With this, the WAV loader is a completely modern LibAudio loader:
- Own type header for RIFF data structures
- custom stream read functions for the types
- Final removal of legacy I/O error checking
- clearer error messages
- clean handling of header chunks
The latter will allow proper handling of other chunks (before "data") in
the future, such as metadata :^)
That's what this class really is; in fact that's what the first line of
the comment says it is.
This commit does not rename the main files, since those will contain
other time-related classes in a little bit.
`WavWriter` can be constructed without a file, which should probably be
made impossible at some point. For now, let's not crash `Piano` when you
close the application.
This was used in exactly one place, to avoid sending multiple
CustomEvents to the enqueuer thread in Audio::ConnectionToServer.
Instead of this, we now just send a CustomEvent and wake the enqueuer
thread. If it wakes up and has multiple CustomEvents, they get delivered
and ignored in no time anyway. Since they only get ignored if there's
no work to be done, this seems harmless.
This is a special case of the sample count field in the header which we
treated as a format error before. Now we just take care to check stream
EOF before reading chunks.
This makes the final FLAC spec test pass, making us one of the most
compliant loaders! :^)
We report a rounded up PCM sample format to the outside, but use the
exact bit depth as specified in header and frames.
This makes the three FLAC spec tests with a a bit depth of 20 pass.
"Improve" is an understatement, since this commit makes all FLAC files
seek without errors, mostly with high accuracy, and partially even fast:
- A new generic seek table type is introduced, which keeps an
always-sorted list of seek points, which allows it to use binary
search and fast insertion.
- Automatic seek points are inserted according to two heuristics
(distance between seek points and minimum seek precision), which not
only builds a seek table for already-played sections of the file, but
improves seek precision even for files with an existing seek table.
- Manual seeking by skipping frames works properly now and is still used
as a last resort.
This container has several design goals:
- Represent all common and relevant metadata fields of audio files in a
unified way.
- Allow perfect recreation of any metadata format from the in-memory
structure. This requires that we allow non-detected fields to reside
in an "untyped" miscellaneous collection.
Like with pictures, plugins are free to store their metadata into the
m_metadata field whenever they read it. It is recommended that this
happens on loader creation; however failing to read metadata should not
cause an error in the plugin.
Similar to POSIX read, the basic read and write functions of AK::Stream
do not have a lower limit of how much data they read or write (apart
from "none at all").
Rename the functions to "read some [data]" and "write some [data]" (with
"data" being omitted, since everything here is reading and writing data)
to make them sufficiently distinct from the functions that ensure to
use the entire buffer (which should be the go-to function for most
usages).
No functional changes, just a lot of new FIXMEs.
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
Turns out that, if we don't use functions that ensure reading until the
very end of the buffer, we only end up getting the very beginning of
samples and fill the rest with uninitialized data.
While at it, make sure that we read the data that is little endian as a
LittleEndian.
For example, consider cases where we want to propagate errors only in
specific instances:
auto result = read_data(); // something like ErrorOr<ByteBuffer>
if (result.is_error() && result.error().code() != EINTR)
continue;
auto bytes = TRY(result);
The TRY invocation will currently copy the byte buffer when the
expression (in this case, just a local variable) is stored into
_temporary_result.
This patch binds the expression to a reference to prevent such copies.
In less trival invocations (such as TRY(some_function()), this will
incur only temporary lifetime extensions, i.e. no functional change.
`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.
