diff options
Diffstat (limited to 'tests/examples/rtp')
| -rw-r--r-- | tests/examples/rtp/Makefile.am | 22 | ||||
| -rw-r--r-- | tests/examples/rtp/Makefile.in | 796 | ||||
| -rwxr-xr-x | tests/examples/rtp/client-H263p-AMR.sh | 23 | ||||
| -rw-r--r-- | tests/examples/rtp/client-H263p-PCMA.sdp | 12 | ||||
| -rwxr-xr-x | tests/examples/rtp/client-H263p-PCMA.sh | 28 | ||||
| -rw-r--r-- | tests/examples/rtp/client-H264-PCMA.sdp | 12 | ||||
| -rwxr-xr-x | tests/examples/rtp/client-H264-PCMA.sh | 64 | ||||
| -rw-r--r-- | tests/examples/rtp/client-PCMA.c | 237 | ||||
| -rwxr-xr-x | tests/examples/rtp/client-PCMA.sh | 39 | ||||
| -rwxr-xr-x | tests/examples/rtp/server-VTS-H263p-ATS-PCMA.sh | 16 | ||||
| -rw-r--r-- | tests/examples/rtp/server-alsasrc-PCMA.c | 221 | ||||
| -rwxr-xr-x | tests/examples/rtp/server-alsasrc-PCMA.sh | 35 | ||||
| -rwxr-xr-x | tests/examples/rtp/server-v4l2-H263p-alsasrc-AMR.sh | 25 | ||||
| -rwxr-xr-x | tests/examples/rtp/server-v4l2-H264-alsasrc-PCMA.sh | 78 |
14 files changed, 1608 insertions, 0 deletions
diff --git a/tests/examples/rtp/Makefile.am b/tests/examples/rtp/Makefile.am new file mode 100644 index 0000000..f636e81 --- /dev/null +++ b/tests/examples/rtp/Makefile.am @@ -0,0 +1,22 @@ +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 \ + client-PCMA.sh \ + server-alsasrc-PCMA.sh \ + server-v4l2-H263p-alsasrc-AMR.sh \ + server-v4l2-H264-alsasrc-PCMA.sh \ + server-VTS-H263p-ATS-PCMA.sh + +EXTRA_DIST= $(noinst_SCRIPTS) \ + client-H263p-PCMA.sdp \ + client-H264-PCMA.sdp diff --git a/tests/examples/rtp/Makefile.in b/tests/examples/rtp/Makefile.in new file mode 100644 index 0000000..56e5948 --- /dev/null +++ b/tests/examples/rtp/Makefile.in @@ -0,0 +1,796 @@ +# Makefile.in generated by automake 1.11.3 from Makefile.am. +# @configure_input@ + +# Copyright (C) 1994, 1995, 1996, 1997, 1998, 1999, 2000, 2001, 2002, +# 2003, 2004, 2005, 2006, 2007, 2008, 2009, 2010, 2011 Free Software +# Foundation, Inc. +# This Makefile.in is free software; the Free Software Foundation +# gives unlimited permission to copy and/or distribute it, +# with or without modifications, as long as this notice is preserved. + +# This program is distributed in the hope that it will be useful, +# but WITHOUT ANY WARRANTY, to the extent permitted by law; without +# even the implied warranty of MERCHANTABILITY or FITNESS FOR A +# PARTICULAR PURPOSE. + +@SET_MAKE@ + + +VPATH = @srcdir@ +pkgdatadir = $(datadir)/@PACKAGE@ +pkgincludedir = $(includedir)/@PACKAGE@ +pkglibdir = $(libdir)/@PACKAGE@ +pkglibexecdir = $(libexecdir)/@PACKAGE@ +am__cd = CDPATH="$${ZSH_VERSION+.}$(PATH_SEPARATOR)" && cd +install_sh_DATA = $(install_sh) -c -m 644 +install_sh_PROGRAM = 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mode 100755 index 0000000..bb0b795 --- /dev/null +++ b/tests/examples/rtp/client-H263p-AMR.sh @@ -0,0 +1,23 @@ +#!/bin/sh +# +# A simple RTP receiver +# + +VIDEO_CAPS="application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H263-1998" +AUDIO_CAPS="application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)AMR,encoding-params=(string)1,octet-align=(string)1" + +VIDEO_DEC="rtph263pdepay ! ffdec_h263" +AUDIO_DEC="rtpamrdepay ! amrnbdec" + +VIDEO_SINK="ffmpegcolorspace ! autovideosink" +AUDIO_SINK="audioconvert ! audioresample ! autoaudiosink" + +gst-launch -v gstrtpbin name=rtpbin latency=100 \ + udpsrc caps=$VIDEO_CAPS port=5000 ! rtpbin.recv_rtp_sink_0 \ + rtpbin. ! $VIDEO_DEC ! $VIDEO_SINK \ + udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 \ + rtpbin.send_rtcp_src_0 ! udpsink port=5005 sync=false async=false \ + udpsrc caps=$AUDIO_CAPS port=5002 ! rtpbin.recv_rtp_sink_1 \ + rtpbin. ! $AUDIO_DEC ! $AUDIO_SINK \ + udpsrc port=5003 ! rtpbin.recv_rtcp_sink_1 \ + rtpbin.send_rtcp_src_1 ! udpsink port=5007 sync=false async=false diff --git a/tests/examples/rtp/client-H263p-PCMA.sdp b/tests/examples/rtp/client-H263p-PCMA.sdp new file mode 100644 index 0000000..8d2264f --- /dev/null +++ b/tests/examples/rtp/client-H263p-PCMA.sdp @@ -0,0 +1,12 @@ +v=0 +o=- 1188340656180883 1 IN IP4 127.0.0.1 +s=Session streamed by GStreamer +i=server.sh +t=0 0 +a=tool:GStreamer +a=type:broadcast +m=video 5000 RTP/AVP 96 +c=IN IP4 127.0.0.1 +a=rtpmap:96 H263-1998/90000 +m=audio 5002 RTP/AVP 8 +c=IN IP4 127.0.0.1 diff --git a/tests/examples/rtp/client-H263p-PCMA.sh b/tests/examples/rtp/client-H263p-PCMA.sh new file mode 100755 index 0000000..9db0d7e --- /dev/null +++ b/tests/examples/rtp/client-H263p-PCMA.sh @@ -0,0 +1,28 @@ +#!/bin/sh +# +# A simple RTP receiver +# + +VIDEO_CAPS="application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H263-1998" +AUDIO_CAPS="application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)PCMA" + +#DEST=192.168.1.126 +DEST=localhost + +VIDEO_DEC="rtph263pdepay ! ffdec_h263" +AUDIO_DEC="rtppcmadepay ! alawdec" + +VIDEO_SINK="ffmpegcolorspace ! autovideosink" +AUDIO_SINK="audioconvert ! audioresample ! autoaudiosink" + +LATENCY=100 + +gst-launch -v gstrtpbin name=rtpbin latency=$LATENCY \ + udpsrc caps=$VIDEO_CAPS port=5000 ! rtpbin.recv_rtp_sink_0 \ + rtpbin. ! $VIDEO_DEC ! $VIDEO_SINK \ + udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 \ + rtpbin.send_rtcp_src_0 ! udpsink host=$DEST port=5005 sync=false async=false \ + udpsrc caps=$AUDIO_CAPS port=5002 ! rtpbin.recv_rtp_sink_1 \ + rtpbin. ! $AUDIO_DEC ! $AUDIO_SINK \ + udpsrc port=5003 ! rtpbin.recv_rtcp_sink_1 \ + rtpbin.send_rtcp_src_1 ! udpsink host=$DEST port=5007 sync=false async=false diff --git a/tests/examples/rtp/client-H264-PCMA.sdp b/tests/examples/rtp/client-H264-PCMA.sdp new file mode 100644 index 0000000..85aa75f --- /dev/null +++ b/tests/examples/rtp/client-H264-PCMA.sdp @@ -0,0 +1,12 @@ +v=0 +o=- 1188340656180883 1 IN IP4 127.0.0.1 +s=Session streamed by GStreamer +i=server.sh +t=0 0 +a=tool:GStreamer +a=type:broadcast +m=video 5000 RTP/AVP 96 +c=IN IP4 127.0.0.1 +a=rtpmap:96 H264/90000 +m=audio 5002 RTP/AVP 8 +c=IN IP4 127.0.0.1 diff --git a/tests/examples/rtp/client-H264-PCMA.sh b/tests/examples/rtp/client-H264-PCMA.sh new file mode 100755 index 0000000..7b104ab --- /dev/null +++ b/tests/examples/rtp/client-H264-PCMA.sh @@ -0,0 +1,64 @@ +#!/bin/sh +# +# A simple RTP receiver +# +# receives H264 encoded RTP video on port 5000, RTCP is received on port 5001. +# the receiver RTCP reports are sent to port 5005 +# 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 | |h264depay| |h264dec| |xvimagesink| +# port=5000 | src->recv_rtp recv_rtp->sink src->sink src->sink | +# '-------' | | '---------' '-------' '-----------' +# | | +# | | .-------. +# | | |udpsink| RTCP +# | send_rtcp->sink | port=5005 +# .-------. | | '-------' sync=false +# RTCP |udpsrc | | | async=false +# port=5001 | src->recv_rtcp | +# '-------' | | +# | | +# .-------. | | .---------. .-------. .