Let's replace this bool with an `enum class` in order to enhance
readability. This is done by repurposing `MappedFile`'s `OpenMode` into
a shared `enum` simply called `Mode`.
This is 10-20% of a speed increase on platforms with fast I/O (Linux)
and not a slowdown on Serenity. Again, the file system layer is the
limit for us :^)
The `Loader` uses a `Vector<Sample>` to store any samples that the
actual audio decoder returned that the loader client did not request
yet. However, it did not take that buffer into account when those
clients asked for the number of samples loaded. The buffer is an
internal implementation detail, and should not be reflected on the
external API.
This allows LibWeb to keep better track of audio position without any
desync caused by the buffer. When using the Serenity `PlaybackStream`
implementation as the back end for an `HTMLAudioElement`, this allows
us to perfectly stop on the exact last sample of the audio file, so it
will not stop before the media element can see that it has finished
playback.
Note that `Loader::get_more_samples()` also calls `loaded_samples()` to
determine how many samples are remaining to load into the output
buffer. However, this change appears to be correct even there, given
that the samples copied from the internal sample buffer are included in
that count.
Bytes will implicitly cast to StringView, but not to ReadonlyBytes. That
means that if you call
`Audio::Loader::create_plugin(mapped_file->bytes())`
it will silently use the `create_plugin(StringView path)` overload.
Reading audio data does not require that data to be writable, so let's
use ReadonlyBytes for it and avoid the footgun.
Especially FLAC had an issue here before, but the loader infrastructure
itself wouldn't handle end of stream properly if the "available samples"
information didn't match up.
It's no longer needed now that this code uses ErrorOr instead of Result.
Ran:
rg -lw LOADER_TRY Userland/Libraries/LibAudio \
| xargs sed -i '' 's/LOADER_TRY/TRY/g'
...and then manually fixed up Userland/Libraries/LibAudio/LoaderError.h
to not redefine TRY but instead remove the now-unused LOADER_TRY,
and ran clang-format.
This removes a lot of duplicated stream creation code from the plugins,
and also simplifies the way that the appropriate plugin is found. This
mirrors the ImageDecoderPlugin design and necessitates new sniffing
methods on the loaders.
Before, some loader plugins implemented their own buffering (FLAC&MP3),
some didn't require any (WAV), and some didn't buffer at all (QOA). This
meant that in practice, while you could load arbitrary amounts of
samples from some loader plugins, you couldn't do that with some others.
Also, it was ill-defined how many samples you would actually get back
from a get_more_samples call.
This commit fixes that by introducing a layer of abstraction between the
loader and its plugins (because that's the whole point of having the
extra class!). The plugins now only implement a load_chunks() function,
which is much simpler to implement and allows plugins to play fast and
loose with what they actually return. Basically, they can return many
chunks of samples, where one chunk is simply a convenient block of
samples to load. In fact, some loaders such as FLAC and QOA have
separate internal functions for loading exactly one chunk. The loaders
*should* load as many chunks as necessary for the sample count to be
reached or surpassed (the latter simplifies loading loops in the
implementations, since you don't need to know how large your next chunk
is going to be; a problem for e.g. FLAC). If a plugin has no problems
returning data of arbitrary size (currently WAV), it can return a single
chunk that exactly (or roughly) matches the requested sample count. If a
plugin is at the stream end, it can also return less samples than was
requested! The loader can handle all of these cases and may call into
load_chunk multiple times. If the plugin returns an empty chunk list (or
only empty chunks; again, they can play fast and loose), the loader
takes that as a stream end signal. Otherwise, the loader will always
return exactly as many samples as the user requested. Buffering is
handled by the loader, allowing any underlying plugin to deal with any
weird sample count requirement the user throws at it (looking at you,
SoundPlayer!).
This (not accidentally!) makes QOA work in SoundPlayer.
Brought to you by the inventor of QOI, QOA is a lossy audio codec that
is, as the name says, quite okay in compressing audio data reasonably
well without frequency transformation, mostly introducing some white
noise in the background. This implementation of QOA is fully compatible
with the qoa.h reference implementation as of 2023-02-25. Note that
there may be changes to the QOA format before a specification is
finalized, and there is currently no information on when that will
happen and which changes will be made.
This implementation of QOA can handle varying sample rate and varying
channel count files. The reference implementation does not produce these
files and cannot handle them, so their implementation is untested.
The QOA loader is capable of seeking in constant-bitrate streams.
QOA links:
https://phoboslab.org/log/2023/02/qoa-time-domain-audio-compressionhttps://github.com/phoboslab/qoa
`Stream` will be qualified as `AK::Stream` until we remove the
`Core::Stream` namespace. `IODevice` now reuses the `SeekMode` that is
defined by `SeekableStream`, since defining its own would require us to
qualify it with `AK::SeekMode` everywhere.
This doesn't have any immediate uses, but this adapts the code a bit
more to `Core::Stream` conventions (as most functions there use
NonnullOwnPtr to handle streams) and it makes it a bit clearer that this
pointer isn't actually supposed to be null. In fact, MP3LoaderPlugin
and FlacLoaderPlugin apparently forgot to check for that completely
before starting to decode data.
This now prepares all the needed (fallible) components before actually
constructing a LoaderPlugin object, so we are no longer filling them in
at an arbitrary later point in time.
`Bytes` is very slim, so the memory and/or performance gains from
passing it by reference isn't that big, and it passing it by value is
more compatible with xvalues, which is handy for things like
`::try_create(buffer.bytes())`.
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.
This commit adds a loader for the FLAC audio codec, the Free Lossless
Audio codec by the Xiph.Org foundation. LibAudio will automatically
read and parse FLAC files, so users do not need to adjust.
This implementation is bare-bones and needs to be improved upon.
There are many bugs, verbatim subframes and any kind of seeking is
not supported. However, stereo files exported by libavcodec on
highest compression setting seem to work well.
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 *
