+/* GStreamer
+ *
+ * Copyright (C) 2016 Igalia S.L.
+ * @author Philippe Normand <philn@igalia.com>
+ *
+ * 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., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/check/gstcheck.h>
+#include <gst/check/gstconsistencychecker.h>
+#include <gst/check/gsttestclock.h>
+#include <gst/rtp/gstrtpbuffer.h>
+
+static GMainLoop *main_loop;
+
+static void
+message_received (GstBus * bus, GstMessage * message, GstPipeline * bin)
+{
+ GST_INFO ("bus message from \"%" GST_PTR_FORMAT "\": %" GST_PTR_FORMAT,
+ GST_MESSAGE_SRC (message), message);
+
+ switch (message->type) {
+ case GST_MESSAGE_EOS:
+ g_main_loop_quit (main_loop);
+ break;
+ case GST_MESSAGE_WARNING:{
+ GError *gerror;
+ gchar *debug;
+
+ gst_message_parse_warning (message, &gerror, &debug);
+ gst_object_default_error (GST_MESSAGE_SRC (message), gerror, debug);
+ g_error_free (gerror);
+ g_free (debug);
+ break;
+ }
+ case GST_MESSAGE_ERROR:{
+ GError *gerror;
+ gchar *debug;
+
+ gst_message_parse_error (message, &gerror, &debug);
+ gst_object_default_error (GST_MESSAGE_SRC (message), gerror, debug);
+ g_error_free (gerror);
+ g_free (debug);
+ fail ("Error!");
+ break;
+ }
+ default:
+ break;
+ }
+}
+
+static void
+on_rtpbinreceive_pad_added (GstElement * element, GstPad * new_pad,
+ gpointer data)
+{
+ GstElement *pipeline = GST_ELEMENT (data);
+ gchar *pad_name = gst_pad_get_name (new_pad);
+
+ if (g_str_has_prefix (pad_name, "recv_rtp_src_")) {
+ GstCaps *caps = gst_pad_get_current_caps (new_pad);
+ GstStructure *s = gst_caps_get_structure (caps, 0);
+ const gchar *media_type = gst_structure_get_string (s, "media");
+ gchar *depayloader_name = g_strdup_printf ("%s_rtpdepayloader", media_type);
+ GstElement *rtpdepayloader =
+ gst_bin_get_by_name (GST_BIN (pipeline), depayloader_name);
+ GstPad *sinkpad;
+
+ g_free (depayloader_name);
+ fail_unless (rtpdepayloader != NULL, NULL);
+
+ sinkpad = gst_element_get_static_pad (rtpdepayloader, "sink");
+ gst_pad_link (new_pad, sinkpad);
+ gst_object_unref (sinkpad);
+ gst_object_unref (rtpdepayloader);
+
+ gst_caps_unref (caps);
+ }
+ g_free (pad_name);
+}
+
+static guint
+on_bundled_ssrc (GstElement * rtpbin, guint ssrc, gpointer user_data)
+{
+ static gboolean create_session = FALSE;
+ guint session_id = 0;
+
+ if (create_session) {
+ session_id = 1;
+ } else {
+ create_session = TRUE;
+ /* use existing session 0, a new session will be created for the next discovered bundled SSRC */
+ }
+ return session_id;
+}
+
+static GstCaps *
+on_request_pt_map (GstElement * rtpbin, guint session_id, guint pt,
+ gpointer user_data)
+{
+ GstCaps *caps = NULL;
+ if (pt == 96) {
+ caps =
+ gst_caps_from_string
+ ("application/x-rtp,media=(string)audio,encoding-name=(string)PCMA,clock-rate=(int)8000");
+ } else if (pt == 100) {
+ caps =
+ gst_caps_from_string
+ ("application/x-rtp,media=(string)video,encoding-name=(string)RAW,clock-rate=(int)90000,sampling=(string)\"YCbCr-4:2:0\",depth=(string)8,width=(string)320,height=(string)240");
+ }
+ return caps;
+}
+
+
+static GstElement *
+create_pipeline (gboolean send)
+{
