Commit Graph

1681 Commits

Author SHA1 Message Date
Geoffrey McRae
1cfbcba813 [client] main: fix failure to check KVMFR udata at connect 2022-02-08 15:50:22 +11:00
Paul Hollinsky
3890c72159 [client] egl: use texture sampler for desktop
The desktop doesn't need its own sampler, there is already an identically
configured one in the `desktop->texture`.

For some reason, using the texture sampler fixes a black screen issue
with my GTX 660 using the 470.86 driver. Maybe hitting some limit
for how many samplers can be allocated?
2022-02-08 15:34:54 +11:00
Paul Hollinsky
4223a5e38f [client] egl: remove extra sampler from TextureBuffer
It was unused, there is a sampler in EGL_Texture (base).
2022-02-08 15:34:54 +11:00
Tudor Brindus
fd28d0604e [host/client] kvmfr: request activation based on guest state 2022-02-08 15:27:27 +11:00
Tudor Brindus
9cd8027901 [client] main: request WM activation on first frame 2022-02-08 14:54:55 +11:00
Tudor Brindus
969ac4d1d1 [client] wayland: activate our window if we're allowed 2022-02-08 14:54:55 +11:00
Tudor Brindus
da548e3858 [client] ds: expose activation requesting 2022-02-08 14:54:55 +11:00
Tudor Brindus
21a349343b [client] wayland: implement activation request 2022-02-08 14:54:55 +11:00
Tudor Brindus
4ee6bdf198 [client] wayland: bind xdg_activation_v1 when available 2022-02-08 14:54:55 +11:00
Tudor Brindus
b13582a911 [client] wayland: build xdg-activation-v1 header 2022-02-08 14:54:55 +11:00
Chris Spencer
05ca59ed48 [client] audio/pw: increase startup latency
PipeWire startup latency varies wildly depending on what else is, or was
last using the audio device. In the worst case, PipeWire can request two
full buffers within a very short period of time immediately at the start of
playback, so make sure we've got enough data in the buffer to support this.
2022-02-04 16:27:12 +11:00
Chris Spencer
e1e60fdaa6 [client] audio: tune target latency
The target latency is now based upon the device maximum period size
(which may be configured by setting the `PIPEWIRE_LATENCY` environment
variable if using PipeWire), with some allowance for timing jitter from
Spice and the audio device.

PipeWire can change the period size dynamically at any time which must be
taken into account when selecting the target latency to avoid underruns
when the period size is increased. This is explained in detail within the
commit body.
2022-02-04 16:27:12 +11:00
Chris Spencer
ca29fe80a6 Revert "[client] audio: tune the target latency based on the latency jitter"
This reverts commit febd081202.

This causes severe underruns when the quantum size increases.
2022-02-04 16:27:12 +11:00
Geoffrey McRae
febd081202 [client] audio: tune the target latency based on the latency jitter 2022-01-28 12:11:56 +11:00
Geoffrey McRae
5bbc1d44bf [client] audio/pw: get the period size to determine when to start audio
Previously this was hardcoded to 100ms which is far too high in most
instances, instead we get the initial period size and use whichever is
greater out of 50ms or the period size.

The idea is to reduce the amount of time it takes for the latency to
come down after initial stream start.
2022-01-28 12:08:58 +11:00
Geoffrey McRae
22b968ff53 [client] audio: change the audio latency graph sample point
This removes the need for locking while also giving a better result in
the graph output. Also when the graph is disabled via the overlay
options it will no longer cause redraws.
2022-01-28 10:59:12 +11:00
Geoffrey McRae
a0477466d2 Revert "[client] audio: allow the audiodev to return the periodFrames"
This reverts commit 41884bfcc5.

