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Previously, a libc-like out-of-line error information was used in the
loader and its plugins. Now, all functions that may fail to do their job
return some sort of Result. The universally-used error type ist the new
LoaderError, which can contain information about the general error
category (such as file format, I/O, unimplemented features), an error
description, and location information, such as file index or sample
index.
Additionally, the loader plugins try to do as little work as possible in
their constructors. Right after being constructed, a user should call
initialize() and check the errors returned from there. (This is done
transparently by Loader itself.) If a constructor caused an error, the
call to initialize should check and return it immediately.
This opportunity was used to rework a lot of the internal error
propagation in both loader classes, especially FlacLoader. Therefore, a
couple of other refactorings may have sneaked in as well.
The adoption of LibAudio users is minimal. Piano's adoption is not
important, as the code will receive major refactoring in the near future
anyways. SoundPlayer's adoption is also less important, as changes to
refactor it are in the works as well. aplay's adoption is the best and
may serve as an example for other users. It also includes new buffering
behavior.
Buffer also gets some attention, making it OOM-safe and thereby also
propagating its errors to the user.
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This consists of two changes: First, a utility function create_empty
allows the user to quickly create an empty buffer. Second, most creation
functions now return a NonnullRefPtr, as their failure causes a VERIFY
crash anyways.
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"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>
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The conversion from a linear scale (how we think about audio) to a
logarithmic scale (how audio actually works) will be useful for other
operations, so let's extract it to its own utility function. Its inverse
will also allow reversible operations to be written more easily.
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With logarithmic volume scaling, the delay effect can sound more
natural.
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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.
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LibDSP is a library for digital signal processing, and is primarily
intended to support the future DAW version of Piano.
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All audio applications (aplay, Piano, Sound Player) respect the ability
of the system to have theoretically any sample rate. Therefore, they
resample their own audio into the system sample rate.
LibAudio previously had its loaders resample their own audio, even
though they expose their sample rate. This is now changed. The loaders
output audio data in their file's sample rate, which the user has to
query and resample appropriately. Resampling code from Buffer, WavLoader
and FlacLoader is removed.
Note that these applications only check the sample rate at startup,
which is reasonable (the user has to restart applications when changing
the sample rate). Fully dynamic adaptation could both lead to errors and
will require another IPC interface. This seems to be enough for now.
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Floating-point ratios are inherently imprecise, and can lead to
unpredictable or nondeterministic behavior when resampling and expecting
a certain number of resulting samples. Therefore, the resampler now uses
integer ratios, with almost identical but fully predictable behavior.
This also introduces the reset() function that the FLAC loader will use
in the future.
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Previously, ResampleHelper was fixed on handling double's, which makes
it unsuitable for the upcoming FLAC loader that needs to resample
integers. For this reason, ResampleHelper is templated to support
theoretically any type of sample, though only the necessary i32 and
double are templated right now.
The ResampleHelper implementations are moved from WavLoader.cpp to
Buffer.cpp.
This also improves some imports in the WavLoader files.
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The signed 32-bit PCM sample format is required for the FLAC standard.
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LibAudio's WavLoader plugin for loading WAV files now supports loading
audio files with 32-bit float or 64-bit float samples.
By supporting these new non-int sample formats, Audio::Buffer now stores
the sample format (out of a list of supported formats) instead of the
raw bit depth. (The bit depth is easily calculated with
pcm_bits_per_sample)
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This makes it more symmetrical with adopt_own() (which is used to
create a NonnullOwnPtr from the result of a naked new.)
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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 *
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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.
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This was the last remaining user of shbufs! :^)
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