-------------. +# RTP |udpsrc | | rtpbin | |pcmadepay| |alawdec| |autoaudiosink| +# 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 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. +DEST=127.0.0.1 + +# this adjusts the latency in the receiver +LATENCY=200 + +# the caps of the sender RTP stream. This is usually negotiated out of band with +# SDP or RTSP. normally these caps will also include SPS and PPS but we don't +# have a mechanism to get this from the sender with a -launch line. +VIDEO_CAPS="application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H264" +AUDIO_CAPS="application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)PCMA" + +VIDEO_DEC="rtph264depay ! ffdec_h264" +AUDIO_DEC="rtppcmadepay ! alawdec" + +VIDEO_SINK="ffmpegcolorspace ! autovideosink" +AUDIO_SINK="audioconvert ! audioresample ! autoaudiosink" + +gst-launch -v gstrtpbin name=rtpbin latency=$LATENCY \ + udpsrc caps=$VIDEO_CAPS port=5000 ! rtpbin.recv_rtp_sink_0 \ + rtpbin. ! $VIDEO_DEC ! $VIDEO_SINK \ + udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 \ + rtpbin.send_rtcp_src_0 ! udpsink port=5005 host=$DEST sync=false async=false \ + udpsrc caps=$AUDIO_CAPS port=5002 ! rtpbin.recv_rtp_sink_1 \ + rtpbin. ! $AUDIO_DEC ! $AUDIO_SINK \ + udpsrc port=5003 ! rtpbin.recv_rtcp_sink_1 \ + rtpbin.send_rtcp_src_1 ! udpsink port=5007 host=$DEST sync=false async=false diff --git a/tests/examples/rtp/client-PCMA.c b/tests/examples/rtp/client-PCMA.c new file mode 100644 index 0000000..885b826 --- /dev/null +++ b/tests/examples/rtp/client-PCMA.c @@ -0,0 +1,237 @@ +/* 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; +} diff --git a/tests/examples/rtp/client-PCMA.sh b/tests/examples/rtp/client-PCMA.sh new file mode 100755 index 0000000..aad6502 --- /dev/null +++ b/tests/examples/rtp/client-PCMA.sh @@ -0,0 +1,39 @@ +#!/bin/sh +# +# 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| |autoaudiosink| +# 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. +AUDIO_CAPS="application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)PCMA" + +AUDIO_DEC="rtppcmadepay ! alawdec" + +AUDIO_SINK="audioconvert ! audioresample ! 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. +DEST=127.0.0.1 + +gst-launch -v gstrtpbin name=rtpbin \ + udpsrc caps=$AUDIO_CAPS port=5002 ! rtpbin.recv_rtp_sink_0 \ + rtpbin. ! $AUDIO_DEC ! $AUDIO_SINK \ + udpsrc port=5003 ! rtpbin.recv_rtcp_sink_0 \ + rtpbin.send_rtcp_src_0 ! udpsink port=5007 host=$DEST sync=false async=false diff --git a/tests/examples/rtp/server-VTS-H263p-ATS-PCMA.sh b/tests/examples/rtp/server-VTS-H263p-ATS-PCMA.sh new file mode 100755 index 0000000..ff2614d --- /dev/null +++ b/tests/examples/rtp/server-VTS-H263p-ATS-PCMA.sh @@ -0,0 +1,16 @@ +#!/bin/sh +# +# A simple RTP server +# + +VCAPS="video/x-raw-yuv,width=352,height=288,framerate=15/1" + +gst-launch -v gstrtpbin name=rtpbin \ + videotestsrc ! $VCAPS ! ffenc_h263p ! rtph263ppay ! rtpbin.send_rtp_sink_0 \ + rtpbin.send_rtp_src_0 ! udpsink port=5000 \ + rtpbin.send_rtcp_src_0 ! udpsink port=5001 sync=false async=false \ + udpsrc port=5005 ! rtpbin.recv_rtcp_sink_0 \ + audiotestsrc samplesperbuffer=1000 ! alawenc ! rtppcmapay ! rtpbin.send_rtp_sink_1 \ + rtpbin.send_rtp_src_1 ! udpsink port=5002 \ + rtpbin.send_rtcp_src_1 ! udpsink port=5003 sync=false async=false \ + udpsrc port=5007 ! rtpbin.recv_rtcp_sink_1 diff --git a/tests/examples/rtp/server-alsasrc-PCMA.c b/tests/examples/rtp/server-alsasrc-PCMA.c new file mode 100644 index 0000000..d61ec30 --- /dev/null +++ b/tests/examples/rtp/server-alsasrc-PCMA.c @@ -0,0 +1,221 @@ +/* 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" + +/* print the stats of a source */ +static void +print_source_stats (GObject * source) +{ + GstStructure *stats; + gchar *str; + + /* 