Prevent the player from locking up in certain circumstances, by not
locking chainQueue the entire time this function is processing.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Remove a single .inc include from CogAudio build phase, as it's included
but not compiled as Pascal like Xcode thinks. Also remove a bunch of
files from being copied into the resulting .framework and .bundle files
during link stage, as we don't need to distribute that stuff.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
The deinterleaved format was being specified incorrectly. Now it asks
for the correct format, which is deinterleaved, and the bytes per frame
or packet sizes are relative to a single channel's buffer, not all
buffers. Oops, that could have been more clear in the documentation.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Moved external cover art reader to a place where it can be used for any
format, even formats unsupported by Metadata Reader interfaces.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
This callback should be unregistered when plugin loading completes,
otherwise we could end up processing bundles loaded by external stuff,
like Audio Units loading for MIDI playback.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Equalizer was copying the output of the equalizer repeatedly to the
first output channel, instead of copying each channel correctly. This
had the effect of making the equalizer output adjusted audio to only the
left channel in stereo output, and possibly render the stream sounding
weird.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
If somehow a plugin doesn't load, skip cuesheet should skip it anyway,
as we don't want any recursive loops.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
When restarting playback on the current track, restart the correct
track, in case restarting near the end of it.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Track play counts for the correct track, even on short tracks. Also
correctly track the play count of the last played item in the play queue
which stops with bufferChain set to nil, so the previous iteration was
not tracking it.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Previously, the cleanup thread was not being run. Also, only reset the
metadata deduplication store when the cache is first emptied.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Wait for the equalizer to be shut down properly by the main thread
before destroying it. Otherwise, the main thread could crash on stop,
due to accessing the equalizer handle while it's being torn down in the
output thread.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Now the API makes both PCM and FFT data optional, and will do nothing if
neither are requested. Also, it now supports a latency offset in seconds
with floating point precision. The two built-in visualizations currently
request zero larency. Increasing the latency asks for even older samples
while specifying a negative count requests samples from the "future"
relative to what the listener is hearing.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Cuesheets can now expose which URLs they contain, which may help with
sandbox path configuration. That is, if the CUE sheets are already
readable.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
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>
All optional fallback code for older versions has also been removed, and
everything now assumes 10.13.0 or newer. Some cases are still included
for point releases, such as 10.13.2.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Redesign the code signing from the ground up. Now all bundles and their
embedded frameworks import the Shared.xcconfig file and enable its
settings, so they may be signed with Apple Development instead of sign
to run locally. This apparently isn't necessary for frameworks which are
embedded in the main app bundle directly, only for the bundles and their
frameworks.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Require asking user consent for data transmission on first launch, or
otherwise disable sending crash reports by default.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Automatically format any XML escapes of file type association names.
Adjust Info.plist to account for this change.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Plist generator now emits the output to the temporary folder, which we
have write permission to.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Add dealloc function to close the file container, in case the caller
neglected to do so on its own.
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>
This code turned out to be somewhat of a mistake to employ, so it's now
being removed, and shall not be re-added, as it doesn't really work.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Add play count data collection, including first seen times for every
file first added to the playlist. Data is indexed by album, artist, and
title, or by filename, whichever matches first. Add interfaces to
AppleScript automation definition as well.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
As it doesn't seem to work properly on Intel machines, anyway. It just
leads to pointless crashes, and doesn't seem to serve any purpose.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Restrict the use of workgroup joining and workgroup intervals to macOS
Monterey or newer, as it seems the way I use it, it's completely broken
on macOS Big Sur, which was the original minimum target for the API.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Apply changes to exit the thread if workgroup initialization or joining
fails, instead of attempting to continue executing the thread.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Add an extra step to the workgroup exit call, so that it only calls to
leave if the join token is valid, or at least initialized.
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>
It was a fun ride, but I think I want to try something different. Users,
please be sure not to have DNS blocking for Crashlytics if you want me
to have any useful bug reporting info if it crashes on you, or otherwise
blows up. Otherwise, I don't get any useful data to help me fix crashes.
