element_factory_make ("audiorate", "audiorate");
+  fail_unless (audiorate != NULL);
+
+  sink = gst_element_factory_make ("fakesink", "fakesink");
+  fail_unless (sink != NULL);
+
+  g_object_set (sink, "signal-handoffs", TRUE, NULL);
+
+  g_signal_connect (sink, "handoff", G_CALLBACK (got_buf), &bufs);
+
+  gst_bin_add_many (GST_BIN (pipe), src, conv, filter, audiorate, sink, NULL);
+  gst_element_link_many (src, conv, filter, audiorate, sink, NULL);
+
+  fail_unless_equals_int (gst_element_set_state (pipe, GST_STATE_PLAYING),
+      GST_STATE_CHANGE_ASYNC);
+
+  fail_unless_equals_int (gst_element_get_state (pipe, NULL, NULL, -1),
+      GST_STATE_CHANGE_SUCCESS);
+
+  msg = gst_bus_poll (GST_ELEMENT_BUS (pipe),
+      GST_MESSAGE_EOS | GST_MESSAGE_ERROR, -1);
+  fail_unless_equals_string (GST_MESSAGE_TYPE_NAME (msg), "eos");
+
+  for (l = bufs; l != NULL; l = l->next) {
+    GstBuffer *buf = GST_BUFFER (l->data);
+    guint num_samples;
+
+    fail_unless (GST_BUFFER_TIMESTAMP_IS_VALID (buf));
+    fail_unless (GST_BUFFER_DURATION_IS_VALID (buf));
+    fail_unless (GST_BUFFER_OFFSET_IS_VALID (buf));
+    fail_unless (GST_BUFFER_OFFSET_END_IS_VALID (buf));
+
+    GST_LOG ("buffer: ts=%" GST_TIME_FORMAT ", end_ts=%" GST_TIME_FORMAT
+        " off=%" G_GINT64_FORMAT ", end_off=%" G_GINT64_FORMAT,
+        GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
+        GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf) + GST_BUFFER_DURATION (buf)),
+        GST_BUFFER_OFFSET (buf), GST_BUFFER_OFFSET_END (buf));
+
+    if (GST_CLOCK_TIME_IS_VALID (next_time)) {
+      fail_unless_equals_uint64 (next_time, GST_BUFFER_TIMESTAMP (buf));
+    }
+    if (next_offset != GST_BUFFER_OFFSET_NONE) {
+      fail_unless_equals_uint64 (next_offset, GST_BUFFER_OFFSET (buf));
+    }
+
+    /* check buffer size for sanity */
+    fail_unless_equals_int (GST_BUFFER_SIZE (buf) % (width / 8), 0);
+
+    /* check there is actually as much data as there should be */
+    num_samples = GST_BUFFER_OFFSET_END (buf) - GST_BUFFER_OFFSET (buf);
+    fail_unless_equals_int (GST_BUFFER_SIZE (buf), num_samples * (width / 8));
+
+    next_time = GST_BUFFER_TIMESTAMP (buf) + GST_BUFFER_DURATION (buf);
+    next_offset = GST_BUFFER_OFFSET_END (buf);
+  }
+
+  gst_message_unref (msg);
+  gst_element_set_state (pipe, GST_STATE_NULL);
+  gst_object_unref (pipe);
+
+  g_list_foreach (bufs, (GFunc) gst_mini_object_unref, NULL);
+  g_list_free (bufs);
+
+  gst_caps_unref (caps);
+}
+
+static const guint rates[] = { 8000, 11025, 16000, 22050, 32000, 44100,
+  48000, 3333, 33333, 66666, 9999
+};
+
+GST_START_TEST (test_perfect_stream_drop0)
+{
+  guint i;
+
+  for (i = 0; i < G_N_ELEMENTS (rates); ++i) {
+    do_perfect_stream_test (rates[i], 8, 0.0);
+    do_perfect_stream_test (rates[i], 16, 0.0);
+  }
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_perfect_stream_drop10)
+{
+  guint i;
+
+  for (i = 0; i < G_N_ELEMENTS (rates); ++i) {
+    do_perfect_stream_test (rates[i], 8, 0.10);
+    do_perfect_stream_test (rates[i], 16, 0.10);
+  }
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_perfect_stream_drop50)
+{
+  guint i;
+
+  for (i = 0; i < G_N_ELEMENTS (rates); ++i) {
+    do_perfect_stream_test (rates[i], 8, 0.50);
+    do_perfect_stream_test (rates[i], 16, 0.50);
+  }
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_perfect_stream_drop90)
+{
+  guint i;
+
+  for (i = 0; i < G_N_ELEMENTS (rates); ++i) {
+    do_perfect_stream_test (rates[i], 8, 0.90);
+    do_perfect_stream_test (rates[i], 16, 0.90);
+  }
+}
+
+GST_END_TEST;
+
+static Suite *
+audiorate_suite (void)
+{
+  Suite *s = suite_create ("audiorate");
+  TCase *tc_chain = tcase_create ("general");
+
+  suite_add_tcase (s, tc_chain);
+
+  tcase_add_test (tc_chain, test_perfect_stream_drop0);
+  tcase_add_test (tc_chain, test_perfect_stream_drop10);
+  tcase_add_test (tc_chain, test_perfect_stream_drop50);
+  tcase_add_test (tc_chain, test_perfect_stream_drop90);
+
+  return s;
+}
+
+GST_CHECK_MAIN (audiorate);