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rtspsrc: Remove some dead code

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stop is not used after this point, nor do we create a new segment
here since 84725d62b5

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/940>
This commit is contained in:
Nirbheek Chauhan 2021-04-13 09:23:12 +05:30 committed by GStreamer Marge Bot
parent fb97ca9458
commit c071cbbe30
1 changed files with 0 additions and 9 deletions
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9
gst/rtsp/gstrtspsrc.c
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@ -2861,11 +2861,6 @@ gst_rtspsrc_perform_seek (GstRTSPSrc * src, GstEvent * event)
gst_segment_do_seek (&seeksegment, rate, format, flags,
cur_type, cur, stop_type, stop, &update);
/* figure out the last position we need to play. If it's configured (stop !=
* -1), use that, else we play until the total duration of the file */
if ((stop = seeksegment.stop) == -1)
stop = seeksegment.duration;
/* if we were playing, pause first */
playing = (src->state == GST_RTSP_STATE_PLAYING);
if (playing) {
@ -2897,10 +2892,6 @@ gst_rtspsrc_perform_seek (GstRTSPSrc * src, GstEvent * event)
src->segment.format, src->segment.position));
}
/* now create the newsegment */
GST_DEBUG_OBJECT (src, "Creating newsegment from %" G_GINT64_FORMAT
" to %" G_GINT64_FORMAT, src->segment.position, stop);
/* mark discont when needed */
if (!(rate_change_only && rate_change_same_direction)) {
GST_DEBUG_OBJECT (src, "mark DISCONT, we did a seek to another position");