how to save video to file in mp4 format recieved from udp src
gagankumarnigam
gagankumarnigam at bel.co.in
Fri Jul 17 20:44:40 PDT 2015
Hi all,
I am able to recieve video over udp . but i dont know how to save this video
to file in mp4 format. though i had downloaded an example from net it is
both displaying video and save it to avi format but it created video size in
MBs and length of video is just few seconds. Here is my client code
#define AVINAME "camera.avi"
#define PORT 9996
#define VIDEO_SINK "autovideosink"
/* 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);
}
int main (int argc, char *argv[])
{
GstElement *rtpbin, *rtpsrc, *rtcpsrc, *rtcpsink;
GstElement *videodepay,
*videodec,
*videoqueue,
//*videores,
*videoconv,
*videosink,
*tee,
*aviqueue,
*aviconv,
*avidenc,
*avifmux,
*avifilesink;
GstElement *pipeline;
GMainLoop *loop;
GstCaps *caps;
gboolean res1,res2;
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", PORT, NULL);
/* we need to set caps on the udpsrc for the RTP data */
caps = gst_caps_from_string (VIDEO_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", 9997, NULL);
rtcpsink = gst_element_factory_make ("udpsink", "rtcpsink");
g_assert (rtcpsink);
g_object_set (rtcpsink, "port", 9999, "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 */
videodepay = gst_element_factory_make ("rtpvrawdepay", "videodepay");
g_assert (videodepay);
videoqueue=gst_element_factory_make ("queue","videoqueue");
g_assert(videoqueue);
tee = gst_element_factory_make ("tee","tee");
g_assert(tee);
aviqueue=gst_element_factory_make ("queue","aviqueue");
g_assert(aviqueue);
// videodec = gst_element_factory_make ("ffmpegcolorspace",
"videodec");
// g_assert (videodec);
/* the audio playback and format conversion */
videoconv = gst_element_factory_make ("ffmpegcolorspace",
"videoconv");
g_assert (videoconv);
/*
audiores = gst_element_factory_make ("audioresample", "audiores");
g_assert (audiores);
*/
videosink = gst_element_factory_make (VIDEO_SINK, "videosink");
g_assert (videosink);
aviconv = gst_element_factory_make ("ffmpegcolorspace","avicinv");
g_assert (aviconv);
avidenc = gst_element_factory_make ("jpegenc","avidenc");
g_assert (aviconv);
avifmux = gst_element_factory_make ("avimux","avifmux");
g_assert (avifmux);
avifilesink = gst_element_factory_make ("filesink","avifilesink");
g_assert (avifilesink);
g_object_set(avifilesink,"location",AVINAME,NULL);
/* add depayloading and playback to the pipeline and link */
// gst_bin_add_many (GST_BIN (pipeline), videodepay, videoconv,
/*videores,*/videoqueue, videosink,
aviconv,avidenc,avifmux,avifilename,NULL);
gst_bin_add_many (GST_BIN (pipeline),
videodepay,tee,videoqueue,videoconv,videosink,
aviqueue,aviconv,avidenc,avifmux,avifilesink,NULL);
res1 = gst_element_link_many (videodepay,
tee,videoqueue,videoconv,videosink, NULL);
g_assert (res1 == TRUE);
res2 = gst_element_link_many
(tee,aviqueue,aviconv,avidenc,avifmux,avifilesink,NULL);
g_assert (res2 == TRUE);
/* the rtpbin element */
rtpbin = gst_element_factory_make ("gstrtpbin", "rtpbin");
g_assert (rtpbin);
g_object_set (G_OBJECT (rtpbin),"latency",200,NULL);
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),
videodepay);
/* give some stats when we receive RTCP */
//g_signal_connect (rtpbin, "on-ssrc-active", G_CALLBACK
(on_ssrc_active_cb),videodepay);
/* 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;
}
--
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