Instead of checking for the gstreamer-video-1.0 package is installed,
just assume it is since we already check for the -base dependency.
With this replace the GST_VIDEO_* variables in makefiles and directly
link with libgstvideo.
https://bugzilla.gnome.org/show_bug.cgi?id=753820
Also implement framerate handling correctly by borrowing the code from
ximagesrc. GstBaseSrc::get_times() can't be used for that, we have to
implement proper waiting ourselves.
The block that is dispatched async to the main thread assumed the
wrapping GstAvSampleVideoSink to be alive. However, at the time of
the block execution the GstObject instance that is deferenced to access
the CA layer might already be freed, which caused occasional crashes.
Instead, we now only pass the CoreAnimation layer that needs to be
released to the block. We use __block to make sure the block is not
increasing the refcount of the CA layer again on its own.
https://bugzilla.gnome.org/show_bug.cgi?id=753081
CMBlockBuffer offers a model similar to GstBuffer, as it can
consist of multiple non-consecutive memory blocks.
Prior to this change, what we were doing was:
1) Incorrect:
CMBlockBufferCreateWithMemoryBlock does not copy the data,
but we gst_buffer_unmap'd right away.
2) Inefficient:
If the GstBuffer consisted of non-contiguous memory blocks,
gst_buffer_map resulted in malloc / memcpy.
With this change, we construct a CMBlockBuffer out of individual mapped
GstMemory objects. CMBlockBuffer is made to retain the GstMemory
objects (through the use of CMBlockBufferCustomBlockSource), so the
original GstBuffer can be unref'd.
https://bugzilla.gnome.org/show_bug.cgi?id=751241
All goto fail happen before ret is set. ret must be NULL, and the only
thing the fail statement block does is return NULL. Replacing the jumps to
do this return directly.
CID #1311329
CMBlockBufferGetDataLength would return the entire data length, while
size of individual blocks can be smaller. Iterate over the block buffer
and add the individual (possibly non-contiguous) memory blocks.
https://bugzilla.gnome.org/show_bug.cgi?id=751071
When AVFoundation indicates a supported frame rate range, add it to
the caps. This is important for devices such as the iPhone 6, which
indicate a single AVFrameRateRange of 2fps - 60fps.
https://bugzilla.gnome.org/show_bug.cgi?id=751048
In JNI_OnLoad() we will already get the Java VM passed and could
just directly use that. gstreamer_android-1.0.c will now provide
this to us.
Reason for this is that apparently not all Android system are
providing the JNI functions to get the currently running Java VMs, so
we would fail to get. With this we will always be able to get the Java
VM on such systems.
We only need that if no Java VM is running yet, and all usual cases,
i.e. when calling GStreamer from an actual Android app, there will already
be a Java VM we can just use.
It seems like some phones come without that symbol, let's hope they come
with the other symbol but for now don't make a missing JNI_CreateJavaVM fatal.
This allows us to signal what kind of audio we are expecting to record,
which should tell the system to apply filters (such as echo
cancellation, noise suppression, etc.) if required.
Even when we fail to encode frame, we should still enqueue it so
it could be passed into handle_frame (with output_buffer == NULL).
Otherwise, we risk GstVideoEncoder's queue of frames growing unbounded.
Note: We're slightly changing the renegotiation code to accommodate for
frames without output buffers, but this commit takes no ownership over
the way negotiation is being done.
https://bugzilla.gnome.org/show_bug.cgi?id=750669
VTCompressionSessionEncodeFrame retains the CVPixelBuffer during
encoding, and will release it as soon as it can (e.g. before it even
calls our callback). This means we can safely release input buffer
at this point, possibly allowing the system to reuse it sooner.
https://bugzilla.gnome.org/show_bug.cgi?id=750671
Copying arbitrary metas is going to cause problems and this should really be
handled by the base class. It overrides most other things already anyway,
including timestamp and duration. Those are just set here now so we can
insert the frame sorted into the queue.
https://bugzilla.gnome.org/show_bug.cgi?id=748922
OMX.Exynos. codecs are existing on some devices like the
Galaxy S5 mini, and cause random crashes (of the device,
not the app!) and generally misbehave. That specific device
has other codecs that work with a different name, but let's
just give them marginal rank in case there are devices that
have no other codecs and these are actually the only working
ones
On some devices there are codecs that don't start with OMX., while
there are also some that do. And on some of these devices the ones
that don't start with OMX. just crash during initialization while
the others work. To make things even more complicated other devices
have codecs with the same name that work and no alternatives.
So just give a lower rank to these non-OMX codecs and hope that
there's an alternative with a higher rank.
Also stagefright gives codecs starting with OMX. a higher rank too and
considers other codecs that don't start with OMX. as software codecs.
