The recent `pwnkit` exploit brought this to my attention, not that we
are a setuid process we should still do this properly... who knows where
this code might get used in the future.
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.
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.
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.
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
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.
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.
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.
To avoid client showing "Using : NVFBC (NVidia Frame Buffer Capt".
This happens because the string is truncated to 31 characters to fit
in the char capture[32]; member of KVMFRRecord_VMInfo.
For Windows 10, it so happens that the major.minor is 10.0. This is not
usually a given, e.g. on Windows 7 where it would read 6.1, on
Windows 8 it would read 6.2, and on Windows 8.1 it would read 6.3.
This is obviously undesirable, so we should just read the ProductName
from registry if possible. This results in something like:
OS Name: Windows 10 Pro for Workstations (Build: 19043)
