When the sink goes from PLAYING to READY and then back to PLAYING,
the initialization of the audioclient in prepare() fails with the
error AUDCLNT_E_ALREADY_INITIALIZED. As a result, the playback
stops.
To fix this, we need to drop the AudioClient in unprepare() and
grab a new one in prepare() to be able to initialize it again
with the new buffer spec.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2096>
The functionality now resides in
gst_wasapi_util_get_device() and
gst_wasapi_util_get_audio_client().
This is a preparatory patch. It will be used in the following
patch to init/deinit the AudioClient separately from the device.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2096>
An oddness of wasapi loopback feature is that capture client will not
produce any data if there's no outputting sound to corresponding
render client. In other words, if there's no sound to render,
capture task will stall. As an option to solve such issue, we can
add timeout to wake up from capture thread if there's no incoming data
within given time interval. But it seems to be glitch prone.
Another approach is that we can keep pushing silence data into
render client so that capture client can keep capturing data
(even if it's just silence).
This patch will choose the latter one because it's more straightforward
way and it's likely produce glitchless sound than former approach.
A bonus point of this approach is that loopback capture on Windows7/8
will work with this patch. Note that there's an OS bug prior to Windows10
when loopback capture client is running with event-driven mode.
To work around the bug, event signalling should be handled manually
for read thread to wake up.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1588>
We don't need to duplicate a method for HRESULT error code to string
conversion. This patch is intended to
* Remove duplicated code
* Ensure FormatMessageW (Unicode version) and avoid FormatMessageA
(ANSI version), as the ANSI format is not portable at all.
Note that if "UNICODE" is not defined, FormatMessageA will be aliased
as FormatMessage by default.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1442>
The whole `src_read()` function is a hot loop since the ringbuffer
thread is waiting on us, and printing to the console from inside it
can easily cause us to miss our deadline.
F.ex., if you had GST_DEBUG=3 and we accidentally missed a device
period, we'd trigger the "reported glitch" warning, which would cause
us to miss another device period, and so on. Let's reduce the log
level so that GST_DEBUG=3 is more usable, and only print buffer flag
info when it's actually relevant.
Some audio drivers return varying amounts of data per ::GetBuffer
call, instead of following the device period that they've told us
about in `src_prepare()`.
Previously, we would just drop those extra buffers hoping that the
extra buffers were temporary (f.ex., a startup 'burst' of audio data).
However, it seems that some audio drivers, particularly on older
Windows versions (such as Windows 10 1703 and older) consistently
return varying amounts of data.
Use GstAdapter to smooth that out, and hope that the audio driver is
locally varying but globally periodic.
Initially reported in https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/808
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.