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); +} + +/* this function is called every second and dumps the RTP manager stats */ +static gboolean +print_stats (GstElement * rtpbin) +{ + GObject *session; + GValueArray *arr; + GValue *val; + guint i; + + g_print ("***********************************\n"); + + /* get session 0 */ + g_signal_emit_by_name (rtpbin, "get-internal-session", 0, &session); + + /* print all the sources in the session, this includes the internal source */ + g_object_get (session, "sources", &arr, NULL); + + for (i = 0; i < arr->n_values; i++) { + GObject *source; + + val = g_value_array_get_nth (arr, i); + source = g_value_get_object (val); + + print_source_stats (source); + } + g_value_array_free (arr); + + g_object_unref (session); + + return TRUE; +} + +/* 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; + 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); + + if (!gst_element_link_many (audiosrc, audioconv, audiores, audioenc, + audiopay, NULL)) { + g_error ("Failed to link audiosrc, audioconv, audioresample, " + "audio encoder and audio payloader"); + } + + /* 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"); + if (gst_pad_link (srcpad, sinkpad) != GST_PAD_LINK_OK) + g_error ("Failed to link audio payloader to rtpbin"); + 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"); + if (gst_pad_link (srcpad, sinkpad) != GST_PAD_LINK_OK) + g_error ("Failed to link rtpbin to rtpsink"); + 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"); + if (gst_pad_link (srcpad, sinkpad) != GST_PAD_LINK_OK) + g_error ("Failed to link rtpbin to rtcpsink"); + 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"); + if (gst_pad_link (srcpad, sinkpad) != GST_PAD_LINK_OK) + g_error ("Failed to link rtcpsrc to rtpbin"); + gst_object_unref (srcpad); + + /* set the pipeline to playing */ + g_print ("starting sender pipeline\n"); + gst_element_set_state (pipeline, GST_STATE_PLAYING); + + /* print stats every second */ + g_timeout_add (1000, (GSourceFunc) print_stats, rtpbin); + + /* 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; +} diff --git a/tests/examples/rtp/server-alsasrc-PCMA.sh b/tests/examples/rtp/server-alsasrc-PCMA.sh new file mode 100755 index 0000000..5340434 --- /dev/null +++ b/tests/examples/rtp/server-alsasrc-PCMA.sh @@ -0,0 +1,35 @@ +#!/bin/sh +# +# A simple RTP server +# sends the output of autoaudiosrc 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 +# +# .--------. .-------. .-------. .----------. .-------. +# |audiosrc| |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 +DEST=127.0.0.1 + +#AELEM=autoaudiosrc +AELEM=audiotestsrc + +# PCMA encode from an the source +ASOURCE="$AELEM ! audioconvert" +AENC="alawenc ! rtppcmapay" + +gst-launch -v gstrtpbin name=rtpbin \ + $ASOURCE ! $AENC ! rtpbin.send_rtp_sink_0 \ + rtpbin.send_rtp_src_0 ! udpsink port=5002 host=$DEST \ + rtpbin.send_rtcp_src_0 ! udpsink port=5003 host=$DEST sync=false async=false \ + udpsrc port=5007 ! rtpbin.recv_rtcp_sink_0 diff --git a/tests/examples/rtp/server-v4l2-H263p-alsasrc-AMR.sh b/tests/examples/rtp/server-v4l2-H263p-alsasrc-AMR.sh new file mode 100755 index 0000000..d8c13a7 --- /dev/null +++ b/tests/examples/rtp/server-v4l2-H263p-alsasrc-AMR.sh @@ -0,0 +1,25 @@ +#!