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.
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.
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.
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.
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.
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.
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.
This changes client methods so that they return the IPC response's
return value directly - instead of the response struct - for IPC
methods which only have a single return value.
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)
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 *
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.
It was using has_any_error, which causes an assertion failure when
destroying the stream. Instead, use handle_any_error, as the
WAV loader does handle errors.
(...and ASSERT_NOT_REACHED => VERIFY_NOT_REACHED)
Since all of these checks are done in release builds as well,
let's rename them to VERIFY to prevent confusion, as everyone is
used to assertions being compiled out in release.
We can introduce a new ASSERT macro that is specifically for debug
checks, but I'm doing this wholesale conversion first since we've
accumulated thousands of these already, and it's not immediately
obvious which ones are suitable for ASSERT.
A C++ source file containing just
#include <LibFoo/Bar.h>
should always compile cleanly.
This patch adds missing header inclusions that could have caused weird error
messages if they were used in a different context. Also, this confused QtCreator.