+ GstElement *pipeline, *rtpbin, *audiosrc, *audio_encoder,
+ *audio_rtppayloader, *sendrtp_udpsink, *recv_rtp_udpsrc,
+ *send_rtcp_udpsink, *recv_rtcp_udpsrc, *sendrtcp_funnel, *sendrtp_funnel;
+ GstElement *audio_rtpdepayloader, *audio_decoder, *audio_sink;
+ GstElement *videosrc, *video_rtppayloader, *video_rtpdepayloader, *video_sink;
+ gboolean res;
+ GstPad *funnel_pad, *rtp_src_pad;
+ GstCaps *rtpcaps;
+ gint rtp_udp_port = 5001;
+ gint rtcp_udp_port = 5002;
+
+ pipeline = gst_pipeline_new (send ? "pipeline_send" : "pipeline_receive");
+
+ rtpbin =
+ gst_element_factory_make ("rtpbin",
+ send ? "rtpbin_send" : "rtpbin_receive");
+ g_object_set (rtpbin, "latency", 200, NULL);
+
+ if (!send) {
+ g_signal_connect (rtpbin, "on-bundled-ssrc",
+ G_CALLBACK (on_bundled_ssrc), NULL);
+ g_signal_connect (rtpbin, "request-pt-map",
+ G_CALLBACK (on_request_pt_map), NULL);
+ }
+
+ g_signal_connect (rtpbin, "pad-added",
+ G_CALLBACK (on_rtpbinreceive_pad_added), pipeline);
+
+ gst_bin_add (GST_BIN (pipeline), rtpbin);
+
+ if (send) {
+ audiosrc = gst_element_factory_make ("audiotestsrc", NULL);
+ audio_encoder = gst_element_factory_make ("alawenc", NULL);
+ audio_rtppayloader = gst_element_factory_make ("rtppcmapay", NULL);
+ g_object_set (audio_rtppayloader, "pt", 96, NULL);
+ g_object_set (audio_rtppayloader, "seqnum-offset", 1, NULL);
+
+ videosrc = gst_element_factory_make ("videotestsrc", NULL);
+ video_rtppayloader = gst_element_factory_make ("rtpvrawpay", NULL);
+ g_object_set (video_rtppayloader, "pt", 100, "seqnum-offset", 1, NULL);
+
+ g_object_set (audiosrc, "num-buffers", 5, NULL);
+ g_object_set (videosrc, "num-buffers", 5, NULL);
+
+ /* muxed rtcp */
+ sendrtcp_funnel = gst_element_factory_make ("funnel", "send_rtcp_funnel");
+ send_rtcp_udpsink = gst_element_factory_make ("udpsink", NULL);
+ g_object_set (send_rtcp_udpsink, "host", "127.0.0.1", NULL);
+ g_object_set (send_rtcp_udpsink, "port", rtcp_udp_port, NULL);
+ g_object_set (send_rtcp_udpsink, "sync", FALSE, NULL);
+ g_object_set (send_rtcp_udpsink, "async", FALSE, NULL);
+
+ /* outgoing bundled stream */
+ sendrtp_funnel = gst_element_factory_make ("funnel", "send_rtp_funnel");
+ sendrtp_udpsink = gst_element_factory_make ("udpsink", NULL);
+ g_object_set (sendrtp_udpsink, "host", "127.0.0.1", NULL);
+ g_object_set (sendrtp_udpsink, "port", rtp_udp_port, NULL);
+
+ gst_bin_add_many (GST_BIN (pipeline), audiosrc, audio_encoder,
+ audio_rtppayloader, sendrtp_udpsink, send_rtcp_udpsink,
+ sendrtp_funnel, sendrtcp_funnel, videosrc, video_rtppayloader, NULL);
+
+ res = gst_element_link (audiosrc, audio_encoder);
+ fail_unless (res == TRUE, NULL);
+ res = gst_element_link (audio_encoder, audio_rtppayloader);
+ fail_unless (res == TRUE, NULL);
+ res =
+ gst_element_link_pads_full (audio_rtppayloader, "src", rtpbin,
+ "send_rtp_sink_0", GST_PAD_LINK_CHECK_NOTHING);
+ fail_unless (res == TRUE, NULL);
+
+ res = gst_element_link (videosrc, video_rtppayloader);
+ fail_unless (res == TRUE, NULL);
+ res =
+ gst_element_link_pads_full (video_rtppayloader, "src", rtpbin,
+ "send_rtp_sink_1", GST_PAD_LINK_CHECK_NOTHING);
+ fail_unless (res == TRUE, NULL);
