/* GStreamer * Copyright (C) 2009 Wim Taymans * * 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. */ #include #include #include #include #include "shared-video.h" /* * A simple RTP receiver * * receives alaw encoded RTP audio on port 5002, RTCP is received on port 5003. * the receiver RTCP reports are sent to port 5007 * * .-------. .----------. .---------. .-------. .--------. * RTP |udpsrc | | rtpbin | | | | | | | * port=5002 | src->recv_rtp recv_rtp->sink src->sink src->sink | * '-------' | | '---------' '-------' '--------' * | | * | | .-------. * | | |udpsink| RTCP * | send_rtcp->sink | port=5007 * .-------. | | '-------' sync=false * RTCP |udpsrc | | | async=false * port=5003 | src->recv_rtcp | * '-------' '----------' */ /* the caps of the sender RTP stream. This is usually negotiated out of band with * SDP or RTSP. */ /* the destination machine to send RTCP to. This is the address of the sender and * is used to send back the RTCP reports of this receiver. If the data is sent * from another machine, change this address. */ #define DEST_HOST "127.0.0.1" typedef struct _App App; struct _App { GstElement *pipeline; ScenicSharedVideo::Writer *writer; std::string socketName; }; App app; /* print the stats of a source */ static void print_source_stats (GObject * source) { GstStructure *stats; gchar *str; g_return_if_fail (source != NULL); /* get the source stats */ g_object_get (source, "stats", &stats, NULL); /* simply dump the stats structure */ str = gst_structure_to_string (stats); g_print ("source stats: %s\n", str); gst_structure_free (stats); g_free (str); } /* will be called when gstrtpbin signals on-ssrc-active. It means that an RTCP*/ /* packet was received from another source. */ static void on_ssrc_active_cb (GstElement * rtpbin, guint sessid, guint ssrc, gpointer user_data/*GstElement * depay*/) { GObject *session, *isrc, *osrc; g_print ("got RTCP from session %u, SSRC %u\n", sessid, ssrc); /* get the right session */ g_signal_emit_by_name (rtpbin, "get-internal-session", sessid, &session); /* get the internal source (the SSRC allocated to us, the receiver */ g_object_get (session, "internal-source", &isrc, NULL); print_source_stats (isrc); /* get the remote source that sent us RTCP */ g_signal_emit_by_name (session, "get-source-by-ssrc", ssrc, &osrc); print_source_stats (osrc); } static void pad_removed_cb (GstElement * rtpbin, GstPad * new_pad, gpointer user_data/*GstElement * depay*/) { g_print ("pad removed: %s\n", GST_PAD_NAME (new_pad)); } /* will be called when rtpbin has validated a payload that we can depayload */ static void pad_added_cb (GstElement * rtpbin, GstPad * new_pad, gpointer user_data/*GstElement * depay*/) { GstPad *sinkpad; GstPadLinkReturn lres; App *app = (App *)user_data; g_print ("new payload on pad: %s\n", GST_PAD_NAME (new_pad)); /* the depayloading and decoding */ GstElement *gstdepay = gst_element_factory_make ("rtpgstdepay" , NULL); g_assert (gstdepay); gst_element_set_state (gstdepay,GST_STATE_PLAYING); /* add depayloading and playback to the pipeline and link */ gst_bin_add (GST_BIN (app->pipeline), gstdepay); sinkpad = gst_element_get_static_pad (gstdepay, "sink"); g_assert (sinkpad); lres = gst_pad_link (new_pad, sinkpad); g_assert (lres == GST_PAD_LINK_OK); gst_object_unref (sinkpad); app->writer = new ScenicSharedVideo::Writer (app->pipeline,gstdepay,app->socketName); } void leave(int sig) { gst_element_set_state (app.pipeline, GST_STATE_NULL); gst_object_unref (GST_OBJECT (app.pipeline)); exit(sig); } /* build a pipeline equivalent to: * * gst-launch -v gstrtpbin name=rtpbin \ * udpsrc caps=$AUDIO_CAPS port=5002 ! rtpbin.recv_rtp_sink_0 \ * rtpbin. ! rtpgstdepay ! appsink \ * udpsrc