mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-24 02:31:03 +00:00
237 lines
8.4 KiB
C
237 lines
8.4 KiB
C
/* GStreamer
|
|
* Copyright (C) 2009 Wim Taymans <wim.taymans@gmail.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., 59 Temple Place - Suite 330,
|
|
* Boston, MA 02111-1307, 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 "127.0.0.1"
|
|
|
|
/* 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 gstrtpbin 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, *isrc, *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);
|
|
|
|
/* get the internal source (the SSRC allocated to us, the receiver */
|
|
g_object_get (session, "internal-source", &isrc, NULL);
|
|
print_source_stats (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);
|
|
}
|
|
|
|
/* 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 -v gstrtpbin name=rtpbin \
|
|
* udpsrc caps=$AUDIO_CAPS port=5002 ! rtpbin.recv_rtp_sink_0 \
|
|
* rtpbin. ! rtppcmadepay ! alawdec ! audioconvert ! audioresample ! alsasink \
|
|
* udpsrc port=5003 ! rtpbin.recv_rtcp_sink_0 \
|
|
* rtpbin.send_rtcp_src_0 ! udpsink port=5007 host=$DEST sync=false async=false
|
|
*/
|
|
int
|
|
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 ("gstrtpbin", "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),
|
|
audiodepay);
|
|
|
|
/* 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;
|
|
}
|