Not only if the video sink is set to PLAYING so far. Also give more
useful debug output about why we don't start, and don't start if already
started.
Also refactor the function to early-return instead of having a huge
if-else block over the whole function.
https://bugzilla.gnome.org/show_bug.cgi?id=790114
The Decklink and GstAudioBaseSink APIs don't fit very well together,
which causes various problems due to inaccuracies in the clock
calculations and the actual ringbuffer and GStreamer's copy getting of
sync.
Problems are audio drop-outs and A/V sync getting wrong after
pausing/seeking.
https://bugzilla.gnome.org/show_bug.cgi?id=790114
The "fields" flag is ignored because currently GStreamer doesn't support
having only top or only bottom fields inside a frame. The "drop frame"
flag is ignored because some occurrences have been spotted where it
wasn't set while it should have been. In practice, when we have 29.97 or
59.94 FPS, it's always drop-frame.
https://bugzilla.gnome.org/show_bug.cgi?id=790112
When we receive a video or audio buffer, we calculate the next stream
time based on the current stream time + buffer duration. If the next
buffer's stream time is after that, we issue a warning.
This happens because the stream time incoming from Decklink should be
really constant and without gaps. If there is a gap, it means that
something went wrong, e.g. the internal buffer pool is empty (too many
buffers queued up downstream).
https://bugzilla.gnome.org/show_bug.cgi?id=781776
If we drop many frames at once, printing one message per video frame and
one per audio packet would cause a lot of disk IO. Just print a total at
the end.
https://bugzilla.gnome.org/show_bug.cgi?id=788780
The buffer itself is 128 bytes into the allocated memory area, to be
able to store the size and other metadata before it. Freeing the buffer
directly will make malloc moderately unhappy.
HRESULT is unsigned long on Windows, but the Decklink headers define
it to 'int' on Linux. Confusingly, the defines that talk about the
possible return values for it use long constants. The easy fix would
be to change the linux/LinuxCOM.h header, but that's copied from the
decklink SDK.
Change the logging to always upcast to unsigned long while printing
HRESULT for consistency across platforms.
gstdecklinkvideosrc.cpp:425:7: warning: format '%x' expects argument of type 'unsigned int', but argument 8 has type 'HRESULT {aka long int}' [-Wformat]
[and so on]
gstdecklinkaudiosink.cpp:155:19: error: conflicting type attributes specified for 'virtual HRESULT GStreamerAudioOutputCallback::QueryInterface(const IID&, void**)'
In file included from /var/lib/jenkins/workspace/cerbero-cross-mingw32/workdir/mingw/w32/bin/../lib/gcc/i686-w64-mingw32/4.7.3/../../../../i686-w64-mingw32/include/objbase.h:153:0,
from /var/lib/jenkins/workspace/cerbero-cross-mingw32/workdir/mingw/w32/bin/../lib/gcc/i686-w64-mingw32/4.7.3/../../../../i686-w64-mingw32/include/ole2.h:16,
from /var/lib/jenkins/workspace/cerbero-cross-mingw32/workdir/mingw/w32/bin/../lib/gcc/i686-w64-mingw32/4.7.3/../../../../i686-w64-mingw32/include/windows.h:94,
from /var/lib/jenkins/workspace/cerbero-cross-mingw32/workdir/mingw/w32/bin/../lib/gcc/i686-w64-mingw32/4.7.3/../../../../i686-w64-mingw32/include/rpc.h:16,
from win/DeckLinkAPI.h:27,
from gstdecklink.h:35,
from gstdecklinkaudiosink.h:27,
from gstdecklinkaudiosink.cpp:25:
/var/lib/jenkins/workspace/cerbero-cross-mingw32/workdir/mingw/w32/bin/../lib/gcc/i686-w64-mingw32/4.7.3/../../../../i686-w64-mingw32/include/unknwn.h:67:25: error: overriding 'virtual HRESULT IUnknown::QueryInterface(const IID&, void**)'
(and many more)
https://ci.gstreamer.net/job/cerbero-cross-mingw32/6407/console
The default memory allocator of the decklink library allocates
a fixed pool of buffers, and the number of buffers is unknown.
This makes it impossible do useful queuing downstream. The new
memory allocator can create an unlimited number of buffers,
giving all queuing features one would expect from a live source.
https://bugzilla.gnome.org/show_bug.cgi?id=782556
This is basically a frame counter provided by the driver and it's
advancing at the speed of the HDMI/SDI input. Having this available on
each buffer allows to know what constant-framerate-based timestamp each
frame is corresponding to and can be used e.g. to write out files
accordingly without having the local pipeline clock timestamps used.
https://bugzilla.gnome.org/show_bug.cgi?id=779213
The audio packet times can be completely unrelated to the video stream
time, depending on the card. While this looks like a bug in the driver,
just always using the video stream time (which is correct) works as a
workaround for now.
This reverts commit 845832263b.
The commit broke cross-mingw CI:
https://ci.gstreamer.net/job/GStreamer-master/8659/console
It seems that cross-mingw on Autotools and native-mingw on Meson
disagree about the size of HRESULT. Revert for now till I can
investigate the Meson side of things some more.
MinGW does not provide comsupp.lib, so there's no implementation of
_com_util::ConvertBSTRToString. Use a fallback implementation that
uses wcstombs() instead.
On MinGW we also truncate the name to 100 chars which should be fine.
This is basically a frame counter provided by the driver and it's
advancing at the speed of the HDMI/SDI input. Having this available on
each buffer allows to know what constant-framerate-based timestamp each
frame is corresponding to and can be used e.g. to write out files
accordingly without having the local pipeline clock timestamps used.
https://bugzilla.gnome.org/show_bug.cgi?id=779213
"meson encountered an error in file
sys/decklink/meson.build, line 33, column 2:
Invalid use of addition: must be str, not list"
Also remove nonsensical linker flags on windows.
https://bugzilla.gnome.org/show_bug.cgi?id=781156
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.
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.
