- Sep 07, 2020
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Tim-Philipp Müller authored
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- Aug 20, 2020
- Aug 03, 2020
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Jordan Petridіs authored
clang 10 is complaining about incompatible types due to the glib typesystem. ``` ../subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-thread-pool.c:534:10: error: incompatible pointer types passing 'typeof ((((void *)0))) *' (aka 'void **') to parameter of type 'GThreadPool **' (aka 'struct _GThreadPool **') [-Werror,-Wincompatible-pointer-types] ``` Part-of: <gstreamer/gst-rtsp-server!145>
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Jordan Petridіs authored
clang 10 is complaining about incompatible types due to the glib typesystem. ``` ../subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-thread-pool.c:534:10: error: incompatible pointer types passing 'typeof ((((void *)0))) *' (aka 'void **') to parameter of type 'GThreadPool **' (aka 'struct _GThreadPool **') [-Werror,-Wincompatible-pointer-types] ``` Part-of: <gstreamer/gst-rtsp-server!145>
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- Jul 15, 2020
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Srimanta Panda authored
Fixed a resource leak for mikey message while adding crypto session failed. Part-of: <gstreamer/gst-rtsp-server!144>
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- Jul 08, 2020
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Tim-Philipp Müller authored
Part-of: <!143>
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- Jul 06, 2020
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This causes them to send caps events before data flow, which is usually a pretty correct thing to do! Not doing so manifested in a bug where ssrcdemux wouldn't forward the caps it had received with an extra ssrc field, as it hadn't received any caps event. Fixes #85 Part-of: <gstreamer/gst-rtsp-server!141>
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- Jul 03, 2020
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Tim-Philipp Müller authored
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- Jul 02, 2020
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Tim-Philipp Müller authored
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- Jun 23, 2020
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Thibault Saunier authored
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- Jun 22, 2020
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Sebastian Dröge authored
Part-of: <gstreamer/gst-rtsp-server!138>
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Sebastian Dröge authored
It's deprecated, unneeded and doesn't do anything anymore. Part-of: <gstreamer/gst-rtsp-server!138>
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- Jun 19, 2020
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Tim-Philipp Müller authored
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Tim-Philipp Müller authored
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- Jun 15, 2020
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Sebastian Dröge authored
Otherwise the transports are not set up yet during the PLAY request handling when unsuspending (and thus unblocking) the media. In case of live pipelines this then causes the first few packets to go to the sinks before they know what to do with them, and they simply discard them which is rather suboptimal in case of keyframes. For non-live pipelines this is not a problem because the sink will still be PAUSED and as such not send out the data yet but wait until it goes to PLAYING, which is late enough. Adding the transports multiple times is not a problem: if the transport is already added it won't be added another time and TRUE will be returned. This fixes a regression introduced by a7732a68 before 1.14.0. Fixes gstreamer/gst-rtsp-server#107 Part-of: <gstreamer/gst-rtsp-server!135>
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Sebastian Dröge authored
Part-of: <gstreamer/gst-rtsp-server!135>
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Sebastian Dröge authored
The pad probes are not needed anymore at this point and later when reaching buffering 100% only the state is changed, no unblocking happens. Part-of: <gstreamer/gst-rtsp-server!135>
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Sebastian Dröge authored
It does literally the same as media_streams_set_blocked(FALSE). Part-of: <gstreamer/gst-rtsp-server!135>
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- Jun 12, 2020
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Lenny Jorissen authored
Part-of: <!134>
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- Jun 10, 2020
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Thibault Saunier authored
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And have the factory in the onvif-server example inherit from GstRTSPOnvifMediaFactory. Part-of: <!133>
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- Jun 09, 2020
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Thibault Saunier authored
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- Jun 08, 2020
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Guillaume Desmottes authored
Test was not enforcing a video format on videotestsrc. I420 was picked as it was the first format in GST_VIDEO_FORMATS_ALL which will no longer be true (gst-plugins-base!689). Part-of: <gstreamer/gst-rtsp-server!129>
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- Jun 05, 2020
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Mathieu Duponchelle authored
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- Jun 03, 2020
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Thibault Saunier authored
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- May 30, 2020
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Sebastian Dröge authored
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Sebastian Dröge authored
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- May 27, 2020
