blob: a1aa7f176b13beb27d4c370f113ef5c27a6c7ef2 [file] [log] [blame]
/* 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
* 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 <string.h>
#include <math.h>
#include <gst/gst.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 | |pcmadepay| |alawdec| |alsasink|
* 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. */
#define AUDIO_CAPS "application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)PCMA"
#define AUDIO_DEPAY "rtppcmadepay"
#define AUDIO_DEC "alawdec"
#define AUDIO_SINK "autoaudiosink"
/* 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 ""
/* 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 rtpbin 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,
GstElement * depay)
GObject *session, *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);
#if 0
/* FIXME: This is broken in rtpbin */
/* get the internal source (the SSRC allocated to us, the receiver */
g_object_get (session, "internal-source", &isrc, NULL);
print_source_stats (isrc);
g_object_unref (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);
g_object_unref (osrc);
g_object_unref (session);
/* will be called when rtpbin has validated a payload that we can depayload */
static void
pad_added_cb (GstElement * rtpbin, GstPad * new_pad, GstElement * depay)
GstPad *sinkpad;
GstPadLinkReturn lres;
g_print ("new payload on pad: %s\n", GST_PAD_NAME (new_pad));
sinkpad = gst_element_get_static_pad (depay, "sink");
g_assert (sinkpad);
lres = gst_pad_link (new_pad, sinkpad);
g_assert (lres == GST_PAD_LINK_OK);
gst_object_unref (sinkpad);
/* build a pipeline equivalent to:
* gst-launch-1.0 -v rtpbin name=rtpbin \
* udpsrc caps=$AUDIO_CAPS port=5002 ! rtpbin.recv_rtp_sink_0 \
* rtpbin. ! rtppcmadepay ! alawdec ! audioconvert ! audioresample ! autoaudiosink \
* udpsrc port=5003 ! rtpbin.recv_rtcp_sink_0 \
* rtpbin.send_rtcp_src_0 ! udpsink port=5007 host=$DEST sync=false async=false
main (int argc, char *argv[])
GstElement *rtpbin, *rtpsrc, *rtcpsrc, *rtcpsink;
GstElement *audiodepay, *audiodec, *audiores, *audioconv, *audiosink;
GstElement *pipeline;
GMainLoop *loop;
GstCaps *caps;
gboolean res;
GstPadLinkReturn lres;
GstPad *srcpad, *sinkpad;
/* always init first */
gst_init (&argc, &argv);
/* the pipeline to hold everything */
pipeline = gst_pipeline_new (NULL);
g_assert (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 = gst_caps_from_string (AUDIO_CAPS);
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 (pipeline), rtpsrc, rtcpsrc, rtcpsink, NULL);
/* the depayloading and decoding */
audiodepay = gst_element_factory_make (AUDIO_DEPAY, "audiodepay");
g_assert (audiodepay);
audiodec = gst_element_factory_make (AUDIO_DEC, "audiodec");
g_assert (audiodec);
/* the audio playback and format conversion */
audioconv = gst_element_factory_make ("audioconvert", "audioconv");
g_assert (audioconv);
audiores = gst_element_factory_make ("audioresample", "audiores");
g_assert (audiores);
audiosink = gst_element_factory_make (AUDIO_SINK, "audiosink");
g_assert (audiosink);
/* add depayloading and playback to the pipeline and link */
gst_bin_add_many (GST_BIN (pipeline), audiodepay, audiodec, audioconv,
audiores, audiosink, NULL);
res = gst_element_link_many (audiodepay, audiodec, audioconv, audiores,
audiosink, NULL);
g_assert (res == TRUE);
/* the rtpbin element */
rtpbin = gst_element_factory_make ("rtpbin", "rtpbin");
g_assert (rtpbin);
gst_bin_add (GST_BIN (pipeline), rtpbin);
/* now link all to the rtpbin, start by getting an RTP sinkpad for session 0 */
srcpad = gst_element_get_static_pad (rtpsrc, "src");
sinkpad = gst_element_get_request_pad (rtpbin, "recv_rtp_sink_0");
lres = gst_pad_link (srcpad, sinkpad);
g_assert (lres == GST_PAD_LINK_OK);
gst_object_unref (srcpad);
/* 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);
/* 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), audiodepay);
/* give some stats when we receive RTCP */
g_signal_connect (rtpbin, "on-ssrc-active", G_CALLBACK (on_ssrc_active_cb),
/* set the pipeline to playing */
g_print ("starting receiver pipeline\n");
gst_element_set_state (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 (pipeline, GST_STATE_NULL);
gst_object_unref (pipeline);
return 0;