+
+ res =
+ gst_element_link_pads_full (sendrtp_funnel, "src", sendrtp_udpsink,
+ "sink", GST_PAD_LINK_CHECK_NOTHING);
+ fail_unless (res == TRUE, NULL);
+
+ funnel_pad = gst_element_get_request_pad (sendrtp_funnel, "sink_%u");
+ rtp_src_pad = gst_element_get_static_pad (rtpbin, "send_rtp_src_0");
+ res = gst_pad_link (rtp_src_pad, funnel_pad);
+ gst_object_unref (funnel_pad);
+ gst_object_unref (rtp_src_pad);
+
+ funnel_pad = gst_element_get_request_pad (sendrtp_funnel, "sink_%u");
+ rtp_src_pad = gst_element_get_static_pad (rtpbin, "send_rtp_src_1");
+ res = gst_pad_link (rtp_src_pad, funnel_pad);
+ gst_object_unref (funnel_pad);
+ gst_object_unref (rtp_src_pad);
+
+ res =
+ gst_element_link_pads_full (sendrtcp_funnel, "src", send_rtcp_udpsink,
+ "sink", GST_PAD_LINK_CHECK_NOTHING);
+ fail_unless (res == TRUE, NULL);
+
+ funnel_pad = gst_element_get_request_pad (sendrtcp_funnel, "sink_%u");
+ rtp_src_pad = gst_element_get_request_pad (rtpbin, "send_rtcp_src_0");
+ res =
+ gst_pad_link_full (rtp_src_pad, funnel_pad, GST_PAD_LINK_CHECK_NOTHING);
+ gst_object_unref (funnel_pad);
+ gst_object_unref (rtp_src_pad);
+
+ funnel_pad = gst_element_get_request_pad (sendrtcp_funnel, "sink_%u");
+ rtp_src_pad = gst_element_get_request_pad (rtpbin, "send_rtcp_src_1");
+ res =
+ gst_pad_link_full (rtp_src_pad, funnel_pad, GST_PAD_LINK_CHECK_NOTHING);
+ gst_object_unref (funnel_pad);
+ gst_object_unref (rtp_src_pad);
+
+ } else {
+ recv_rtp_udpsrc = gst_element_factory_make ("udpsrc", NULL);
+ g_object_set (recv_rtp_udpsrc, "port", rtp_udp_port, NULL);
+ rtpcaps = gst_caps_from_string ("application/x-rtp");
+ g_object_set (recv_rtp_udpsrc, "caps", rtpcaps, NULL);
+ gst_caps_unref (rtpcaps);
+
+ recv_rtcp_udpsrc = gst_element_factory_make ("udpsrc", NULL);
+ g_object_set (recv_rtcp_udpsrc, "port", rtcp_udp_port, NULL);
+
+ audio_rtpdepayloader =
+ gst_element_factory_make ("rtppcmadepay", "audio_rtpdepayloader");
+ audio_decoder = gst_element_factory_make ("alawdec", NULL);
+ audio_sink = gst_element_factory_make ("fakesink", NULL);
+ g_object_set (audio_sink, "sync", TRUE, NULL);
+
+ video_rtpdepayloader =
+ gst_element_factory_make ("rtpvrawdepay", "video_rtpdepayloader");
+ video_sink = gst_element_factory_make ("fakesink", NULL);
+ g_object_set (video_sink, "sync", TRUE, NULL);
+
+ gst_bin_add_many (GST_BIN (pipeline), recv_rtp_udpsrc, recv_rtcp_udpsrc,
+ audio_rtpdepayloader, audio_decoder, audio_sink, video_rtpdepayloader,
+ video_sink, NULL);
+
+ res =
+ gst_element_link_pads_full (audio_rtpdepayloader, "src", audio_decoder,
+ "sink", GST_PAD_LINK_CHECK_NOTHING);
+ fail_unless (res == TRUE, NULL);
+ res = gst_element_link (audio_decoder, audio_sink);
+ fail_unless (res == TRUE, NULL);
+
+ res =
+ gst_element_link_pads_full (video_rtpdepayloader, "src", video_sink,
+ "sink", GST_PAD_LINK_CHECK_NOTHING);
+ fail_unless (res == TRUE, NULL);
+
+ /* request a single receiving RTP session. */
+ res =
+ gst_element_link_pads_full (recv_rtcp_udpsrc, "src", rtpbin,
+ "recv_rtcp_sink_0", GST_PAD_LINK_CHECK_NOTHING);
+ fail_unless (res == TRUE, NULL);
+ res =
