webrtcbin: implement support for FEC and RTX

https://bugzilla.gnome.org/show_bug.cgi?id=795044

4 files changed, 295 insertions, 2 deletions

diff --git a/tests/check/elements/webrtcbin.c b/tests/check/elements/webrtcbin.c
index b7f42e0f3..756f38021 100644
--- a/tests/check/elements/webrtcbin.c
+++ b/tests/check/elements/webrtcbin.c
@@ -822,6 +822,67 @@ GST_START_TEST (test_media_direction)
 GST_END_TEST;
 
 static void
+on_sdp_media_payload_types (struct test_webrtc *t, GstElement * element,
+    GstWebRTCSessionDescription * desc, gpointer user_data)
+{
+  const GstSDPMedia *vmedia;
+  guint j;
+
+  fail_unless_equals_int (gst_sdp_message_medias_len (desc->sdp), 2);
+
+  vmedia = gst_sdp_message_get_media (desc->sdp, 1);
+
+  for (j = 0; j < gst_sdp_media_attributes_len (vmedia); j++) {
+    const GstSDPAttribute *attr = gst_sdp_media_get_attribute (vmedia, j);
+
+    if (!g_strcmp0 (attr->key, "rtpmap")) {
+      if (g_str_has_prefix (attr->value, "97")) {
+        fail_unless_equals_string (attr->value, "97 VP8/90000");
+      } else if (g_str_has_prefix (attr->value, "96")) {
+        fail_unless_equals_string (attr->value, "96 red/90000");
+      } else if (g_str_has_prefix (attr->value, "98")) {
+        fail_unless_equals_string (attr->value, "98 ulpfec/90000");
+      } else if (g_str_has_prefix (attr->value, "99")) {
+        fail_unless_equals_string (attr->value, "99 rtx/90000");
+      } else if (g_str_has_prefix (attr->value, "100")) {
+        fail_unless_equals_string (attr->value, "100 rtx/90000");
+      }
+    }
+  }
+}
+
+/* In this test we verify that webrtcbin will pick available payload
+ * types when it needs to, in that example for RTX and FEC */
+GST_START_TEST (test_payload_types)
+{
+  struct test_webrtc *t = create_audio_video_test ();
+  struct validate_sdp offer = { on_sdp_media_payload_types, NULL };
+  GstWebRTCRTPTransceiver *trans;
+  GArray *transceivers;
+
+  t->offer_data = &offer;
+  t->on_offer_created = validate_sdp;
+  t->on_ice_candidate = NULL;
+  /* We don't really care about the answer here */
+  t->on_answer_created = NULL;
+
+  g_signal_emit_by_name (t->webrtc1, "get-transceivers", &transceivers);
+  fail_unless_equals_int (transceivers->len, 2);
+  trans = g_array_index (transceivers, GstWebRTCRTPTransceiver *, 1);
+  g_object_set (trans, "fec-type", GST_WEBRTC_FEC_TYPE_ULP_RED, "do-nack", TRUE,
+      NULL);
+  g_array_unref (transceivers);
+
+  test_webrtc_create_offer (t, t->webrtc1);
+
+  test_webrtc_wait_for_answer_error_eos (t);
+  fail_unless_equals_int (STATE_ANSWER_CREATED, t->state);
+  test_webrtc_free (t);
+}
+
+GST_END_TEST;
+
+static void
 on_sdp_media_setup (struct test_webrtc *t, GstElement * element,
     GstWebRTCSessionDescription * desc, gpointer user_data)
 {
@@ -1367,6 +1428,7 @@ webrtcbin_suite (void)
     tcase_add_test (tc, test_get_transceivers);
     tcase_add_test (tc, test_add_recvonly_transceiver);
     tcase_add_test (tc, test_recvonly_sendonly);
+    tcase_add_test (tc, test_payload_types);
   }
 
   if (nicesrc)
diff --git a/tests/examples/webrtc/Makefile.am b/tests/examples/webrtc/Makefile.am
index 520942d7f..6323263bf 100644
--- a/tests/examples/webrtc/Makefile.am
+++ b/tests/examples/webrtc/Makefile.am
@@ -1,5 +1,5 @@
 
-noinst_PROGRAMS = webrtc webrtcbidirectional webrtcswap
+noinst_PROGRAMS = webrtc webrtcbidirectional webrtcswap webrtctransceiver
 
