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>
This should not interact with the Audio Player object on a background
thread, but instead the main thread queue.
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>
Two playback event items were set to queue a playback start to a
background thread, when playback should instead be queued on the main
thread. Fix this in a simple way.
This crash was easily reproducible by skipping through tracks rapidly.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Sometimes the play count data only includes the filenames, and thus will
fail a query for just the tags. Also, a file query may be stored without
the subsong fragment tag, which will also break the tags.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Play Count sorting was entirely missing, and sample rate and bits per
sample sorting caused exceptions due to the capitalization of the fields
versus the column identifiers.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
The TagLib framework build process leaves several
key fields empty. This breaks App Store submission.
Fix it again. Dang.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Remove deprecated functions, make use of free functions that clear the
pointers before returning, etc.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Remove deprecated functions, make use of free functions that clear the
pointers before returning, etc.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
When adding tracks to the playlist, clear the search filter first, so
the playlist doesn't become all jumbled, or so we don't overflow the
playlist indexes.
Also add some bug fixes for reversing the arranged to disarranged index
lists, so if an arranged index is past the end of the arranged list, as
is the case for appending, we shift the indexes forward past the end of
the diarranged object list.
Extra exception handling was added as well, so these things will only
cause a failure to add playlist items at worst, instead of crashing the
player entirely.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Update the readers to support the newly added tag fields, and also read
the supported format list from the library itself.
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>
In case playlist setup is reset or not, move the reset above the menu
setup code, so the menu is set up correctly if a reset occurs.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Reset to defaults if no columns are visible. Also log this situation in
Firebase events, in case it becomes relevant.
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>
Play Count cannot be updated for tracks which have been deleted before
the update was added to them. This was another cause of a rare crash.
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>
Fixes timestamps in several cases where they were being processed
incorrectly, which was causing some chunked audio files to mis-report
timestamps into the past or the future, which caused the seekbar to jump
around in an unpredictable way.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Stream timestamps were correctly being converted from the monotonically
increasing frame count, but the AudioChunk parameter was being set from
the frame count rather than the converted seconds count.
Fixes#418
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Timed wait for 500us is kind of stupid and makes the threads wake up way
too much, and use way more CPU time. Reduce this, as the semaphores are
signaled appropriately, and the waiter should not wake up constantly.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
