fssession: Remove the "method" from the telephone_event methods

Now, it will always try event then fall back on sound. If one wants to
ensure that only sound is sent, one must remove the events from the remote
codecs.

3 files changed, 25 insertions, 41 deletions

diff --git a/tests/check/raw/conference.c b/tests/check/raw/conference.c
index 81529ebe..a0a490c8 100644
--- a/tests/check/raw/conference.c
+++ b/tests/check/raw/conference.c
@@ -1367,9 +1367,8 @@ GST_START_TEST (test_rawconference_dispose)
 
   fs_session_destroy (session);
 
-  fail_if (fs_session_start_telephony_event (session, 1, 2,
-          FS_DTMF_METHOD_AUTO));
-  fail_if (fs_session_stop_telephony_event (session, FS_DTMF_METHOD_AUTO));
+  fail_if (fs_session_start_telephony_event (session, 1, 2));
+  fail_if (fs_session_stop_telephony_event (session));
 
   fail_if (fs_session_set_send_codec (session, NULL, &error));
   fail_unless (error->domain == FS_ERROR &&
diff --git a/tests/check/rtp/conference.c b/tests/check/rtp/conference.c
index 290dc8b0..f2bc9676 100644
--- a/tests/check/rtp/conference.c
+++ b/tests/check/rtp/conference.c
@@ -1281,9 +1281,8 @@ GST_START_TEST (test_rtpconference_dispose)
 
   g_object_run_dispose (G_OBJECT (session));
 
-  fail_if (fs_session_start_telephony_event (session, 1, 2,
-          FS_DTMF_METHOD_AUTO));
-  fail_if (fs_session_stop_telephony_event (session, FS_DTMF_METHOD_AUTO));
+  fail_if (fs_session_start_telephony_event (session, 1, 2));
+  fail_if (fs_session_stop_telephony_event (session));
 
   fail_if (fs_session_set_send_codec (session, NULL, &error));
   fail_unless (error->domain == FS_ERROR && error->code == FS_ERROR_DISPOSED);
diff --git a/tests/check/rtp/sendcodecs.c b/tests/check/rtp/sendcodecs.c
index 71d65d67..6471cc11 100644
--- a/tests/check/rtp/sendcodecs.c
+++ b/tests/check/rtp/sendcodecs.c
@@ -32,13 +32,13 @@
 
 GMainLoop *loop = NULL;
 
-FsDTMFMethod method = FS_DTMF_METHOD_AUTO;
 guint dtmf_id = 0;
 gint digit = 0;
 gboolean sending = FALSE;
 gboolean received = FALSE;
 gboolean ready_to_send = FALSE;
 gboolean change_codec = FALSE;
+gboolean filter_telephone_event = FALSE;
 
 struct SimpleTestConference *dat = NULL;
 FsStream *stream = NULL;
@@ -118,21 +118,24 @@ _bus_callback (GstBus *bus, GstMessage *message, gpointer user_data)
                   NULL));
 
           ts_fail_unless (codec != NULL);
-          ts_fail_unless (secondary_codec_list != NULL);
-
-          for (item = secondary_codec_list; item; item = item->next)
+          if (!filter_telephone_event)
           {
-            FsCodec *codec = item->data;
+            ts_fail_unless (secondary_codec_list != NULL);
 
-            if (codec->clock_rate == 8000 &&
-                !g_strcasecmp ("telephone-event", codec->encoding_name))
+            for (item = secondary_codec_list; item; item = item->next)
             {
-              ts_fail_unless (codec->id == dtmf_id);
-              ready_to_send = TRUE;
+              FsCodec *codec = item->data;
+
+              if (codec->clock_rate == 8000 &&
+                  !g_strcasecmp ("telephone-event", codec->encoding_name))
+              {
+                ts_fail_unless (codec->id == dtmf_id);
+                ready_to_send = TRUE;
+              }
             }
-          }
 
