From 9f46e70477b728cb6cfb429169b90efa9b889e78 Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Fri, 2 Jan 2009 15:20:48 +0000 Subject: tests/examples/rtp/: Add two C examples of using gstrtpbin as a sender and a receiver. Original commit message from CVS: * tests/examples/rtp/.cvsignore: * tests/examples/rtp/Makefile.am: * tests/examples/rtp/client-PCMA.c: (pad_added_cb), (main): * tests/examples/rtp/server-alsasrc-PCMA.c: (main): Add two C examples of using gstrtpbin as a sender and a receiver. --- ChangeLog | 8 ++ tests/examples/rtp/.gitignore | 2 + tests/examples/rtp/Makefile.am | 10 ++ tests/examples/rtp/client-PCMA.c | 191 +++++++++++++++++++++++++++++++ tests/examples/rtp/server-alsasrc-PCMA.c | 168 +++++++++++++++++++++++++++ 5 files changed, 379 insertions(+) create mode 100644 tests/examples/rtp/.gitignore create mode 100755 tests/examples/rtp/client-PCMA.c create mode 100755 tests/examples/rtp/server-alsasrc-PCMA.c diff --git a/ChangeLog b/ChangeLog index 89259836..a13169c9 100644 --- a/ChangeLog +++ b/ChangeLog @@ -1,3 +1,11 @@ +2009-01-02 Wim Taymans + + * tests/examples/rtp/.cvsignore: + * tests/examples/rtp/Makefile.am: + * tests/examples/rtp/client-PCMA.c: (pad_added_cb), (main): + * tests/examples/rtp/server-alsasrc-PCMA.c: (main): + Add two C examples of using gstrtpbin as a sender and a receiver. + 2008-12-31 Jan Schmidt * ChangeLog: diff --git a/tests/examples/rtp/.gitignore b/tests/examples/rtp/.gitignore new file mode 100644 index 00000000..6b195b7b --- /dev/null +++ b/tests/examples/rtp/.gitignore @@ -0,0 +1,2 @@ +client-PCMA +server-alsasrc-PCMA diff --git a/tests/examples/rtp/Makefile.am b/tests/examples/rtp/Makefile.am index f0b033bd..f636e816 100644 --- a/tests/examples/rtp/Makefile.am +++ b/tests/examples/rtp/Makefile.am @@ -1,3 +1,13 @@ +noinst_PROGRAMS = server-alsasrc-PCMA client-PCMA + +server_alsasrc_PCMA_SOURCES = server-alsasrc-PCMA.c +server_alsasrc_PCMA_CFLAGS = $(GST_CFLAGS) +server_alsasrc_PCMA_LDADD = $(GST_LIBS) $(LIBM) + +client_PCMA_SOURCES = client-PCMA.c +client_PCMA_CFLAGS = $(GST_CFLAGS) +client_PCMA_LDADD = $(GST_LIBS) $(LIBM) + noinst_SCRIPTS=client-H263p-AMR.sh \ client-H263p-PCMA.sh \ client-H264-PCMA.sh \ diff --git a/tests/examples/rtp/client-PCMA.c b/tests/examples/rtp/client-PCMA.c new file mode 100755 index 00000000..0c895a23 --- /dev/null +++ b/tests/examples/rtp/client-PCMA.c @@ -0,0 +1,191 @@ +/* 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 + * 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 +#include + +#include + +/* + * 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; +} diff --git a/tests/examples/rtp/server-alsasrc-PCMA.c b/tests/examples/rtp/server-alsasrc-PCMA.c new file mode 100755 index 00000000..3af888a0 --- /dev/null +++ b/tests/examples/rtp/server-alsasrc-PCMA.c @@ -0,0 +1,168 @@ +/* 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 + * 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 +#include + +#include + +/* + * 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; +} -- cgit