This decoder does not work if width and height field are not set
in the sinkpad caps. Let's make this explicit by adding them to
the template caps.
https://bugzilla.gnome.org/show_bug.cgi?id=749655
It is incorrect to modify the frame properties after passing them, since
VTCompressionSessionEncodeFrame takes reference and we have no control
over when it's being used.
In fact, the code can be simplified. We just preallocate the frame
properties for keyframe requests, and pass NULL otherwise.
https://bugzilla.gnome.org/show_bug.cgi?id=748467
Unless stopRequest is set, we should unlock conditionally -- otherwise,
the 'create:' method can wake up to an empty buffer queue
and pull a nil buffer.
https://bugzilla.gnome.org/show_bug.cgi?id=748054
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.
This API has been deprecated for eternities and microsoft
stopped shipping the headers in 2010 accoding to wikipedia,
so let's just remove it and focus on bringing the plugins
based on the newer APIs up to snuff.
This fixes handling of flushing seeks, where we will get a PAUSED->PLAYING
state transition after the previous one without actually going to PAUSED
first.
Otherwise we will overflow the internal buffer of the hardware
with useless frames and run into an error. This is necessary until
this bug in basesink is fixed:
https://bugzilla.gnome.org/show_bug.cgi?id=742916
decklinkvideosink might be added later to the pipeline, or its state might
be handled separately from the pipeline. In which case the running time when
we (last) went into PLAYING state will be different from the pipeline's.
However we need our own start time to tell the Decklink API, which running
time should be displayed at the moment we go to PLAYING and start scheduled
rendering.
... and hope that everything will be fine. This shouldn't really happen but
previously happened during shutdown. It should be fixed in videoencoder now,
but better be on the safe side here.
Use AVF provided timings to timestamp output buffers. Use the running time at
the time the first buffer is produced to base timestamps on. Report 1-frame
latency based on the negotiated framerate instead of hardcoding 4ms latency.
The property is in kbit/s and we store it in bit/s, so just multiply and
divide by 1000. No need to put a factor of 8 in there.
kVTCompressionPropertyKey_AverageBitRate is also in bit/s according to
its documentation.
This updates the dshowvideosink to work with the GStreamer 1.0.x APIs, and
avoids the use of deprecated GLib threading API that are now used since
GLib 2.32+
https://bugzilla.gnome.org/show_bug.cgi?id=699364
No idea where the DecklinkAPIDispatch.cpp comes from on Windows,
but this should still work. Will just become a problem once we
use other parts of the API.
Otherwise we're going to starve other elements if the decklink clock
is slower than the pipeline clock, or starts much later.
Of course this will still cause problems if the decklink clock and ours are
completely out of sync, or running at a very different rate. But this at least
works better now.
If we just count the frames and calculate timestamps from that, all frames
will arrive late in the sink as we have a live source here. Instead take
the pipeline clock at capture time as reference.
We have to handle the callback object a bit different:
a) it needs a virtual destructor
b) we need to set the callback to NULL when we're done with the output
c) create a new one every time
https://bugzilla.gnome.org/show_bug.cgi?id=740616
We will run into an assertion in set_caps() if we try to change
caps while the source is already running. Don't try to find new
caps in GstBaseSrc::negotiate() to prevent caps changes.
The object lock only protects the session, as we modify
the session from other threads when the bitrate property
is changed. Don't hold it much longer than for session
related things.
And we need to release the video decoder stream lock before
enqueueing a frames. It might wait for our callback to dequeue
a frame from another thread, which will then take the stream
lock too and deadlock.
It is not required on OSX apparently and was only added in 10.9.6 there.
Calculating the correct level from the configuration is not trivial, so let's
just not set a level at all here.
DVB-T2 supports 5, 10 and 1.712 MHz
Order of the enum values (new values after _AUTO)
has been kept congruent with the one in the v4l
API for consistency
Previously known as DMB-T/H, this is the
terrestial DTV broadcast standard currently
used by the People's Republic of China,
Hong Kong, Laos and Macau (officially),
and by Malaysia, Iraq, Jordan, Syria and
Lebanon (experimentally).
These apply to ISDB-T, DVB-T2 and DTMB
Order of the enum values (new rates after _AUTO)
has been kept congruent with the one in the v4l
API for consistency.
According to the v4l-dvb API docs, these are only
used for DVB-T2 at the moment.
Order of the enum values (new rates after _AUTO)
has been kept congruent with the one in the v4l
API for consistency.
iOS has special stride requirements that we don't know yet, so copy
input buffers into buffers allocated by iOS for now.
Later we should check the stride and probably provide a buffer pool for these
buffers so upstream can directly write in there.
