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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.)
:^)
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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.
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This little functional change uses the most common algorithm for panning
audio, known as constant power panning. It makes it so that the total
output power (not directly the sample value, i.e. the peak) stays the
same no matter how the audio is panned.
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This makes the code easier to read.
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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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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.
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Note: While ClientAudioStream has had a volume property, it is only now
used in the mixer.
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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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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. :^)
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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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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.
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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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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.
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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.
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This is more idiomatic :^)
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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
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The value was always zero.
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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.
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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.
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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 845d403b8c175ff99793239404ca8657d95da104 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.
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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 :^).
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This makes aplay show current playback position.
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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.
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This fixes crash when reading the end of the file.
The logic is mostly borrowed from WavLoader.
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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.
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This might allow the program to return an error a bit quicker.
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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.
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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.
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async_enqueue() is a wrapper over the async_enqueue_buffer() call
to AudioServer. This allows users to asyncronously enqueue audio
samples, when the program requires non-blocking audio streaming.
This also makes ClientConnection use east-const everywhere.
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We don't know what is a good time to wait after an audio buffer fails to
be processed by AudioServer. However, it seems like decreasing the wait
time to 10ms after such a failure should improve latency and has not
caused issues in my testing. After all, 10ms is quite some time in audio
sample magnitudes.
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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.
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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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Previously, error_string() returned char* which is bad Serenity style
and caused issues when other error handling methods were tried. As both
WavLoader and (future) FLAC loader store a String internally for the
error message, it makes sense to return a String reference instead.
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The signed 32-bit PCM sample format is required for the FLAC standard.
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This commit addresses two issues:
1. If you play a 96 KHz Wave file, the slider position is incorrect,
because it is assumed all files are 44.1 KHz.
2. For high-bitrate files, there are audio dropouts due to not
buffering enough audio data.
Issue 1 is addressed by scaling the number of played samples by the
ratio between the source and destination sample rates.
Issue 2 is addressed by buffering a certain number of milliseconds
worth of audio data (instead of a fixed number of bytes).
This makes the the buffer size independent of the source sample rate.
Some of the code is redesigned to be simpler. The code that did the
book-keeping of which buffers need to be loaded and which have been
already played has been removed. Instead, we enqueue a new buffer based
on a low watermark of samples remaining in the audio server queue.
Other small fixes include:
1. Disable the stop button when playback is finished.
2. Remove hard-coded instances of 44100.
3. Update the GUI every 50 ms (was 100), which improves visualizations.
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When using `aplay` to play audio files with a sample rate of 96000,
there were occasional one-second gaps in playback. This is
because the Audio::ClientConnection sleeps for a full second when
the audio buffer is full.
One second is too long to sleep, especially for high-bitrate files.
Changing the sleep to a more reasonable value like 100 ms ensures
we attempt to enqueue again before the audio buffer runs empty.
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Prior code in `WavLoader::get_more_samples()` would attempt to
read the requested number of samples without actually checking
whether that many samples were remaining in the stream.
This was the cause of an audible pop at the end of a track, due
to reading non-audio data that is sometimes at the end of a Wave file.
Now we only attempt to read up to the end of sample data, but no
further.
Also, added comments to clarify the meaning of "sample", and how it
should be independent of the number of channels.
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This enables support for playing float32 and float64
WAVE_FORMAT_EXTENSIBLE files.
The PCM data format is encoded in the
first two bytes of the SubFormat GUID inside of the
WAVE_FORMAT_EXTENSIBLE `fmt` chunk.
Also, fixed the RIFF header size check to allow up to
maximum_wav_size (currently defined as 1 GiB).
The RIFF header size is the size of the entire
file, so it should be checked against the largest Wave size.
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This fixes a bug where if you try to play a Wave file a second
time (or loop with `aplay -l`), the second time will be pure
noise.
The function `Audio::Loader::seek` is meant to seek to a specific
audio sample, e.g. seek(0) should go to the first audio sample.
However, WavLoader was interpreting seek(0) as the beginning
of the file or stream, which contains non-audio header data.
This fixes the bug by capturing the byte offset of the start of the
audio data, and offseting the raw file/stream seek by that amount.
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When samples are requested in `Audio::Loader::get_more_samples`,
the request comes in as a max number of bytes to read.
However, the requested number of bytes may not be an even multiple
of the bytes per sample of the loaded file. If this is the case, and
the bytes are read from the file/stream, then
the last sample will be a partial/runt sample, which then offsets
the remainder of the stream, causing white noise in playback.
This bug was discovered when trying to play 24-bit Wave files, which
happened to have a sample size that never aligned with the number
of requested bytes.
This commit fixes the bug by only reading a multiple of
"bytes per sample" for the loaded file.
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IODeviceStreamReader isn't pulling its weight.
It's essentially a subset of InputFileStream with only one user
(WavLoader).
This refactors WavLoader to use InputFileStream instead.
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This is no longer used by any of our IPC pairs.
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