When computing the 'output mix', the Mixer iterates over all client
audio streams and computes a 'mixed sample' taking into account mainly
the client's volume.
This new member and methods will allow us to ignore a muted client
when computing that mix.
The `m_remaining_samples` attribute was underflowing at the end of an
audio stream. This fix guards against the underflow by only decrementing
the attribute when it is greater than zero.
I found this bug because the SoundPlayer userland application was not
correctly detecting when an audio stream was completed. This was
happening because the remaining samples being returned from the client
audio connection was an underflowed 16 bit integer instead of zero.
Executing `asctl set r 96000` no longer results in weird sample rates
being set on the audio devices. SB16 checks for a sample rate between 1
and 44100 Hz, while AC97 implements double-rate support which allows
sample rates between 8kHz and 96kHZ.
"Frame" is an MPEG term, which is not only unintuitive but also
overloaded with different meaning by other codecs (e.g. FLAC).
Therefore, use the standard term Sample for the central audio structure.
The class is also extracted to its own file, because it's becoming quite
large. Bundling these two changes means not distributing similar
modifications (changing names and paths) across commits.
Co-authored-by: kleines Filmröllchen <malu.bertsch@gmail.com>
Derivatives of Core::Object should be constructed through
ClassName::construct(), to avoid handling ref-counted objects with
refcount zero. Fixing the visibility means that misuses like this are
more difficult.
This commit is separate from the other Servives changes because it
required additional adaption of the code. Note that the old code did
precisely what these changes try to prevent: Create and handle a
ref-counted object with a refcount of zero.
Previously, AudioServer would deadlock when trying to play another audio
stream, i.e. creating a queue. The m_pending_cond condition was used
improperly, and the condition wait now happens independently of querying
the pending queue for new clients if the mixer is running.
To make the mixer's concurrency-safety code more readable, the use of
raw POSIX mutex and condition syscalls is replaced with Threading::Mutex
and Threading::ConditionVariable.
Across the entire audio system, audio now works in 0-1 terms instead of
0-100 as before. Therefore, volume is now a double instead of an int.
The master volume of the AudioServer changes smoothly through a
FadingProperty, preventing clicks. Finally, volume computations are done
with logarithmic scaling, which is more natural for the human ear.
Note that this could be 4-5 different commits, but as they change each
other's code all the time, it makes no sense to split them up.
AudioServer loads its settings, currently volume and mute state, from a
user config file "Audio.ini". Additionally, the current settings are
stored every ten seconds, if necessary. This allows for persistent audio
settings in between boots.
This way we don't have to allocate this at runtime. I'm intentionally
not using static constexpr here because that would put the variable
into the .rodata segment and would therefore increase the binary by
4kB.
The old code also failed to free() the buffer in the destructor, however
that wasn't much of an issue because the Mixer object exists throughout
the program's entire lifetime.
SPDX License Identifiers are a more compact / standardized
way of representing file license information.
See: https://spdx.dev/resources/use/#identifiers
This was done with the `ambr` search and replace tool.
ambr --no-parent-ignore --key-from-file --rep-from-file key.txt rep.txt *
Because it's what it really is. A frame is composed of 1 or more samples, in
the case of SerenityOS 2 (stereo). This will make it less confusing for
future mantainability.
