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>
The dynamic metadata functions should only activate for unseekable
streams, not seekable streams, and not local files.
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>
There were some leftover modifications to the 0.5.x tree I was using,
which were supposed to be speed optimizations, but they're probably
pointless anyway. The NEON optimizations were especially pointless,
since this plugin version isn't used on any macOS version that runs on
Apple Silicon.
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>
TrueAudio will now read APE tags, and if I should start writing tags
some day, will prefer creating APE tags if no tags exist.
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 fixes the dynamic length field not updating correctly after a track
info reload, which breaks the seekbar when reloading the current track.
Fixes#227
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Channel config should not contain duplicate channels, or unsupported
channels. Also fix a memory leak from not freeing the AudioChannelLayout
structure allocated previously.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
This implements channel masks for inputs where applicable, namely the
CoreAudio decoder, FFmpeg, FLAC, and WavPack. All others will still use
guessing from the channel number.
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>
Import a feature change that improves decoding speed on Haswell or newer
Intel processors. No effect on Apple Silicon, which doesn't support the
feature under Rosetta 2, and the ARM64 build doesn't need the changes
anyway, as the code already did the right thing.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
This fixes a possible crash with seeking operations, especially with
Audio Unit plugins. Fix implemented in foo_midi and imported here.
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>
Tiny files complete fetching within the scope of the open function, so
the URL session task would have completed already. Now the function will
accept the data, and allow reading it.
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>
When deleting the currently playing track, stop playback, because the
player engine doesn't like dealing with the current playing track not
actually having a playlist reference to go with it. Better stop playback
instead, as a safety check.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
This fixes the ability to configure the MIDI synthesizer properly, after
commit 7e5107d431 was applied.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
The timing of block based mode was kind of off. Now it should be just
fine. Thanks to testing on Windows in foo_midi.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
This bug prevented zero length or unknown length files, such as FLAC
files with no sample count in the header, or audio streams, from playing
properly, and clipped their output to the 0 samples indicated by the
field. Now it will simply allow wrapped files to decode until they stop
producing output.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Christopher Snowhill