Error::from_string_literal now takes direct char const*s, while
Error::from_string_view does what Error::from_string_literal used to do:
taking StringViews. This change will remove the need to insert `sv`
after error strings when returning string literal errors once
StringView(char const*) is removed.
No functional changes.
This has been overkill from the start, and it has been bugging me for a
long time. With this change, we're probably a bit slower on most
platforms but save huge amounts of space with all in-memory sample
datastructures.
The file is now renamed to Queue.h, and the Resampler APIs with
LegacyBuffer are also removed. These changes look large because nobody
actually needs Buffer.h (or Queue.h). It was mostly transitive
dependencies on the massive list of includes in that header, which are
now almost all gone. Instead, we include common things like Sample.h
directly, which should give faster compile times as very few files
actually need Queue.h.
Previously, we were sending Buffers to the server whenever we had new
audio data for it. This meant that for every audio enqueue action, we
needed to create a new shared memory anonymous buffer, send that
buffer's file descriptor over IPC (+recfd on the other side) and then
map the buffer into the audio server's memory to be able to play it.
This was fine for sending large chunks of audio data, like when playing
existing audio files. However, in the future we want to move to
real-time audio in some applications like Piano. This means that the
size of buffers that are sent need to be very small, as just the size of
a buffer itself is part of the audio latency. If we were to try
real-time audio with the existing system, we would run into problems
really quickly. Dealing with a continuous stream of new anonymous files
like the current audio system is rather expensive, as we need Kernel
help in multiple places. Additionally, every enqueue incurs an IPC call,
which are not optimized for >1000 calls/second (which would be needed
for real-time audio with buffer sizes of ~40 samples). So a fundamental
change in how we handle audio sending in userspace is necessary.
This commit moves the audio sending system onto a shared single producer
circular queue (SSPCQ) (introduced with one of the previous commits).
This queue is intended to live in shared memory and be accessed by
multiple processes at the same time. It was specifically written to
support the audio sending case, so e.g. it only supports a single
producer (the audio client). Now, audio sending follows these general
steps:
- The audio client connects to the audio server.
- The audio client creates a SSPCQ in shared memory.
- The audio client sends the SSPCQ's file descriptor to the audio server
with the set_buffer() IPC call.
- The audio server receives the SSPCQ and maps it.
- The audio client signals start of playback with start_playback().
- At the same time:
- The audio client writes its audio data into the shared-memory queue.
- The audio server reads audio data from the shared-memory queue(s).
Both sides have additional before-queue/after-queue buffers, depending
on the exact application.
- Pausing playback is just an IPC call, nothing happens to the buffer
except that the server stops reading from it until playback is
resumed.
- Muting has nothing to do with whether audio data is read or not.
- When the connection closes, the queues are unmapped on both sides.
This should already improve audio playback performance in a bunch of
places.
Implementation & commit notes:
- Audio loaders don't create LegacyBuffers anymore. LegacyBuffer is kept
for WavLoader, see previous commit message.
- Most intra-process audio data passing is done with FixedArray<Sample>
or Vector<Sample>.
- Improvements to most audio-enqueuing applications. (If necessary I can
try to extract some of the aplay improvements.)
- New APIs on LibAudio/ClientConnection which allows non-realtime
applications to enqueue audio in big chunks like before.
- Removal of status APIs from the audio server connection for
information that can be directly obtained from the shared queue.
- Split the pause playback API into two APIs with more intuitive names.
I know this is a large commit, and you can kinda tell from the commit
message. It's basically impossible to break this up without hacks, so
please forgive me. These are some of the best changes to the audio
subsystem and I hope that that makes up for this :yaktangle: commit.
:yakring:
With the following change in how we send audio, the old Buffer type is
not really needed anymore. However, moving WavLoader to the new system
is a bit more involved and out of the scope of this PR. Therefore, we
need to keep Buffer around, but to make it clear that it's the old
buffer type which will be removed soon, we rename it to LegacyBuffer.
Most of the users will be gone after the next commit anyways.
A mistake I've repeatedly made is along these lines:
```c++
auto nread = TRY(source_file->read(buffer));
TRY(destination_file->write(buffer));
```
It's a little clunky to have to create a Bytes or StringView from the
buffer's data pointer and the nread, and easy to forget and just use
the buffer. So, this patch changes the read() function to return a
Bytes of the data that were just read.
The other read_foo() methods will be modified in the same way in
subsequent commits.
Fixes#13687
Apologies for the enormous commit, but I don't see a way to split this
up nicely. In the vast majority of cases it's a simple change. A few
extra places can use TRY instead of manual error checking though. :^)
FixedArray now doesn't expose any infallible constructors anymore.
Rather, it exposes fallible methods. Therefore, it can be used for
OOM-safe code.
This commit also converts the rest of the system to use the new API.
However, as an example, VMObject can't take advantage of this yet,
as we would have to endow VMObject with a fallible static
construction method, which would require a very fundamental change
to VMObject's whole inheritance hierarchy.
Previously, FlacLoader would read the data for each frame into a
separate vector, which are then combined via extend() in the end. This
incurs an avoidable copy per frame. By having the next_frame() function
write into a given Span, there's only one vector allocated per call to
get_more_samples().
This increases performance by at least 100% realtime, as measured by
abench, from about 1200%-1300% to (usually) 1400% on complex test files.
