| /* 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/gst.h> |
| |
| /* |
| * RTP bundle receiver |
| * |
| * In this example we initially create one RTP session but the incoming RTP |
| * and RTCP streams actually bundle 2 different media type, one audio stream |
| * and one video stream. We are notified of the discovery of the streams by |
| * the on-bundled-ssrc rtpbin signal. In the handler we decide to assign the |
| * first SSRC to the (existing) audio session and the second SSRC to a new |
| * session (id: 1). |
| * |
| * .-------. .----------. .-----------. .-------. .-------------. |
| * RTP |udpsrc | | rtpbin | | pcmadepay | |alawdec| |autoaudiosink| |
| * port=5001 | src->recv_rtp_0 recv_rtp_0->sink src->sink src->sink | |
| * '-------' | | '-----------' '-------' '-------------' |
| * | | |
| * | | .-------. |
| * | | |udpsink| RTCP |
| * | send_rtcp_0->sink | port=5003 |
| * .-------. | | '-------' sync=false |
| * RTCP |udpsrc | | | async=false |
| * port=5002 | src->recv_rtcp_0 | |
| * '-------' | | |
| * | | |
| * | | .---------. .-------------. |
| * | | |vrawdepay| |autovideosink| |
| * | recv_rtp_1->sink src->sink | |
| * | | '---------' '-------------' |
| * | | |
| * | | .-------. |
| * | | |udpsink| RTCP |
| * | send_rtcp_1->sink | port=5004 |
| * | | '-------' sync=false |
| * | | async=false |
| * | | |
| * '----------' |
| * |
| */ |
| |
| static gboolean |
| plug_video_rtcp_sender (gpointer user_data) |
| { |
| gint send_video_rtcp_port = 5004; |
| GstElement *rtpbin = GST_ELEMENT_CAST (user_data); |
| GstElement *send_video_rtcp_udpsink; |
| GstElement *pipeline = |
| GST_ELEMENT_CAST (gst_object_get_parent (GST_OBJECT (rtpbin))); |
| |
| send_video_rtcp_udpsink = gst_element_factory_make ("udpsink", NULL); |
| g_object_set (send_video_rtcp_udpsink, "host", "127.0.0.1", NULL); |
| g_object_set (send_video_rtcp_udpsink, "port", send_video_rtcp_port, NULL); |
| g_object_set (send_video_rtcp_udpsink, "sync", FALSE, NULL); |
| g_object_set (send_video_rtcp_udpsink, "async", FALSE, NULL); |
| gst_bin_add (GST_BIN (pipeline), send_video_rtcp_udpsink); |
| gst_element_link_pads (rtpbin, "send_rtcp_src_1", send_video_rtcp_udpsink, |
| "sink"); |
| gst_element_sync_state_with_parent (send_video_rtcp_udpsink); |
| |
| gst_object_unref (pipeline); |
| gst_object_unref (rtpbin); |
| return G_SOURCE_REMOVE; |
| } |
| |
| static void |
| on_rtpbinreceive_pad_added (GstElement * rtpbin, 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); |
| |
| 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); |
| |
| if (g_str_has_prefix (pad_name, "recv_rtp_src_1")) { |
| g_timeout_add (0, plug_video_rtcp_sender, gst_object_ref (rtpbin)); |
| } |
| } |
| 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 (void) |
| { |
| GstElement *pipeline, *rtpbin, *recv_rtp_udpsrc, *recv_rtcp_udpsrc, |
| *audio_rtpdepayloader, *audio_decoder, *audio_sink, *video_rtpdepayloader, |
| *video_sink, *send_audio_rtcp_udpsink; |
| GstCaps *rtpcaps; |
| gint rtp_udp_port = 5001; |
| gint rtcp_udp_port = 5002; |
| gint send_audio_rtcp_port = 5003; |
| |
| pipeline = gst_pipeline_new (NULL); |
| |
| rtpbin = gst_element_factory_make ("rtpbin", NULL); |
| g_object_set (rtpbin, "latency", 200, NULL); |
| |
| 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); |
| |
| 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 ("autoaudiosink", NULL); |
| |
| video_rtpdepayloader = |
| gst_element_factory_make ("rtpvrawdepay", "video_rtpdepayloader"); |
| video_sink = gst_element_factory_make ("autovideosink", 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); |
| |
| gst_element_link_pads (audio_rtpdepayloader, "src", audio_decoder, "sink"); |
| gst_element_link (audio_decoder, audio_sink); |
| |
| gst_element_link_pads (video_rtpdepayloader, "src", video_sink, "sink"); |
| |
| /* request a single receiving RTP session. */ |
| gst_element_link_pads (recv_rtcp_udpsrc, "src", rtpbin, "recv_rtcp_sink_0"); |
| gst_element_link_pads (recv_rtp_udpsrc, "src", rtpbin, "recv_rtp_sink_0"); |
| |
| send_audio_rtcp_udpsink = gst_element_factory_make ("udpsink", NULL); |
| g_object_set (send_audio_rtcp_udpsink, "host", "127.0.0.1", NULL); |
| g_object_set (send_audio_rtcp_udpsink, "port", send_audio_rtcp_port, NULL); |
| g_object_set (send_audio_rtcp_udpsink, "sync", FALSE, NULL); |
| g_object_set (send_audio_rtcp_udpsink, "async", FALSE, NULL); |
| gst_bin_add (GST_BIN (pipeline), send_audio_rtcp_udpsink); |
| gst_element_link_pads (rtpbin, "send_rtcp_src_0", send_audio_rtcp_udpsink, |
| "sink"); |
| |
| return pipeline; |
| } |
| |
| /* |
| * Used to generate informative messages during pipeline startup |
| */ |
| static void |
| cb_state (GstBus * bus, GstMessage * message, gpointer data) |
| { |
| GstObject *pipe = GST_OBJECT (data); |
| GstState old, new, pending; |
| gst_message_parse_state_changed (message, &old, &new, &pending); |
| if (message->src == pipe) { |
| g_print ("Pipeline %s changed state from %s to %s\n", |
| GST_OBJECT_NAME (message->src), |
| gst_element_state_get_name (old), gst_element_state_get_name (new)); |
| if (old == GST_STATE_PAUSED && new == GST_STATE_PLAYING) |
| GST_DEBUG_BIN_TO_DOT_FILE (GST_BIN (pipe), GST_DEBUG_GRAPH_SHOW_ALL, |
| GST_OBJECT_NAME (message->src)); |
| } |
| } |
| |
| int |
| main (int argc, char **argv) |
| { |
| GstElement *pipe; |
| GstBus *bus; |
| GMainLoop *loop; |
| |
| gst_init (&argc, &argv); |
| |
| loop = g_main_loop_new (NULL, FALSE); |
| |
| pipe = create_pipeline (); |
| bus = gst_element_get_bus (pipe); |
| g_signal_connect (bus, "message::state-changed", G_CALLBACK (cb_state), pipe); |
| gst_bus_add_signal_watch (bus); |
| gst_object_unref (bus); |
| |
| g_print ("starting server pipeline\n"); |
| gst_element_set_state (pipe, GST_STATE_PLAYING); |
| |
| g_main_loop_run (loop); |
| |
| g_print ("stopping server pipeline\n"); |
| gst_element_set_state (pipe, GST_STATE_NULL); |
| |
| gst_object_unref (pipe); |
| g_main_loop_unref (loop); |
| |
| return 0; |
| } |
