/* 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 /* * A simple RTP server * sends the output of alsasrc as alaw encoded RTP on port 5002, RTCP is sent on * port 5003. The destination is 127.0.0.1. * the receiver RTCP reports are received on port 5007 * * .-------. .-------. .-------. .----------. .-------. * |alsasrc| |alawenc| |pcmapay| | rtpbin | |udpsink| RTP * | src->sink src->sink src->send_rtp send_rtp->sink | port=5002 * '-------' '-------' '-------' | | '-------' * | | * | | .-------. * | | |udpsink| RTCP * | send_rtcp->sink | port=5003 * .-------. | | '-------' sync=false * RTCP |udpsrc | | | async=false * port=5007 | src->recv_rtcp | * '-------' '----------' */ /* change this to send the RTP data and RTCP to another host */ #define DEST_HOST "127.0.0.1" /* #define AUDIO_SRC "alsasrc" */ #define AUDIO_SRC "audiotestsrc" /* the encoder and payloader elements */ #define AUDIO_ENC "alawenc" #define AUDIO_PAY "rtppcmapay" /* print the stats of a source */ static void print_source_stats (GObject * source) { GstStructure *stats; gchar *str; /* 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); } /* this function is called every second and dumps the RTP manager stats */ static gboolean print_stats (GstElement * rtpbin) { GObject *session; GValueArray *arr; GValue *val; guint i; g_print ("***********************************\n"); /* get session 0 */ g_signal_emit_by_name (rtpbin, "get-internal-session", 0, &session); /* print all the sources in the session, this includes the internal source */ g_object_get (session, "sources", &arr, NULL); for (i = 0; i < arr->n_values; i++) { GObject *source; val = g_value_array_get_nth (arr, i); source = g_value_get_object (val); print_source_stats (source); } g_value_array_free (arr); g_object_unref (session); return TRUE; } /* build a pipeline equivalent to: * * gst-launch -v gstrtpbin name=rtpbin \ * $AUDIO_SRC ! audioconvert ! audioresample ! $AUDIO_ENC ! $AUDIO_PAY ! rtpbin.send_rtp_sink_0 \ * rtpbin.send_rtp_src_0 ! udpsink port=5002 host=$DEST \ * rtpbin.send_rtcp_src_0 ! udpsink port=5003 host=$DEST sync=false async=false \ * udpsrc port=5007 ! rtpbin.recv_rtcp_sink_0 */ int main (int argc, char *argv[]) { GstElement *audiosrc, *audioconv, *audiores, *audioenc, *audiopay; GstElement *rtpbin, *rtpsink, *rtcpsink, *rtcpsrc; GstElement *pipeline; GMainLoop *loop; GstPad *srcpad, *sinkpad; /* always init first */ gst_init (&argc, &argv); /* the pipeline to hold everything */ pipeline = gst_pipeline_new (NULL); g_assert (pipeline); /* the audio capture and format conversion */ audiosrc = gst_element_factory_make (AUDIO_SRC, "audiosrc"); g_assert (audiosrc); audioconv = gst_element_factory_make ("audioconvert", "audioconv"); g_assert (audioconv); audiores = gst_element_factory_make ("audioresample", "audiores"); g_assert (audiores); /* the encoding and payloading */ audioenc = gst_element_factory_make (AUDIO_ENC, "audioenc"); g_assert (audioenc); audiopay = gst_element_factory_make (AUDIO_PAY, "audiopay"); g_assert (audiopay); /* add capture and payloading to the pipeline and link */ gst_bin_add_many (GST_BIN (pipeline), audiosrc, audioconv, audiores, audioenc, audiopay, NULL); if (!gst_element_link_many (audiosrc, audioconv, audiores, audioenc, audiopay, NULL)) { g_error ("Failed to link audiosrc, audioconv, audioresample, " "audio encoder and audio payloader"); } /* the rtpbin element */ rtpbin = gst_element_factory_make ("gstrtpbin", "rtpbin"); g_assert (rtpbin); gst_bin_add (GST_BIN (pipeline), rtpbin); /* the udp sinks and source we will use for RTP and RTCP */ rtpsink = gst_element_factory_make ("udpsink", "rtpsink"); g_assert (rtpsink); g_object_set (rtpsink, "port", 5002, "host", DEST_HOST, NULL); rtcpsink = gst_element_factory_make ("udpsink", "rtcpsink"); g_assert (rtcpsink); g_object_set (rtcpsink, "port", 5003, "host", DEST_HOST, NULL); /* no need for synchronisation or preroll on the RTCP sink */ g_object_set (rtcpsink, "async", FALSE, "sync", FALSE, NULL); rtcpsrc = gst_element_factory_make ("udpsrc", "rtcpsrc"); g_assert (rtcpsrc); g_object_set (rtcpsrc, "port", 5007, NULL); gst_bin_add_many (GST_BIN (pipeline), rtpsink, rtcpsink, rtcpsrc, NULL); /* now link all to the rtpbin, start by getting an RTP sinkpad for session 0 */ sinkpad = gst_element_get_request_pad (rtpbin, "send_rtp_sink_0"); srcpad = gst_element_get_static_pad (audiopay, "src"); if (gst_pad_link (srcpad, sinkpad) != GST_PAD_LINK_OK) g_error ("Failed to link audio payloader to rtpbin"); gst_object_unref (srcpad); /* get the RTP srcpad that was created when we requested the sinkpad above and * link it to the rtpsink sinkpad*/ srcpad = gst_element_get_static_pad (rtpbin, "send_rtp_src_0"); sinkpad = gst_element_get_static_pad (rtpsink, "sink"); if (gst_pad_link (srcpad, sinkpad) != GST_PAD_LINK_OK) g_error ("Failed to link rtpbin to rtpsink"); gst_object_unref (srcpad); gst_object_unref (sinkpad); /* get an RTCP srcpad for sending RTCP to the receiver */ srcpad = gst_element_get_request_pad (rtpbin, "send_rtcp_src_0"); sinkpad = gst_element_get_static_pad (rtcpsink, "sink"); if (gst_pad_link (srcpad, sinkpad) != GST_PAD_LINK_OK) g_error ("Failed to link rtpbin to rtcpsink"); gst_object_unref (sinkpad); /* we also want to receive RTCP, request an RTCP sinkpad for session 0 and * link it to the srcpad of the udpsrc for RTCP */ srcpad = gst_element_get_static_pad (rtcpsrc, "src"); sinkpad = gst_element_get_request_pad (rtpbin, "recv_rtcp_sink_0"); if (gst_pad_link (srcpad, sinkpad) != GST_PAD_LINK_OK) g_error ("Failed to link rtcpsrc to rtpbin"); gst_object_unref (srcpad); /* set the pipeline to playing */ g_print ("starting sender pipeline\n"); gst_element_set_state (pipeline, GST_STATE_PLAYING); /* print stats every second */ g_timeout_add (1000, (GSourceFunc) print_stats, rtpbin); /* 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 sender pipeline\n"); gst_element_set_state (pipeline, GST_STATE_NULL); return 0; }