[gst-devel] how to implement a Server to forward rtp packet
xuxin04072129
xuxin04072129 at 126.com
Sun Jun 7 14:39:23 CEST 2009
hi all :
First, thanks to all who helped to solve my problem. Now the "Sender" works fine. I've attached my sample-application for Sender . Now I want to implement a "Server" ,which receive udp packets from "Sender" and forward these packets to Receiver. I have used the following commends to build a Server.But when I send the video again, I should restart the Server. I want the Server to work all the time, and i don't have to restart it each time I send the video.
Any suggestions?
regards
xuxin
Server commends:
gst-launch -v gstrtpbin name=rtpbin latency=200 \
udpsrc caps="application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H264" port=5000 ! rtpbin.recv_rtp_sink_0 \
rtpbin. ! udpsink port=5000 host=172.21.29.177 sync=false ts-offset=0 \
udpsrc caps="application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)PCMA" port=5002 ! rtpbin.recv_rtp_sink_1 \
rtpbin. ! udpsink port=5002 host=172.21.29.177 sync=false ts-offset=0
###########################
Sender:
#include <gst/gst.h>
#include <glib.h>
#include <unistd.h>
#include <stdlib.h>
#define SOURCE "/home/xuxin/desktop/g_p/test/a.avi"
#define DEST_HOST "172.21.29.169"
#define VIDEO_PORT 5000
#define AUDIO_PORT 5002
// global declaration
GstElement *pipeline,*udpsink0,*udpsink1;
GstElement *audio_bin,*video_bin;
//
gboolean res;//element link
GstPadLinkReturn lres;//pad link
//bus
static gboolean
bus_call (GstBus *bus,
GstMessage *msg,
gpointer data)
{
GMainLoop *loop = (GMainLoop *) data;
switch (GST_MESSAGE_TYPE (msg)) {
case GST_MESSAGE_EOS:
g_print ("End of stream\n");
g_main_loop_quit (loop);
break;
case GST_MESSAGE_ERROR: {
gchar *debug;
GError *error;
gst_message_parse_error (msg, &error, &debug);
g_free (debug);
g_printerr ("Error: %s\n", error->message);
g_error_free (error);
g_main_loop_quit (loop);
break;
}
default:
break;
}
return TRUE;
}
static void on_pad_added_rtpbin(GstElement *element, GstPad *pad, gpointer data)
{
GstCaps *caps=NULL;
GstStructure *str=NULL;
GstPad *t_pad=NULL;
const gchar* media_type=NULL;
gint clock_rate;
gboolean judgment;
const gchar* encoding_name=NULL;
const gchar* name=NULL;
g_print("**************enter into rtpbin callback ****************\n");
g_assert(pad!=NULL);
caps=gst_pad_get_caps(pad);
g_assert(caps!=NULL);
str=gst_caps_get_structure(caps,0);
g_assert(str!=NULL);
media_type=gst_structure_get_string(str,"media");
g_assert(media_type!=NULL);
g_print("the media type is %s\n",media_type);
//**************test*****************************
name=gst_structure_get_name(str);
g_print("the caps is %s\n",name);
judgment=gst_structure_get_int(str,"clock-rate",&clock_rate);
g_print("clock_rate is :%d\n",clock_rate);
encoding_name=gst_structure_get_string(str,"encoding-name");
g_assert(encoding_name!=NULL);
g_print("the encoding_name is %s\n",encoding_name);
//***********************************************
//media type audio or video
if (g_strrstr (media_type, "video"))
{
g_print("rtp catch audio\n");
t_pad = gst_element_get_static_pad (udpsink0, "sink");
g_assert(t_pad!=NULL);
