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
This commit creates a new utility library, eglutil.h, which contains code
to detect and use EGL_KHR_swap_buffers_with_damage or its EXT equivalent.
This logic used to be duplicated between the X11 and Wayland display servers,
which is not ideal.
When linking against libbfd.so, just passing libbfd.so to the compiler is
sufficient. When linking against the static version libbfd.a, however,
we must additionally link against libiberty.a and libz.a.
This commit adds a CMake helper to find the correct libraries that need
to be passed to link against libbfd correctly.
Using a macro ENABLE_OPENGL just like ENABLE_EGL to optionally remove
OpenGL implementation code. This is mostly because on Wayland it's just
a rehash of the EGL code (as EGL is the only way to create OpenGL
contexts on Wayland).
-Wno-sign-compare is used to suppress warnings related to comparing signed
values with unsigned ones. It's too pedantic.
-Wunused-parameter is also too pedantic, especially since all parameters
have to be named in C.
Otherwise, -Wextra lets us catch bugs, such as x < 0 for unsigned x.
On gcc, we pass -Wimplicit-fallthrough=2 so it will recognize our fall
through comment.
