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297 lines
10 KiB
C
297 lines
10 KiB
C
/* GStreamer
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* Copyright (C) 2013 Collabora Ltd.
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* @author Torrie Fischer <torrie.fischer@collabora.co.uk>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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#include <gst/gst.h>
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#include <gst/rtp/rtp.h>
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/*
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* An RTP server
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* creates two sessions and streams audio on one, video on the other, with RTCP
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* on both sessions. The destination is 127.0.0.1.
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*
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* In both sessions, we set "rtprtxsend" as the session's "aux" element
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* in rtpbin, which enables RFC4588 retransmission for that session.
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*
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* .-------. .-------. .-------. .------------. .-------.
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* |audiots| |alawenc| |pcmapay| | rtpbin | |udpsink|
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* | src->sink src->sink src->send_rtp_0 send_rtp_0->sink |
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* '-------' '-------' '-------' | | '-------'
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* | |
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* .-------. .---------. .---------. | | .-------.
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* |audiots| |theoraenc| |theorapay| | | |udpsink|
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* | src->sink src->sink src->send_rtp_1 send_rtp_1->sink |
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* '-------' '---------' '---------' | | '-------'
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* | |
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* .------. | |
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* |udpsrc| | | .-------.
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* | src->recv_rtcp_0 | |udpsink|
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* '------' | send_rtcp_0->sink |
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* | | '-------'
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* .------. | |
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* |udpsrc| | | .-------.
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* | src->recv_rtcp_1 | |udpsink|
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* '------' | send_rtcp_1->sink |
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* '------------' '-------'
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*
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* To keep the set of ports consistent across both this server and the
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* corresponding client, a SessionData struct maps a rtpbin session number to
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* a GstBin and is used to create the corresponding udp sinks with correct
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* ports.
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*/
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typedef struct _SessionData
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{
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int ref;
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guint sessionNum;
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GstElement *input;
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} SessionData;
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static SessionData *
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session_ref (SessionData * data)
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{
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g_atomic_int_inc (&data->ref);
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return data;
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}
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static void
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session_unref (gpointer data)
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{
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SessionData *session = (SessionData *) data;
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if (g_atomic_int_dec_and_test (&session->ref)) {
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g_free (session);
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}
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}
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static SessionData *
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session_new (guint sessionNum)
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{
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SessionData *ret = g_new0 (SessionData, 1);
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ret->sessionNum = sessionNum;
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return session_ref (ret);
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}
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/*
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* Used to generate informative messages during pipeline startup
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*/
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static void
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cb_state (GstBus * bus, GstMessage * message, gpointer data)
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{
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GstObject *pipe = GST_OBJECT (data);
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GstState old, new, pending;
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gst_message_parse_state_changed (message, &old, &new, &pending);
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if (message->src == pipe) {
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g_print ("Pipeline %s changed state from %s to %s\n",
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GST_OBJECT_NAME (message->src),
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gst_element_state_get_name (old), gst_element_state_get_name (new));
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}
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}
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/*
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* Creates a GstGhostPad named "src" on the given bin, pointed at the "src" pad
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* of the given element
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*/
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static void
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setup_ghost (GstElement * src, GstBin * bin)
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{
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GstPad *srcPad = gst_element_get_static_pad (src, "src");
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GstPad *binPad = gst_ghost_pad_new ("src", srcPad);
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gst_element_add_pad (GST_ELEMENT (bin), binPad);
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}
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static SessionData *
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make_audio_session (guint sessionNum)
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{
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SessionData *session;
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GstBin *audioBin = GST_BIN (gst_bin_new (NULL));
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GstElement *audioSrc = gst_element_factory_make ("audiotestsrc", NULL);
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GstElement *encoder = gst_element_factory_make ("alawenc", NULL);
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GstElement *payloader = gst_element_factory_make ("rtppcmapay", NULL);
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g_object_set (audioSrc, "is-live", TRUE, NULL);
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gst_bin_add_many (audioBin, audioSrc, encoder, payloader, NULL);
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gst_element_link_many (audioSrc, encoder, payloader, NULL);
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setup_ghost (payloader, audioBin);
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session = session_new (sessionNum);
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session->input = GST_ELEMENT (audioBin);
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return session;
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}
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static SessionData *
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make_video_session (guint sessionNum)
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{
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GstBin *videoBin = GST_BIN (gst_bin_new (NULL));
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GstElement *videoSrc = gst_element_factory_make ("videotestsrc", NULL);
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GstElement *encoder = gst_element_factory_make ("theoraenc", NULL);
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GstElement *payloader = gst_element_factory_make ("rtptheorapay", NULL);
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GstCaps *videoCaps;
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SessionData *session;
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g_object_set (videoSrc, "is-live", TRUE, "horizontal-speed", 1, NULL);
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g_object_set (payloader, "config-interval", 2, NULL);
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gst_bin_add_many (videoBin, videoSrc, encoder, payloader, NULL);
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videoCaps = gst_caps_new_simple ("video/x-raw",
