Properly handle protocol version in the connection
Add the following headers types:
* Pipelined-Request
* Media-Properties
* Seek-Style
* Accept-Ranges
https://bugzilla.gnome.org/show_bug.cgi?id=781446
This is the same code that is in decklinkaudiosrc, audioringbuffer,
audiomixer and various other places. Have it once instead of copying it
everywhere.
https://bugzilla.gnome.org/show_bug.cgi?id=787560
+ Refactor previous constructor to call on that new constructor
+ Reimplement is_passthrough to strictly check whether the matrix
is an identity matrix, comparing channel-masks was incorrect:
the mixer may be remixing from a list of positions to the same
list of positions, but ordered differently, and reciprocally,
the mixer may be remixing from a list of positions to another
list of positions identically ordered
+ Remove unused tmp field, must have been a refactoring leftover
https://bugzilla.gnome.org/show_bug.cgi?id=785471
remove_format_info was a bit confusing to read, this removes
it in favor of standard gst_caps_map_in_place calls.
This no longer simplifies the resulting caps, but I
consider this should be the job of basetransform.
https://bugzilla.gnome.org/show_bug.cgi?id=785471
If someone calls gst_app_sink_try_pull_sample they are
probably no longer interested in any preroll samples.
Useful if the user has not registered a preroll appsink callback.
Also added unit test 'test_do_not_care_preroll'
make elements/appsink.check
that fails without this patch.
https://bugzilla.gnome.org/show_bug.cgi?id=786740
There is no reason for appsink to hang onto the preroll buffer.
If needed, the application can just keep a ref on this buffer
after calling gst_app_sink_try_pull_preroll.
Also added unit test 'test_pull_preroll'
make elements/appsink.check
https://bugzilla.gnome.org/show_bug.cgi?id=786740
This is used to indicate upstream the requirement in buffers
while no buffer pool can be provided. In this case, only
configure the pool with caps/size/min/max if we have caps,
which we only parsed when there was no allocation pool.
https://bugzilla.gnome.org/show_bug.cgi?id=730758
__gst_video_element_proxy_caps is called by
__gst_video_element_proxy_getcaps with caps set to the caps
allowed downstream. As we didn't set colorimetry or
chroma-site on the resulting caps, upstream considered it
possible to use whatever values it wanted, leading to
not negotiated errors later on.
As the description for that function is:
"Takes caps and copies its video fields to tmpl_caps",
it seems legitimate to set these fields there.
https://bugzilla.gnome.org/show_bug.cgi?id=786172
Use the intended sequence for re-using elements:
* EOS
* STREAM_START if element is to be re-used
This avoids having elements (such as queue/multiqueue/queue2) not
properly resetting themselves.
When delaying EOS propagation (because we want to wait until all
streams of a group are done for example), we re-trigger them by
first sending the cached STREAM_START and then EOS (which will
cause elements to re-set themselves if needed and accept new
buffers/events).
https://bugzilla.gnome.org/show_bug.cgi?id=785951
Only adjusting the base_ts might lead to a negative ts and as such integer
overflow into a huge timestamp which then propagates into the granulepos
and so on. Instead, resync to incoming buffer timestamp using both base_ts
and sample count rather than only base_ts.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=785948
It is forwarding messages to the playbin bus, thus forwarding messages
that contain a floating reference to the application. This generally
makes bindings unhappy, we must not leak floating references to them.
channels=1 is always mono, having it 'unpositioned' does not make
sense.
This fixes pipeline such as:
gst-validate-1.0 audiotestsrc ! audio/x-raw,channels=2,rate=44100,layout=interleaved ! audioconvert ! audioresample ! audio/x-raw, rate=44100, channels=1 ! avenc_mp2 ! fakesink
https://bugzilla.gnome.org/show_bug.cgi?id=785407
