- Jan 12, 2017
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Sebastian Dröge authored
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- Jan 10, 2017
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This bug was accidentally introduced while fixing a segfault in _get_server_port() function. https://bugzilla.gnome.org/show_bug.cgi?id=776345
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- Jan 09, 2017
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Calling function gst_rtsp_stream_get_server_port() results in segmenation fault in the RTP/RTSP/TCP case. Port that the server will use to receive RTCP makes only sense in the UDP case, however the function should handle the TCP case in a nicer way. https://bugzilla.gnome.org/show_bug.cgi?id=776345
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- Jan 05, 2017
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Guillaume Desmottes authored
Generating those files is useful for users building the GStreamer stack using meson and having to link it to another project which is still using the autotools. https://bugzilla.gnome.org/show_bug.cgi?id=776810
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Guillaume Desmottes authored
pcfiledir was never defined so the paths were wrong. https://bugzilla.gnome.org/show_bug.cgi?id=776867
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- Dec 22, 2016
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Make sure that the appsink element is actually added to the bin before trying to link it with the elements in it. https://bugzilla.gnome.org/show_bug.cgi?id=776343
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- Dec 16, 2016
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Tim-Philipp Müller authored
Likely extremely bitrotten, and we should not ship this anyway.
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- Dec 03, 2016
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Edward Hervey authored
From f980fd9 to 39ac2f5
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- Dec 02, 2016
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Since decryption is handled within rtpbin, all outcoming stream caps will be application/x-rtp (i.e. regular rtp) Fixes RECORD with SRTP streams
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The function called immediately afterwards (collect_streams()) will need it to work properly
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Sebastian Dröge authored
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- Dec 01, 2016
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Sebastian Dröge authored
We're going to put a pipeline into a pipeline otherwise, which is not exactly ideal.
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- Nov 30, 2016
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Jan Schmidt authored
Fix a warning on shutdown - don't keep a pointer to an alread-unreffed object.
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- Nov 26, 2016
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- Nov 23, 2016
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Matthew Waters authored
85c52e19 introduced a more correct detection of the srtp rollover counter to add to the SDP. Unfortunately, it was incomplete for live pipelines where the logic blocks the source bin before creating the SDP and thus would never have the necessary informaiton to create a correct SDP with srtp encryption. Move the pad blocks to rtpbin's output pads instead so that the necessary information can be created before we need the information for the SDP. https://bugzilla.gnome.org/show_bug.cgi?id=770239
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The RTSP server will not timeout an idle RTSP connection (note this is different from doing timeout on a RTSP session). At least for Apache this is a problem when running RTSP over HTTPS since it uses one of the threads (there is a rather limited number) that are available for handling requests. https://bugzilla.gnome.org/show_bug.cgi?id=771830
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Tim-Philipp Müller authored
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- Nov 22, 2016
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With this RTSP server can use the sockets independent on the udpsrc state. When the udp src is finalized it will unref socket and when g_socket is finalized the socket will be closed. https://bugzilla.gnome.org/show_bug.cgi?id=765673
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- Nov 19, 2016
- Nov 18, 2016
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Thibault Saunier authored
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- Nov 17, 2016
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This type mismatch fails building with MSVC https://bugzilla.gnome.org/show_bug.cgi?id=774640
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- Nov 11, 2016
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Sebastian Dröge authored
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- Nov 10, 2016
- Nov 01, 2016
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These signals let the application validate the requests, configure the media/stream in a certain way and also generate error status code in case of error or bad request. https://bugzilla.gnome.org/show_bug.cgi?id=758062
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Tim-Philipp Müller authored
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Sebastian Dröge authored
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Sebastian Dröge authored
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- Oct 31, 2016
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Tim-Philipp Müller authored
Causes problems with client multicast tests otherwise if tests are run in parallel. https://bugzilla.gnome.org/show_bug.cgi?id=773640
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- Oct 28, 2016
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Tim-Philipp Müller authored
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- Oct 25, 2016
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Call session filter with filter_session_media as paramer in client_unwatch_session if using drop_backlog = FALSE. In client_unwatch_session its allowed to grow the watchs backlog. If using drop_backlog = FALSE and the backlog is full it will cause a deadlock when setting session media state to NULL if the backlog is not allowed to grow. https://bugzilla.gnome.org/show_bug.cgi?id=771983
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- Oct 20, 2016
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Tim-Philipp Müller authored
For gst-all.
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- Oct 06, 2016
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When using dynamic elements, gst_rtsp_stream_join_bin() is called from "pad-added" signal. In that case priv->srcpad could already have its caps, and they'll be sent to priv->send_src[0] pad. That means that when it connects "notify::caps" signal, that pad could already have received its caps and the signal won't be emitted anymore. In that case priv->caps stay to NULL and when building the SDP that stream gets ignored. Leading to missing video or audio when playing in client side. https://bugzilla.gnome.org/show_bug.cgi?id=772478
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- Sep 30, 2016
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Tim-Philipp Müller authored
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Sebastian Dröge authored
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- Sep 18, 2016
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To prevent any possibly confusion with IPs or anything else. https://bugzilla.gnome.org/show_bug.cgi?id=771530
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