- Jun 07, 2015
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Sebastian Dröge authored
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- Jun 03, 2015
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When calling gst_rtsp_watch_write_data in gstrtspconnection.c and backlog is empty it can happen that just a part of a message will be sent and rest is in backlog queue. If then flush during teardown just a part of message will be sent.This can lead to client miss teardown response since it expect to get the last part of message. The flushing during teardown was introduced to fix a deadlock that now is fixed more generally in handle_request by temporary setting backlog size to unlimited. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749845
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- May 27, 2015
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Tim-Philipp Müller authored
Needed by the new automake test runner and the current version of the common submodule.
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- May 20, 2015
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Sebastian Dröge authored
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- May 05, 2015
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Sebastian Dröge authored
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Sebastian Dröge authored
Otherwise we're going to lose a few packets for live streams during DESCRIBE.
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- May 04, 2015
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Sebastian Dröge authored
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Sebastian Dröge authored
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- Apr 27, 2015
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clock-rate in last-sample's caps is integer, not unsigned. To get this value properly, variable needs to be type-casted to int. https://bugzilla.gnome.org/show_bug.cgi?id=747614
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- Apr 26, 2015
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Tim-Philipp Müller authored
Not just because there happens to be a po directory. https://bugzilla.gnome.org/show_bug.cgi?id=748058
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Tim-Philipp Müller authored
This reverts commit 1545d8fe. We don't need a gettext setup here and there's no po directory either, so no reason why autopoint would be run in the first place. See https://bugzilla.gnome.org/show_bug.cgi?id=748058
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- Apr 23, 2015
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Tim-Philipp Müller authored
Make sure the test environment is set up. https://bugzilla.gnome.org//show_bug.cgi?id=747624
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Tim-Philipp Müller authored
This is only required for builds from git, people can still build tarballs if they only have older autotools. https://bugzilla.gnome.org//show_bug.cgi?id=747624
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- Apr 20, 2015
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Vincent Penquerc'h authored
Fixes autogen.sh. It would run autopoint, which would complain that it could not find the gettext version in configure.ac. https://bugzilla.gnome.org/show_bug.cgi?id=748058
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- Apr 15, 2015
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Setting wrong payload type causes failure to do retransmission through audio stream https://bugzilla.gnome.org/show_bug.cgi?id=747839
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Because of duplicated g_signal_connect for request-aux-sender signal, wrong stream pointer is passed to the signal handler. Instead of passing each stream, pass stream array and get the relevant stream. https://bugzilla.gnome.org/show_bug.cgi?id=747839
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- Apr 06, 2015
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Tim-Philipp Müller authored
Fixes build after aclocal_check etc. helpers have been removed.
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- Apr 03, 2015
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Tim-Philipp Müller authored
From bc76a8b to c8fb372
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- Mar 23, 2015
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Sebastian Dröge authored
We only need them to be able to pre-roll, queueing up more data here is only going to harm latency and memory usage.
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Sebastian Dröge authored
We have a queue in front of the udpsink too to prevent the pipeline from locking up.
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- Mar 21, 2015
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Nicolas Dufresne authored
Instead we where returning first GstBuffer timestamp. This would result in clock skew and unwanted behaviour in RTSP playback. https://bugzilla.gnome.org/show_bug.cgi?id=746479
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- Mar 18, 2015
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Nicolas Dufresne authored
If the seq is NULL, the RTP buffer was left mapped. We should always unmap the buffer.
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- Mar 15, 2015
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Sebastian Dröge authored
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- Mar 10, 2015
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Tim-Philipp Müller authored
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- Mar 09, 2015
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Sebastian Dröge authored
This gives more accurate values than asking the payloader. There might be queueing happening between the payloader and the sink. https://bugzilla.gnome.org/show_bug.cgi?id=745704
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Sebastian Dröge authored
When the sdp media attribute framesize are converted to caps the <payload> should not be included. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725335 Based on the patch for rtspsrc by Linus Svensson <linussn@axis.com>
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The sdp framesize attribute is desribed in RFC6064. It is specified for payloading of H263 and has the following form a=framesize:<payload type> <width>-<height>. The <width>-<height> part should be added to the caps in a payloader and the <payload type> should be added by the rtsp-server. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725334
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- Mar 03, 2015
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Luis de Bethencourt authored
Insignificant but this keeps Coverity happy. CID #1268404
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Jan Schmidt authored
An example server and client that works for synchronising live streams only - as it can't support pause/play.
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Jan Schmidt authored
Allow specifying the GType of a GstRtspMedia subclass to create as a simpler way to get the factory to create a custom GstRtspMedia sub-class, without subclassing GstRtspMediaFactory.
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- Mar 02, 2015
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Fixes an abort when calling gst_rtsp_media_get_buffer_size() because of double g_mutex_unlock () usage. https://bugzilla.gnome.org/show_bug.cgi?id=745434
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- Feb 19, 2015
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Changed RTSP session timeout handling to monotonic time and deprecating the API for current system time. This fixes timeouts when the system time changes. https://bugzilla.gnome.org/show_bug.cgi?id=743346
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- Feb 13, 2015
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Sebastian Dröge authored
If the media was just not seekable, we continue from whatever position we are and let the client decide if that is what is wanted or not. Only if the actual seek failed, we can't really recover and should error out.
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- Feb 12, 2015
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Sebastian Dröge authored
Instead error out properly the same way as if the SEEKING query already failed.
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- Feb 11, 2015
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Tim-Philipp Müller authored
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- Feb 10, 2015
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Luis de Bethencourt authored
Fix the logic of gst_rtsp_media_collect_streams() so after looping collecting all streams it knows if it got any, and can check if the transport mode is OK. CID #1268400
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- Feb 09, 2015
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Sebastian Dröge authored
Just print a warning if the one that was set before disagrees with what elements we found. It must already be set to something before as this function is called after we received the SDP from ANNOUNCE in RECORD mode, and we would reject ANNOUNCE if the RECORD flag was not set.
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