/bin/sh +# +# A simple RTP server +# + +# change these to change the server sync. This causes the server to send the +# packets largly out-of-sync, the client should use the RTCP SR packets to +# restore proper lip-sync between the streams. +AOFFSET=0 +VOFFSET=0 + +VCAPS="video/x-raw-yuv,width=352,height=288,framerate=15/1" + +#DEST=192.168.1.126 +DEST=localhost + +gst-launch -v gstrtpbin name=rtpbin \ + v4l2src ! videorate ! ffmpegcolorspace ! $VCAPS ! ffenc_h263p ! rtph263ppay ! rtpbin.send_rtp_sink_0 \ + rtpbin.send_rtp_src_0 ! queue ! udpsink host=$DEST port=5000 ts-offset=$AOFFSET \ + rtpbin.send_rtcp_src_0 ! udpsink host=$DEST port=5001 sync=false async=false \ + udpsrc port=5005 ! rtpbin.recv_rtcp_sink_0 \ + autoaudiosrc ! audioconvert ! amrnbenc ! rtpamrpay ! rtpbin.send_rtp_sink_1 \ + rtpbin.send_rtp_src_1 ! queue ! udpsink host=$DEST port=5002 ts-offset=$VOFFSET \ + rtpbin.send_rtcp_src_1 ! udpsink host=$DEST port=5003 sync=false async=false \ + udpsrc port=5007 ! rtpbin.recv_rtcp_sink_1 diff --git a/tests/examples/rtp/server-v4l2-H264-alsasrc-PCMA.sh b/tests/examples/rtp/server-v4l2-H264-alsasrc-PCMA.sh new file mode 100755 index 0000000..8dc4d70 --- /dev/null +++ b/tests/examples/rtp/server-v4l2-H264-alsasrc-PCMA.sh @@ -0,0 +1,78 @@ +#!/bin/sh +# +# A simple RTP server +# sends the output of v4l2src as h264 encoded RTP on port 5000, RTCP is sent on +# port 5001. The destination is 127.0.0.1. +# the video receiver RTCP reports are received on port 5005 +# sends the output of autoaudiosrc 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 +# +# .-------. .-------. .-------. .----------. .-------. +# |v4lssrc| |h264enc| |h264pay| | rtpbin | |udpsink| RTP +# | src->sink src->sink src->send_rtp send_rtp->sink | port=5000 +# '-------' '-------' '-------' | | '-------' +# | | +# | | .-------. +# | | |udpsink| RTCP +# | send_rtcp->sink | port=5001 +# .-------. | | '-------' sync=false +# RTCP |udpsrc | | | async=false +# port=5005 | src->recv_rtcp | +# '-------' | | +# | | +# .--------. .-------. .-------. | | .-------. +# |audiosrc| |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 | +# '-------' '----------' +# +# ideally we should transport the properties on the RTP udpsink pads to the +# receiver in order to transmit the SPS and PPS earlier. + +# change this to send the RTP data and RTCP to another host +DEST=127.0.0.1 + +# tuning parameters to make the sender send the streams out of sync. Can be used +# ot test the client RTCP synchronisation. +#VOFFSET=900000000 +VOFFSET=0 +AOFFSET=0 + +# H264 encode from the source +VELEM="v4l2src" +#VELEM="videotestsrc is-live=1" +VCAPS="video/x-raw-yuv,width=352,height=288,framerate=15/1" +VSOURCE="$VELEM ! queue ! videorate ! ffmpegcolorspace ! $VCAPS" +VENC="x264enc tune=zerolatency byte-stream=true bitrate=300 ! rtph264pay" + +VRTPSINK="udpsink port=5000 host=$DEST ts-offset=$VOFFSET name=vrtpsink" +VRTCPSINK="udpsink port=5001 host=$DEST sync=false async=false name=vrtcpsink" +VRTCPSRC="udpsrc port=5005 name=vrtpsrc" + +# PCMA encode from the source +AELEM="autoaudiosrc" +#AELEM="audiotestsrc is-live=1" +ASOURCE="$AELEM ! queue ! audioresample ! audioconvert" +AENC="alawenc ! rtppcmapay" + +ARTPSINK="udpsink port=5002 host=$DEST ts-offset=$AOFFSET name=artpsink" +ARTCPSINK="udpsink port=5003 host=$DEST sync=false async=false name=artcpsink" +ARTCPSRC="udpsrc port=5007 name=artpsrc" + +gst-launch -v gstrtpbin name=rtpbin \ + $VSOURCE ! $VENC ! rtpbin.send_rtp_sink_0 \ + rtpbin.send_rtp_src_0 ! $VRTPSINK \ + rtpbin.send_rtcp_src_0 ! $VRTCPSINK \ + $VRTCPSRC ! rtpbin.recv_rtcp_sink_0 \ + $ASOURCE ! $AENC ! rtpbin.send_rtp_sink_1 \ + rtpbin.send_rtp_src_1 ! $ARTPSINK \ + rtpbin.send_rtcp_src_1 ! $ARTCPSINK \ + $ARTCPSRC ! rtpbin.recv_rtcp_sink_1 |