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221 lines
7.7 KiB
C
221 lines
7.7 KiB
C
/* GStreamer
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* Copyright (C) 2009 Wim Taymans <wim.taymans@gmail.com>
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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., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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#include <string.h>
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#include <math.h>
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#include <gst/gst.h>
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/*
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* A simple RTP server
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* sends the output of alsasrc as alaw encoded RTP on port 5002, RTCP is sent on
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* port 5003. The destination is 127.0.0.1.
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* the receiver RTCP reports are received on port 5007
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*
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* .-------. .-------. .-------. .----------. .-------.
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* |alsasrc| |alawenc| |pcmapay| | rtpbin | |udpsink| RTP
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* | src->sink src->sink src->send_rtp send_rtp->sink | port=5002
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* '-------' '-------' '-------' | | '-------'
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* | |
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* | | .-------.
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* | | |udpsink| RTCP
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* | send_rtcp->sink | port=5003
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* .-------. | | '-------' sync=false
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* RTCP |udpsrc | | | async=false
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* port=5007 | src->recv_rtcp |
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* '-------' '----------'
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*/
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/* change this to send the RTP data and RTCP to another host */
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#define DEST_HOST "127.0.0.1"
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/* #define AUDIO_SRC "alsasrc" */
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#define AUDIO_SRC "audiotestsrc"
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/* the encoder and payloader elements */
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#define AUDIO_ENC "alawenc"
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#define AUDIO_PAY "rtppcmapay"
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/* print the stats of a source */
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static void
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print_source_stats (GObject * source)
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{
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GstStructure *stats;
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gchar *str;
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/* get the source stats */
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g_object_get (source, "stats", &stats, NULL);
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/* simply dump the stats structure */
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str = gst_structure_to_string (stats);
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g_print ("source stats: %s\n", str);
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gst_structure_free (stats);
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g_free (str);
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}
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/* this function is called every second and dumps the RTP manager stats */
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static gboolean
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print_stats (GstElement * rtpbin)
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{
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GObject *session;
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GValueArray *arr;
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GValue *val;
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guint i;
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g_print ("***********************************\n");
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/* get session 0 */
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g_signal_emit_by_name (rtpbin, "get-internal-session", 0, &session);
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/* print all the sources in the session, this includes the internal source */
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g_object_get (session, "sources", &arr, NULL);
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for (i = 0; i < arr->n_values; i++) {
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GObject *source;
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val = g_value_array_get_nth (arr, i);
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source = g_value_get_object (val);
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print_source_stats (source);
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}
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g_value_array_free (arr);
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g_object_unref (session);
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return TRUE;
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}
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/* build a pipeline equivalent to:
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*
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* gst-launch -v gstrtpbin name=rtpbin \
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* $AUDIO_SRC ! audioconvert ! audioresample ! $AUDIO_ENC ! $AUDIO_PAY ! rtpbin.send_rtp_sink_0 \
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* rtpbin.send_rtp_src_0 ! udpsink port=5002 host=$DEST \
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* rtpbin.send_rtcp_src_0 ! udpsink port=5003 host=$DEST sync=false async=false \
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* udpsrc port=5007 ! rtpbin.recv_rtcp_sink_0
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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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GstElement *audiosrc, *audioconv, *audiores, *audioenc, *audiopay;
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GstElement *rtpbin, *rtpsink, *rtcpsink, *rtcpsrc;
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GstElement *pipeline;
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GMainLoop *loop;
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GstPad *srcpad, *sinkpad;
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/* always init first */
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gst_init (&argc, &argv);
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/* the pipeline to hold everything */
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pipeline = gst_pipeline_new (NULL);
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g_assert (pipeline);
