Same changes as done for wasapisink in cbe2fc40a. Turns out this is
sometimes also needed for capture. Reported by Mathieu_Du.
Also improve logging in that case for easier debugging.
Sometimes the minimum period advertised by a card results in an
unaligned buffer size error during initialization in exclusive mode.
In that case, we can fetch the actual buffer size in frames and
calculate the period from that.
We can't do this pre-emptively because we can't call GetBufferSize
till Initialize has been called at least once.
https://bugzilla.gnome.org/show_bug.cgi?id=793289
This reduces the chances of startup glitches, and also reduces the
chances that we'll get garbled output due to driver bugs.
Recommended by the WASAPI documentation.
https://bugzilla.gnome.org/show_bug.cgi?id=793289
So far, we have been completely discarding the values of latency-time
and buffer-time and trying to always open the device in the lowest
latency mode possible. However, sometimes this is a bad idea:
1. When we want to save power/CPU and don't want low latency
2. When the lowest latency setting causes glitches
3. Other audio-driver bugs
Now we will try to follow the user-set values of latency-time and
buffer-time in shared mode, and only latency-time in exclusive mode (we
have no control over the hardware buffer size, and there is no use in
setting GstAudioRingBuffer size to something larger).
The elements will still try to open the devices in the lowest latency
mode possible if you set the "low-latency" property to "true".
https://bugzilla.gnome.org/show_bug.cgi?id=793289
This requires using allocated strings, but it's the best option. For
instance, a call could fail because CoInitialize() wasn't called, or
because some other thing in the stack failed.
https://bugzilla.gnome.org/show_bug.cgi?id=793289
This is particularly important when running in exclusive mode because
any delays will immediately cause glitching.
The MinGW version in Cerbero is too old, so we can only enable this when
building with MSVC or when people build GStreamer for MSYS2 or other
MinGW-based distributions.
To force-enable this code when building with MinGW, build with
CFLAGS="-DGST_FORCE_WIN_AVRT -lavrt".
https://bugzilla.gnome.org/show_bug.cgi?id=793289
This provides much lower latency compared to opening in shared mode,
but it also means that the device cannot be opened by any other
application. The advantage is that the achievable latency is much
lower.
In shared mode, WASAPI's engine period is 10ms, and so that is the
lowest latency achievable.
In exclusive mode, the limit is the device period itself, which in my
testing with USB DACs, on-board PCI sound-cards, and HDMI cards is
between 2ms and 3.33ms.
We set our audioringbuffer limits to match the device, so the
achievable sink latency is 6-9ms. Further improvements can be made if
needed.
https://bugzilla.gnome.org/show_bug.cgi?id=793289
We will use ->device for storing a pointer to the IMMDevice structure
which is needed for fetching the caps supported by devices in
exclusive mode.
https://bugzilla.gnome.org/show_bug.cgi?id=793289
This will set the actual-latency-time and actual-buffer-time of the sink
and source.
We completely ignore the latency-time/buffer-time values set
on the element because WASAPI is happiest when it is reading/writing at
the default period. Improving this will likely require the use of the
IAudioClient3 interfaces which are not available in MinGW yet.
https://bugzilla.gnome.org/show_bug.cgi?id=792897
Currently only does probing and does not handle messages from
endpoints/devices. In the future we want to do proper monitoring which
is well-supported in WASAPI.
https://bugzilla.gnome.org/show_bug.cgi?id=792897
We need to parse the WAVEFORMATEXTENSIBLE structure, figure out what
positions the channels have (if they are positional), and reorder them
as necessary.
https://bugzilla.gnome.org/show_bug.cgi?id=792897
There is no fixed limitation for the number of devices on the
decklink API side according to BlackMagic. Many PC motherboards
are able support 6 decklink cards each with up to 8 inputs so
a limit of 16 might well be too low.
https://bugzilla.gnome.org/show_bug.cgi?id=777239
Both the source and the sink elements were broken in a number of ways:
* prepare() was assuming that the format was always S16LE 2ch 44.1KHz.
We now probe the preferred format with GetMixFormat().
* Device initialization was done with the wrong buffer size
(buffer_time is in microseconds, not nanoseconds).
* sink_write() and src_read() were just plain wrong and would never
write or read anything useful.
* Some functions in prepare() were always returning FALSE which meant
trying to use the elements would *always* fail.
* get_caps() and delay() were not implemented at all.
TODO: support for >2 channels
TODO: pro-audio low-latency
TODO: SPDIF and other encoded passthroughs
Three new properties are now implemented: role, mute, and device.
* 'role' designates the stream role of the initialized device, see:
https://msdn.microsoft.com/en-us/library/windows/desktop/dd370842(v=vs.85).aspx
* 'device' is a system-wide GUIDesque string for a specific device.
* 'mute' is a sink property and simply mutes it.