 webrtc_SOURCES = webrtc.c
 webrtc_CFLAGS=\
@@ -39,3 +39,16 @@ webrtcswap_LDADD=\
 	$(GST_LIBS) \
 	$(GST_SDP_LIBS) \
 	$(top_builddir)/gst-libs/gst/webrtc/libgstwebrtc-@GST_API_VERSION@.la
+
+webrtctransceiver_SOURCES = webrtctransceiver.c
+webrtctransceiver_CFLAGS=\
+	-I$(top_srcdir)/gst-libs \
+	-I$(top_builddir)/gst-libs \
+	$(GST_PLUGINS_BASE_CFLAGS) \
+	$(GST_CFLAGS) \
+	$(GST_SDP_CFLAGS)
+webrtctransceiver_LDADD=\
+	$(GST_PLUGINS_BASE_LIBS) \
+	$(GST_LIBS) \
+	$(GST_SDP_LIBS) \
+	$(top_builddir)/gst-libs/gst/webrtc/libgstwebrtc-@GST_API_VERSION@.la
diff --git a/tests/examples/webrtc/meson.build b/tests/examples/webrtc/meson.build
index 7c2aab72e..90c15e5b7 100644
--- a/tests/examples/webrtc/meson.build
+++ b/tests/examples/webrtc/meson.build
@@ -1,4 +1,4 @@
-examples = ['webrtc', 'webrtcbidirectional', 'webrtcswap']
+examples = ['webrtc', 'webrtcbidirectional', 'webrtcswap', 'webrtctransceiver']
 