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/* the audio capture and format conversion */
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audiosrc = gst_element_factory_make (AUDIO_SRC, "audiosrc");
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g_assert (audiosrc);
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audioconv = gst_element_factory_make ("audioconvert", "audioconv");
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g_assert (audioconv);
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audiores = gst_element_factory_make ("audioresample", "audiores");
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g_assert (audiores);
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/* the encoding and payloading */
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audioenc = gst_element_factory_make (AUDIO_ENC, "audioenc");
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g_assert (audioenc);
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audiopay = gst_element_factory_make (AUDIO_PAY, "audiopay");
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g_assert (audiopay);
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/* add capture and payloading to the pipeline and link */
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gst_bin_add_many (GST_BIN (pipeline), audiosrc, audioconv, audiores,
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audioenc, audiopay, NULL);
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if (!gst_element_link_many (audiosrc, audioconv, audiores, audioenc,
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audiopay, NULL)) {
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g_error ("Failed to link audiosrc, audioconv, audioresample, "
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"audio encoder and audio payloader");
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}
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/* the rtpbin element */
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rtpbin = gst_element_factory_make ("gstrtpbin", "rtpbin");
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g_assert (rtpbin);
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gst_bin_add (GST_BIN (pipeline), rtpbin);
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/* the udp sinks and source we will use for RTP and RTCP */
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rtpsink = gst_element_factory_make ("udpsink", "rtpsink");
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g_assert (rtpsink);
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g_object_set (rtpsink, "port", 5002, "host", DEST_HOST, NULL);
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rtcpsink = gst_element_factory_make ("udpsink", "rtcpsink");
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g_assert (rtcpsink);
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g_object_set (rtcpsink, "port", 5003, "host", DEST_HOST, NULL);
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/* no need for synchronisation or preroll on the RTCP sink */
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g_object_set (rtcpsink, "async", FALSE, "sync", FALSE, NULL);
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rtcpsrc = gst_element_factory_make ("udpsrc", "rtcpsrc");
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g_assert (rtcpsrc);
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g_object_set (rtcpsrc, "port", 5007, NULL);
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gst_bin_add_many (GST_BIN (pipeline), rtpsink, rtcpsink, rtcpsrc, NULL);
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/* now link all to the rtpbin, start by getting an RTP sinkpad for session 0 */
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sinkpad = gst_element_get_request_pad (rtpbin, "send_rtp_sink_0");
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srcpad = gst_element_get_static_pad (audiopay, "src");
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if (gst_pad_link (srcpad, sinkpad) != GST_PAD_LINK_OK)
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g_error ("Failed to link audio payloader to rtpbin");
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gst_object_unref (srcpad);
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/* get the RTP srcpad that was created when we requested the sinkpad above and
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* link it to the rtpsink sinkpad*/
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srcpad = gst_element_get_static_pad (rtpbin, "send_rtp_src_0");
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sinkpad = gst_element_get_static_pad (rtpsink, "sink");
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if (gst_pad_link (srcpad, sinkpad) != GST_PAD_LINK_OK)
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g_error ("Failed to link rtpbin to rtpsink");
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gst_object_unref (srcpad);
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gst_object_unref (sinkpad);
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/* get an RTCP srcpad for sending RTCP to the receiver */
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srcpad = gst_element_get_request_pad (rtpbin, "send_rtcp_src_0");
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sinkpad = gst_element_get_static_pad (rtcpsink, "sink");
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if (gst_pad_link (srcpad, sinkpad) != GST_PAD_LINK_OK)
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g_error ("Failed to link rtpbin to rtcpsink");
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gst_object_unref (sinkpad);
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/* we also want to receive RTCP, request an RTCP sinkpad for session 0 and
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* link it to the srcpad of the udpsrc for RTCP */
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srcpad = gst_element_get_static_pad (rtcpsrc, "src");
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sinkpad = gst_element_get_request_pad (rtpbin, "recv_rtcp_sink_0");
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if (gst_pad_link (srcpad, sinkpad) != GST_PAD_LINK_OK)
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g_error ("Failed to link rtcpsrc to rtpbin");
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gst_object_unref (srcpad);
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/* set the pipeline to playing */
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g_print ("starting sender pipeline\n");
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gst_element_set_state (pipeline, GST_STATE_PLAYING);
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/* print stats every second */
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g_timeout_add (1000, (GSourceFunc) print_stats, rtpbin);
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/* we need to run a GLib main loop to get the messages */
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loop = g_main_loop_new (NULL, FALSE);
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g_main_loop_run (loop);
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g_print ("stopping sender pipeline\n");
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gst_element_set_state (pipeline, GST_STATE_NULL);
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return 0;
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}
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