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