PipeWire can change it's period size on the fly on us making this
approach invalid.
2022-01-28 10:00:35 +11:00
Geoffrey McRae
c2a766c2ee [client] audio: fix setfault due to failure to properly reset 2022-01-27 19:20:16 +11:00
Geoffrey McRae
a560a610d9 [client] audio: allow building without any audio support 2022-01-27 18:03:11 +11:00
Geoffrey McRae
a7db3d3a0f [client] audio: check for malloc failure 2022-01-27 18:03:11 +11:00
Geoffrey McRae
016001da67 [client] audio: cosmetics 2022-01-27 18:03:11 +11:00
Geoffrey McRae
41884bfcc5 [client] audio: allow the audiodev to return the periodFrames
This change allows the audiodevs to return the minimum period frames
needed to start playback instead of having to rely on a pull to obtain
these details.

Additionally we are using this information to select an initial start
latency as well as to train the desired latency in order to keep it as
low as possible.
2022-01-27 18:03:11 +11:00
Chris Spencer
dd2d84a080 [client] audio: adjust playback speed to match audio device clock
This change is based on the techniques described in [1] and [2].

The input audio stream from Spice is not synchronised to the audio playback
device. While the input and output may be both nominally running at 48 kHz,
when compared against each other, they will differ by a tiny fraction of a
percent. Given enough time (typically on the order of a few hours), this
will result in the ring buffer becoming completely full or completely
empty. It will stay in this state permanently, periodically resulting in
glitches as the buffer repeatedly underruns or overruns.

To address this, adjust the speed of the received data to match the rate at
which it is being consumed by the audio device. This will result in a
slight pitch shift, but the changes should be small and smooth enough that
this is unnoticeable to the user.

The process works roughly as follows:
1. Every time audio data is received from Spice, or consumed by the audio
   device, sample the current time. These are fed into a pair of delay
   locked loops to produce smoothed approximations of the two clocks.
2. Compute the difference between the two clocks and compare this against
   the target latency to produce an error value. This error value will be
   quite stable during normal operation, but can change quite rapidly due
   to external factors, particularly at the start of playback. To smooth
   out any sudden changes in playback speed, which would be noticeable to
   the user, this value is also filtered through another delay locked loop.
3. Feed this error value into a PI controller to produce a ratio value.
   This is the target playback speed in order to bring the error value
   towards zero.
4. Resample the input audio using the computed ratio to apply the speed
   change. The output of the resampler is what is ultimately inserted into
   the ring buffer for consumption by the audio device.

Since this process targets a specific latency value, rather than simply
trying to rate match the input and output, it also has the effect of
'correcting' latency issues. If a high latency application (such as a media
player) is already running, the time between requesting the start of
playback and the audio device actually starting to consume samples can be
very high, easily in the hundreds of milliseconds. The changes here will
automatically adjust the playback speed over the course of a few minutes to
bring the latency back down to the target value.

[1] https://kokkinizita.linuxaudio.org/papers/adapt-resamp.pdf
[2] https://kokkinizita.linuxaudio.org/papers/usingdll.pdf
2022-01-27 18:03:11 +11:00
Chris Spencer
599fdd6ffd [common] ringbuffer: add unbounded mode
In unbounded mode, the read and write pointers are free to move
independently of one another. This is useful where the input and output
streams are progressing at the same rate on average, and we want to keep
the latency stable in the event than an underrun or overrun occurs.

If an underrun occurs (i.e., there is not enough data in the buffer to
satisfy a read request), the missing values with be filled with zeros. When
the writer catches up, the same number of values will be skipped from the
input.

If an overrun occurs (i.e., there is not enough free space in the buffer to
satisfy a write request), excess values will be discarded. When the reader
catches up, the same number of values will be zeroed in the output.

Unbounded mode is currently unused since our audio input and output
streams are not synchronised. This will be implemented in a later commit.