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>
The changes include no longer leaving the workgroup for seeking or for
converter format changes, and also still leaving the workgroup on thread
termination if there was an error with intervals starting or finishing.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
An impulse cache reduces any glitching from format channel count changes
to near insignificant levels, resulting in a more pleasant experience
when there are different mixed formats playing, or even a file which
changes format mid-playback.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Ensure that dynamic info updates, even on static files, only update the
exact track they apply to, by atomically assigning the userInfo property
before opening the decoder, so that callbacks to the player indicate the
correct track and don't assume it's the one that's currently visibly
playing. Fixes start of track metadata notifications from overwriting
the previously playing track.
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>
DFT float happens to clobber one extra sample on forward translate, so
allocate one extra for every complex buffer.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Work back to a vDSP implementation, this time using overlap-save instead
of overlap-add, also accumulating the results as complex values, only
inversing them once at the end, and finally, replacing the FFT method
with the newer DFT API.
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>
When leaving the workgroup, clear the token, as the join call requires
the token to be uninitialized.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Errors should stop all attempts to further use the audio thread priority
code, so there won't be debug breakpoints called on older OSes.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
As the decimator has shown to be twice as loud as it should be, the
volume should be reduced by half when converting DSD to PCM with it.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Pure downsampling is slower, but may or may not be more accurate. Though
probably not worth it. It did help me realize a minor error, though.
The decimator's volume is twice as loud as it should be.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
This prevents crashes where inputs were not returning either properties
or metadata blocks and the file open cache was attempting to cache the
resulting nil pointer as if it were valid. Also prevent the metadata
redundant string coalescing from processing nil objects as well, in case
it's used that way somewhere else.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Set baseline real-time priority for audio threads even on old macOS,
since that API is available there. Only set it once, and do not attempt
again if it fails, only once per thread.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
I'm not sure about macOS Ventura, but stable releases of macOS, at
least on Intel, require that threads joining Audio Interval
workgroups already be set to run as real-time before joining. Not
doing this results in an uncaught exception and a crash.
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>
Update all project files with new upgrade version number, and add the
dead code stripping option. Don't touch MASShortcut because it's not my
project.
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>
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>
When resampling the impulse according to the playback rate, it becomes
necessary to normalize the resulting impulse by the inverse of the
sample ratio, as resampling adds more or less loudness by virtue of
interpolating samples.
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>
Moved the string encoding guesser/converter to the Plugin.h header, so
it may be accessible from any plugin. I may make it a global member of
something eventually, but a static inline for such a simple function
should be fine for now.
This function facilitates converting arbitrary 8 bit encoded strings to
Unicode NSString objects. It should be used anywhere that UTF-8 is
expected, but not necessarily guaranteed, and where other 8-bit
encodings may also be supplied by a user's files.
Not using this setup for string inputs has already led to failed UTF-8
decoding resulting in nil NSStrings being passed to the inline array or
dictionary initializers, which results in crashes due to uncaught
exceptions.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Update the Info.plist generator to emit file type definitions which use
system generated icons in place of the legacy icons in the app bundle.
Also include the new LSHandlerRank field. And also add a definition for
the scripting definition, which I accidentally added to the Info.plist
manually when I fixed scripting.
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>
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>
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>
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>
The cache thread should have an autoreleasepool around the release loop,
because it will be freeing Objective C objects periodically.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Promote the Plugin Controller source file to Objective-C++, and add a
simple data cache that holds on to requests for up to 5 seconds after
their last access, for preventing spammed requests from hitting files
over and over. This is apparently really relevant to the CUESheet reader
and its embedded CUESheet handling, as that tends to reread the same
file over and over as it populates the playlist with tracks. The nested
reader can also lead to repeated reading even on files without CUESheets
embedded.
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>
When the option is enabled, and playback comes to a completion, the
player will quit on its own.
