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>
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>
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>
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().
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
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.
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.
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 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
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
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
Original commit message from CVS:
* sys/Makefile.am:
* sys/wasapi/Makefile.am:
* sys/wasapi/gstwasapi.c:
* sys/wasapi/gstwasapisink.c:
* sys/wasapi/gstwasapisink.h:
* sys/wasapi/gstwasapisrc.c:
* sys/wasapi/gstwasapisrc.h:
* sys/wasapi/gstwasapiutil.c:
* sys/wasapi/gstwasapiutil.h:
New plugin for audio capture and playback using Windows Audio Session
API (WASAPI) available with Vista and newer (#520901).
Comes with hardcoded caps and obviously needs lots of love. Haven't
had time to work on this code since it was written, was initially just
a quick experiment to play around with this new API.