Also reimplemented as a lock-free queue which is safer for use in audio
device callbacks.
2022-01-27 18:03:11 +11:00
Chris Spencer
b34b253814 [client] audio: stop playback immediately if new playback is started
If a new playback is started while the previous playback is still flushing,
we simply allow the stream to continue playing and effectively cancel the
flush. In general this is not safe because there may not be enough data in
the buffer to avoid underrunning. We could handle this better later by
trying to insert the right number of silent samples into the buffer, but
for now just completely stop the previous stream before starting the new
one.
2022-01-27 18:03:11 +11:00
Chris Spencer
68b42e1c1a [client] audio/pw: drop restarting state
Automatically restarting playback once draining has completed could result
in playback starting too early (i.e., before there is enough data in the
ring buffer to avoid underrunning). `audio_playbackData` will keep invoking
`start` until it returns true anyway, so we can just allow draining to
complete normally and wait for `start` to be called again.
2022-01-27 18:03:11 +11:00
Chris Spencer
8580978321 [client] audio/pw: drop redundant flushing state
We do not stop the audio device until after the internal buffer has already
been flushed, so this state does nothing useful.
2022-01-27 18:03:11 +11:00
Geoffrey McRae
b0568ca404 [client] egl: use a sigmoid curve for nv 2022-01-26 23:03:35 +11:00
Geoffrey McRae
3c9b9e6370 [client] main: fix heap-buffer-overflow on cursor update 2022-01-26 20:05:51 +11:00
Geoffrey McRae
efa49391fc [client] fix race segfault on pointer queue unsubscribe/timeout 2022-01-26 17:20:12 +11:00
Geoffrey McRae
fb4bdaee2b [client] egl: set the active preset if specified at launch 2022-01-26 16:07:17 +11:00
Geoffrey McRae
c7389285f9 [client] egl: fix null pointer access when no preset is set 2022-01-26 16:04:15 +11:00
Geoffrey McRae
aa426d13a7 [client] egl: added egl:preset to load a default preset at startup 2022-01-26 16:00:07 +11:00
Geoffrey McRae
89c83dafc1 [client] egl: make egl less noisy unless debug is specified 2022-01-26 15:42:33 +11:00
Geoffrey McRae
05e363e009 [client] x11: cleanup duplicated code 2022-01-26 15:30:38 +11:00
Geoffrey McRae
e17b289759 [client] x11: sync the mouse on meta resize 2022-01-26 14:55:30 +11:00
Geoffrey McRae
79e986cc60 [client] x11: fix failure to unfocus when performing meta resize/move 2022-01-26 14:43:11 +11:00
Geoffrey McRae
22f3cf5ba6 [client] egl: fix masked color cursor blend operation for xor drawing 2022-01-26 12:23:05 +11:00
Geoffrey McRae
3067bdaa15 [client] egl: properly apply xor mask to masked color cursors 2022-01-26 12:11:43 +11:00
Quantum
11800029f0 [client] core: redraw cursor after warping guest cursor 2022-01-24 17:07:15 +11:00
Quantum
71901414d1 [client] overlay: realign cursor when turning off overlay
This is only done in non-capture mode to avoid messing up games.
2022-01-24 17:07:15 +11:00
Geoffrey McRae
96fa8891c8 [client] egl: fixed incorrect drawing of masked color cursors 2022-01-24 06:56:32 +11:00
Geoffrey McRae
1082875b8e [client] opengl: fix startup since ImGui upgrade 2022-01-24 06:42:16 +11:00
Geoffrey McRae
dc918c55b6 [client] main: only copy the needed cursor data instead of everything 2022-01-24 06:36:15 +11:00
Geoffrey McRae
a8ba014b52 [client] main: lgmpClientMessageDone is not idempotent 2022-01-24 04:10:49 +11:00
Quantum
9a6aa3ce66 [client] egl: remove duplicate #include "app.h" 2022-01-23 08:49:15 +11:00
Quantum
f2fbb2b27c [client] opengl: make draw functions static 2022-01-23 08:49:01 +11:00
Geoffrey McRae
9601bc677f [client] audio: report the buffered frames and not the buffer length 2022-01-22 16:06:16 +11:00
Geoffrey McRae
c84879717f [client] audio: fix failure to properly handle restart if draining 2022-01-20 07:02:00 +11:00
Geoffrey McRae
b3c81bcedf [client] audio: fix use after free race 2022-01-19 18:50:39 +11:00