gst_ks_device_provider_probe() is a no-braier, just runs ks_enumerate_devices()
and reports the results.
Monitoring is a bit more tricky. We have to create a dummy message-processing
window and register device change notifications for it.
As kernel streaming can (and should) be used for audio capture and audio
playback, this change also has certain placeholders for such.
https://bugzilla.gnome.org/show_bug.cgi?id=747757
The autodetection mode was broken because a race condition in the input mode
setting. The mode could be reverted back when it was replaced in
the streaming thread by the old mode in the middle of mode changed callback.
We now fill GErrors for everything that could throw an exception, and method
calls now always return a gboolean and their value in an out-parameter to
distinguish failures from other values.
The shm-area-property tells the name of the shm area used by the element. This
is useful for cases where shmsink is not able to clean up (calling
shm_unlink()), e.g. if it is in a sandbox.
https://bugzilla.gnome.org/show_bug.cgi?id=675134
while having the default vtdec at secondary rank. This allows decodebin/playbin
to prefer the hardware based decoders, and if that fails to initialize because
hardware resources are busy to fall back to e.g. the libav based h264 decoder
instead of the software based vtdec (which is slower), and only fall back to
the software based vtdec if there is no higher ranked decoder available.
Using requestMediaDataWhenReadyOnQueue the layer will execute a block
when it would like more frames. Using this we can provide the current
frame and avoid needlessly filling the layer's buffer queue causing
older frames to be displayed when under resource pressure.
Otherwise we might set bogus values or GST_CLOCK_TIME_NONE.
Also make sure to reset the caps field to NULL after unreffing
the caps to prevent accidential use afterwards, and unref any
old caps before we remember new caps.
Otherwise we will still have a reference to the surface left, which would
prevent activating the sink again later. E.g. after we lost the device.
Hopefully fixes https://bugzilla.gnome.org/show_bug.cgi?id=744615
Add the diff between the external time when we went to playing and
the external time when the pipeline went to playing. Otherwise we
will always start outputting from 0 instead of the current running
time.
gstdecklink.cpp: In member function 'virtual HRESULT GStreamerDecklinkInputCallback::VideoInputFrameArrived(IDeckLinkVideoInputFrame*, IDeckLinkAudioInputPacket*)':
gstdecklink.cpp:498:22: error: comparison between signed and unsigned integer expressions [-Werror=sign-compare]
if (capture_time > m_input->clock_start_time)
^
gstdecklink.cpp:503:22: error: comparison between signed and unsigned integer expressions [-Werror=sign-compare]
if (capture_time > m_input->clock_offset)
^
The driver has an internal buffer of unspecified and unconfigurable size, and
it will pull data from our ring buffer as fast as it can until that is full.
Unfortunately that means that we pull silence from the ringbuffer unless its
size is by conincidence larger than the driver's internal ringbuffer.
The good news is that it's not required to completely fill the buffer for
proper playback. So we now throttle reading from the ringbuffer whenever
the driver has buffered more than half of our ringbuffer size by waiting
on the clock for the amount of time until it has buffered less than that
again.
The ringbuffer's acquire() is too early, and ringbuffer's start() will only be
called after the clock has advanced a bit... which it won't unless we start
scheduled playback.
Not from the decklink clock. Both will return exactly the same time once the
decklink clock got slaved to the pipeline clock and received the first
observation, but until then it will return bogus values. But as both return
exactly the same values, we can as well use the pipeline clock directly.
There is no reason to pre-roll more buffers here as we have our own ringbuffer
with more segments around it, and we can immediately provide more buffers to
OpenSL ES when it requests that from the callback.
Pre-rolling a single buffer before starting is necessary though, as otherwise
we will only output silence.
Lowers latency a bit, depending on latency-time and buffer-time settings.
4 is the "typical" number of buffers defined by Android's OpenSL ES
implementation, and its code is optimized for this. Also because we
have our own ringbuffer around this, we will always have enough
buffering on our side already.
Allows for more efficient processing.
The pseudo buffer pool code was using gst_buffer_is_writable()
alone to try and figure-out if cached buffer could be reused.
It needs to check for memory writability too. Also check map
result and fix map flags.
https://bugzilla.gnome.org/show_bug.cgi?id=734264
Use YUV instead of RGB textures, then convert using the new apple specific
shader in GstGLColorConvert. Also use GLMemory directly instead of using the
GL upload meta, avoiding an extra texture copy we used to have before.
When doing texture sharing we don't need to call CVPixelBufferLockBaseAddress to
map the buffer in CPU. This cuts about 10% relative cpu time from a vtdec !
glimagesink pipeline.
Otherwise we might start the scheduled playback before the audio or video streams are
actually enabled, and then error out later because they are enabled to late.
We enable the streams when getting the caps, which might be *after* we were
set to PLAYING state.
Otherwise we might start the streams before the audio or video streams are
actually enabled, and then error out later because they are enabled to late.
We enable the streams when getting the caps, which might be *after* we were
set to PLAYING state.
