Access rtpsession stats property
joakim
joakim.magnusson at spiideo.com
Wed Jan 30 10:57:29 UTC 2019
Hi,
I'm building a pipeline in c with rtspsrc. I want to access the stats
property in the rtpsession object created internally with rtspsrc. Is this
possible? This is my code for now:
#include <gst/gst.h>
#include <glib.h>
typedef struct myDataTag {
GstElement *pipeline;
GstElement *source;
GstElement *depayloader;
GstElement *decoder;
GstElement *sink;
} myData_t;
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 *decoder = (GstElement *) data;
/* We can now link this pad with the vorbis-decoder sink pad */
g_print ("Dynamic pad created, linking demuxer/decoder\n");
sinkpad = gst_element_get_static_pad (decoder, "sink");
gst_pad_link (pad, sinkpad);
gst_object_unref (sinkpad);
}
/* pad added handler */
static void pad_added_handler (GstElement *src, GstPad *new_pad, myData_t
*pThis) {
GstPad *sink_pad = gst_element_get_static_pad (pThis->depayloader,
"sink");
GstPadLinkReturn ret;
GstCaps *new_pad_caps = NULL;
GstStructure *new_pad_struct = NULL;
const gchar *new_pad_type = NULL;
g_print ("Received new pad '%s' from '%s':\n", GST_PAD_NAME (new_pad),
GST_ELEMENT_NAME (src));
/* Check the new pad's name */
if (!g_str_has_prefix (GST_PAD_NAME (new_pad), "recv_rtp_src_")) {
g_print (" It is not the right pad. Need recv_rtp_src_.
Ignoring.\n");
goto exit;
}
/* If our converter is already linked, we have nothing to do here */
if (gst_pad_is_linked (sink_pad)) {
g_print (" Sink pad from %s already linked. Ignoring.\n",
GST_ELEMENT_NAME (src));
goto exit;
}
/* Check the new pad's type */
new_pad_caps = gst_pad_query_caps (new_pad, NULL);
new_pad_struct = gst_caps_get_structure (new_pad_caps, 0);
new_pad_type = gst_structure_get_name (new_pad_struct);
/* Attempt the link */
ret = gst_pad_link (new_pad, sink_pad);
if (GST_PAD_LINK_FAILED (ret)) {
g_print (" Type is '%s' but link failed.\n", new_pad_type);
} else {
g_print (" Link succeeded (type '%s').\n", new_pad_type);
}
exit:
/* Unreference the new pad's caps, if we got them */
if (new_pad_caps != NULL)
gst_caps_unref (new_pad_caps);
/* Unreference the sink pad */
gst_object_unref (sink_pad);
}
int
main (int argc,
char *argv[])
{
argv[0] = "gstreamer-test";
argv[1] = "rtsp://ipadress/axis-media/media.amp";
//argv[2] = "para2";
myData_t appData;
GMainLoop *loop;
GstElement *rtpbin;
GstBus *bus;
guint bus_watch_id;
GstStructure *stats;
/* Initialisation */
gst_init (&argc, &argv);
loop = g_main_loop_new (NULL, FALSE);
/* Check input arguments */
if (argc != 2) {
g_printerr ("Usage: %s <Ogg/Vorbis filename>\n", argv[0]);
return -1;
}
/* Create gstreamer elements */
appData.pipeline = gst_pipeline_new ("pipeline");
appData.source = gst_element_factory_make ("rtspsrc", "rtspsrc");
g_object_set (G_OBJECT (appData.source), "location",
"rtsp://root:spiideo2017@10.0.3.69/axis-media/media.amp", NULL);
appData.depayloader = gst_element_factory_make
("rtph264depay","depayloader");
appData.decoder = gst_element_factory_make ("avdec_h264",
"avh264decoder");
appData.sink = gst_element_factory_make ("autovideosink", "sink");
if (!appData.pipeline || !appData.source || !appData.depayloader ||
!appData.decoder || !appData.sink) {
g_printerr ("One element could not be created. Exiting.\n");
return -1;
}
g_signal_connect (appData.source, "pad-added", G_CALLBACK
(pad_added_handler), &appData);
/* Set up the pipeline */
/* we set the input filename to the source element */
g_object_set (G_OBJECT (appData.source), "location", argv[1], NULL);
/* we add a message handler */
bus = gst_pipeline_get_bus (GST_PIPELINE (appData.pipeline));
bus_watch_id = gst_bus_add_watch (bus, bus_call, loop);
gst_object_unref (bus);
/* we add all elements into the pipeline */
/* file-source | ogg-demuxer | vorbis-decoder | converter | alsa-output
*/
gst_bin_add_many (GST_BIN (appData.pipeline),
appData.source, appData.depayloader, appData.decoder,
appData.sink, NULL);
/* we link the elements together */
/* file-source -> ogg-demuxer ~> vorbis-decoder -> converter ->
alsa-output */
gst_element_link_many (appData.depayloader,
appData.decoder, appData.sink, NULL);
g_signal_connect (appData.source, "pad-added", G_CALLBACK
(pad_added_handler), &appData);
/* Set the pipeline to "playing" state*/
g_print ("Now playing: %s\n", argv[1]);
gst_element_set_state (appData.pipeline, GST_STATE_PLAYING);
/* Iterate */
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 (appData.pipeline, GST_STATE_NULL);
g_print ("Deleting pipeline\n");
gst_object_unref (GST_OBJECT (appData.pipeline));
g_source_remove (bus_watch_id);
g_main_loop_unref (loop);
return 0;
--
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