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kAudioFormatFlagIsSignedInteger is a format flags

Closes #394

In 2018 khronos changed the gl header guards. If we don't detect
this properly we would end up with plenty of symbol redifinition
(since we would be importing twice the "same" header).

Instead detect if the "newer" header was already included and if
so define the "old" define to avoid this situation

Fixes #523

For the demuxer we have to select line offset 0 for the time being as
this information is not passed over MOV.

When the EOS event is received, run all timers immediately and avoid
pushing the EOS downstream before this has been run. This ensures that
the lost packet statistics are accurate.

The teardown of the pads checks the refcount, but there are timers
inside the jitterbuffer that can push things, so if we're not lucky,
things could be pushed while the pads are being shut down. Putting the
jitterbuffer to NULL first avoids this.

After EOS is received, it is pointless to wait for further events,
specially waiting on timers. This patches fixes two cases where we could
wait instead of returning GST_FLOW_EOS and trigger a spin of the loop
function when EOS is queued, regardless if this EOS is the queue head or
not.

If the decoder signals metadata change we need to update the output
format and negotiate with downstream elements.

Any call to set_format() could mean that the stream type changed so we
reset the decoder and mark got_headers FALSE.

Draining the parser is not enough here, on caps change we need to
reset it so it is ready to accept new caps.

If the inlined caps change while using the same CV we need to update the
source pad caps.

Otherwise, it fails to link.

gst-build#13

Pull in video frame fields into local variables. Without this the
compiler must assume that they could've changed on every use and read
them from memory again.

This reduces the inner loop from 6 memory reads per pixels to 4, and the
number of writes stays at 3.

Using the same calculation used for video streams.

From ed78bee to 59cb678

This causes 'redefinition of typedef ...' errors on GCC 4.5.3

stream.segment should be updated with the values of the current edit
list, also when a new `moov` is received. Unfortunately this was not
being the case because of an early return.

As a consequence of this bugs, no end of movie clipping was being
performed on the new moov and no segment event was being emitted.

When performing stream switching (e.g. in MSE) the new moov may have a
different edit list. This is often the case when switching between
baseline H.264 (which lacks B-frames) and more demanding profiles. For
this reason it's important to emit a new segment in order to be able
to get matching stream times.

This patch moves the initialization of QtDemuxStream.segment from
gst_qtdemux_add_stream() to _create_stream(). This ensures the segment
is always initialized when the stream is created.

Otherwise the segment format is left as GST_FORMAT_UNDEFINED in the case
were a track is reparsed and qtdemux_reuse_and_configure_stream() is
called instead of gst_qtdemux_add_stream(). (See
qtdemux_expose_streams() in the non streams-aware case.)

- Consider GST_CLOCK_TIME_NONE as not to be used.
- Complete "rtcp-feedback-retention-window" property getter/setter
  implementation.

Closes #522

This is an extra internal recurisve lock use to avoid having to take
both sink pad streams lock all the time. This patch renamed it
INTERLNAL_STREAM_LOCK/UNLOCK() to avoid confusion with possible upstream
GST_PAD API.

This reverts "6f3734c3 rtpssrcdemux: Only forward stick events while
holding the sinkpad stream lock" and actually hold on the internal
stream lock. This prevents in some needed case having a second
streaming thread poping in and messing up event ordering.

This the first unit test of this element. It adds a test that verify
that events are forwarded correctly.

Pass riff codec-data as strf, not strd, which is where
gst_riff_create_audio_caps() expects the WAVEFORMATEXTENSIBLE
data.

https://bugzilla.gnome.org/show_bug.cgi?id=757583
Fixes #234

This is required before we enabled an indent test in the CI.

gstreamer/gstreamer-project#33

Otherwise it might lead to deadlocks

See https://gitlab.gnome.org/GNOME/pitivi/issues/2259

Closes #518

While forwarding serialized event, we use gst_pad_forward() function.
In the forward callback (GstPadForwardFunction) we always return
TRUE. Returning true there will stop the dispatching procedure. As a
side effect, only one events is receiving the events. This breaks
when sending EOS from the applicaiton, it also breaks the latency
tracer.

Use build arguments consistent with core and -base. This can also
remove noisy "C4819" warning of non-us locale MSVC.

The previous patch did not even compile on any possible platform or C
standard. That commit also didn't have a proper commit message.

Android ships Linux with a different signature for ioctl. They first
released an ioctl with int as request type, and later "fixed" it by
adding an override with unsign, which is still not matching Linux and
BSD implementation which uses unsigned long int.

PulseAudio defines PA_RATE_MAX as the maximum sampling rate that it
supports. We were previously exposing a maximum rate of INT_MAX, which
is incorrect, but worked because nothing was really using a rate greater
than 384000 kHz.

While playing DSD data, we hit a case where there might be very high
sample rates (>1MHz), and pulsesink fails during stream creation with
such streams because it erroneously advertises that it supports such
rates.

Since PA_RATE_MAX is #define'd to (8*48000U), we can't just use it in
the caps string. Instead, we fix up the rate to what we actually support
whenever we use our macro caps.