g_print("get the pad of udpsink0\n");
lres=gst_pad_link (pad, t_pad);
g_assert (lres == GST_PAD_LINK_OK);
g_print("Dynamic pad created, linking rtpbin/udp1\n");
}
else if(g_strrstr (media_type, "audio"))
{
g_print("rtp catch video\n");
t_pad = gst_element_get_static_pad (udpsink1, "sink");
g_assert(t_pad!=NULL);
g_print("get the pad of udpsink1\n");
lres=gst_pad_link (pad, t_pad);
g_assert (lres == GST_PAD_LINK_OK);
g_print("Dynamic pad created, linking rtpbin/udp0\n");
}
else
{
gst_caps_unref (caps);
return;
}
}
static void on_pad_added_decodebin(GstElement *decodebin,GstPad *pad,gboolean last,gpointer data)
{
GstCaps *caps=NULL;
GstStructure *str=NULL;
GstPad *t_pad=NULL;
const gchar* media_type=NULL;
g_print("**************** enter into the decodebin signal callback **********\n");
g_assert(pad!=NULL);
caps = gst_pad_get_caps (pad);
g_assert(caps!=NULL);
str = gst_caps_get_structure (caps, 0);
g_assert(str!=NULL);
media_type=gst_structure_get_name (str);
g_assert(media_type!=NULL);
g_print("the media type is %s\n",media_type);
//check media type audio or video
if (g_strrstr (media_type, "audio"))
{
g_print("catch audio in decodebin\n");
t_pad = gst_element_get_static_pad (audio_bin, "sink");
g_assert(t_pad!=NULL);
lres=gst_pad_link (pad, t_pad);
g_assert (lres == GST_PAD_LINK_OK);
g_print("Dynamic pad created, linking decoderbin/queue2\n");
}
else if(g_strrstr (media_type, "video"))
{
g_print("catch video in decodebin\n");
t_pad = gst_element_get_static_pad (video_bin, "sink");
g_assert(t_pad!=NULL);
lres=gst_pad_link (pad, t_pad);
g_assert (lres == GST_PAD_LINK_OK);
g_print("Dynamic pad created, linking decoderbin/queue1\n");
}
else
{
g_print("have not get suitable type\n");
gst_caps_unref (caps);
return;
}
}
int main(int argc, char **argv)
{
GMainLoop *loop;
GstBus *bus;
//pad of rtpbin
GstPad *sink_pad;
GstPad *bin_pad;
GstPad *videopad_src,*videopad_sink;
GstPad *audiopad_src,*audiopad_sink;
//declare elements
GstElement *filesrc,*decodebin,*gstrtpbin;
//element for video
GstElement *queue1,*x264enc,*rtph264pay;
//element for audio
GstElement *queue2,*audioresample,*audioconvert,*alawenc,*rtppcmapay;
gst_init(&argc, &argv);
loop = g_main_loop_new(NULL, FALSE);
//generate elements
pipeline = gst_pipeline_new("server");
filesrc=gst_element_factory_make("filesrc", "filesrc");
decodebin=gst_element_factory_make("decodebin", "decodebin");
gstrtpbin = gst_element_factory_make("gstrtpbin", "gstrtpbin");
//for video
queue1=gst_element_factory_make("queue", "queue1");
x264enc=gst_element_factory_make("x264enc", "x264enc");
rtph264pay=gst_element_factory_make("rtph264pay", "rtph264pay");
udpsink0=gst_element_factory_make("udpsink", "udpsink0");
//for audio
queue2 = gst_element_factory_make("queue", "queue2");
audioresample = gst_element_factory_make("audioresample", "audioresample");
audioconvert = gst_element_factory_make("audioconvert", "audioconvert");
alawenc = gst_element_factory_make("alawenc", "alawenc");
rtppcmapay = gst_element_factory_make("rtppcmapay", "rtppcmapay");
udpsink1=gst_element_factory_make("udpsink", "udpsink1");
bus = gst_pipeline_get_bus(GST_PIPELINE(pipeline));
