Since in one case, it probably wasn't combining them into one stack
allocation, it probably blew up the stack allocation quite a bit.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Change to the future render api, hopefully float support
will be available eventually. Also change to allocate the
sample buffers from the heap instead of the stack.
This is imported by the player, though it doesn't fail when it's
missing, so that would explain why I didn't spot this error sooner.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
They'll be imported by the plugin on startup anyway, and they don't
really have any exported functions we can use.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Correctly scale the AudioChunk frame counts. This more and more makes me
think I should be scaling this in the AudioChunk code instead, but then
code may not know about the special case of every 8 frames only being
one byte per channel.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
DSD wasn't tracking the correct sample count, because DSD
Audio Chunks store the byte count, rather than the bit count.
This may be changed in the future, so I'll have to remember.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
If transfer completes quickly, do not hang waiting for it to achieve
reading state. Also add some comments indicating what we're doing.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Based on the C++11 code by Joel Yliluoma / bisqwit, which in turn is
based on information from NX-Engine. I have also taken the liberty of
bundling the required wavetable bank and PixTone drums. Contrary to the
documentation provided with the code, my version of dou_1006.zip, as
downloaded over a decade ago, had the wavetable bank at offset 635,816
bytes into the file, not 1,115,748 bytes. Possibly a difference of
having applied the translation patch? My copy is the original version,
so I had to use a real resource parser to locate the waveforms.
The player will obey the configured sample rate, loop count, and fade
time for synthesizers, and also obey Repeat One to play indefinitely.
The code should be quite robust to minor abuses, though I can't imagine
how well it would hold up to random bad files, other than playing
outright garbage.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Remove shell script that was modifying the import path of a framework,
as we are no longer using libvorbis like that.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
This should fix some potential initialization errors it may have had
before, but this doesn't fix the broken Sound Canvas VA plugin. Roland
says it's supposed to be broken on macOS 12+ and/or Apple silicon
anyway, so I guess there's no dodging that.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Tag reading can read cue sheets as either a single NSString, or an
NSArray of NSStrings, so handle either case.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
There has been an API in GME to detect tracks ending for quite some time
now, and this just adds a little bit to the existing comment, which
previously noted that there was no way to detect if a track had ended,
which may have been true several major versions of GME long past.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
The output has been assigning twice as many samples as it was supposed
to ever since commit 8d851e5bda, which
ended up generating the correct 1024 samples (2048 per GME parameter),
but assigned 2048 to the AudioChunk, which resulted in over-reading the
audio buffer, and thankfully not crashing, but instead causing an awful
sound distortion effect as random memory contents were played as PCM
audio.
Fixes#320
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Allow .mp3 and such to fall back to the FFmpeg container handler, in
case there are chapters in a renamed file.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
The input isn't supposed to close its own sources, as it did not open
them itself, and they should be cleaned up automatically when they are
released to zero reference count.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Let the FFmpeg decoder handle RIFF files, if they happen to be named
.mp3 and not something like .wav.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
The SID builder needs a static initializer, otherwise multiple instances
created simultaneously, such as during populating info on adding a lot
of tracks, will race and crash the player.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
The subdirectory parser, the CUEsheet reader, and the legacy XML
playlist reader were missing grants for Sandbox access.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Buffer up to 5 milliseconds of audio, or at minimum 1024 samples, each
call. Also pre-allocate the buffer, rather than using a stack buffer.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
