--- /dev/null
+/* 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"
+
+/* 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);
+
+ /* 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;
+}
--- /dev/null
+/* 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 server
+ * sends the output of alsasrc as alaw encoded RTP on port 5002, RTCP is sent on
+ * port 5003. The destination is 127.0.0.1.
+ * the receiver RTCP reports are received on port 5007
+ *
+ * .-------. .-------. .-------. .----------. .-------.
+ * |alsasrc| |alawenc| |pcmapay| | rtpbin | |udpsink| RTP
+ * | src->sink src->sink src->send_rtp send_rtp->sink | port=5002
+ * '-------' '-------' '-------' | | '-------'
+ * | |
+ * | | .-------.
+ * | | |udpsink| RTCP
+ * | send_rtcp->sink | port=5003
+ * .-------. | | '-------' sync=false
+ * RTCP |udpsrc | | | async=false
+ * port=5007 | src->recv_rtcp |
+ * '-------' '----------'
+ */
+
+/* change this to send the RTP data and RTCP to another host */
+#define DEST_HOST "127.0.0.1"
+
+/* #define AUDIO_SRC "alsasrc" */
+#define AUDIO_SRC "audiotestsrc"
+
+/* the encoder and payloader elements */
+#define AUDIO_ENC "alawenc"
+#define AUDIO_PAY "rtppcmapay"
+
+/* build a pipeline equivalent to:
+ *
+ * gst-launch -v gstrtpbin name=rtpbin \
+ * $AUDIO_SRC ! audioconvert ! audioresample ! $AUDIO_ENC ! $AUDIO_PAY ! rtpbin.send_rtp_sink_0 \
+ * rtpbin.send_rtp_src_0 ! udpsink port=5002 host=$DEST \
+ * rtpbin.send_rtcp_src_0 ! udpsink port=5003 host=$DEST sync=false async=false \
+ * udpsrc port=5007 ! rtpbin.recv_rtcp_sink_0
+ */
+int
+main (int argc, char *argv[])
+{
+ GstElement *audiosrc, *audioconv, *audiores, *audioenc, *audiopay;
+ GstElement *rtpbin, *rtpsink, *rtcpsink, *rtcpsrc;
+ GstElement *pipeline;
+ GMainLoop *loop;
+ 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 audio capture and format conversion */
+ audiosrc = gst_element_factory_make (AUDIO_SRC, "audiosrc");
+ g_assert (audiosrc);
+ audioconv = gst_element_factory_make ("audioconvert", "audioconv");
+ g_assert (audioconv);
+ audiores = gst_element_factory_make ("audioresample", "audiores");
+ g_assert (audiores);
+ /* the encoding and payloading */
+ audioenc = gst_element_factory_make (AUDIO_ENC, "audioenc");
+ g_assert (audioenc);
+ audiopay = gst_element_factory_make (AUDIO_PAY, "audiopay");
+ g_assert (audiopay);
+
+ /* add capture and payloading to the pipeline and link */
+ gst_bin_add_many (GST_BIN (pipeline), audiosrc, audioconv, audiores,
+ audioenc, audiopay, NULL);
+
+ res = gst_element_link_many (audiosrc, audioconv, audiores, audioenc,
+ audiopay, 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);
+
+ /* the udp sinks and source we will use for RTP and RTCP */
+ rtpsink = gst_element_factory_make ("udpsink", "rtpsink");
+ g_assert (rtpsink);
+ g_object_set (rtpsink, "port", 5002, "host", DEST_HOST, NULL);
+
+ rtcpsink = gst_element_factory_make ("udpsink", "rtcpsink");
+ g_assert (rtcpsink);
+ g_object_set (rtcpsink, "port", 5003, "host", DEST_HOST, NULL);
+ /* no need for synchronisation or preroll on the RTCP sink */
+ g_object_set (rtcpsink, "async", FALSE, "sync", FALSE, NULL);
+
+ rtcpsrc = gst_element_factory_make ("udpsrc", "rtcpsrc");
+ g_assert (rtcpsrc);
+ g_object_set (rtcpsrc, "port", 5007, NULL);
+
+ gst_bin_add_many (GST_BIN (pipeline), rtpsink, rtcpsink, rtcpsrc, NULL);
+
+ /* now link all to the rtpbin, start by getting an RTP sinkpad for session 0 */
+ sinkpad = gst_element_get_request_pad (rtpbin, "send_rtp_sink_0");
+ srcpad = gst_element_get_static_pad (audiopay, "src");
+ lres = gst_pad_link (srcpad, sinkpad);
+ g_assert (lres == GST_PAD_LINK_OK);
+ gst_object_unref (srcpad);
+
+ /* get the RTP srcpad that was created when we requested the sinkpad above and
+ * link it to the rtpsink sinkpad*/
+ srcpad = gst_element_get_static_pad (rtpbin, "send_rtp_src_0");
+ sinkpad = gst_element_get_static_pad (rtpsink, "sink");
+ 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 to the receiver */
+ 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);
+
+ /* we also want to receive RTCP, request an RTCP sinkpad for session 0 and
+ * link it to the srcpad of the udpsrc for RTCP */
+ 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);
+
+ /* set the pipeline to playing */
+ g_print ("starting sender 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 sender pipeline\n");
+ gst_element_set_state (pipeline, GST_STATE_NULL);
+
+ return 0;
+}
projects / platform / upstream / gstreamer.git / commitdiff