gstreamer/gst-libs/gst
Julien Isorce 8af7b1f70b appsink: fix end condition of query drain handler
The while loop should end when all buffers "and" the preroll
buffer are consumed but this means to continue waiting if there
are still some pending buffers "or" preroll buffer.

The unit test was correct but racy because of this mistake.
I.e. because of the wrong "and" the while could finish too early.

cd tests/check && GST_CHECKS=test_query_drain make elements/appsink.forever

https://bugzilla.gnome.org/show_bug.cgi?id=789763
2017-11-29 15:09:04 +00:00
..
allocators meson: remove vs_module_defs 2017-10-05 13:53:14 +01:00
app appsink: fix end condition of query drain handler 2017-11-29 15:09:04 +00:00
audio audiobasesink: Print signed time offset as a signed number 2017-11-08 19:24:55 +02:00
fft meson: remove vs_module_defs 2017-10-05 13:53:14 +01:00
pbutils discoverer: Don't remove element when switching to PLAYING 2017-11-15 10:55:56 +01:00
riff riff-media: Handle strf_data being NULL 2017-11-02 09:19:21 +01:00
rtp rtp: Require gconstpointer instead of gpointer for gst_rt[c]p_buffer_new_copy_data() 2017-11-17 14:14:55 +02:00
rtsp rtspconnection: Allow setting a custom accept-certificate function for manually checking a TLS certificate for validity 2017-11-01 13:41:42 +02:00
sdp sdpmessage: add_attribute accepts NULL value 2017-11-03 17:56:39 +11:00
tag vorbistag: Fix previous comment 2017-11-02 10:43:02 +01:00
video video: add missing GST_EXPORT 2017-11-26 18:14:39 +00:00
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