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>
vDSP functions expect their input and output pointers to be aligned to
an even four values. Correct this by aligning all pointers. The
allocated buffers used for one parameter should already be aligned
somewhat, but align the incremented positions used on some of them so
that the vDSP functions don't misbehave. Also align the volume scaler
input by doing scalar math until the pointer is aligned prior to calling
vDSP_vsmul. Also, change 16-bit and 32-bit scale to use vsdiv instead of
vsmul with a really small number already divided into one.
Fixes the test vectors that were sent in extrapolating incorrectly due
to their final blocks having uneven sample counts, resulting in
unaligned pointers.
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 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>
In the rare event that we're somehow playing decimated DSD at full
sample rate instead of resampling, only the start needs to be skipped,
and the end needs the input to the decimator padded to flush it, but
nothing needs to be truncated from the end of the output in that case.
Still, mostly pointless, since next to nobody will be outputting 384 kHz
from their Macs, in any case, much less unprocessed DSD.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
We should be extrapolating right over top of the DSD decimator latency,
rather than in front of it. Yeah, that'll do.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Metadata versus properties merging, correctly merge over empty fields if
they are assigned with empty strings or zeroed numbers, instead of only
merging over completely missing fields.
Fixes emailed bug, CUE Sheet metadata reading, primarily.
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>
The reader should have been skipping the properties of CUE sheets when
reading the referenced data for the inner files.
Fixes#235
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>
For big endian sample formats, endianness can be swapped using Clang
specific byte swap functions, which are present in all supported
versions of Xcode.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Simple upmixing algorithms now use Accelerate framework functions
instead of complex loops, and the HRIR filter now supports forcing
stereo output to any channel output configuration that has at least
front stereo speakers.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Where TagLib is not being employed, use FFmpeg to read tags where
possible. This allows reading tags from files like IFF. It reads it
through properties, otherwise allowing tag readers to function like
usual.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
When decoder is redirected to the internal silence decoder, show an icon
on the playlist indicating a playback error.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
This resulted in horrible things, the generic N to N upmixer was leaving
unmapped channels as uninitialized memory. This fixes horrible things
happening for people with interfaces with more channels than the source
file, frequently when the source file is stereo, or if the file is mono
and a center channel is present.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
HTTP Reader now supports limited seeking backwards even in streams, so
seek back to file start for repeated file tests, since there are at
least a few inputs that all claim to support things like Ogg containers.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
_mm_malloc and _mm_free are apparently based on intrinsic functions,
and only exist on Intel or older macOS targets. So removing them in
favor of posix_memalign.
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>
The ChunkList wasn't clearing the remover entered flag when the chain
was empty. Now it does, so it will shut down correctly.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Code ordering was wrong, it was writing the output samples repeatedly
for each input speaker, now it will only write them once.
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>
The volume should have been twice what it was, because I got this scale
wrong. The correct scale for Accelerate inverse FFT is 1/4 per sample,
not 1/8 like I accidentally misread while rewriting a convolver for the
umpteenth time from scratch.
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 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>
The end of input queueing, which can go nuts when there are a lot of
short files, should be terminated when the user asks the player to stop.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Flush the resampler when the source file terminates, so that it outputs
delayed samples properly. This fixes gapless decoding of resampled
files.
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 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>
Added a logging method that indicates starting playback of a given URL,
and added a debug build only logging of every metadata load event.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
7.0 downmix was passing parameters to cblas_scopy backwards, and WAV
files report "host" endian, not "native".
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>
This should fix some coding issues, and also fix some potential memory
leaks in the file verifier, assuming it didn't already release the
files it was pulling the stats from.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
The filter wasn't properly freeing its FFT setup state, and also was
unnecessarily null checking the pointers before passing them to the
aligned free function, which already does null checking.
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>
Implement the ability to configure and select an HRIR preset to use with
the HRIR filter, or remove the preset. It will validate the file's
usefulness before setting it for the player to use.
Also, fixed back center channel filtering for 7.0 format audio.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
This is needed for HeSuVi no-echo impulses, which are only one channel
per input channel, and mapping uses symmetrical mirroring of the input
set to create the surround effect, since there's no side-to-side delay
in these impulses.
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.