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"width", G_TYPE_INT, 352,
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"height", G_TYPE_INT, 288, "framerate", GST_TYPE_FRACTION, 15, 1, NULL);
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gst_element_link_filtered (videoSrc, encoder, videoCaps);
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gst_element_link (encoder, payloader);
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setup_ghost (payloader, videoBin);
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session = session_new (sessionNum);
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session->input = GST_ELEMENT (videoBin);
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return session;
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}
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static GstElement *
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request_aux_sender (GstElement * rtpbin, guint sessid, SessionData * session)
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{
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GstElement *rtx, *bin;
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GstPad *pad;
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gchar *name;
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GstStructure *pt_map;
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GST_INFO ("creating AUX sender");
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bin = gst_bin_new (NULL);
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rtx = gst_element_factory_make ("rtprtxsend", NULL);
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pt_map = gst_structure_new ("application/x-rtp-pt-map",
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"8", G_TYPE_UINT, 98, "96", G_TYPE_UINT, 99, NULL);
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g_object_set (rtx, "payload-type-map", pt_map, NULL);
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gst_structure_free (pt_map);
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gst_bin_add (GST_BIN (bin), rtx);
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pad = gst_element_get_static_pad (rtx, "src");
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name = g_strdup_printf ("src_%u", sessid);
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gst_element_add_pad (bin, gst_ghost_pad_new (name, pad));
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g_free (name);
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gst_object_unref (pad);
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pad = gst_element_get_static_pad (rtx, "sink");
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name = g_strdup_printf ("sink_%u", sessid);
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gst_element_add_pad (bin, gst_ghost_pad_new (name, pad));
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g_free (name);
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gst_object_unref (pad);
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return bin;
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}
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/*
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* This function sets up the UDP sinks and sources for RTP/RTCP, adds the
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* given session's bin into the pipeline, and links it to the properly numbered
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* pads on the rtpbin
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*/
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static void
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add_stream (GstPipeline * pipe, GstElement * rtpBin, SessionData * session)
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{
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GstElement *rtpSink = gst_element_factory_make ("udpsink", NULL);
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GstElement *rtcpSink = gst_element_factory_make ("udpsink", NULL);
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GstElement *rtcpSrc = gst_element_factory_make ("udpsrc", NULL);
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GstElement *identity = gst_element_factory_make ("identity", NULL);
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int basePort;
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gchar *padName;
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basePort = 5000 + (session->sessionNum * 6);
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gst_bin_add_many (GST_BIN (pipe), rtpSink, rtcpSink, rtcpSrc, identity,
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session->input, NULL);
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/* enable retransmission by setting rtprtxsend as the "aux" element of rtpbin */
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g_signal_connect (rtpBin, "request-aux-sender",
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(GCallback) request_aux_sender, session);
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g_object_set (rtpSink, "port", basePort, "host", "127.0.0.1", NULL);
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g_object_set (rtcpSink, "port", basePort + 1, "host", "127.0.0.1", "sync",
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FALSE, "async", FALSE, NULL);
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g_object_set (rtcpSrc, "port", basePort + 5, NULL);
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/* this is just to drop some rtp packets at random, to demonstrate
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* that rtprtxsend actually works */
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g_object_set (identity, "drop-probability", 0.01, NULL);
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padName = g_strdup_printf ("send_rtp_sink_%u", session->sessionNum);
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gst_element_link_pads (session->input, "src", rtpBin, padName);
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g_free (padName);
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/* link rtpbin to udpsink directly here if you don't want
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* artificial packet loss */
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padName = g_strdup_printf ("send_rtp_src_%u", session->sessionNum);
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gst_element_link_pads (rtpBin, padName, identity, "sink");
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gst_element_link (identity, rtpSink);
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g_free (padName);
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padName = g_strdup_printf ("send_rtcp_src_%u", session->sessionNum);
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gst_element_link_pads (rtpBin, padName, rtcpSink, "sink");
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g_free (padName);
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padName = g_strdup_printf ("recv_rtcp_sink_%u", session->sessionNum);
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gst_element_link_pads (rtcpSrc, "src", rtpBin, padName);
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g_free (padName);
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g_print ("New RTP stream on %i/%i/%i\n", basePort, basePort + 1,
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basePort + 5);
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session_unref (session);
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}
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int
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main (int argc, char **argv)
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{
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GstPipeline *pipe;
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GstBus *bus;
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SessionData *videoSession;
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SessionData *audioSession;
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GstElement *rtpBin;
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GMainLoop *loop;
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gst_init (&argc, &argv);
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loop = g_main_loop_new (NULL, FALSE);
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pipe = GST_PIPELINE (gst_pipeline_new (NULL));
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bus = gst_element_get_bus (GST_ELEMENT (pipe));
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g_signal_connect (bus, "message::state-changed", G_CALLBACK (cb_state), pipe);
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gst_bus_add_signal_watch (bus);
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gst_object_unref (bus);
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rtpBin = gst_element_factory_make ("rtpbin", NULL);
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g_object_set (rtpBin, "rtp-profile", GST_RTP_PROFILE_AVPF, NULL);
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gst_bin_add (GST_BIN (pipe), rtpBin);
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videoSession = make_video_session (0);
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audioSession = make_audio_session (1);
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add_stream (pipe, rtpBin, videoSession);
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add_stream (pipe, rtpBin, audioSession);
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g_print ("starting server pipeline\n");
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gst_element_set_state (GST_ELEMENT (pipe), GST_STATE_PLAYING);
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g_main_loop_run (loop);
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g_print ("stopping server pipeline\n");
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gst_element_set_state (GST_ELEMENT (pipe), GST_STATE_NULL);
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gst_object_unref (pipe);
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g_main_loop_unref (loop);
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return 0;
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}
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