Limiting a RTSP recording after X seconds
Jesper Taxbøl
jesper at taxboel.dk
Fri Dec 1 11:20:10 UTC 2017
You guys are right. My solution is not safe.
The signal works but the file is messed up in such a way that when
transcoding the file gstreamer often stalls generating much larger files
than the input. When running ffprobe against the file duration is strange.
I was thinking that the problem was that the pipeline did not have time to
close the file properly.
Therefore I found the example from
https://gstreamer.freedesktop.org/documentation/tutorials/basic/time-management.html.I
could insert my configuration and piggyback on the message flow.
*mpegtsmux ! filesink location=two.ts \ rtspsrc
location=rtsp://root:hest1234@192.168.130.200/axis-media/media.amp
<http://root:hest1234@192.168.130.200/axis-media/media.amp> ntp-sync=true
protocols=GST_RTSP_LOWER_TRANS_TCP ! \ queue ! capsfilter
caps=\"application/x-rtp,media=video\" ! \ rtph264depay ! mpegtsmux0. \
rtspsrc location=rtsp://root:hest1234@192.168.130.201/axis-media/media.amp
<http://root:hest1234@192.168.130.201/axis-media/media.amp> ntp-sync=true
protocols=GST_RTSP_LOWER_TRANS_TCP ! \ queue ! capsfilter
caps=\"application/x-rtp,media=video\" ! \ rtph264depay ! mpegtsmux0.*
In the message handler I count timeouts and stop after a given time. Full
source added below.
The approach works as such, but it still gives me funny readings. ffprobe
says:
Invalid return value 0 for stream protocol
Input #0, mpegts, from 'two.ts':
Duration: 00:00:11.95, start: 3600.698056, bitrate: 37229 kb/s
Program 1
Stream #0:0[0x41]: Video: h264 (High) (HDMV / 0x564D4448), yuvj420p(pc,
bt709, progressive), 3840x2160 [SAR 1:1 DAR 16:9], 29.97 fps, 29.97 tbr,
90k tbn, 59.94 tbc
Stream #0:1[0x42]: Video: h264 (High) (HDMV / 0x564D4448), yuvj420p(pc,
bt709, progressive), 3840x2160 [SAR 1:1 DAR 16:9], 29.97 fps, 29.97 tbr,
90k tbn, 59.94 tbc
I notice that duration is somewhat plausible, but start is 3600, which I
guess could be part of the problem.
Any Ideas on how to fix time in my output file.
Kind regards
Jesper
#include <stdio.h>
#include <gst/gst.h>
/* Structure to contain all our information, so we can pass it around */
typedef struct _CustomData {
GstElement *playbin; /* Our one and only element */
gboolean playing; /* Are we in the PLAYING state? */
gboolean terminate; /* Should we terminate execution? */
gboolean seek_enabled; /* Is seeking enabled for this media? */
gboolean seek_done; /* Have we performed the seek already? */
gint64 duration; /* How long does this media last, in nanoseconds */
} CustomData;
/* Forward definition of the message processing function */
static void handle_message (CustomData *data, GstMessage *msg);
int main(int argc, char *argv[]) {
CustomData data;
GstBus *bus;
GstMessage *msg;
GstStateChangeReturn ret;
data.playing = FALSE;
data.terminate = FALSE;
data.seek_enabled = FALSE;
data.seek_done = FALSE;
data.duration = GST_CLOCK_TIME_NONE;
/* Initialize GStreamer */
gst_init (&argc, &argv);
/* Create the elements */
//data.playbin = gst_element_factory_make ("playbin", "playbin");
char* cmd = "mpegtsmux ! filesink location=two.ts \
rtspsrc location=rtsp://root:hest1234@192.168.130.200/axis-media/media.amp
ntp-sync=true protocols=GST_RTSP_LOWER_TRANS_TCP ! \
queue ! capsfilter caps=\"application/x-rtp,media=video\" ! \
rtph264depay ! mpegtsmux0. \
rtspsrc location=rtsp://root:hest1234@192.168.130.201/axis-media/media.amp
ntp-sync=true protocols=GST_RTSP_LOWER_TRANS_TCP ! \
queue ! capsfilter caps=\"application/x-rtp,media=video\" ! \
rtph264depay ! mpegtsmux0.";
printf(cmd);
data.playbin = gst_parse_launch (cmd, NULL);
if (!data.playbin) {
g_printerr ("Not all elements could be created.\n");
return -1;
}
/* Set the URI to play */
//g_object_set (data.playbin, "uri", //"
https://www.freedesktop.org/software/gstreamer-sdk/data/media/sintel_trailer-480p.webm",
NULL);
/* Start playing */
ret = gst_element_set_state (data.playbin, GST_STATE_PLAYING);
if (ret == GST_STATE_CHANGE_FAILURE) {
g_printerr ("Unable to set the pipeline to the playing state.\n");
gst_object_unref (data.playbin);
return -1;
}
/* Listen to the bus */
bus = gst_element_get_bus (data.playbin);
int t = 0;
do {
msg = gst_bus_timed_pop_filtered (bus, 100 * GST_MSECOND,
