By this new property, user can select physical port to connect,
and element will pick requested port instead of random ones.
User should provide full port name including "client_name:" prefix.
An example is
jackaudiosrc port-names="system:capture_1,system:capture_3" ! ...
jackaudiosink port-names="system:playback_2"
In addition to "port-names" property, a new connect type "explicit"
is added so that element can post error message if requested
"port-names" contains invalid port(s).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1037>
Enhance open gop detection to drop B-frame which are invalid
before the first reference frame.
In stream such
gst-integration-testsuites/medias/defaults/mxf/op2b-mpeg2-wave_hd.mxf,
the two first frames must be dropped as we detect an open GOP situation
but in another media, such as http://col.la/1920X1080IXDCAMEX5MIN, the
first frames should not be dropped as we are in a closed GOP situation.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-ugly/-/merge_requests/84>
The relevant CI log
dyld: Library not loaded: /Users/gst-ci/builds/gstreamer/cerbero/cerbero-build/dist/darwin_universal/x86_64/lib/liborc-0.4.0.dylib
Referenced from: /Users/gst-ci/builds/xhaakon/gstreamer-sharp/cerbero-build/dist/darwin_universal/x86_64/bin/orcc
Reason: image not found
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-ci/-/merge_requests/410>
Default timer precision of Windows is dependent on system, but
usually it's known to be about 15ms in worst case.
That's not an enough precision for multimedia application.
Enable high-resolution clock in gst-launch to demonstrate
the usage of Windows high-precision clock for application developers.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/817>
We need to consider the first field of the last picture when the
last picture can not enter the DPB.
Another change is, when prev field's frame_num is not equal to the
current field's frame_num, we should also return FASLE because it
is also a case of losing some field.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2430>
For interlaced streams, it is also possible that the last frame is
not able to be inserted into DPB when the DPB is full and the last
frame is a non ref. For this case, we need to hold a extra ref for
the first field of the last frame and wait for the complete frame
with both top and bottom fields. For the progressive stream, the
behaviour is unchanged.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2430>
E.g. a pipeline like qmlglsrc ! gldownload ! ... would currently fail to
run because the OpenGL context are not created in the correct order.
The QtWindow also needs to know the OpenGL context used by downstream
elements in order to set optimize for the correct GstGLSyncMeta for
synchonisation purposes.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1036>
When reaching the end of non-frame wrapping track in pull mode, we want to force
the switch to the next non-eos pad. This is similar to when we exceed the
maximum drift.
Fixes issues on EOS where not everything would be drained out and stray errors
would pop out.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2434>
Adds a new plugin for ASIO devices.
Although there is a standard low-level audio API, WASAPI, on Windows,
ASIO is still being broadly used for audio devices which are aiming to
professional use case. In case of such devices, ASIO API might be able
to show better quality and latency performance depending on manufacturer's
driver implementation.
In order to build this plugin, user should provide path to
ASIO SDK as a build option, "asio-sdk-path".
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2309>
Normally uri is get from user input and invalid user input should not
be treated as critical error. Moved gst_uri_is_valid outside of
g_return_val_if_fail.
NULL uri is checked inside of gst_uri_is_valid and is correctly
treated as critical error, removed unneeded checks of NULL uri outside
of gst_uri_is_valid function.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/816>
... for user to be able to set the number of required samples.
For instance, our default value is 240 samples
(about 5ms latency in case that sample rate is 48000), which might
be larger than actual buffer size of audio capture device.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2307>
Prevent a condition where splitmuxsink won't go back to NULL state
after a child element fails to change state by making sure that
a READY->READY state change doesn't fail, and by returning
GST_FLOW_ERROR or GST_FLOW_FLUSHING upstream to shut down streaming
as quickly as possible.
This can happen after (for example) setting an invalid filename
on the sink element. In that case, the READY->PAUSED transition
fails, but with internal elements still in the NULL state. Trying
to set splitmuxsink back to NULL then ends up trying to bring
those NULL elements up to READY with a READY->READY transition,
(which fails, prevent splitmuxsink from getting to NULL)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1023>
The print_ref_pic_list_b now not only needs to trace the ref_pic_list_b0/1,
but also need to trace the ref_frame_list_0_short_term. We need to pass the
name directly to it rather than an index to refer to ref_pic_list_b0/1.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2425>