port=5003 ! rtpbin.recv_rtcp_sink_0 \ * rtpbin.send_rtcp_src_0 ! udpsink port=5007 host=$DEST sync=false async=false */ int main (int argc, char *argv[]) { GstElement *rtpbin, *rtcpsrc, *rtcpsink; GstElement *rtpsrc; GMainLoop *loop; GstCaps *caps; GstPadLinkReturn lres; GstPad *srcpad, *sinkpad; /* Check input arguments */ if (argc != 2) { g_printerr ("Usage: %s \n", argv[0]); return -1; } app.socketName.append (argv[1]); (void) signal(SIGINT,leave); /* always init first */ gst_init (&argc, &argv); /* the pipeline to hold everything */ app.pipeline = gst_pipeline_new (NULL); g_assert (app.pipeline); /* the udp src and source we will use for RTP and RTCP */ rtpsrc = gst_element_factory_make ("udpsrc", "rtpsrc"); g_assert (rtpsrc); g_object_set (rtpsrc, "port", 5002, NULL); /* we need to set caps on the udpsrc for the RTP data */ // caps inside caps is obtained with "echo application/nicolas | base64", removing the "0=" at the end caps = gst_caps_from_string ("application/x-rtp,media=(string)application,payload=(int)96,clock-rate=(int)90000,encoding-name=(string)X-GST,caps=(string)YXBwbGljYXRpb24vc2NlbmljZGF0YV8K"); g_object_set (rtpsrc, "caps", caps, NULL); gst_caps_unref (caps); rtcpsrc = gst_element_factory_make ("udpsrc", "rtcpsrc"); g_assert (rtcpsrc); g_object_set (rtcpsrc, "port", 5003, NULL); rtcpsink = gst_element_factory_make ("udpsink", "rtcpsink"); g_assert (rtcpsink); g_object_set (rtcpsink, "port", 5007, "host", DEST_HOST, NULL); /* no need for synchronisation or preroll on the RTCP sink */ g_object_set (rtcpsink, "async", FALSE, "sync", FALSE, NULL); gst_bin_add_many (GST_BIN (app.pipeline), rtpsrc, rtcpsrc, rtcpsink, NULL); /* the rtpbin element */ rtpbin = gst_element_factory_make ("gstrtpbin", "rtpbin"); g_assert (rtpbin); gst_bin_add (GST_BIN (app.pipeline), rtpbin); g_print ("rtp sinkpad\n"); /* now link all to the rtpbin, start by getting an RTP sinkpad for session 0 */ srcpad = gst_element_get_static_pad (rtpsrc, "src"); g_assert (srcpad); sinkpad = gst_element_get_request_pad (rtpbin, "recv_rtp_sink_0"); g_assert (sinkpad); lres = gst_pad_link (srcpad, sinkpad); g_print ("%d \n",lres); g_assert (lres == GST_PAD_LINK_OK); gst_object_unref (srcpad); g_print ("rtcp sinkpad\n"); /* get an RTCP sinkpad in session 0 */ srcpad = gst_element_get_static_pad (rtcpsrc, "src"); sinkpad = gst_element_get_request_pad (rtpbin, "recv_rtcp_sink_0"); lres = gst_pad_link (srcpad, sinkpad); g_assert (lres == GST_PAD_LINK_OK); gst_object_unref (srcpad); gst_object_unref (sinkpad); g_print ("rtcp srcpad\n"); /* get an RTCP srcpad for sending RTCP back to the sender */ srcpad = gst_element_get_request_pad (rtpbin, "send_rtcp_src_0"); sinkpad = gst_element_get_static_pad (rtcpsink, "sink"); lres = gst_pad_link (srcpad, sinkpad); g_assert (lres == GST_PAD_LINK_OK); gst_object_unref (sinkpad); /* the RTP pad that we have to connect to the depayloader will be created * dynamically so we connect to the pad-added signal, pass the depayloader as * user_data so that we can link to it. */ g_signal_connect (rtpbin, "pad-added", G_CALLBACK (pad_added_cb), &app); g_signal_connect (rtpbin, "pad-removed", G_CALLBACK (pad_removed_cb), &app); /* give some stats when we receive RTCP */ g_signal_connect (rtpbin, "on-ssrc-active", G_CALLBACK (on_ssrc_active_cb), NULL); /* set the pipeline to playing */ g_print ("starting receiver pipeline\n"); gst_element_set_state (app.pipeline, GST_STATE_PLAYING); /* we need to run a GLib main loop to get the messages */ loop = g_main_loop_new (NULL, FALSE); g_main_loop_run (loop); g_print ("stopping receiver pipeline\n"); gst_element_set_state (app.pipeline, GST_STATE_NULL); gst_object_unref (app.pipeline); return 0; }