On my Windows 8.1 system, the lowest latency that works is:
wasapisrc buffer-time=20000
wasapisink buffer-time=10000
aka, 20ms and 10ms respectively. These values are close to the lowest
possible with the IAudioClient interface. Further improvements require
porting to IAudioClient2 or IAudioClient3.
https://docs.microsoft.com/en-us/windows-hardware/drivers/audio/low-latency-audio
Sometimes we might get an audio packet without a corresponding video
frame. In these cases, the stream and hardware reference timestamps
would be missing, because they're called on the video frame. Instead of
potentially breaking stuff downstream that might depend on these, we now
extrapolate them.
https://bugzilla.gnome.org/show_bug.cgi?id=792042
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
Sometimes we might get an audio packet without a corresponding video
frame. In these cases, the stream and hardware reference timestamps
would be missing, because they're called on the video frame. Instead of
potentially breaking stuff downstream that might depend on these, we now
extrapolate them.
https://bugzilla.gnome.org/show_bug.cgi?id=792042
The correct behaviour of anything stuck in the ->render() function
between ->unlock() and ->unlock_stop() is to call
gst_base_sink_wait_preroll() and only return an error if this returns an
error, otherwise, it must continue where it left off!
https://bugzilla.gnome.org/show_bug.cgi?id=774950
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
When we cannot scale, we need to enforce the pixel aspect ratio.
This was partly implemented in the previous patch. Doing this
simplify some of the code.
https://bugzilla.gnome.org/show_bug.cgi?id=784599
1. Similar to 880f3d8, don't consider not getting an output buffer as
an error during flushing. I've seen the following sometimes when
encoding:
W GStreamer+amcvideoenc: java.lang.IllegalStateException
W GStreamer+amcvideoenc: at android.media.MediaCodec.getBuffer(Native Method)
W GStreamer+amcvideoenc: at android.media.MediaCodec.getOutputBuffer(MediaCodec.java:2886)
2. For amcvideodec/enc, call _find_nearest_frame (which grabs a fresh
reference on a GstVideoCodecFrame) after we have an output buffer,
so as to not leak the reference, in case getting an output buffer
fails.
Otherwise, if we get an error grabbing the output buffer, we leak
the reference to the frame. This can cause issues with a
v4l2bufferpool feeding the encoder not being able to clean itself
up properly due to buffers still being marked as in-use.
https://bugzilla.gnome.org/show_bug.cgi?id=791258
This is to be used with gst_video_overlay_set_render_rectangle()
so the application can calculate a rectangle that fits inside
the display. The property changes are notify in a way that you
can watch either notify::display-width or notify::display-height
and both will be up-to-data when this is called back. Before the
element is started, the size will be 0x0.
https://bugzilla.gnome.org/show_bug.cgi?id=784599
Implement videooverlay interface in kmssink, divided into two cases:
when driver supports scale, then we do refresh in show_frame(); if
not, send a reconfigure event to upstream and re-negotiate, using the
new size.
https://bugzilla.gnome.org/show_bug.cgi?id=784599
If the driver requires more data, just unref the frame at the moment
then retreive/finish the frame after encoding is finished.
This also fixes a memory leak.
https://bugzilla.gnome.org/show_bug.cgi?id=790312
Fixes outputted frame sequence when performing a seek
i.e. when seeking backwards, the first frame after the seek was a frame
from the future. This would result in GstVideoDecoder essentially
marking all the timestamps as essentially bogus and the base class would
attempt to compensate. A visible indication of this was 'decreasing timestamp'
warning after a seek.
https://bugzilla.gnome.org/show_bug.cgi?id=790478
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
Now that we are doing lazy allocation, we may endup calling _stop()
before the allocator was created. As a side effect, we need to nul-check
the pointer before calling it's method (_clear_cache()).
https://bugzilla.gnome.org/show_bug.cgi?id=787593
DRM_RDWR was not defined until libdrm 2.4.68. However,
in configure.ac we only require libdrm >= 2.4.55.
Seems silly to to bump minimum libdrm version for a simple
define. Thus, define DRM_RDWR if it's not defined.
This fixes compilation error introduced in:
commit 922031b0f9
Author: Nicolas Dufresne <nicolas.dufresne@collabora.com>
Date: Tue Sep 12 12:07:13 2017 -0400
kms: Export DMABuf from Dumb buffer when possible
https://bugzilla.gnome.org/show_bug.cgi?id=787593
Signed-off-by: U. Artie Eoff <ullysses.a.eoff@intel.com>
When we guess the strides, we need to also update the GstVideoInfo.size
otherwise the memory size will be set to something smaller then needed.
This was causing crash with the DMABuf exportation, since we would not
mmap() a large enough buffer.
https://bugzilla.gnome.org/show_bug.cgi?id=787593
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.
