These strings were added to the InfoPlist translations in a
previous commit, but likely need to be added to the
Info.plist file directly as well.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
This reverts usage of the AVFoundation output to use
the previous lower latency CoreAudio output, and
paves the way for a change I am cooking up soon.
Fixes several issues with playback and seeking latency.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Do this by serializing the background thread actions against
the AudioPlayer object, so we don't start playback multiple
times at once.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
The other translations currently have placeholders, awaiting translation.
I may end up using machine translation from Deepl just for Russian,
Polish, and Turkish, until I get human translations from a contributor.
Spanish will be provided by our team later tonight.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Perform playback start and seeking operations in the background, instead
of on the main thread, which should help prevent them from stalling the
user interface.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
These column header names were missing from the strings
tables. Add them based on existing translations of the same
field names by our translators.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Default repeat mode should now be Repeat All, and the menu items should
now all function, fixed by removing a bunch of pointless attributes from
each affected menu item.
Fixes#371
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
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>
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>
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>
Instead of clear playlist and play. This was confusing people,
apparently. I should have changed it sooner, since this is what
I use normally anyway.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
