This makes it possible to set any controls that can be set with
VIDIOC_S_CTRL.
The controls are set when the property is set (if the device is open)
and when the device is opened.
https://bugzilla.gnome.org/show_bug.cgi?id=698837
It is not needed to do a state change from the _play() function on
ourselves. The state change function already did that and we don't want to
interfere with that (or use hacks to avoid interference).
Also send stream-start and segment event on the RTCP pad.
We don't need to send anything on the sync_src pad because we
already forwarded all incomming events.
The previous implementation had the formatting of SDP attributes happen
in each RTP payloader, now instead the constituent values are propagated
as caps fields. This allows for applications to do SDP offer/answer
based on caps negotiation.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700748
The previous implementation had the formatting of SDP attributes happen
in each RTP payloader, now instead the constituent values are propagated
as caps fields. This allows for applications to do SDP offer/answer
based on caps negotiation.
Keep parsing a-framerate, x-framerate and x-dimensions in rtpjpegdepay
to be backwards compatible with previous payloaders.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700748
Without this, a queued buffer may be required, filled and queued before it
is dequeued.
Calling gst_buffer_pool_acquire_buffer() ensures that the buffer is set up
correctly and gst_buffer_unref() calls buffer_release().
https://bugzilla.gnome.org/show_bug.cgi?id=700781
This can happen if other parts of the pipeline are reconfigured.
Stop streaming even for a short amount of time can be quite visible, so it
should be avoided if possible.
https://bugzilla.gnome.org/show_bug.cgi?id=700503
There is no reason to send a flush-stop when receiving a seek event.
In the case of a flushing seek, we could eventually want to, but in
the code path were we check if the seek is "flushing", we have the
following comment that makes sense:
"we can't send FLUSH_STOP here since upstream could start pushing data
after we unlock mix->collect.
We set flush_stop_pending to TRUE instead and send FLUSH_STOP after
forwarding the seek upstream or from gst_videomixer_collected,
whichever happens first."
https://bugzilla.gnome.org/show_bug.cgi?id=684237
In case qtdemux is handling a mss stream, do not mark the stream to wait
for EOS after a segment. Even if it seems to be the last one according to
the current streams information.
MSS handling is different here because there is another demuxer driving
the pipeline
The samplerate field in the STSD atom is not right for some ALAC files
(usually when audio is 96kHz/24bits), so the audio caps must be
extracted from the codec data.
https://bugzilla.gnome.org/show_bug.cgi?id=700382
