This man page was referenced from some places. This is mostly a
condensed version of the POSIX behavior that the system call
implementation already has, only documenting the obviously visible
errors (in source code) we do actually report.
This change was a long time in the making ever since we obtained sample
rate awareness in the system. Now, each client has its own sample rate,
accessible via new IPC APIs, and the device sample rate is only
accessible via the management interface. AudioServer takes care of
resampling client streams into the device sample rate. Therefore, the
main improvement introduced with this commit is full responsiveness to
sample rate changes; all open audio programs will continue to play at
correct speed with the audio resampled to the new device rate.
The immediate benefits are manifold:
- Gets rid of the legacy hardware sample rate IPC message in the
non-managing client
- Removes duplicate resampling and sample index rescaling code
everywhere
- Avoids potential sample index scaling bugs in SoundPlayer (which have
happened many times before) and fixes a sample index scaling bug in
aplay
- Removes several FIXMEs
- Reduces amount of sample copying in all applications (especially
Piano, where this is critical), improving performance
- Reduces number of resampling users, making future API changes (which
will need to happen for correct resampling to be implemented) easier
I also threw in a simple race condition fix for Piano's audio player
loop.
Previously, strings would exit immediately if there was an error
changing file ownership. We now print an error to stderr and
continue when an error occurs.
Previously, the `-p` option printed the path of the file being
processed before any strings for that file. The `-f` prints the file
path before each string . This matches the behavior of strings on
Linux and FreeBSD.
This is a sensible separation of concerns that mirrors the WindowServer
IPC split. On the one hand, there is the "normal" audio interface, used
for clients that play audio, which is the primary service of
AudioServer. On the other hand, there is the management interface,
which, like the WindowManager endpoint, provides higher-level control
over clients and the server itself.
The reasoning for this split are manifold, as mentioned we are mirroring
the WindowServer split. Another indication to the sensibility of the
split is that no single audio client used the APIs of both interfaces.
Also, useless audio queues are no longer created for managing clients
(since those don't even exist, just like there's no window backing
bitmap for window managing clients), eliminating any bugs that may occur
there as they have in the past.
Implementation-wise, we just move all the APIs and implementations from
the old AudioServer into the AudioManagerServer (and respective clients,
of course). There is one point of duplication, namely the hardware
sample rate. This will be fixed in combination with per-client sample
rate, eliminating client-side resampling and the related update bugs.
For now, we keep one legacy API to simplify the transition.
The new AudioManagerServer also gains a hardware sample rate change
callback to have exact symmetry on the main server parameters (getter,
setter, and callback).
While this is a useful piece of information it means that diff is
producing hunks that are not of a valid normal diff format. This breaks
the ability to redirect the output of diff to a file to generate a
patch.
If more than one file is specified on the command line and the `-L`
option is used, the totals field will show the longest line
encountered; it is not a sum like the other values.
The intention for this utility is to eventually become a general-purpose
multimedia conversion tool like ffmpeg (except probably not with as many
supported formats, stream mappings and filters). For now, we can not
write any video format so the added complexity is not necessary at the
moment.
The JS::VM now owns the one Bytecode::Interpreter. We no longer have
multiple bytecode interpreters, and there is no concept of a "current"
bytecode interpreter.
If you ask for VM::bytecode_interpreter_if_exists(), it will return null
if we're not running the program in "bytecode enabled" mode.
If you ask for VM::bytecode_interpreter(), it will return a bytecode
interpreter in all modes. This is used for situations where even the AST
interpreter switches to bytecode mode (generators, etc.)
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.
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.
`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.
If we don't paint, SVG-as-image documents don't get laid out, and so
have 0x0 size throughout.
This change is also generally nice, as it makes the painting code run
on all the layout tests, increasing coverage. :^)
Previously, we would wait for the ping interval after the last ping
before displaying the closing statistics. We now display the closing
statistics and exit as soon as the required number of pings has been
performed.