We were miscalculating the device period, i.e. the number of frames
we'll get from WASAPI in each IAudioClient::GetBuffer call, due to
a calculation mistake (truncate instead of round).
For example, on my machine when the aux input is set to 44.1KHz, the
reported device period is 101587, which comes out to 447.998 frames
per ::GetBuffer call. In reality we will, of course, get 448 frames
per call, but we were truncating, so we expected 447 and were
discarding one frame every time. This led to glitching, and skew over
time.
Interestingly, I can only see this with 44.1Khz. 48Khz/96Khz are fine,
because the device period is a more 'even' number.
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/806
gstwasapiutil.c(173) : warning C4715: 'gst_wasapi_device_role_to_erole': not all control paths return a value
gstwasapiutil.c(188) : warning C4715: 'gst_wasapi_erole_to_device_role': not all control paths return a value
The GstDeviceProvider isn't subclass of GstElement.
(gst-device-monitor-1.0:49356): GLib-GObject-WARNING **: 20:21:18.651:
invalid cast from 'GstWasapiDeviceProvider' to 'GstElement'
gstladspa.c:360:5: error: zero-length ms_printf format string [-Werror=format-zero-length]
vad_private.c:108:3: error: this decimal constant is unsigned only in ISO C90 [-Werror]
gstdecklinkvideosink.cpp:478:32: error: comparison between 'BMDTimecodeFormat {aka enum _BMDTimecodeFormat}' and 'enum GstDecklinkTimecodeFormat' [-Werror=enum-compare]
win/DeckLinkAPI_i.c:72:8: error: extra tokens at end of #endif directive [-Werror]
win/DeckLinkAPIDispatch.cpp:35:10: error: unused variable 'res' [-Werror=unused-variable]
gstwasapiutil.c:733:3: error: format '%x' expects argument of type 'unsigned int', but argument 8 has type 'DWORD' [-Werror=format]
gstwasapiutil.c:733:3: error: format '%x' expects argument of type 'unsigned int', but argument 9 has type 'guint64' [-Werror=format]
kshelpers.c:446:3: error: missing braces around initializer [-Werror=missing-braces]
kshelpers.c:446:3: error: (near initialization for 'known_property_sets[0].guid.Data4') [-Werror=missing-braces]
'channel-mask' field should not be put in caps if channel mask is 0x0
Mapping WASAPI channel mask to GST equivalent was going only over
first nChannels elements of wasapi_to_gst_pos array, translating, for
example, WASAPI's 0x63f to GST's 0x3f instead of 0xc3f.
When 'channel-mask' is specified as NULL, it signifies that there's
need to do downmix or upmix and it makes caps negotiation with
audioconvert element impossible. Just omit it.
Signed-off-by: Nirbheek Chauhan <nirbheek@centricular.com>
When the audio device goes away during playback or capture, we were
going into an infinite loop of AUDCLNT_E_DEVICE_INVALIDATED. Return -1
and post an error message so the ringbuffer thread exits with an error.
When either the source or sink goes from PLAYING -> NULL -> PLAYING,
we call _reset() which sets client_needs_restart, and then we call
prepare() which calls IAudioClient_Start(), so we don't need to call
it again in src_read() or sink_write(). Unlike when we're just going
PLAYING -> PAUSED -> PLAYING.
This is now handled directly in gstaudiosrc/sink, and we were setting
it in the wrong thread anyway. prepare() is not the same thread as
sink_write() or src_read().
With the Windows 8.1 SDK, the v1 of the AUDCLNT_STREAMOPTIONS enum is
defined which only has NONE and RAW, so it's not only defined when
AudioClient3 is available.
Add a meson check for the symbol. This is not needed for Autotools
because there we build against the MinGW audioclient.h which is still
at v1 of the AudioClient interface.
In case the wasapi buffer levels got low in shared mode we would still wait until
more buffer is available until writing something in it, which means we could never
catch up and recover.
Instead only wait for a new buffer in case the existing one is full and always write
what we can. Also don't loop until all data is written since the base class can handle
that for us and under normal circumstances this doesn't happen anyway.
This only works in shared mode, as in exclusive mode we have to exactly
fill the buffer and always have to wait first.
This fixes noisy (buffer underrun) playback with the wasapisink under load.
https://bugzilla.gnome.org/show_bug.cgi?id=796354
The calculation for the frame count in the non-aligned case resulted in
a one too low buffer frame count.
This resulted in:
1) exclusive mode not working as the frame count has to match
exactly there.
2) Buffer underruns in shared mode as the current write() code doesn't
handle catching up to low buffer levels (fixed in the next commit)
To fix just use the wasapi API to get the buffer size which will always
be correct.
https://bugzilla.gnome.org/show_bug.cgi?id=796354
S_FALSE is a valid return value which does not indicate an error.
For example IAudioClient_Stop() returns S_FALSE when it is already stopped.
Use the FAILED macro instead which just checks if an error occured or not.
This fixes spurious warnings when using the wasapisink element.
https://bugzilla.gnome.org/show_bug.cgi?id=796280
The clock seems to have a lot of drift (or we're using it incorrectly)
which causes buffers to be late on the sink and get dropped.
Disable till someone can investigate whether our usage of the API is
incorrect (it looked correct to me) or if something is wrong.
We can just return the template caps till the device is opened when
going from READY -> PAUSED. This fixes a CRITICAL when calling
ELEMENT_ERROR before the ringbuffer is allocated.
Also fixes a couple of leaks in error conditions.
https://bugzilla.gnome.org/show_bug.cgi?id=794611
Now, when you set loopback=true on wasapisrc, the `device` property
should refer to a sink (render) device for loopback recording.
If the `device` property is not set, the default sink device is used.
Directsoundsrc/sink have multiple issues, most of which cannot be
fixed at all because the API is deprecated and is implemented as a
compatibility wrapper around WASAPI since Vista.
Users and developers should now use the wasapisrc/sink elements, and
future development efforts should go towards that.
The low-latency property is *always* safe to enable, so applications
that do realtime communication should set it, and the elements will
automatically configure WASAPI to use the lowest possible device
period, and the audioringbuffer in audiobasesink will also be
configured accordingly.
Applications can also use exclusive mode during capture and playback
for the lowest possible latency if they know that the device will not
be used by any other application.
In this mode, the latency-time and buffer-time properties will be
completely ignored.
The AudioClient3 API is only available on Windows 10, and we will
automatically detect when it is available and use it.
However, using it for capturing audio with low latency and without
glitches seems to require setting the realtime priority of the entire
pipeline to "critical", which we cannot do from inside the element.
Hence, we can only enable that by default for wasapisink since
apps should be able to safely set the low-latency property to TRUE if
they need low-latency capture or playback.
This allows us to request ultra-low-latency device periods even in
shared mode. However, this requires good drivers and Windows 10, so
we only enable this when we detect that we are running on Windows 10
at runtime.
You can forcibly disable this feature on Windows 10 by setting
GST_WASAPI_DISABLE_AUDIOCLIENT3=1 in the environment.