As long as possible, entire decoded frame sample vectors are moved into
the output vector, leading to up to 20% speedups by avoiding memmoves on
take_first.
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.
Decoding the residual in FLAC subframes is by far the most I/O-heavy
operation in FLAC decoding, as the residual data makes up the majority
of subframe data in LPC subframes. As the residual consists of many
Rice-encoded numbers with different bit sizes for differently large
numbers, the residual decoder frequently reads only one or two bytes at
a time. As we use a normal FileInputStream, that directly translates to
many calls to the read() syscall. We can see that the I/O overhead while
FLAC decoding is quite large, and much time is spent in the read()
syscall's kernel code.
This is optimized by using a Buffered<FileInputStream> instead, leading
to 4K blocks being read at once and a large reduction in I/O overhead.
Benchmarking with the new abench utility gives a 15-20% speedup on
identical files, usually pushing FLAC decoding to 10-15x realtime speed
on common sample rates.
Some nuances in the FLAC loading code can do well with an explanation,
as these non-obvious insights are often the result of long and painful
debugging and nobody should touch the affected code without careful
deliberation.
(Of course, secretly I just want people to maintain my loader code.)
:^)
This fixes all current code smells, bugs and issues reported by
SonarCloud static analysis. Other issues are almost exclusively false
positives. This makes much code clearer, and some minor benefits in
performance or bug evasion may be gained.
"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>
They're mostly used in literal random data, so it isn't like
there is a high demand for it, but it's cool to have more complete
implementation anyway. :^)
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.
FlacLoader initialized, but never used its resampler; this is now fixed
and all subframes are resampled before decorrelation occurs. FLAC files
with non-44100-Hz sample rates now play properly.
When computing sample values from a linear predictor, the repeated
multiplication and addition can lead to very large values that may
overflow a 32-bit integer. This was never discovered with 16-bit FLAC
test files used to create and validate the first version of the FLAC
loader. However, 24-bit audio, especially with large LPC shifts, will
regularly exceed and overflow i32. Therefore, we now use 64 bits
temporarily. If the resulting value is too large for 32 bits, something
else has gone wrong :^)
This fixes playback noise on 24-bit FLACs.
The FLAC samples are signed, so we need to rescale them not by their bit
depth, but by half of the bit depth. For example, a 24-bit sample
extends from -2^23 to 2^23-1, and therefore needs to be rescaled by 2^23
to conform to the [-1, 1] double sample range.
Playing a lossy flac file resulted in hearing something
you'd not like to play. Instead of your lovely bass, you had sounds
as if you put a CD-ROM disc to a CD player.
It turned out that the size for making signed values was too big,
making all the values unsigned.
I've used lossyWav[1] (the posix port[2] to be exact)
to generate such files.
[1]: https://wiki.hydrogenaud.io/index.php?title=LossyWAV
[2]: https://github.com/MoSal/lossywav-for-posix
Prior this change, decoding fixed subframes produced "unpleasant
crackling noices".
While the type doesn't appear so often when using the default settings,
encoding files in flac(1) with --fast option uses fixed subframes
almost every time.
This also applies the logic to the constant subframes,
which isn't so important, as the type is generally for the silence,
but let's use it as well to avoid inconsistency.
Before this change the file stream was generated two times:
one time in the parse_header(), and another time for the whole class
in the constructor.
The previous commit moved the m_stream initialization before
executing the parse_header function, so we can now reuse that here.
Before this change opening the file in the system resulted in crash
caused by assertion saying:
SoundPlayer(32:32): ASSERTION FAILED: m_ptr
../.././AK/OwnPtr.h:139
[#0 SoundPlayer(32:32)]: Terminating SoundPlayer(32) due to signal 6
[#0 FinalizerTask(4:4)]: 0xdeadc0de
The issue was that 845d403b8c started
using m_stream in the parse_header() function, but that variable wasn't
initialized if the Loader plugin was created using a file path
(which is used everywhere except for the fuzz testing),
resulting in a crash mentioned above.
The FlacLoader already has numerous checks for invalid data reads and
for invalid stream states, but it never actually handles the stream
errors on the stream object. By handling them properly we can actually
run FuzzFlacLoader for longer than a few seconds before it hits the
first assertion :^).
This fixes stucking in a loop at the end of the file, as
(a) custom block sizes are usually placed there, as the remaining
size might not be simply calculated as a power of two, and
(b) the number of bytes to read was incorrect (the program said
the block size was 32525, where flac -a said it's actually 3200).
Unfortunately, I couldn't trigger the bug for the sample rates,
so it may be not true, but I'd doubt it, giving the fact that flac
almost everywhere uses big endian numbers.
The problem here was that the multi-byte UTF-8 encoded characters
were taking one byte too much, misaligning the data completely
and eventually crashing the program on the 128th frame.
This change reduces the for loop by one, as it has been already
calculated from the start_byte variable.
AK's version should see better inlining behaviors, than the LibM one.
We avoid mixed usage for now though.
Also clean up some stale math includes and improper floatingpoint usage.
This fixes an crash caused by using the type from
FlacSubframeHeader::order (unsigned 8-bit), which after overflowing
the integer, converting it back to u32, and decrementing by one
resulted in accessing an array waaay out of bounds.
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.
