File channel remapping was incorrectly only working for 1-7 channel
files, not 8 channel files. Fixed that.
PLEASE NOTE: This will be my last commit and build for over a week, I
promised myself I would stop, and this is the last straw. These are the
last two, I promise. No more bug fixes this week, until at least the 7th
of February. Please, I'm begging you.
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>
We're not using Gitlab's CI anyway, and they don't offer free Mac
build bots at this time, either. It just keeps spamming me with build
failure notices every time I push commits to that mirror.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
The patch is now updated to exactly what I sent to the ffmpeg-devel
mailing list for hopeful inclusion in the latest release of FFmpeg.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
The old quadra method was broken, too. Now it uses a simple linear
difference sliding scale for the extra points, which seems to work much
better for the 20 kHz band. And the 16 kHz band doesn't get predicted to
heck any more, either.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Added a keyboard shortcut for the equalizer menu item, added a user
contributed button for an equalizer toolbar button, and replaced the
randomize playlist button with something different from the same user.
I had to redo the randomize button, because template PDFs don't support
white cutout shapes, but instead need to be formatted as hollow spaces
in the paths. Naturally, I couldn't figure out how to do this with
Inkscape the right way, so I rendered the original to a 1024x1024
bitmap, then imported that into a new document and traced it. Meh. I
need a better icon anyway, but this looks okay on the toolbar for now.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Track entries are updated when files are added and their metadata is
later loaded. The refresh cycle was failing, because dbIndex was unset
in the new tracks, for some reason. Now it will set this when updating
the track item, assuming it read from the database in the first place.
This fixes metadata being loaded properly on adding tracks to the
playlist, as well as reloading track data manually.
Fixes#218, hopefully.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
... also disable use of AudioToolbox codecs, and use only bundled codecs
and libfdk-aac for AAC input. This is required for HLS at least, as
Apple's system codecs didn't really like the network streams that were
provided by HLS streaming stations.
Also reshuffle the input priorities between Core Audio input and FFmpeg
input, so that they were the way they were before I messed with things a
while back. This puts FFmpeg back at the top, using bundled codecs where
supported.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Also disable AudioToolbox codecs, as they don't really like network
streaming so much. Use bundled codecs instead.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
The built-in M3U container parser should not be used for HLS playlists,
so they should end up in the playlist as-is, so that inputs can parse
them in real time.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
For Playlist Entries that are not File URLs, return the full absolute
URL for path queries, so the playlist path column will show the full
URL instead of a useless reduced path string.
Fixes#214.
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>
If there's no configured SoundFont bank, or if the selected bank has
gone missing, and the user has configured the player to use the
FluidSynth driver, fall back to the system DLS Synthesizer, which has
its own Roland bank to fall back on if unconfigured.
Also, whether falling back, or already on an AU synthesizer, don't fail
if there's no bank configured or found. DLS doesn't explicitly require a
bank, and most other synthesizers of interest would not require a bank
either.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
The file prober in FFmpeg expects that when the read function reaches
end of file, it returns AVERROR_EOF, not zero. Otherwise, it will loop
endlessly until the process is terminated.
Fixes#217.
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>
For some incredibly dumb reason, PSF files get into FFmpeg 5.0, then
just sit there and lock up, reading them forever and ever, doing nothing
useful. Add a bodge to detect PSF files by signature and ignore them in
the VGMStream container parser, and all other parts of VGMStream in Cog.
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>
Added a link to a downloadable set of HRIR presets, borrowed from the
HeSuVi project, which is on SourceForge, but inside of an executable,
therefore possibly a pain to extract.
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.
Dzmitry Neviadomski