This should fix an exception being thrown because the observer wasn't
registered, or known to be registered. Only register it when it will be
used, and only unregister it if it was registered.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
And disabled by default, at that. I can't actually hear the difference
of Peak Extension in the Rock track I have that claims to use it. And
Low Level Range Extension is more trouble than it's worth on tracks that
use it by mistake, or maliciously, if the case may be. I may add track
tag level control in the future.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
This should be included, for safety purposes, in case the rounding up to
the nearest multiple of 256 samples doesn't bump the buffer size enough.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
This affects User Defaults, but only has any effect on ChunkLists which
are being used for conversion, and only if they're processing DSD source
material. Thus, the observer should only be added on the one stream that
is converting DSD, and should definitely be removed when the object is
deallocated.
This fixes a serious crash bug that mostly appears to only affect Intel
Macs, and has no major side effects on Apple Silicon that I can tell.
It's a good thing I still own an Intel Mac or two to test on, even if
they are both trapped on older releases of macOS.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Apparently I somehow didn't notice this situation because I still had
Rubber Band enabled, and existing users kept it enabled ever since I
introduced it.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
The converter doesn't just require an output format call, it also
requires this input format change callback to actually signal it to
reopen the converter process with a new format setup.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Move the DSPs to the output node, so they don't get closed and reopened
across each file. Also restructure the output handler to buffer a little
on its own, to account for track switch activity.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
The filter uses a pre-buffer of input audio, so extrapolate from the
actual input to fill the buffer. Fixes clicking on non-zero-crossing
track endings.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Buffers were being treated as empty before they were actually processed,
due to races between the current node's end of stream marker and
actually feeding the output buffer.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
This optional code, disabled at compile time by default, allows finding
weird issues with the sample decoding chain.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Apparently, this isn't needed, and on two users reporting crashes,
actually causes exceptions to be thrown somewhere.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
This required some minor workarounds to deal with the play time counting
that works toward play count reporting.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
There is a race condition with the next Node in the chain and the End of
Stream marker, considering how tiny the buffering is for these DSPs. Set
End of Stream instead after inserting the end of stream flush chunk.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
DSPs should not be performing a cascading reset when resetting just
their own buffers, for example, on init or shutdown of just that one
DSP filter.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Cleaned up project settings to current defaults, except for the macOS
deployment version, which is still 10.13. Cleaned up a lot of headers
and such to include with angle braces instead of double quotes. Enabled
build sandbox in a lot of places. Disabled subproject signing in several
places, for libraries and frameworks which will be stripped and signed
when they are copied into place in the final build.
Also, while trying to solve compilation issues, the visualization
controller was reverted to the Objective C implementation, which is
probably faster anyway. Stupid Swift/Objective-C language mixing issues.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Seeking should clear the buffers completely now, and will be nearly
instant, depending on how fast the input can decode.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
This includes setting and unsetting the equalizer DSP chain objects on
track change and advancing on track playback end, and also bugs with
applying equalizer presets to the band configuration items when the
equalizer is disabled or when playback is stopped.
Fixes#420
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Seeking now mutes properly, and will not leave the audio muted across
other operations. Audio output changes should also mute and destroy the
buffers of the input chain, so that the audio resets properly.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Fixes output volume setting on seek or audio output restart on format
change. Also safeguards these setters so they don't go off if the nodes
aren't actually allocated.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Now buffer twice as much audio as would be requested for a single
visualization PCM/FFT chunk, which should hopefully prevent it from
flickering due to running out of audio because of too low latency.
Now it buffers up to two chunks at the current hard coded visualization
sample rate, which works out to about 186 milliseconds.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
We implement this function to return the current latency buffered,
regardless of how often this function may be called. In practice, it is
only called on track completion, to time the reporting of the next track
display. We also avoid using Rubber Band's latency function, as in most
cases, this function will be called from other threads, and also, it
currently only gets called after Rubber Band has been emptied out, so it
would otherwise calculate zero samples buffered. And thirdly, Rubber
Band's latency function doesn't account for the buffered samples already
removed from it and waiting to be fed out.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
The code was polling the input chunk duration after emptying out the
chunk's samples, which resulted in an input duration account sitting at
exactly zero, so the end overrun flush would not be cut short properly,
resulting in gaps between tracks.
Correct the input sum to tabulate before emptying the input chunk, so
output remains properly gapless.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
This code was being duplicated across three different playback functions
which basically did most of the same things.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Check all audio chain elements for allocation failures, and also dispose
of all of the previous handles in reverse order, including nulling the
final node handle so the output does not attempt to poll for audio while
the chain is being rebuilt.
Also set up output node to handle the new null finalNode state, and
return an empty chunk to the caller.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
We should not be processing a potential playback restart when the chain
is being torn down for shutdown.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
This should be perfectly safe to use in all situations now. It may have
been unstable due to mishandling return values, or not supporting
requesting more sample data from the library without feeding in more
input first.
Also, still signaling the End of Stream flag on chunk reading should be
correct, as downstream processors only react to it when the buffer runs
empty.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
The Downmixer wasn't updating its output format correctly, so it was
prone to outputting the wrong format for a while, which could confuse
the output device and produce garbage output.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
The delay value should be scaled by the resampling ratio, similar to
how it already is when allocating the impulse buffer. This went
undetected, as it scribbled over other memory without causing immediate
crashes, but instead later heap corruption.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Check for paused processing state in various places, so that startup
playback works properly, and resume playback at seek offset works
properly and doesn't hang the player.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
The samples available function returns a signed integer, so it can
apparently return negative on error, and the DSP was incorrectly casting
this to an unsigned type, and thus attempting to buffer an inordinate
number of samples and crashing.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
The merge function should be able to tell when the caller has no audio
left to process, such as on end of stream.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
FreeSurround needs more buffering from its input, so increase buffering
of previous node to 100ms.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
