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>
Various warnings related to uninitialized variables, or setting values
to variables that would not be used later or would be overwritten by per
loop initializers.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Surprised I didn't catch this sooner. This could have resulted in a
division by zero error if either sample rate somehow was zero.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
DFT should use aligned memory blocks for best results. Also allocate one
extra sample for DFT output, just in case DFT zop is as bad as zrop.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Apparently, PFFFT double is much faster than vDSP, and I didn't even
notice. Thanks to Aleksey Vaneev for testing this properly.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
This will be proper at least unless I get this commit merged upstream.
Squashed the changes to a single commit, and removed extraneous
whitespace that crept into the code.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
The function I added only works for non-interleaved real/imaginary pairs
and not the interleaved setup that r8brain expects. Fix that by removing
the multiply implementation and using the original one.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
The used fork of r8brain now uses the Accelerate vDSP FFT functions for
resampling, which should provide a slight speedup, or significant for
large sample ratios.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
The resampler wasn't being given enough room to flush its final output,
so a function was added to determine the current output latency, and
more sample data is requested, allowing the full output flush to occur.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Just in case anything using the implementation ever needs to request
less sample data than would be returned by the resampler, it should be
able to return a remainder and keep extra remaining samples, if any.
However, the way Cog currently uses it, it would not be likely to run
into this scenario.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Fixes to the resampler wrapper, such that it will survive some close
encounters with the edge of the buffer, if necessary. Also so it will
obey the buffer size limit for the output buffer.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
This rename is more in line with what R8Brain does in its example code.
No actual behavioral changes to the code, however.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
This should improve performance slightly again, as there were some ARM
code paths that weren't being enabled for ARM64.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Replace libsoxr dylib with a static library, and also build the two
architectures separately, to allow for platform-specific optimizations
to be employed for both. This also reduces the size of the CogAudio
framework by a few hundred kilobytes, as we eliminate unused code paths
better this way.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
The original didn't really handle backwards versus forwards differently,
as far as the predictor coefficients should have been, as they probably
should have been reversed for a different direction window.
This didn't fix my problem, though, but did possibly expose something
else to mess with.
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>
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 is a fixed point implementation identical to Microsoft's original
algorithm. Or at least I assume it's Microsoft's. It was actually
adapted from hdcd_decode.exe, which was adapted from somewhere else.
It's entirely in fixed point math now, so it's fairly deterministic.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Replace individual virtual buffers with large _mm_malloc blocks at a
time, then dole out chunks of those buffers as the nodes need them.
Should reduce memory contention a little bit.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
This new virtualizer uses the Accelerate framework to process samples.
I've bundled a HeSuVi impulse for now, and will add an option to select
an impulse in the future. It will validate the selection before sending
it to the actual filter, which outright fails if it receives invalid
input. Impulses will be supported in any arbitrary format that Cog
supports, but let's not go too hog wild, it requires HeSuVi 14 channel
presets.
Christopher Snowhill