[farsight2/master] Add tests for the codec profiles

[Olivier Crête]

---
 tests/check/rtp/conference.c |  132 +++++++++++++++++++++++++++++++++++++++++-
 1 files changed, 130 insertions(+), 2 deletions(-)

diff --git a/tests/check/rtp/conference.c b/tests/check/rtp/conference.c
index 84d356b..4f30034 100644
--- a/tests/check/rtp/conference.c
+++ b/tests/check/rtp/conference.c
@@ -973,8 +973,6 @@ GST_START_TEST (test_rtpconference_no_rtcp)
 }
 GST_END_TEST;
 
-
-
 GST_START_TEST (test_rtpconference_three_way_no_source_assoc)
 {
   GParameter param = {0};
@@ -988,6 +986,128 @@ GST_START_TEST (test_rtpconference_three_way_no_source_assoc)
 GST_END_TEST;
 
 
+static void
+_simple_profile_init (void)
+{
+  struct SimpleTestStream *st1 = dats[0]->streams->data;
+  struct SimpleTestStream *st2 = dats[1]->streams->data;
+  GList *prefs = NULL;
+  FsCodec *codec = NULL;
+  gboolean ret;
+
+  codec = fs_codec_new (0, "PCMU", FS_MEDIA_TYPE_AUDIO, 8000);
+  fs_codec_add_optional_parameter (codec, "farsight_send_profile",
+      "audioconvert ! audioresample ! audioconvert ! mulawenc ! rtppcmupay");
+  prefs = g_list_append (NULL, codec);
+
+  ret = fs_session_set_codec_preferences (st1->dat->session, prefs,
+      NULL);
+  ts_fail_unless (ret, "set codec prefs");
+  ret = fs_session_set_codec_preferences (st2->dat->session, prefs,
+      NULL);
+  ts_fail_unless (ret, "set codec prefs");
+
+  fs_codec_list_destroy (prefs);
+
+}
+
+
+GST_START_TEST (test_rtpconference_simple_profile)
+{
+  nway_test (2, _simple_profile_init, 0, NULL);
+}
+GST_END_TEST;
+
+
+static void
+_double_codec_handoff_handler (GstElement *element, GstBuffer *buffer,
+    GstPad *pad, gpointer user_data)
+{
+  static int buffer_count [2][2] = {{0,0},{0,0}};
+  static gpointer sts[2] = {NULL, NULL};
+  GstPad *peer = gst_pad_get_peer (pad);
+  gchar *name;
+  gint session, ssrc, pt;
+  guint id = 0xFFFFFF;
+
+  if (!(sts[0] == user_data || sts[1] == user_data))
+  {
+    if (!sts[0])
+      sts[0] = user_data;
+    else if (!sts[1])
+      sts[1] = user_data;
+    else
+      ts_fail ("Already have two streams");
+  }
+
+  if (sts[0] == user_data)
+    id = 0;
+  else if (sts[1] == user_data)
+    id = 1;
+  else
+    ts_fail ("Should not be here");
+
+  ts_fail_if (peer == NULL);
+  name = gst_pad_get_name (peer);
+  ts_fail_if (name == NULL);
+  gst_object_unref (peer);
+
+  ts_fail_unless (sscanf (name, "src_%d_%d_%d", &session, &ssrc, &pt) == 3);
+  g_free (name);
+
+  if (pt == 0)
+    buffer_count[0][id]++;
+  else if (pt == 8)
+    buffer_count[1][id]++;
+  else
+    ts_fail ("Wrong PT: %d", pt);
+
+  if (buffer_count[0][0] > 20 &&
+      buffer_count[0][1] > 20  &&
+      buffer_count[1][0] > 20 &&
+      buffer_count[1][1] > 20 )
+  {
+    g_main_loop_quit (loop);
+  }
+}
+
+static void
+_double_profile_init (void)
+{
+  struct SimpleTestStream *st1 = dats[0]->streams->data;
+  struct SimpleTestStream *st2 = dats[1]->streams->data;
+  GList *prefs = NULL;
+  FsCodec *codec = NULL;
+  gboolean ret;
+
+  st1->handoff_handler = G_CALLBACK (_double_codec_handoff_handler);
+  st2->handoff_handler = G_CALLBACK (_double_codec_handoff_handler);
+
+  codec = fs_codec_new (0, "PCMU", FS_MEDIA_TYPE_AUDIO, 8000);
+  fs_codec_add_optional_parameter (codec, "farsight_send_profile",
+      "tee name=t "
+      "t. ! audioconvert ! audioresample ! audioconvert ! mulawenc ! rtppcmupay "
+      "t. ! audioconvert ! audioresample ! audioconvert ! alawenc ! rtppcmapay");
+  prefs = g_list_append (NULL, codec);
+
+  ret = fs_session_set_codec_preferences (st1->dat->session, prefs,
+      NULL);
+  ts_fail_unless (ret, "set codec prefs");
+
+  ret = fs_session_set_codec_preferences (st2->dat->session, prefs,
+      NULL);
+  ts_fail_unless (ret, "set codec prefs");
+
+  fs_codec_list_destroy (prefs);
+}
+
+GST_START_TEST (test_rtpconference_double_codec_profile)
+{
+  nway_test (2, _double_profile_init, 0, NULL);
+}
+GST_END_TEST;
+
+
 static Suite *
 fsrtpconference_suite (void)
 {
@@ -1051,6 +1171,14 @@ fsrtpconference_suite (void)
   tcase_add_test (tc_chain, test_rtpconference_three_way_no_source_assoc);
   suite_add_tcase (s, tc_chain);
 
+  tc_chain = tcase_create ("fsrtpconference_simple_profile");
+  tcase_add_test (tc_chain, test_rtpconference_simple_profile);
+  suite_add_tcase (s, tc_chain);
+
+  tc_chain = tcase_create ("fsrtpconference_double_codec_profile");
+  tcase_add_test (tc_chain, test_rtpconference_double_codec_profile);
+  suite_add_tcase (s, tc_chain);
+
   return s;
 }
 
-- 
1.5.6.5