+ gst_element_link_pads_full (recv_rtp_udpsrc, "src", rtpbin,
+ "recv_rtp_sink_0", GST_PAD_LINK_CHECK_NOTHING);
+ fail_unless (res == TRUE, NULL);
+ }
+
+ return pipeline;
+}
+
+GST_START_TEST (test_simple_rtpbin_bundle)
+{
+ GstElement *send_pipeline, *recv_pipeline;
+ GstBus *send_bus, *recv_bus;
+ GstStateChangeReturn state_res = GST_STATE_CHANGE_FAILURE;
+ GstElement *rtpbin_receive;
+ GObject *rtp_session;
+
+ main_loop = g_main_loop_new (NULL, FALSE);
+
+ send_pipeline = create_pipeline (TRUE);
+ recv_pipeline = create_pipeline (FALSE);
+
+ send_bus = gst_element_get_bus (send_pipeline);
+ gst_bus_add_signal_watch_full (send_bus, G_PRIORITY_HIGH);
+
+ g_signal_connect (send_bus, "message::error", (GCallback) message_received,
+ send_pipeline);
+ g_signal_connect (send_bus, "message::warning", (GCallback) message_received,
+ send_pipeline);
+ g_signal_connect (send_bus, "message::eos", (GCallback) message_received,
+ send_pipeline);
+
+ recv_bus = gst_element_get_bus (recv_pipeline);
+ gst_bus_add_signal_watch_full (recv_bus, G_PRIORITY_HIGH);
+
+ g_signal_connect (recv_bus, "message::error", (GCallback) message_received,
+ recv_pipeline);
+ g_signal_connect (recv_bus, "message::warning", (GCallback) message_received,
+ recv_pipeline);
+ g_signal_connect (recv_bus, "message::eos", (GCallback) message_received,
+ recv_pipeline);
+
+ state_res = gst_element_set_state (recv_pipeline, GST_STATE_PLAYING);
+ ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
+
+ state_res = gst_element_set_state (send_pipeline, GST_STATE_PLAYING);
+ ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
+
+ GST_INFO ("enter mainloop");
+ g_main_loop_run (main_loop);
+ GST_INFO ("exit mainloop");
+
+ rtpbin_receive =
+ gst_bin_get_by_name (GST_BIN (recv_pipeline), "rtpbin_receive");
+ fail_if (rtpbin_receive == NULL, NULL);
+
+ /* Check that 2 RTP sessions where created while only one was explicitely requested. */
+ g_signal_emit_by_name (rtpbin_receive, "get-internal-session", 0,
+ &rtp_session);
+ fail_if (rtp_session == NULL, NULL);
+ g_object_unref (rtp_session);
+ g_signal_emit_by_name (rtpbin_receive, "get-internal-session", 1,
+ &rtp_session);
+ fail_if (rtp_session == NULL, NULL);
+ g_object_unref (rtp_session);
+
+ gst_object_unref (rtpbin_receive);
+
+ state_res = gst_element_set_state (send_pipeline, GST_STATE_NULL);
+ ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
+
+ state_res = gst_element_set_state (recv_pipeline, GST_STATE_NULL);
+ ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
+
+ /* cleanup */
+ g_main_loop_unref (main_loop);
+
+ gst_bus_remove_signal_watch (send_bus);
+ gst_object_unref (send_bus);
+ gst_object_unref (send_pipeline);
+
+ gst_bus_remove_signal_watch (recv_bus);
+ gst_object_unref (recv_bus);
+ gst_object_unref (recv_pipeline);
+
+}
+
+GST_END_TEST;
+
+static Suite *
+rtpbundle_suite (void)
+{
+ Suite *s = suite_create ("rtpbundle");
+ TCase *tc_chain = tcase_create ("general");
+
+ tcase_set_timeout (tc_chain, 10000);
+
+ suite_add_tcase (s, tc_chain);
+
+ tcase_add_test (tc_chain, test_simple_rtpbin_bundle);
+
+ return s;
+}
+
+GST_CHECK_MAIN (rtpbundle);