This output may prove to have lower latency, but the results are too
glitchy to really be usable. Not even visualization latency is handled
correctly.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
And default it to disabled. As was pointed out to me by a user, DSD is
apparently mastered to a level of -6 dB, so double its level on output
by default.
Also reorder all preferences dialog controls so they are instantiated in
display order, which should help screen readers, maybe.
Fixes#368
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Resampler flush may indefinitely produce 1 sample if there is a rounding
error with the buffering calculations. Work around this.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
When the clipped sample rate changes, the resampler needs to be
restarted. This was previously failing because the target sample rate
wasn't changing.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
When reading partial chunks, and when returning partial data, it is
essential to maintain this lossless chunk status across either whole or
partial chunk reads. Otherwise, the converter chain sees the lossless
flag constantly changing on lossless files, such as PCM or DSD, and
causes the DSD decimator and/or resampler to be torn down and reset
repeatedly, causing glitches in the audio.
The glitch was not, in fact, with the decimator itself, and was
occurring to a degree without it, as it would be restarting the
resampler repeatedly as well.
Fixes#367
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
The latency is half of the FIFO, or half the filter size, and each byte
is 8 samples, so return the value accordingly.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
This comment was in the original sample decimator code, I neglected to
include it in my port over to Cog. Doesn't really serve any functional
change, though. It would have clarified that I needed to reduce the gain
level much sooner, though.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Makes volume slider logarithmic when limited to 100% to allow easier changing of volume towards the bottom of the slider.
The tooltip remains as the slider location instead of the logarithmic value of the actual volume.
Input thread now signals when it has stopped and is about to return, in
case the input thread returns before the BufferChain dealloc function
would be waiting for it to terminate. Somehow, even though the Semaphore
is being signaled at this point, the BufferChain still ends up waiting
the default of 2.5 seconds for the signal that apparently never comes,
delaying file stoppage. This prevents the wait action entirely. Must
have been some sort of race condition.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
This new method should cause all stops to default to immediate stoppage,
and only stops that occur after an end of track signal should indicate
to play out the entire buffer.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
External artwork already supported the HEIC format, just not the correct
filename extension for the format.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
This magically fixes the stupid header maps that were pulling the system
semaphore.h into Swift projects, when they shouldn't have been doing
that in the first place. This is the same reason that the FLAC library
has its assert.h renamed to FLAC_assert.h.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
And this is the actual meat of getting it to work properly, the changes
the Swift code needed to actually be fully functional.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
These imports needed to be changed so that Swift bridging didn't import
the system's semaphore.h instead of CogAudio's Semaphore.h, which is a
completely different thing.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Oops, this compare blunder resulted in DSD decimation breaking every
1024 samples or so, owing to block sizes, and caused ticking sounds as a
result. It would also cause HDCD decoding to break completely.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Neither of these two changes is really important, but they do simplify
things, and the division on that one function makes the non-decimating
DSD support actually functional, as the caller expects a specific number
of samples, and that was otherwise octupling the input sample count.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Another large stack buffer was at play here. Consolidated it into an
existing buffer that can perform double duty here, since neither place
it's used conflicts with each other.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Only unregister the listener if it actually has been registered, and
clear the handle upon doing so.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Sample rate changes will now occur on exact sample boundaries, like they
are supposed to. Also, FreeSurround accounts for its output latency.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
The DSPs should not be deinitialized from another thread, possibly while
they are currently processing.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Apparently, all these new changes with FreeSurround have pushed the
default 512KB thread stack size to the limit. And I'm not even using
stack variables, really, except for maybe the autoreleasepools.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Apparently, the LFE channel is not being initialized at all if bass
redirection isn't enabled, and even if it is, it's uninitialized for a
great portion of the spectrum. Clear it all on every iteration.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
The output implementation has a block size of 4096, so the class
implementation should also use that.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
This is the code that actually needs to be added to make it process
audio. This insertion makes the whole app crash when processing audio at
all. Weirdly, simply reverting these two files makes the audio code work
again. I can't explain it.
Also, commenting out CMAudioFormatDescriptionCreate makes it work, too.
There's something weird going on with that function.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
This is a working implementation of FreeSurround, but I can't get it to
work in the Cog code base, as the whole project crashes head over heels
if this code is inserted into the output chain.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Move the Float32 converter to a different location, for any future plans
to support decoding audio files to common data for any other purpose.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
This should also seal up any potential hole for problems if there's an
audio format change and no audio buffered.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
The input chain could hang up indefinitely, and MAD decoder didn't
indicate end of file properly.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
