If upsampling the audio by a significant factor, it may be necessary to
process more than one buffer at a time, rather than lose input.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
The visualization buffer now holds up to 45 seconds of loop, and the
latency measurement code now caps this at 30 seconds, and restarts the
output if latency exceeds 30 seconds, such as if a sound output is
reset.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
For one thing, the example code I followed was Swift and handled auto
releasing handles in the background, while Objective-C requires manual
handle reference management.
For two, there was no autoreleasepool around the block handling the
input audio chunks, which need to be released as they are pulled out and
disposed of. This also contributed to memory leakage.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Correctly configure AVFoundation with the channel layouts supported by
WAVEFORMATEXTENSIBLE speaker position flags, which includes varied
formats supported by FFmpeg and Core Audio inputs.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Stop output when requested, except on natural completion of the last
track in the play queue. Also fix deadlocks with stopping and
restarting.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
The output now uses AVSampleBufferAudioRenderer to play all formats, and
uses that to resample. It also supports Spatial Audio on macOS 12.0 or
newer. Note that there are some outstanding bugs with Spatial Audio
support. Namely that it appears to be limited to only 192 kHz at mono or
stereo, or 352800 Hz at surround configurations. This breaks DSD64
playback at stereo formats, as well as possibly other things. This is
entirely an Apple bug. I have reported it to Apple with reference code
FB10441301 for reference, in case anyone else wants to complain that it
isn't fixed.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Refine the output function a bit, including adding some minor safety
checks, in case the caller requests zero samples, or requests a format
with zero channels.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Move the Core Audio output function block to its own declarative
function, so that its block variables are isolated, and so that debug
traces show up in a more sensible place.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Fix a potential bug where the device enumerator would return a nil
device name string, which would result in a crash. Instead, report an
unknown numbered device.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Synchronize audio setup and audio stopping on the object's own pointer,
to hopefully prevent race conditions with out of sync calls to the stop
function from both the main and the audio thread.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Only uninitialize the equalizer if sound output was successfully started
and the equalizer AudioUnit was successfully ininitialized.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Now it allocates audio workgroups per thread, using work slices like the
Apple documentation describes for asynchronous threads.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
On Big Sur or newer, it is possible to join the audio threads to the
same OS workgroup as the audio output device, improving response.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Replace overlap-add vDSP/Accelerate implementation with a faster PFFFT
overlap-save implementation, using fewer FFT steps as well.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Rewrite some of the output and a lot of the downmixer to use Accelerate
framework instead of dumb for loops.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Add safety check to check if a device is actually alive when enumerating
it, and also add nil pointer checks for the device name before trying to
CFRelease it. Fixes a rare crash on device add/remove cycle, such as
Bluetooth headphones.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
These two changes fix playback issues with either starting in the middle
of the playlist on a really short file terminating immediately instead
of queueing more files (InputNode.m), and issues with starting playback
at all on the end of a playlist on a short file. (OutputCoreAudio.m)
Fixes#246
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
IN A.D. 2101, WAR WAS BEGINNING. *boom*
Yeah, this was a dumb bug, I didn't realize that AUAudioUnit would just
arbitrarily ignore my configured block size and request a different one.
The AirPods Pro will just request 480 instead of the 512 I ask for, so
let's instead support variable block sizes, and only take up to the last
4096 samples of the chunk fed to the output device.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
A bad sample scanner and cleaner will point out in the log whenever a
bad sample, such as infinity, or Not a Number, or even huge values over
±2.0, in case some piece of code, or a decoder, or even a bad file, has
taken over the output.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
The quality of the equalizer dialog is now up to par with what we had
before, minus all the crashes.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Borrowing some DFT code from deadbeef, this implements a simple spectrum
visualization into the main toolbar of the app.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Now the output is restarted on the current file at the current position
if the output format has changed. This should resolve the issue finally.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
This was buggy as hell, and resulted in errors. Now the user should
restart playback if they change output device formats.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Sample format can now change dynamically at play time, and the player
will resample it as necessary, extrapolating edges between changes to
reduce the potential for gaps.
Currently supported formats for this:
- FLAC
- Ogg Vorbis
- Any format supported by FFmpeg, such as MP3 or AAC
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
By applying copious amounts of autorelease pools, memory is freed in a
timely manner. Prior to this, buffer objects were freed, but not being
released, and thus accumulating in memory indefinitely, as the original
threads and functions had autorelease pools that scoped the entire
thread, rather than individual function blocks that utilized the new
buffering system. This fixes memory growth caused by playback.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
This implements the basic output and mixing support for channel config
bits, optionally set by the input plugin.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Rewrite attempt number two. Now using array lists of audio chunks, with
each chunk having its format and optionally losslessness stashed along
with it. This replaces the old virtual ring buffer method. As a result
of this, the HRIR toggle now works instantaneously.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
After all this rewriting, down or upmixing the audio is now handled with
the lowest latency possible, meaning that toggling the HRIR option now
takes effect immediately.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
This seals up a major memory leak of the playback state whenever a chain
is released on stop or on manual track change. CogAudioMulti was
retaining the input node due to its listeners, and InputNode was not
releasing the listeners when asked to stop running. This is fixed now.
Fixes#221
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
The thread wait on shutdown had the potential to lock up waiting for the
thread to shut down. Now it should at least spam the semaphores, so that
the thread should progress to shutdown a lot quicker.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Add a safety fix for pausing and shutting down, so that we don't call
into AUAudioUnit's stopHardware function unless the stream has already
been started by the output.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Reduce the timing at which an end of file notification is sent to the
main thread from 16384 bytes to 8192 bytes. This may help with playback
of a lot of really small files, and skipping tracks.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Christopher Snowhill