giorgino a écrit :
> <pierre.crepieux <at> orange-ftgroup.com> writes:
>
> Dear Pierre could you give me some suggestion about your working example with
> h.263 altough like you I'm interested to work with h.264.
>
> G.
>
Here it is. It's really a simple copy-paste of the sample provided with
the source distribution of gstreamer. I only mimic the first RTP sink to
add the second (also added a preview). I didn't tried again with h264.
Maybe there are a few more constraints this simple code doesn't handle ...
I'll tell you if I succeed.
>
>
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/*
#
# A simple RTP server
#
# .-------. .-------. .-------. .----------. .-------.
# |dshowsrc| |h263enc| |h263pay| | 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 |
# '-------' '----------'
*/
#include <string.h>
#include <math.h>
#include <gst/gst.h>
/* change this to send the RTP data and RTCP to another host */
#define DEST_HOST "127.0.0.1"
/* #define AUDIO_SRC "alsasrc" */
#define VIDEO_SRC "dshowvideosrc"
/* the encoder and payloader elements */
#define VIDEO_ENC "ffenc_h263p"
#define VIDEO_PAY "rtph263ppay"
/* #define AUDIO_SRC "alsasrc" */
#define AUDIO_SRC "directsoundsrc"
/* the encoder and payloader elements */
#define AUDIO_ENC "alawenc"
#define AUDIO_PAY "rtppcmapay"
int
main (int argc, char *argv[])
{
GstElement *pipeline;
GMainLoop *loop;
GstElement *videosrc, *capsfilter,*videoqueue,*encqueue,*videorate,*videoconv, *videotee,*previewconv, *preview,*videoenc, *videopay, *videosendtee;
GstElement *rtpbin, *v_rtpsink, *v_rtcpsink, *v_rtcpsrc;
GstElement *audiosrc, *audioqueue,*audioconv, *audiores, *audioenc, *audiopay;
GstElement *a_rtpsink, *a_rtcpsink, *a_rtcpsrc;
gboolean res;
GstPadLinkReturn lres;
GstPad *srcpad, *sinkpad;
GstCaps *caps;
/* 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 */
videosrc = gst_element_factory_make (VIDEO_SRC, "videosrc");
g_assert (videosrc);
videoqueue = gst_element_factory_make ("queue", "videoqueue");
g_assert (videoqueue);
encqueue = gst_element_factory_make ("queue", "encqueue");
g_assert (encqueue);
capsfilter = gst_element_factory_make ("capsfilter", "capsfilter");
g_assert (capsfilter);
videorate = gst_element_factory_make ("videorate", "videorate");
g_assert (videoqueue);
videoconv = gst_element_factory_make ("ffmpegcolorspace", "videoconv");
g_assert (videoconv);
previewconv = gst_element_factory_make ("ffmpegcolorspace", "previewconv");
g_assert (previewconv);
videotee = gst_element_factory_make ("tee", "videotee");
g_assert (videotee);
preview = gst_element_factory_make ("autovideosink", "preview");
g_assert (preview);
/* the encoding and payloading */
videoenc = gst_element_factory_make (VIDEO_ENC, "videoenc");
g_assert (videoenc);
videopay = gst_element_factory_make (VIDEO_PAY, "videopay");
g_assert (videopay);
videosendtee = gst_element_factory_make ("tee", "videosendtee");
g_assert (videosendtee);
/* add capture and payloading to the pipeline and link */
gst_bin_add_many (GST_BIN (pipeline), videosrc, videoqueue,encqueue,capsfilter,videorate,videoconv, videotee,previewconv,preview,videoenc, videopay,videosendtee, NULL);
caps=gst_caps_from_string("video/x-raw-yuv , width=352, "
"height=288 , framerate=15/1");
g_object_set (capsfilter, "caps", caps, NULL);
res = gst_element_link_many (videosrc,capsfilter,videorate,videoconv, videotee,previewconv, preview,NULL);
g_assert (res == TRUE);
srcpad = gst_element_get_request_pad (videotee, "src%d");
sinkpad = gst_element_get_static_pad (encqueue, "sink");
lres = gst_pad_link (srcpad, sinkpad);
g_assert (lres == GST_PAD_LINK_OK);
gst_object_unref (srcpad);
res = gst_element_link_many (encqueue,videoenc,videopay, videosendtee,NULL);
g_assert (res == TRUE);
/* the audio capture and format conversion */
audiosrc = gst_element_factory_make (AUDIO_SRC, "audiosrc");
g_assert (audiosrc);
audioqueue = gst_element_factory_make ("queue", "audioqueue");
g_assert (audioqueue);
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, audioqueue,audioconv, audiores,
audioenc, audiopay, NULL);
res = gst_element_link_many (audiosrc, audioqueue,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 */
v_rtpsink = gst_element_factory_make ("udpsink", "rtpsink");
g_assert (v_rtpsink);
g_object_set (v_rtpsink, "port", 5000, "host", DEST_HOST, NULL);
v_rtcpsink = gst_element_factory_make ("udpsink", "rtcpsink");
g_assert (v_rtcpsink);
g_object_set (v_rtcpsink, "port", 5001, "host", DEST_HOST, NULL);
/* no need for synchronisation or preroll on the RTCP sink */
g_object_set (v_rtcpsink, "async", FALSE, "sync", FALSE, NULL);
v_rtcpsrc = gst_element_factory_make ("udpsrc", "rtcpsrc");
g_assert (v_rtcpsrc);
g_object_set (v_rtcpsrc, "port", 5007, NULL);
gst_bin_add_many (GST_BIN (pipeline), v_rtpsink, v_rtcpsink, v_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_request_pad (videosendtee, "src%d");
//srcpad = gst_element_get_static_pad (videopay, "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 (v_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 (v_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 (v_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);
/* the udp sinks and source we will use for RTP and RTCP */
a_rtpsink = gst_element_factory_make ("udpsink", "a_rtpsink");
g_assert (a_rtpsink);
g_object_set (a_rtpsink, "port", 5002, "host", DEST_HOST, NULL);
a_rtcpsink = gst_element_factory_make ("udpsink", "a_rtcpsink");
g_assert (a_rtcpsink);
g_object_set (a_rtcpsink, "port", 5003, "host", DEST_HOST, NULL);
/* no need for synchronisation or preroll on the RTCP sink */
g_object_set (a_rtcpsink, "async", FALSE, "sync", FALSE, NULL);
a_rtcpsrc = gst_element_factory_make ("udpsrc", "a_rtcpsrc");
g_assert (a_rtcpsrc);
g_object_set (a_rtcpsrc, "port", 5009, NULL);
gst_bin_add_many (GST_BIN (pipeline), a_rtpsink, a_rtcpsink, a_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_1");
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_1");
sinkpad = gst_element_get_static_pad (a_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_1");
sinkpad = gst_element_get_static_pad (a_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 (a_rtcpsrc, "src");
sinkpad = gst_element_get_request_pad (rtpbin, "recv_rtcp_sink_1");
lres = gst_pad_link (srcpad, sinkpad);
g_assert (lres == GST_PAD_LINK_OK);
gst_object_unref (srcpad);
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);
gst_element_set_state (pipeline, GST_STATE_NULL);
return 0;
}
------------------------------------------------------------------------------
Protect Your Site and Customers from Malware Attacks
Learn about various malware tactics and how to avoid them. Understand
malware threats, the impact they can have on your business, and how you
can protect your company and customers by using code signing.
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