GST_MESSAGE_STATE_CHANGED | GST_MESSAGE_ERROR | GST_MESSAGE_EOS |
GST_MESSAGE_DURATION);
printf("Tick: \r\n%d\r\n", t++);
if(t == 200)
{
gst_element_send_event (GST_ELEMENT (data.playbin),
gst_event_new_eos());
}
/* Parse message */
if (msg != NULL) {
handle_message (&data, msg);
} else {
/* We got no message, this means the timeout expired */
if (data.playing) {
gint64 current = -1;
/* Query the current position of the stream */
if (!gst_element_query_position (data.playbin, GST_FORMAT_TIME,
¤t)) {
g_printerr ("Could not query current position.\n");
}
/* If we didn't know it yet, query the stream duration */
if (!GST_CLOCK_TIME_IS_VALID (data.duration)) {
if (!gst_element_query_duration (data.playbin, GST_FORMAT_TIME,
&data.duration)) {
g_printerr ("Could not query current duration.\n");
}
}
/* Print current position and total duration */
g_print ("Position %" GST_TIME_FORMAT " / %" GST_TIME_FORMAT "\r",
GST_TIME_ARGS (current), GST_TIME_ARGS (data.duration));
/* If seeking is enabled, we have not done it yet, and the time is
right, seek */
if (data.seek_enabled && !data.seek_done && current > 10 *
GST_SECOND) {
g_print ("\nReached 10s, performing seek...\n");
gst_element_seek_simple (data.playbin, GST_FORMAT_TIME,
GST_SEEK_FLAG_FLUSH | GST_SEEK_FLAG_KEY_UNIT, 30 *
GST_SECOND);
data.seek_done = TRUE;
}
}
}
} while (!data.terminate);
/* Free resources */
gst_object_unref (bus);
gst_element_set_state (data.playbin, GST_STATE_NULL);
gst_object_unref (data.playbin);
return 0;
}
static void handle_message (CustomData *data, GstMessage *msg) {
GError *err;
gchar *debug_info;
switch (GST_MESSAGE_TYPE (msg)) {
case GST_MESSAGE_ERROR:
gst_message_parse_error (msg, &err, &debug_info);
g_printerr ("Error received from element %s: %s\n", GST_OBJECT_NAME
(msg->src), err->message);
g_printerr ("Debugging information: %s\n", debug_info ? debug_info :
"none");
g_clear_error (&err);
g_free (debug_info);
data->terminate = TRUE;
break;
case GST_MESSAGE_EOS:
g_print ("End-Of-Stream reached.\n");
data->terminate = TRUE;
break;
case GST_MESSAGE_DURATION:
/* The duration has changed, mark the current one as invalid */
printf("...\r\n");
data->duration = GST_CLOCK_TIME_NONE;
break;
case GST_MESSAGE_STATE_CHANGED: {
GstState old_state, new_state, pending_state;
gst_message_parse_state_changed (msg, &old_state, &new_state,
&pending_state);
if (GST_MESSAGE_SRC (msg) == GST_OBJECT (data->playbin)) {
g_print ("Pipeline state changed from %s to %s:\n",
gst_element_state_get_name (old_state),
gst_element_state_get_name (new_state));
/* Remember whether we are in the PLAYING state or not */
data->playing = (new_state == GST_STATE_PLAYING);
if (data->playing) {
/* We just moved to PLAYING. Check if seeking is possible */
GstQuery *query;
gint64 start, end;
query = gst_query_new_seeking (GST_FORMAT_TIME);
if (gst_element_query (data->playbin, query)) {
gst_query_parse_seeking (query, NULL, &data->seek_enabled,
&start, &end);
if (data->seek_enabled) {
g_print ("Seeking is ENABLED from %" GST_TIME_FORMAT " to %"
GST_TIME_FORMAT "\n",
GST_TIME_ARGS (start), GST_TIME_ARGS (end));
} else {
g_print ("Seeking is DISABLED for this stream.\n");
}
}
else {
g_printerr ("Seeking query failed.");
}
gst_query_unref (query);
}
}
} break;
default:
/* We should not reach here */
g_printerr ("Unexpected message received.\n");
break;
}
gst_message_unref (msg);
}
2017-11-30 11:40 GMT+01:00 Tim Müller <tim at centricular.com>:
> On Thu, 2017-11-30 at 10:22 +0100, Jesper Taxbøl wrote:
>
> > I spent some time today and figured out i can use use the following
> > syntax in my makefile. Its actually the terminal sending a signal to
> > kill the process.
>
> That'll work of course, but the file won't be finalised properly, so
> the headers won't be updated with the duration and there won't be a
> seektable and you might be missing the last few frames/seconds.
>
> Cheers
> -Tim
>
> --
> Tim Müller, Centricular Ltd - http://www.centricular.com
> _______________________________________________
> gstreamer-devel mailing list
> gstreamer-devel at lists.freedesktop.org
> https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
>
--
Jesper Taxbøl
+45 61627501
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