This seems to happen sometimes on some hardware, and is not really
critical as long as the scheduling of the normal frames works fine.
Only post a warning message for this case.
and error out here already otherwise. We currently don't support
reconfiguration here and it can't happen really either unless the auto
mode is selected.
15:18:47 gstdecklinkaudiosrc.cpp:745:45: error: cannot initialize a parameter of type 'int64_t *' (aka 'long long *') with an rvalue of type 'gint64 *' (aka 'long *')
15:18:47 (BMDDeckLinkMaximumAudioChannels, &self->channels_found);
15:18:47 ^~~~~~~~~~~~~~~~~~~~~
15:18:47 ./linux/DeckLinkAPI.h:970:87: note: passing argument to parameter 'value' here
15:18:47 virtual HRESULT GetInt (/* in */ BMDDeckLinkAttributeID cfgID, /* out */ int64_t *value) = 0;
15:18:47 ^
gstdecklink.cpp:821:11: warning: variable 'dtc' is used uninitialized whenever 'if' condition is false [-Wsometimes-uninitialized]
if (m_input->videosrc) {
^~~~~~~~~~~~~~~~~
gstdecklink.cpp:837:41: note: uninitialized use occurs here
stream_time, stream_duration, dtc, no_signal);
^~~
gstdecklink.cpp:821:7: note: remove the 'if' if its condition is always true
if (m_input->videosrc) {
^~~~~~~~~~~~~~~~~~~~~~~
gstdecklink.cpp:810:29: note: initialize the variable 'dtc' to silence this warning
IDeckLinkTimecode *dtc;
^
= NULL
gstdecklink.cpp: In member function ‘virtual HRESULT GStreamerDecklinkInputCallback::VideoInputFrameArrived(IDeckLinkVideoInputFrame*, IDeckLinkAudioInputPacket*)’:
gstdecklink.cpp:766:34: error: ‘base_time’ may be used uninitialized in this function [-Werror=maybe-uninitialized]
capture_time -= base_time;
^
First of all, all the HD and UHD modes should be top-field-first, as
also returned by the Decklink mode iterator API.
Then we should include the caps field "field-order" in the caps of the
source (not the sink due to negotiation problems with optional fields).
And finally we should set the TFF flag on interlaced buffers that are
top-field-first.
On some hardware the first few frames are bogus and not very useful.
Their timestamps are off, they have no timecodes, or there are spurious
black frames / no-signal frames. After a few frames this stabilizes
though.
https://bugzilla.gnome.org/show_bug.cgi?id=774850
Based on this we calculate the actual capture time, which should get us
rid of any capturing jitter by averaging it out.
Also add a output-stream-time property which forces the elements to
output the stream time directly instead of doing any conversion to the
pipeline clock. Use with care.
https://bugzilla.gnome.org/show_bug.cgi?id=774850
The hardware timestamps have no relation to when frames were produced,
only when frames arrived somewhere in the hardware. Especially there is
no guarantee that audio and video will have the same hardware timestamps
although they belong together, and even more important: the rate with
which the hardware timestamps increase is completely unrelated to the
rate with which the frames are captured!
As such we can as well use the pipeline clock directly and stop doing
complicated calculations. Also as a side effect this allows now running
without any pipeline clock, by directly making use of the stream times
as reported by the driver.
https://bugzilla.gnome.org/show_bug.cgi?id=774850
When a frame is found to not have an associated input source (cable
unplugged, wrong mode selected), an element warning will be issued. When
the next frame in the stream is found to have an input source selected
(e.g. cable replugged), an element info will be issued.
https://bugzilla.gnome.org/show_bug.cgi?id=774629
Unfortunately this does not go through the normal state change
machinery, so we don't get notified about this in change_state().
However we need to stop scheduled playback, so that once PLAYING is
reached again we can start scheduled playback with the correct time.
Without this, flushing seeks in PLAYING will not work correctly:
decklinkvideosink will wait before showing the new frames for the amount
of time the pipeline was in PLAYING before.
While gint64 and int64_t are always the same, clang does not agree with that.
/Applications/Xcode.app/Contents/Developer/usr/bin/make -C decklink
CXX libgstdecklink_la-gstdecklinkaudiosink.lo
gstdecklinkaudiosink.cpp:675:79: error: cannot initialize a parameter of type 'int64_t *' (aka 'long long *') with an rvalue of type 'gint64 *' (aka 'long *')
ret = buf->output->attributes->GetInt (BMDDeckLinkMaximumAudioChannels, &max_channels);
^~~~~~~~~~~~~
./linux/DeckLinkAPI.h:692:87: note: passing argument to parameter 'value' here
virtual HRESULT GetInt (/* in */ BMDDeckLinkAttributeID cfgID, /* out */ int64_t *value) = 0;
^
Scale down to milliseconds, otherwise at least some hardware has problems
scheduling the frames (or schedules them too slow) and we run out of available
frames.
https://bugzilla.gnome.org/show_bug.cgi?id=770282
We have no idea which timestamps they are supposed to have so the only thing
we can do at this point is to drop them. Packets without timestamps happen if
audio was captured but no corresponding video, which shouldn't happen under
normal circumstances.
https://bugzilla.gnome.org/show_bug.cgi?id=747633
When the mode of decklinkvideosink is set to "auto", the sink claims to
support the full set of caps that it can support for all modes. Then, every
time new caps are set, the sink will automatically find the correct mode for
these caps and set it.
Caveat: We have no way to know whether a specific mode will actually work for
your hardware. Therefore, if you try sending 4K video to a 1080 screen, it
will silently fail, we have no way to know that in advance. Manually setting
that mode at least gave the user a way to double-check what they are doing.
https://bugzilla.gnome.org/show_bug.cgi?id=759600
Otherwise we're going to return times starting at 0 again after shutting down
an element for a specific input/output and then using it again later.
https://bugzilla.gnome.org/show_bug.cgi?id=755426
We were converting all times to our internal running times, that is the time
the sink itself spent in PLAYING already. But forgot to do that for the
running time calculated from the buffer timestamps. As such, all buffers were
scheduled much later if the pipeline's running time did not start at 0.
This happens for example if a base time is explicitly set on the pipeline.
https://bugzilla.gnome.org/show_bug.cgi?id=754528
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.
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.
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 fixes handling of flushing seeks, where we will get a PAUSED->PLAYING
state transition after the previous one without actually going to PAUSED
first.
