playsinkconvertbin: only intersect with the filter at the end

Otherwise we might change some capsfeatures from ANY to the specific
value from the filter and do not filter those out in case the
sink doesn't support them

https://bugzilla.gnome.org/show_bug.cgi?id=734822
audiodecoder: Don't ignore ::start/stop return values
po: Update translations
Update .po files
Release 1.4.1
textoverlay: Don't hold any mutexes while calling negotiate

It's not done in any other code calling negotiate and will cause deadlocks
as it is sending events and queries in the pipeline.

Specifically this pipeline was deadlocking:
gst-launch-1.0 videotestsrc ! textoverlay ! textoverlay ! fakesink
basetextoverlay: properly fallback to non-overlay caps

When downstream claims to accept the overlay meta but fails to
provide it in the allocation query, properly fallback to setting
a new caps without the overlay meta as that is not going to be used.

Only do this if the original caps doesn't have the overlay already,
otherwise there isn't much that can be done.

https://bugzilla.gnome.org/show_bug.cgi?id=735800
tests: textoverlay: add test to reproduce fakesink scenario

Adds a new test to textoverlay to make sure it can properly handle
elements that have ANY caps but fail to add the overlay meta in
the allocation query.

This test verifies that textoverlay won't use the caps features even
knowing that the overlay meta is accepted when querying the downstream
caps because it also needs downstream to confirm by putting the meta
in the allocation query.

https://bugzilla.gnome.org/show_bug.cgi?id=735800
videorate: GstStructure refcount critical message

s3 is not being initialized when run in a loop
and the same was being freed, which resulted in the crash

https://bugzilla.gnome.org/show_bug.cgi?id=735952
rtspconnection: Protect readsrc, writesrc and controllsrc with a mutex

Fixes a crash when controlsrc, readsrc or writesrc are modified from
gst_rtsp_source_dispatch_read/write and gst_rtsp_watch_reset at the
same time.

https://bugzilla.gnome.org/show_bug.cgi?id=735569
audioresample: don't skip input samples

when downsampling, the output buffer can be filled before all the input
samples are consumed.  this is correct:  when downsampling, several input
samples are needed for each output sample, so when only a small number of
input samples are available the number of output samples produced can be 0.

the resampler, however, was discarding those extra input samples instead of
clocking them into its filter history for the next iteration.  this patch
fixes this by removing the check that the output buffer is full.  the code
now always loops until all input samples are consumed, and relies on the
calling code to have provided a suitably sized location for the output.
note that there are already other checks in place in the calling code to
ensure that this is the case.

https://bugzilla.gnome.org/show_bug.cgi?id=732908
playbin: filter out buffering messages when switching uri

When switching URI from about-to-finish, playbin starts decoding the new
URI and the queue2 inside uridecodebin starts emitting buffering messages
immediately. However, the queue(s) inside playsink still have buffers to
play and the pipeline doesn't need to pause for buffering, so we should
not send those buffering messages up to the application, otherwise there
is an audible glitch caused by pausing the pipeline for a very short time.

https://bugzilla.gnome.org/show_bug.cgi?id=727255
decodebin: protect buffering message handling

Use the object lock to avoid concurrent processing which leads
to small disasters (assertions or crashes)
videofilter: Unref buffers before calling the transform_frame functions

GstVideoFrame has another reference, so the buffer looks unwriteable,
meaning that we can't attach any metas or anything to it

https://bugzilla.gnome.org/show_bug.cgi?id=736118
audiobasesink: Fix deadlock caused by holding object lock while calling clock functions

Issue:
During a PAUSED->PLAYING transition when we are rendering an audio buffer in AudioBaseSink
we make adjustments to the sink's provided clock i.e. fix clock calibration using the external
pipeline clock, within "gst_audio_base_sink_sync_latency function inside gstaudiobasesink.c".
For the calibration adjustment we need to get the sink clock time using "gst_audio_clock_get_time".
But before calling "gst_audio_clock_get_time" we acquire the Object Lock on the Sink. If sink is
a pulsesink, "gst_audio_clock_get_time" internally calls "gst_pulsesink_get_time" which needs to
acquire Pulse Audio Main Loop Lock before querying Pulse Audio for its stream time using
"pa_stream_get_time". Please see "gst_pulsesink_get_time in pulsesink.c".

So the situation here is we have acquired the Object lock on Sink and need PA Main Loop Lock.
Now Pulse Audio Main Thread itself might be in the process of posting a stream status
message after Paused to Playing transition which in turn acquires the PA Main loop lock and
needs the Object Lock on Pulse Sink. This causes a deadlock with the earlier render thread.

Fix:
Do not acquire the object Lock on Sink before querying the time on PulseSink clock. This is
similar to the way we have used get_time at other places in the code. Acquire it after the
get_time call. This way PA Main loop will be able to post its stream status message by
acquiring the Sink Object lock and will eventually release its Main Loop lock needed for
gst_pulsesink_get_time to continue.

https://bugzilla.gnome.org/show_bug.cgi?id=736071
videodecoder: do not leak pool and allocator in error case

https://bugzilla.gnome.org/show_bug.cgi?id=736679
playsink: Add audio/videoconvert in front of the audio/video-filters

audioresample and videoscale is something the application will have to do if
required, but we can at least help here by adding the
audioconvert/videoconvert elements.

https://bugzilla.gnome.org/show_bug.cgi?id=735748
typefind: correct the condition for irap flag

https://bugzilla.gnome.org/show_bug.cgi?id=736779
audiocdsrc: do not leak uid after parsing TOC select event

https://bugzilla.gnome.org/show_bug.cgi?id=736739
basetextoverlay: schedule reconfigure on source pad when negotiation fails

The source pad might be flushing while negotiating, resulting in
set_caps or the ALLOCATION query failing. In this case set the
reconfigure flag on the source pad so that negotiation is retried on the
next buffer.