[gst-devel] problem of sending out vedio througt rtp protocol.
Wim Taymans
wim.taymans at gmail.com
Sun May 31 11:26:27 CEST 2009
On Sun, 2009-05-31 at 16:20 +0800, xuxin04072129 wrote:
Hi,
You connect to the pad-added signal twice. Just connect only once and in
the callback, see which udpsink you need to link.
Wim
>
> hi all
> Hi, I was trying to transfer video and audio using network
>
> sender------->server---------------->receiver_1
>
> when i use the gst-launch tool to test my commends,it succeede. But
> when i wrote the "server" in c language and run the project again , i
> got this
>
> Error: internal data flow error.
>
> on "server".the flowing is my commends and source code of server.
> Please help me,thank you very much
>
> sender:
> gst-launch -v gstrtpbin name=rtpbin \
> filesrc location=filesrc location=/home/xuxin/desktop/g_p/a.avi !
> decodebin name=dec \
> dec. ! queue ! x264enc byte-stream=false ! rtph264pay !
> rtpbin.send_rtp_sink_0 \
> rtpbin.send_rtp_src_0 ! udpsink port=5000 host=172.21.29.177
> name=vrtpsink \
> dec. ! queue ! audioresample ! audioconvert ! alawenc ! rtppcmapay !
> rtpbin.send_rtp_sink_1 \
> rtpbin.send_rtp_src_1 ! udpsink port=5002 host=172.21.29.177
> ts-offset=0 name=artpsink
>
> Server( ip:172.21.29.177)
> 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=224.0.0.1 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=224.0.0.1 sync=false ts-offset=0
>
> receiver (in multigroup)
>
> gst-launch -v gstrtpbin name=rtpbin latency=200 \
> udpsrc multigroup="224.0.0.1"
> caps="application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H264" port=5000 ! rtpbin.recv_rtp_sink_0 \
> rtpbin. ! rtph264depay ! decodebin ! xvimagesink \
> udpsrc multigroup="224.0.0.1"
> caps="application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)PCMA" port=5002 ! rtpbin.recv_rtp_sink_1 \
> rtpbin. ! rtppcmadepay ! decodebin ! audioconvert ! audioresample !
> alsasink
>
> then I write "Server" in C language ,the code is showed below
>
> #include <gst/gst.h>
> #include <glib.h>
> #include <unistd.h>
> #include <stdlib.h>
>
> 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(GstElement *element, GstPad *pad, gpointer
> data)
> {
> GstPad *sinkpad;
> GstElement *udpsink = (GstElement *)data;
>
> g_print("Dynamic pad created, linking demuxer/decoder\n");
> sinkpad = gst_element_get_static_pad(udpsink, "sink");
> gst_pad_link(pad, sinkpad);
> gst_object_unref(sinkpad);
> }
>
> int main(int argc, char **argv)
> {
> GMainLoop *loop;
> GstBus *bus;
> GstPad *pad;
> GstCaps *videocap, *audiocap;
> GstElement *pipeline, *gstrtpbin, *udpsrc1, *udpsrc2,
> *udpsink1, *udpsink2;
>
> gst_init(&argc, &argv);
> loop = g_main_loop_new(NULL, FALSE);
>
> pipeline = gst_pipeline_new("server");
> gstrtpbin = gst_element_factory_make("gstrtpbin", "gst_rtpbin");
> udpsrc1 = gst_element_factory_make("udpsrc", "udpsrc1");
> udpsrc2 = gst_element_factory_make("udpsrc", "udpsrc2");
> udpsink1 = gst_element_factory_make("udpsink", "udpsink1");
> udpsink2 = gst_element_factory_make("udpsink", "udpsink2");
>
> bus = gst_pipeline_get_bus(GST_PIPELINE(pipeline));
> gst_bus_add_watch(bus, bus_call, loop);
> gst_object_unref(bus);
>
> videocap = gst_caps_new_simple("application/x-rtp",
> "media", G_TYPE_STRING, "video",
> "clock-rate", G_TYPE_LONG, 90000,
> "encoding-name", G_TYPE_STRING, "H264", NULL);
>
> audiocap = gst_caps_new_simple("application/x-rtp",
> "media", G_TYPE_STRING, "audio",
> "clock-rate", G_TYPE_LONG, 8000,
> "encoding-name", G_TYPE_STRING, "PCMA", NULL);
>
> g_object_set(G_OBJECT(udpsrc1), "caps", videocap, NULL);
> g_object_set(G_OBJECT(udpsrc2), "caps", audiocap, NULL);
> g_object_set(G_OBJECT(udpsrc1), "port", 5000, NULL);
> g_object_set(G_OBJECT(udpsrc2), "port", 5002, NULL);
> g_object_set(G_OBJECT(udpsink1), "port", 5000, NULL);
> g_object_set(G_OBJECT(udpsink2), "port", 5002, NULL);
> g_object_set(G_OBJECT(udpsink1), "host", "172.21.29.177", NULL);
> g_object_set(G_OBJECT(udpsink2), "host", "172.21.29.177", NULL);
>
> gst_caps_unref(videocap);
> gst_caps_unref(audiocap);
>
> gst_bin_add_many(GST_BIN(pipeline), udpsrc1, udpsrc2, gstrtpbin,
> udpsink1, udpsink2, NULL);
>
> pad = gst_element_get_request_pad(gstrtpbin, "recv_rtp_sink_0");
> gst_pad_link(gst_element_get_pad(udpsrc1, "src"), pad);
>
> pad = gst_element_get_request_pad(gstrtpbin, "recv_rtp_sink_1");
> gst_pad_link(gst_element_get_pad(udpsrc2, "src"), pad);
>
> g_signal_connect(gstrtpbin, "pad-added", G_CALLBACK(on_pad_added),
> udpsink1);
> g_signal_connect(gstrtpbin, "pad_added", G_CALLBACK(on_pad_added),
> udpsink2);
>
> 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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