 foreach example : examples
   exe_name = example
diff --git a/tests/examples/webrtc/webrtctransceiver.c b/tests/examples/webrtc/webrtctransceiver.c
new file mode 100644
index 000000000..df1a18e3d
--- /dev/null
+++ b/tests/examples/webrtc/webrtctransceiver.c
@@ -0,0 +1,218 @@
+#include <gst/gst.h>
+#include <gst/sdp/sdp.h>
+#include <gst/webrtc/webrtc.h>
+
+#include <string.h>
+
+static GMainLoop *loop;
+static GstElement *pipe1, *webrtc1, *webrtc2;
+static GstBus *bus1;
+
+static gboolean
+_bus_watch (GstBus * bus, GstMessage * msg, GstElement * pipe)
+{
+  switch (GST_MESSAGE_TYPE (msg)) {
+    case GST_MESSAGE_STATE_CHANGED:
+      if (GST_ELEMENT (msg->src) == pipe) {
+        GstState old, new, pending;
+
+        gst_message_parse_state_changed (msg, &old, &new, &pending);
+
+        {
+          gchar *dump_name = g_strconcat ("state_changed-",
+              gst_element_state_get_name (old), "_",
+              gst_element_state_get_name (new), NULL);
+          GST_DEBUG_BIN_TO_DOT_FILE_WITH_TS (GST_BIN (msg->src),
+              GST_DEBUG_GRAPH_SHOW_ALL, dump_name);
+          g_free (dump_name);
+        }
+      }
+      break;
+    case GST_MESSAGE_ERROR:{
+      GError *err = NULL;
+      gchar *dbg_info = NULL;
+
+      GST_DEBUG_BIN_TO_DOT_FILE_WITH_TS (GST_BIN (pipe),
+          GST_DEBUG_GRAPH_SHOW_ALL, "error");
+
+      gst_message_parse_error (msg, &err, &dbg_info);
+      g_printerr ("ERROR from element %s: %s\n",
+          GST_OBJECT_NAME (msg->src), err->message);
+      g_printerr ("Debugging info: %s\n", (dbg_info) ? dbg_info : "none");
+      g_error_free (err);
+      g_free (dbg_info);
+      g_main_loop_quit (loop);
+      break;
+    }
+    case GST_MESSAGE_EOS:{
+      GST_DEBUG_BIN_TO_DOT_FILE_WITH_TS (GST_BIN (pipe),
+          GST_DEBUG_GRAPH_SHOW_ALL, "eos");
+      g_print ("EOS received\n");
+      g_main_loop_quit (loop);
+      break;
+    }
+    default:
+      break;
+  }
+
+  return TRUE;
+}
+
+static void
+_webrtc_pad_added (GstElement * webrtc, GstPad * new_pad, GstElement * pipe)
+{
+  GstElement *out;
+  GstPad *sink;
+
+  if (GST_PAD_DIRECTION (new_pad) != GST_PAD_SRC)
+    return;
+
+  out = gst_parse_bin_from_description ("rtpvp8depay ! vp8dec ! "
+      "videoconvert ! queue ! xvimagesink", TRUE, NULL);
+  gst_bin_add (GST_BIN (pipe), out);
+  gst_element_sync_state_with_parent (out);
+
+  sink = out->sinkpads->data;
+
+  gst_pad_link (new_pad, sink);
+}
+
+static void
+_on_answer_received (GstPromise * promise, gpointer user_data)
+{
+  GstWebRTCSessionDescription *answer = NULL;
+  const GstStructure *reply;
+  gchar *desc;
+
+  g_assert (gst_promise_wait (promise) == GST_PROMISE_RESULT_REPLIED);
+  reply = gst_promise_get_reply (promise);
+  gst_structure_get (reply, "answer",
+      GST_TYPE_WEBRTC_SESSION_DESCRIPTION, &answer, NULL);
+  gst_promise_unref (promise);
+  desc = gst_sdp_message_as_text (answer->sdp);
+  g_print ("Created answer:\n%s\n", desc);
+  g_free (desc);
+
+  g_signal_emit_by_name (webrtc1, "set-remote-description", answer, NULL);
+  g_signal_emit_by_name (webrtc2, "set-local-description", answer, NULL);
+
+  gst_webrtc_session_description_free (answer);
+}
+
+static void
+_on_offer_received (GstPromise * promise, gpointer user_data)
+{
+  GstWebRTCSessionDescription *offer = NULL;
+  const GstStructure *reply;
+  gchar *desc;
+
+  g_assert (gst_promise_wait (promise) == GST_PROMISE_RESULT_REPLIED);
+  reply = gst_promise_get_reply (promise);
+  gst_structure_get (reply, "offer",
+      GST_TYPE_WEBRTC_SESSION_DESCRIPTION, &offer, NULL);
+  gst_promise_unref (promise);
+  desc = gst_sdp_message_as_text (offer->sdp);
+  g_print ("Created offer:\n%s\n", desc);
+  g_free (desc);
+
+  g_signal_emit_by_name (webrtc1, "set-local-description", offer, NULL);
+  g_signal_emit_by_name (webrtc2, "set-remote-description", offer, NULL);
+
+  promise = gst_promise_new_with_change_func (_on_answer_received, user_data,
+      NULL);
+  g_signal_emit_by_name (webrtc2, "create-answer", NULL, promise);
+
+  gst_webrtc_session_description_free (offer);
+}
+
+static void
+_on_negotiation_needed (GstElement * element, gpointer user_data)
+{
+  GstPromise *promise;
+
+  promise = gst_promise_new_with_change_func (_on_offer_received, user_data,
+      NULL);
+  g_signal_emit_by_name (webrtc1, "create-offer", NULL, promise);
+}
+
+static void
+_on_ice_candidate (GstElement * webrtc, guint mlineindex, gchar * candidate,
+    GstElement * other)
+{
+  g_signal_emit_by_name (other, "add-ice-candidate", mlineindex, candidate);
+}
+
+static void
+_on_new_transceiver (GstElement * webrtc, GstWebRTCRTPTransceiver * trans)
+{
+  /* If we expected more than one transceiver, we would take a look at
+   * trans->mline, and compare it with webrtcbin's local description */
+  g_object_set (trans, "fec-type", GST_WEBRTC_FEC_TYPE_ULP_RED, NULL);
+}
+
+static void
+add_fec_to_offer (GstElement * webrtc)
+{
+  GstWebRTCRTPTransceiver *trans;
+  GArray *transceivers;
+
+  /* A transceiver has already been created when a sink pad was
+   * requested on the sending webrtcbin */
+
+  g_signal_emit_by_name (webrtc, "get-transceivers", &transceivers);
+
+  trans = g_array_index (transceivers, GstWebRTCRTPTransceiver *, 0);
+
+  g_object_set (trans, "fec-type", GST_WEBRTC_FEC_TYPE_ULP_RED,
+      "fec-percentage", 100, NULL);
+
+  g_array_unref (transceivers);
+}
+
+int
+main (int argc, char *argv[])
+{
+  gst_init (&argc, &argv);
+
+  loop = g_main_loop_new (NULL, FALSE);
+  pipe1 =
+      gst_parse_launch
+      ("videotestsrc pattern=ball ! video/x-raw ! queue ! vp8enc ! rtpvp8pay ! queue ! "
+      "application/x-rtp,media=video,payload=96,encoding-name=VP8 ! "
+      "webrtcbin name=send webrtcbin name=recv", NULL);
+  bus1 = gst_pipeline_get_bus (GST_PIPELINE (pipe1));
+  gst_bus_add_watch (bus1, (GstBusFunc) _bus_watch, pipe1);
+
+  webrtc1 = gst_bin_get_by_name (GST_BIN (pipe1), "send");
+  g_signal_connect (webrtc1, "on-negotiation-needed",
+      G_CALLBACK (_on_negotiation_needed), NULL);
+  add_fec_to_offer (webrtc1);
+
+  webrtc2 = gst_bin_get_by_name (GST_BIN (pipe1), "recv");
+  g_signal_connect (webrtc2, "pad-added", G_CALLBACK (_webrtc_pad_added),
+      pipe1);
+  g_signal_connect (webrtc1, "on-ice-candidate",
+      G_CALLBACK (_on_ice_candidate), webrtc2);
+  g_signal_connect (webrtc2, "on-ice-candidate",
+      G_CALLBACK (_on_ice_candidate), webrtc1);
+  g_signal_connect (webrtc2, "on-new-transceiver",
+      G_CALLBACK (_on_new_transceiver), NULL);
+
+  g_print ("Starting pipeline\n");
+  gst_element_set_state (GST_ELEMENT (pipe1), GST_STATE_PLAYING);
+
+  g_main_loop_run (loop);
+
+  gst_element_set_state (GST_ELEMENT (pipe1), GST_STATE_NULL);
+  g_print ("Pipeline stopped\n");
+
+  gst_object_unref (webrtc1);
+  gst_object_unref (webrtc2);
+  gst_bus_remove_watch (bus1);
+  gst_object_unref (bus1);
+  gst_object_unref (pipe1);
+
+  gst_deinit ();
+
+  return 0;
+}