diff options
Diffstat (limited to 'tests/examples/rtp/server-alsasrc-PCMA.c')
| -rwxr-xr-x | tests/examples/rtp/server-alsasrc-PCMA.c | 168 |
1 files changed, 168 insertions, 0 deletions
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 <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; +} |