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>
When driver return error on update plane request, kmssink
disables the scaling and retries plane update.
While doing so kmssink was matching the source rectangle dimensions
to the target rectangle dimensions which were calculated
as per scaling but this is incorrect, instead what we want here is
that target rectangle dimensions should match the source rectangle
dimensions as scaling is disabled now and so we match result
rectangle dimensions with source rectangle dimensions.
While at it, also match the result rectangle coordinates for
horizontal and vertical offsets with source rectange coordinates,
as since there is no scaling being done so no recentering is
required.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2415>
Some cut VP9 streams begin with a non key frame. The current code
just bail out the parse_process_frame() if not a key frame. Because
of this, we do not set the valid caps before we push the data of the
first frame(even this first frame will be discarded by the downstream
decoder because it is not a key frame).
The pipeline such as:
gst-launch-1.0 filesrc location=some.ivf ! ivfparse ! vp9parse !
vavp9dec ! fakesink
will get a negotiation error and the pipeline can not continue. The
correct behaviour should be: the decoder discard the first frame and
continue to decode later frames successfully.
So, when the parse does not have valid stream info(should be the first
frame case), we should continue and report caps.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2427>
We may need to drop the slices such as RASL pictures with the NoRaslOutputFlag, so
the current picture of h265decoder may be freed. We should not assign the frame->
output_buffer too early until we really output it. Or, the later coming slices will
allocate another picture and trigger the assert of:
gst_video_decoder_allocate_output_frame_with_params:
assertion 'frame->output_buffer == NULL' failed
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2421>
There is currently no way for users to receive incoming events from
appsink while keeping them properly serialized with the buffers flow.
This can be especially useful when application is injecting custom
downstream events into the pipeline and needs to know when they reached
appsink.
Solving this by adding a new signal notifying about new incoming events
and a set of action signals and method to pull those events.
The API is actually pulling the samples and events all together as they
are actually fetched from the same queue.
Having a specific API to pull only events would have the side effect of
discarding samples (and pulling samples would discard events) making
this API not convenient for users.
Partially fix#247
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1046>
In H265, the stream may have odd bit depth such as 9 or 11. And
the bit depth of luma and chroma may differ. For example, the
stream with luma depth of 8 and chroma depth of 9 should use the
10 bit rtformat as the decoded picture format.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2420>
The problem is that EGLNativeWindowSurface and wl_egl_surface are the
same object underneath, so we must recreate both together. As an
optimization, the EGLNativeWindowSurface wrapper is only re-created
if the window_handle changed.
On Mesa, this would cause crash, which will be fixed by:
https://gitlab.freedesktop.org/mesa/mesa/-/merge_requests/11979
And will lead to proper errors in the future or on other GL stack. This
issue was encounter using a permanent GstGLDisplay after cycling one of
multiple independent pipelines through NULL state.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1230>