gst_bus_add_watch(bus, bus_call, loop);
gst_object_unref(bus);
//set properties of elements
g_object_set(G_OBJECT(filesrc), "location",SOURCE, NULL);
g_object_set(G_OBJECT(udpsink0), "host",DEST_HOST,"port",VIDEO_PORT, NULL);
g_object_set(G_OBJECT(udpsink1), "host",DEST_HOST,"port",AUDIO_PORT, NULL);
//g_object_set(G_OBJECT(udpsink0), "sync",FALSE,NULL);
//g_object_set(G_OBJECT(udpsink1), "sync",FALSE,NULL);
//g_object_set(G_OBJECT(udpsink0), "async", FALSE,NULL);
//g_object_set(G_OBJECT(udpsink1), "async", FALSE,NULL);
//create video_bin
video_bin = gst_bin_new ("videobin");
gst_bin_add_many (GST_BIN (video_bin),queue1,x264enc,rtph264pay,NULL);
res=gst_element_link_many(queue1,x264enc,rtph264pay,NULL);
g_assert (res == TRUE);
//add pad to video_bin
videopad_sink = gst_element_get_static_pad (queue1, "sink");
videopad_src= gst_element_get_static_pad (rtph264pay, "src");
gst_element_add_pad (video_bin,gst_ghost_pad_new ("sink", videopad_sink));
gst_element_add_pad (video_bin,gst_ghost_pad_new ("src",videopad_src));
gst_object_unref (videopad_sink);
gst_object_unref (videopad_src);
//create audio_bin
audio_bin = gst_bin_new ("audiobin");
gst_bin_add_many (GST_BIN (audio_bin),queue2,audioresample,audioconvert,alawenc,rtppcmapay,NULL);
res=gst_element_link_many(queue2,audioresample,audioconvert,alawenc,rtppcmapay,NULL);
g_assert (res == TRUE);
//add pad to audio_bin
audiopad_sink = gst_element_get_static_pad (queue2, "sink");
audiopad_src= gst_element_get_static_pad (rtppcmapay, "src");
res=gst_element_add_pad (audio_bin,gst_ghost_pad_new ("sink", audiopad_sink));
g_assert (res == TRUE);
res=gst_element_add_pad (audio_bin,gst_ghost_pad_new ("src", audiopad_src));
g_assert (res == TRUE);
gst_object_unref (audiopad_sink);
gst_object_unref (audiopad_src);
//add elements into pipeline
gst_bin_add_many(GST_BIN(pipeline),
filesrc,decodebin,audio_bin,video_bin,gstrtpbin,udpsink0,udpsink1,NULL);
//static link filesrc and decoderbin
res=gst_element_link(filesrc,decodebin);
g_assert (res == TRUE);
//get request pad from gstrtpbin and connect with video_bin
bin_pad=gst_element_get_pad(video_bin, "src");
g_assert(bin_pad!=NULL);
sink_pad = gst_element_get_request_pad(gstrtpbin, "recv_rtp_sink_0");
g_assert(sink_pad!=NULL);
lres=gst_pad_link(bin_pad, sink_pad);
g_assert (lres == GST_PAD_LINK_OK);
//get request pad from gstrtpbin and connect with audio_bin
bin_pad=gst_element_get_pad(audio_bin, "src");
g_assert(bin_pad!=NULL);
sink_pad = gst_element_get_request_pad(gstrtpbin, "recv_rtp_sink_1");
g_assert(sink_pad!=NULL);
lres=gst_pad_link(bin_pad, sink_pad);
g_assert (lres == GST_PAD_LINK_OK);
//signal link
g_signal_connect(decodebin, "new-decoded-pad", G_CALLBACK(on_pad_added_decodebin),NULL);
g_signal_connect(gstrtpbin, "pad-added", G_CALLBACK(on_pad_added_rtpbin),NULL);
gst_element_set_state(pipeline, GST_STATE_PLAYING);
g_print("Running...\n");
g_main_loop_run(loop);
/* Out of the main loop, clean up nicely */
g_print("Returned, stopping playback\n");
gst_element_set_state(pipeline, GST_STATE_NULL);
g_print("Deleting pipeline\n");
gst_object_unref(GST_OBJECT(pipeline));
return 0;
}
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