-          fail_unless (ready_to_send == TRUE);
+            fail_unless (ready_to_send == TRUE);
+          }
 
           fs_codec_list_destroy (secondary_codec_list);
           fs_codec_destroy (codec);
@@ -235,7 +238,8 @@ set_codecs (struct SimpleTestConference *dat, FsStream *stream)
       ts_fail_unless (dtmf_codec == NULL,
           "More than one copy of telephone-event");
       dtmf_codec = codec;
-      filtered_codecs = g_list_append (filtered_codecs, codec);
+      if (!filter_telephone_event)
+        filtered_codecs = g_list_append (filtered_codecs, codec);
     }
   }
 
@@ -272,7 +276,7 @@ one_way (GstElement *recv_pipeline, gint port)
   digit = 0;
   sending = FALSE;
   received = FALSE;
-  ready_to_send = FALSE;
+  ready_to_send = filter_telephone_event;
 
   loop = g_main_loop_new (NULL, FALSE);
 
@@ -376,7 +380,7 @@ start_stop_sending_dtmf (gpointer data)
 
   if (sending)
   {
-    ts_fail_unless (fs_session_stop_telephony_event (dat->session, method),
+    ts_fail_unless (fs_session_stop_telephony_event (dat->session),
         "Could not stop telephony event");
     sending = FALSE;
   }
@@ -407,7 +411,7 @@ start_stop_sending_dtmf (gpointer data)
 
     received = FALSE;
     ts_fail_unless (fs_session_start_telephony_event (dat->session,
-            digit, digit, method),
+            digit, digit),
         "Could not start telephony event");
     sending = TRUE;
   }
@@ -421,20 +425,6 @@ GST_START_TEST (test_senddtmf_event)
   GstElement *recv_pipeline = build_recv_pipeline (
       G_CALLBACK (send_dmtf_havedata_handler), NULL, &port);
 
-  method = FS_DTMF_METHOD_RTP_RFC4733;
-  g_timeout_add (350, start_stop_sending_dtmf, NULL);
-  one_way (recv_pipeline, port);
-}
-GST_END_TEST;
-
-
-GST_START_TEST (test_senddtmf_auto)
-{
-  gint port;
-  GstElement *recv_pipeline = build_recv_pipeline (
-      G_CALLBACK (send_dmtf_havedata_handler), NULL, &port);
-
-  method = FS_DTMF_METHOD_AUTO;
   g_timeout_add (350, start_stop_sending_dtmf, NULL);
   one_way (recv_pipeline, port);
 }
@@ -499,9 +489,10 @@ GST_START_TEST (test_senddtmf_sound)
   gint port = 0;
   GstElement *recv_pipeline = build_dtmf_sound_recv_pipeline (&port);
 
-  method = FS_DTMF_METHOD_SOUND;
   g_timeout_add (350, start_stop_sending_dtmf, NULL);
+  filter_telephone_event = TRUE;
   one_way (recv_pipeline, port);
+  filter_telephone_event = FALSE;
 }
 GST_END_TEST;
 
@@ -512,7 +503,6 @@ GST_START_TEST (test_senddtmf_change_auto)
   GstElement *recv_pipeline = build_recv_pipeline (
       G_CALLBACK (send_dmtf_havedata_handler), NULL, &port);
 
-  method = FS_DTMF_METHOD_AUTO;
   change_codec = TRUE;
   g_timeout_add (350, start_stop_sending_dtmf, NULL);
   one_way (recv_pipeline, port);
@@ -586,10 +576,6 @@ fsrtpsendcodecs_suite (void)
   tcase_add_test (tc_chain, test_senddtmf_event);
   suite_add_tcase (s, tc_chain);
 
-  tc_chain = tcase_create ("fsrtpsenddtmf_auto");
-  tcase_add_test (tc_chain, test_senddtmf_auto);
-  suite_add_tcase (s, tc_chain);
-
   tc_chain = tcase_create ("fsrtpsenddtmf_sound");
   tcase_add_test (tc_chain, test_senddtmf_sound);
   suite_add_tcase (s, tc_chain);