Don't loop forever if an USB audio device gets disconnected
while in use. Post an error message instead. This is not
enough yet though, we still need to make the base class
and/or the ring buffer bail out.
https://bugzilla.gnome.org/show_bug.cgi?id=690197
Add a limit to the amount of queued bytes or messages we allow on the watch.
API: GstRTSPConnection::gst_rtsp_watch_set_send_backlog()
API: GstRTSPConnection::gst_rtsp_watch_get_send_backlog()
The behaviour is sensibly changed here. Instead of purely falling when a
preset is set on the #GstEncodingProfile, we now make sure that the
element that is plugged corresponds to the one specified as preset. Then,
if we have a preset_name, we use it, if it fails, we fail (we might rather
just keep working even without setting the element properties?)
+ Add tests that it behave correctly
It was possible to decide only what #GstElement implementing #GstPreset
to use during the encoding, we can now let the user select a specific preset previously
saved using #gst_preset_save_preset specifying the name chosen when it was saved
in the gst_encoding_profile_set_preset_name.
Actually loading a preset with %NULL as a name would have always failed, so
in the current state of the API that feature is unusable
API:
gst_encoding_profile_set_preset_name
gst_encoding_profile_get_preset_name
In order for 1.x and 1.(x+1) versions to not invade on each other
we need to have different lib versions.
So we need a consistent and predictable scheme:
library version number = MINOR * 100 + MICRO
Ex:
1.0.0 => 0 (duh)
1.0.3 => 3
1.1.0 => 100
1.1.1 => 101
1.2.0 => 120
1.10.5 => 1005
When the input buffers for a stream don't have a duration set,
timestamp_end might still be GST_CLOCK_TIME_NONE. When advancing
EOSed streams via GAP events (with other streams not yet EOS), we
would then use the invalid timestamp_end to calculate the duration
of the gap. This in turn would make baseaudiosink abort, because it
would try to allocate memory for a trizillion samples.
So if buffers don't have a duration set, assume a duration of
one second for stream catch-up purposes, just so we can still
continue to catch up in those cases. And make sure that
timestamp_end is valid before doing calculations with it.
http://bugzilla.gnome.org/show_bug.cgi?id=678530
And only return the proportion. The earliest time already can be
retrieved from get_max_decode_time() and by renaming we allow this
to be more extensible in the future.
Add a getter for the QoS proportion and earliest_time to help
subclasses do better estimations based on the proportion.
API: gst_video_decoder_get_qos_info()
https://bugzilla.gnome.org/show_bug.cgi?id=687991