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Tim-Philipp Müller authored
Doesn't actually exist. Was fixed differently in Meson so that the user doesn't have to specify it. Part-of: <gstreamer/gst-rtsp-server!127>
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Underscorify the test name before setting GST_REGISTRY, so the registry actually ends up in the current build dir and not some subdir. For consistency with the other modules, but should also avoid problems on windows. Also fix indentation of environment block. Part-of: <gstreamer/gst-rtsp-server!126>
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If core is built as a subproject (e.g. as in gst-build), make sure to use the gst-plugin-scanner from the built subproject. Without this, gstreamer might accidentally use the gst-plugin-scanner from the install prefix if that exists, which in turn might drag in gst library versions we didn't mean to drag in. Those gst library versions might then be older than what our current build needs, and might cause our newly-built plugins to get blacklisted in the test registry because they rely on a symbol that the wrongly-pulled in gst lib doesn't have. This should fix running of unit tests in gst-build when invoking meson test or ninja test from outside the devenv for the case where there is an older or different-version gst-plugin-scanner installed in the install prefix. In case no gst-plugin-scanner is installed in the install prefix, this will fix "GStreamer-WARNING: External plugin loader failed. This most likely means that the plugin loader helper binary was not found or could not be run. You might need to set the GST_PLUGIN_SCANNER environment variable if your setup is unusual." warnings when running the unit tests. In the case where we find GStreamer core via pkg-config we use a newly-added pkg-config var "pluginscannerdir" to get the right directory. This has the benefit of working transparently for both installed and uninstalled pkg-config files/setups. Part-of: <gstreamer/gst-rtsp-server!126>
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Make hard requirement until we have more fine-grained control in the unit tests. Of course the presence of the .pc file doesn't imply that the plugins we need are actually there, but it's at least a step in the right direction. Part-of: <gstreamer/gst-rtsp-server!126>
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Part-of: <gstreamer/gst-rtsp-server!126>
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Make sure rtsp-media have received a GstRTSPStreamBlocking message from each active stream when checking if all streams are blocked. Without this change there will be a race condition when using two or more streams and rtsp-media receives a GstRTSPStreamBlocking message from one of the streams. This is because rtsp-media then checks if all streams are blocked by calling gst_rtsp_stream_is_blocking() for each stream. This function call returns TRUE if the stream has sent a GstRTSPStreamBlocking message, however, rtsp-media may have yet to receive this message. This would then result in that rtsp-media erroneously thinks it is blocking all streams which could result in rtsp-media changing state, from PREPARING to PREPARED. In the case of a preroll, this could result in that rtsp-media thinks that the pipeline is prerolled even though that might not be the case. Part-of: <gstreamer/gst-rtsp-server!124>
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Set expected_async_done to FALSE in default_suspend() if a state change occurs and the return value from set_target_state() is something other than GST_STATE_CHANGE_ASYNC. Without this change there is a risk that expected_async_done will be TRUE even though no asynchronous state change is taking place. This could happen if the pipeline is set to PAUSED using media_set_pipeline_state_locked(), an asynchronous state change starts and then the media is suspended (which could result in a state change, aborting the asynchronous state change). If the media is suspended before the asynchronous state change ends then expected_async_done will be TRUE but no asynchronous state change is taking place. Part-of: <gstreamer/gst-rtsp-server!123>
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Kristofer Björkström authored
There was a race condition where client was being finalized and concurrently in some other thread the rtsp ctrl timout was relying on client data that was being freed. When rtsp ctrl timeout is setup, a WeakRef on Client is set. Part-of: <gstreamer/gst-rtsp-server!121>
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- May 18, 2020
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add dscp_qos setting support at factory and media level to setup IP DSCP field of bounded UDP sinks. Fixes gstreamer/gst-rtsp-server#6 Part-of: <gstreamer/gst-rtsp-server!120>
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- May 14, 2020
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Sebastian Dröge authored
We always need to take the lock while accessing it as otherwise another thread might've removed it in the meantime. Also when destroying and creating a new one, ensure that the mutex is not shortly unlocked in between as during that time another one might potentially be created already. Part-of: <gstreamer/gst-rtsp-server!119>
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- May 03, 2020
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And the same for gst_rtsp_stream_get_rates(). Part-of: <gstreamer/gst-rtsp-server!118>
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Tim-Philipp Müller authored
Fix printf format for 64-bit variables. Part-of: <gstreamer/gst-rtsp-server!117>
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- May 01, 2020
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Sebastian Dröge authored
The old API is preserved now and new API was added that provides the additional parameter to the callback. Fixes gstreamer/gst-rtsp-server#104 Part-of: <gstreamer/gst-rtsp-server!116>
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