Apparently, this doesn't work too well with real HDCD tracks, and causes
all sorts of weird volume issues. Just leave the volume alone, and let
HDCD decoding make tracks louder, possibly, rather than try to halve the
volume automatically on a detector that isn't terribly good.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Now using Accelerate methods to calculate and multiply ramps per channel
and add the remainder if necessary.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
This code did not check the number of samples in a packet before adding
it to the output buffer, which apparently had the potential to cause the
output code to emit up to 512 samples of silence between tracks. This,
as one can guess, is a bad thing, and causes noticeable gapping between
tracks.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Output logging, a debugging feature that is only enabled at build time
if I need to chase down some audio mixing or output bug, was not logging
anything at all. Change to use Cocoa file writing methods, and actually
implement the output writer function again.
This code is left disabled 99% of the time anyway, and especially in
release builds. Like the node logging code elsewhere, it has the
potential to be very noisy and consume massive amounts of disk space.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Output format mostly requires stopping and restarting the output device,
and this also prevents us from using the latency function properly,
which apparently always returns 0 for output devices anyway. These
changes also prevent the output callback from hanging when resets occur.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Fix default output device logging, and also the preferences if no
default device happens to be set.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Fixes visualization latency under virtual machines, at least. Not sure
which local or native systems would be reporting high latency here, but
this should fix them as well.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Move the DSPs to the output node, so they don't get closed and reopened
across each file. Also restructure the output handler to buffer a little
on its own, to account for track switch activity.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Cleaned up project settings to current defaults, except for the macOS
deployment version, which is still 10.13. Cleaned up a lot of headers
and such to include with angle braces instead of double quotes. Enabled
build sandbox in a lot of places. Disabled subproject signing in several
places, for libraries and frameworks which will be stripped and signed
when they are copied into place in the final build.
Also, while trying to solve compilation issues, the visualization
controller was reverted to the Objective C implementation, which is
probably faster anyway. Stupid Swift/Objective-C language mixing issues.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Seeking should clear the buffers completely now, and will be nearly
instant, depending on how fast the input can decode.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Seeking now mutes properly, and will not leave the audio muted across
other operations. Audio output changes should also mute and destroy the
buffers of the input chain, so that the audio resets properly.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Fixes timestamps in several cases where they were being processed
incorrectly, which was causing some chunked audio files to mis-report
timestamps into the past or the future, which caused the seekbar to jump
around in an unpredictable way.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Visualization now buffers in the audio output pipeline, and uses a
container system to delay multiple buffer chains from emitting
visualization data over top of each other. This should stabilize
display output significantly, while introducing minimal lag before
DSP configuration changes take effect.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
We were forcing a resampling ratio to match the HRTF filter supplied
with the app, now we resample the HRTF to match the input audio, which
will be resampled to match the output device settings.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
It's more like the output monitor thread, since it only monitors output,
rather than actually handing the output callbacks.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Audio Chunks now have full timestamp accounting, including DSP playback
speed ratio for the one DSP that can change play ratio, Rubber Band.
Inputs which support looping and actually reporting the absolute play
position now do so.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
The last of the built-in processors is now in the threaded processing
chain, and all DSPs are marked high priority and with short buffers.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
This class can more flexibly process and emit varying chunk sizes than
the previous code could, solving the problem of wide tempo changes.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Now there's a configuration dialog for tweaking the settings
in semi-real time. Everything that can be changed without
restarting is changed without restarting, otherwise the audio
pipeline is reset, which happens quickly enough anyway.
Awaiting translation to Spanish, other languages have been
removed pending their maintainers fixing most of their
problems, which includes me being lazy and AI translating
bits so I could rush updates.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
I will implement the more complex setup of providing options for
most of the configuration that Rubber Band provides, at a later
date, when I feel like creating a complex configuration dialog
for it, and asking for help translating every option and setting.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
It should be deriving its channel count from the file format,
since it's applied before any other filters.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Implements a simple speed control using a resampler
designed for real time changes. A rubberband speed
control will be implemented at a later date.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Shuffle around @autoreleasepool blocks, and also add one
to the audio processing code in the playback callback, so
audio memory is released during playback instead of
accumulating.
Fixes#379
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
This is checked inside the audio thread, it isn't needed in the watcher
thread. Remove the second check.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
A stopped instance of OutputCoreAudio should not continue to feed the
visualization system with stale audio, potentially while another instance
is already starting up and feeding its own audio output.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
This reverts usage of the AVFoundation output to use
the previous lower latency CoreAudio output, and
paves the way for a change I am cooking up soon.
Fixes several issues with playback and seeking latency.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
