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gstreamer/gst-libs/gst
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Arun Raghavan a6742e81b9 convertframe: Fix async video sample conversion with non-default context 
The GSource for dealing with timeouts in
gst_video_convert_sample_async() might be attached to a non-default
context, so we should not be using g_source_remove() on the returned ID.

The correct thing to do is to keep a reference to the actual GSource and
then call g_source_destroy() on it.

https://bugzilla.gnome.org/show_bug.cgi?id=780297
2017-03-20 17:23:58 +05:30
..
allocators docs: Convert gtkdoc comments to markdown 2017-03-10 18:19:17 -03:00
app docs: Convert gtkdoc comments to markdown 2017-03-10 18:19:17 -03:00
audio docs: Convert gtkdoc comments to markdown 2017-03-10 18:19:17 -03:00
fft docs: Convert gtkdoc comments to markdown 2017-03-10 18:19:17 -03:00
pbutils encoding-profile: Mark format caps as transfer-none in profile creation 2017-03-17 16:01:57 +05:30
riff docs: Convert gtkdoc comments to markdown 2017-03-10 18:19:17 -03:00
rtp docs: Convert gtkdoc comments to markdown 2017-03-10 18:19:17 -03:00
rtsp docs: Convert gtkdoc comments to markdown 2017-03-10 18:19:17 -03:00
sdp docs: Convert gtkdoc comments to markdown 2017-03-10 18:19:17 -03:00
tag docs: Convert gtkdoc comments to markdown 2017-03-10 18:19:17 -03:00
video convertframe: Fix async video sample conversion with non-default context 2017-03-20 17:23:58 +05:30
gettext.h Fix FSF address 2012-11-03 23:05:09 +00:00
glib-compat-private.h Fix FSF address 2012-11-03 23:05:09 +00:00
gst-i18n-app.h tools: add simple command-line gst-play utility for testing purposes 2013-08-16 15:45:23 +01:00
gst-i18n-plugin.h Fix FSF address 2012-11-03 23:05:09 +00:00
Makefile.am rtp: build audio library before rtp 2016-02-16 17:42:44 +02:00
meson.build rtsp: Include GstSdp-1.0.gir when generating the gir 2016-11-10 17:43:38 -03:00