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243 lines
8.5 KiB
C
243 lines
8.5 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., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, 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 receiver
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*
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* receives alaw encoded RTP audio on port 5002, RTCP is received on port 5003.
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* the receiver RTCP reports are sent to port 5007
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*
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* .-------. .----------. .---------. .-------. .--------.
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* RTP |udpsrc | | rtpbin | |pcmadepay| |alawdec| |alsasink|
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* port=5002 | src->recv_rtp recv_rtp->sink src->sink src->sink |
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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=5007
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* .-------. | | '-------' sync=false
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* RTCP |udpsrc | | | async=false
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* port=5003 | src->recv_rtcp |
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* '-------' '----------'
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*/
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/* the caps of the sender RTP stream. This is usually negotiated out of band with
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* SDP or RTSP. */
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#define AUDIO_CAPS "application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)PCMA"
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#define AUDIO_DEPAY "rtppcmadepay"
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#define AUDIO_DEC "alawdec"
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#define AUDIO_SINK "autoaudiosink"
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/* the destination machine to send RTCP to. This is the address of the sender and
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* is used to send back the RTCP reports of this receiver. If the data is sent
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* from another machine, change this address. */
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#define DEST_HOST "127.0.0.1"
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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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g_return_if_fail (source != NULL);
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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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/* will be called when rtpbin signals on-ssrc-active. It means that an RTCP
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* packet was received from another source. */
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static void
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on_ssrc_active_cb (GstElement * rtpbin, guint sessid, guint ssrc,
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GstElement * depay)
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{
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GObject *session, *isrc, *osrc;
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g_print ("got RTCP from session %u, SSRC %u\n", sessid, ssrc);
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/* get the right session */
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g_signal_emit_by_name (rtpbin, "get-internal-session", sessid, &session);
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#if 0
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/* FIXME: This is broken in rtpbin */
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/* get the internal source (the SSRC allocated to us, the receiver */
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g_object_get (session, "internal-source", &isrc, NULL);
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print_source_stats (isrc);
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g_object_unref (isrc);
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#endif
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/* get the remote source that sent us RTCP */
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g_signal_emit_by_name (session, "get-source-by-ssrc", ssrc, &osrc);
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print_source_stats (osrc);
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g_object_unref (osrc);
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g_object_unref (session);
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}
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/* will be called when rtpbin has validated a payload that we can depayload */
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static void
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pad_added_cb (GstElement * rtpbin, GstPad * new_pad, GstElement * depay)
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{
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GstPad *sinkpad;
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GstPadLinkReturn lres;
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g_print ("new payload on pad: %s\n", GST_PAD_NAME (new_pad));
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sinkpad = gst_element_get_static_pad (depay, "sink");
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g_assert (sinkpad);
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lres = gst_pad_link (new_pad, sinkpad);
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g_assert (lres == GST_PAD_LINK_OK);
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gst_object_unref (sinkpad);
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}
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/* build a pipeline equivalent to:
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*
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* gst-launch-1.0 -v rtpbin name=rtpbin \
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* udpsrc caps=$AUDIO_CAPS port=5002 ! rtpbin.recv_rtp_sink_0 \
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* rtpbin. ! rtppcmadepay ! alawdec ! audioconvert ! audioresample ! autoaudiosink \
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* udpsrc port=5003 ! rtpbin.recv_rtcp_sink_0 \
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* rtpbin.send_rtcp_src_0 ! udpsink port=5007 host=$DEST sync=false async=false
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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 *rtpbin, *rtpsrc, *rtcpsrc, *rtcpsink;
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GstElement *audiodepay, *audiodec, *audiores, *audioconv, *audiosink;
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GstElement *pipeline;
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GMainLoop *loop;
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GstCaps *caps;
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gboolean res;
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GstPadLinkReturn lres;
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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 udp src and source we will use for RTP and RTCP */
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rtpsrc = gst_element_factory_make ("udpsrc", "rtpsrc");
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g_assert (rtpsrc);
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g_object_set (rtpsrc, "port", 5002, NULL);
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/* we need to set caps on the udpsrc for the RTP data */
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caps = gst_caps_from_string (AUDIO_CAPS);
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g_object_set (rtpsrc, "caps", caps, NULL);
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gst_caps_unref (caps);
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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", 5003, 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", 5007, "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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gst_bin_add_many (GST_BIN (pipeline), rtpsrc, rtcpsrc, rtcpsink, NULL);
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/* the depayloading and decoding */
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audiodepay = gst_element_factory_make (AUDIO_DEPAY, "audiodepay");
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g_assert (audiodepay);
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audiodec = gst_element_factory_make (AUDIO_DEC, "audiodec");
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g_assert (audiodec);
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/* the audio playback and format conversion */
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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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audiosink = gst_element_factory_make (AUDIO_SINK, "audiosink");
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g_assert (audiosink);
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/* add depayloading and playback to the pipeline and link */
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gst_bin_add_many (GST_BIN (pipeline), audiodepay, audiodec, audioconv,
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audiores, audiosink, NULL);
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res = gst_element_link_many (audiodepay, audiodec, audioconv, audiores,
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audiosink, NULL);
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g_assert (res == TRUE);
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/* the rtpbin element */
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rtpbin = gst_element_factory_make ("rtpbin", "rtpbin");
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g_assert (rtpbin);
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gst_bin_add (GST_BIN (pipeline), rtpbin);
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/* now link all to the rtpbin, start by getting an RTP sinkpad for session 0 */
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srcpad = gst_element_get_static_pad (rtpsrc, "src");
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sinkpad = gst_element_get_request_pad (rtpbin, "recv_rtp_sink_0");
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lres = gst_pad_link (srcpad, sinkpad);
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g_assert (lres == GST_PAD_LINK_OK);
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gst_object_unref (srcpad);
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/* get an RTCP sinkpad in session 0 */
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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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lres = gst_pad_link (srcpad, sinkpad);
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g_assert (lres == GST_PAD_LINK_OK);
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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 back to the sender */
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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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lres = gst_pad_link (srcpad, sinkpad);
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g_assert (lres == GST_PAD_LINK_OK);
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gst_object_unref (sinkpad);
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/* the RTP pad that we have to connect to the depayloader will be created
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* dynamically so we connect to the pad-added signal, pass the depayloader as
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* user_data so that we can link to it. */
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g_signal_connect (rtpbin, "pad-added", G_CALLBACK (pad_added_cb), audiodepay);
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/* give some stats when we receive RTCP */
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g_signal_connect (rtpbin, "on-ssrc-active", G_CALLBACK (on_ssrc_active_cb),
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audiodepay);
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/* set the pipeline to playing */
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g_print ("starting receiver pipeline\n");
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gst_element_set_state (pipeline, GST_STATE_PLAYING);
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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 receiver pipeline\n");
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gst_element_set_state (pipeline, GST_STATE_NULL);
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gst_object_unref (pipeline);
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return 0;
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}
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