This implements the ability to drag the timeline and volume buttons on
UA-rendered media controls. The two behave a bit differently:
Volume is updated as the user drags the volume button. This isn't a very
expensive operation, so updating in real-time and hearing the volume
change feels nice.
The current time, on the other hand, is not committed until the user
releases the mouse button. Performing a seek every time we get a mouse-
move event is pretty laggy, especially for video. However, we still want
to render updates on the timeline itself (so the position of the button
and the timestamp update as you drag). To do so, we internally pause the
media and override the timestamp provided to the layout node.
In the future, we may be able to seek video periodically to provide some
visual feedback. For example, we can seek after every N seconds of
scrubbing, or when the user pauses scrubbing for a while.
It's currently possible to seek to the total sample count of an audio
loader. We must limit seeking to one less than that count.
This mistake was duplicated in both AudioCodecPluginSerenity/Ladybird,
so the computation was moved to a helper in the base AudioCodecPlugin.
The `report_time_in_ms` and `speed_update_time_in_ms` variables
weren't previously being respected. This was causing the progress
display to update too frequently, making it difficult to read.
The intent of the spec is that the output of console.dir is interactable
within the console. Our Printer implementation currently just prints the
provided object as a string, and doesn't check the provided `options`
argument. But having console.dir defined prevents exceptions from being
thrown on real websites.
This change adds the TTL value of the inbound packet to the output of
the userland ping program, bringing it more in line with other common
ping utilities. It also adds the (optional) -t option to configure the
TTL of the outgoing packet if desired.
Previously, touch would exit immediately if there was an error
changing file permissions. We now print an error to stderr and
continue when an error occurs.
Instead of a custom struct, use an AK::Variant for flex-basis.
A flex-basis is either `content` or a CSS size value, so we don't need
anything custom for that.
By using a CSS size, we also avoid having to convert in and out of size
in various places, simplifying the code.
This finally gets rid of the "Unsupported main size for flex-basis"
debug spam. :^)
Return error when input svg is not valid and SVGSVGElement is not
present in the tree instead of doing svg_root nullptr dereference.
Fixes crash on https://apps.kde.org/en-gb/
When the default audio device changes on the host, it's convenient to
automatically switch to that device rather than needing to reload the
page to update.
To ensure actual PS2 code is not tied to the i8042 code, we make them
separated in the following ways:
- PS2KeyboardDevice and PS2MouseDevice classes are no longer inheriting
from the IRQHandler class. Instead we have specific IRQHandler derived
class for the i8042 controller implementation, which is used to ensure
that we don't end up mixing PS2 code with low-level interrupt handling
functionality. In the future this means that we could add a driver for
other PS2 controllers that might have only one interrupt handler but
multiple PS2 devices are attached, therefore, making it easier to put
the right propagation flow from the controller driver all the way to
the HID core code.
- A simple abstraction layer is added between the PS2 command set which
devices could use and the actual implementation low-level commands.
This means that the code in PS2MouseDevice and PS2KeyboardDevice
classes is no longer tied to i8042 implementation-specific commands,
so now these objects could send PS2 commands to their PS2 controller
and get a PS2Response which abstracts the given response too.
The HIDController class is removed and instead adding SerialIOController
class. The HIDController class was a mistake - there's no such thing in
real hardware as host controller only for human interface devices
(VirtIO PCI input controller being the exception here, but it could be
technically treated as serial IO controller too).
Instead, we simply add a new abstraction layer - the SerialIO "bus",
which will hold all the code that is related to serial communications
with other devices. A PS2 controller is simply a serial IO controller,
and the Intel 8042 Controller is simply a specific implementation of a
PS2 controller.
`FileSystem::absolute_path()` does `stat` the file, this commit runs
all `absolute_path` calls before touching the veil to make sure this
works as intended.
GCC's build fails in `libisl`'s configure step if `CC` is set to
Homebrew Clang with the message "Link Time Optimisation is not
supported". This is likely due to the fact that it tries to use ranlib
from Xcode, which is not compatible with the newer LLVM version's
bitcode format.
The toolchain build runs after `pick_host_compiler` is called, which
selects Homebrew Clang if the installed Xcode version is too old. We
need to unset `CC` and `CXX` for the toolchain build to sidestep the
issue.
Ideally, we would want the audio controller to run a channel at a
device's initial sample rate instead of hardcoding 44.1 KHz. However,
most audio is provided at 44.1 KHz and as long as `Audio::Resampler`
introduces significant audio artifacts, let's set a sensible sample
rate that offers a better experience for most users.
This can be removed after someone implements a higher quality
`Audio::Resampler`.
Specifying port 0 on the command line causes WebServer to select a
random available port. We now show the port WebServer is actually
using rather than assuming it is the same as the command line argument.
Adding undistributable space right before setting the content width is
incorrect when it's a percentage. Follow the specification and add it to
GRIDMIN and GRIDMAX instead.
We are currently forcing audio to play with a sample size of 16 bits. We
are also feeding the output audio device a hard-set amount of samples
without considering the actual size of its sample buffer. This would
cause a wide array of issues when playing audio elements. On my Linux
machine, we would hear some cracking; on my macOS machine, audio was
quite garbled.
We now write samples of the size requested by the output audio device.
We also limit the samples we provide to the audio device to however many
bytes are available in its buffer.
The main thread in the WebContent process is often busy with layout and
running JavaScript. This can cause audio to sound jittery and crack. To
avoid this behavior, we now drive audio on a secondary thread.
Note: Browser on Serenity uses AudioServer, the connection for which is
already handled on a secondary thread within LibAudio. So this only
applies to Lagom.
Rather than using LibThreading, our hands are tied to QThread for now.
Internally, the Qt media objects use a QTimer, which is forbidden from
running on a thread that is not a QThread (the debug console is spammed
with messages pointing this out). Ideally, in the future AudioServer
will be able to run for non-Serenity platforms, and most of this can be
aligned with the Serenity implementation.
