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 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, 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>
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 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>
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>
Sample block merging code should not be duplicated across the DSPs that
require it, but instead should be a common function. Also added some
optimizations to the Float32 converter function, to bypass conversion if
the audio format needs no conversion.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Audio Chunks now have full timestamp accounting, including DSP playback
speed ratio for the one DSP that can change play ratio, Rubber Band.
Inputs which support looping and actually reporting the absolute play
position now do so.
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>
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>
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>
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 was buggy as hell, and resulted in errors. Now the user should
restart playback if they change output device formats.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Sample format can now change dynamically at play time, and the player
will resample it as necessary, extrapolating edges between changes to
reduce the potential for gaps.
Currently supported formats for this:
- FLAC
- Ogg Vorbis
- Any format supported by FFmpeg, such as MP3 or AAC
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
The ChunkList wasn't clearing the remover entered flag when the chain
was empty. Now it does, so it will shut down correctly.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
By applying copious amounts of autorelease pools, memory is freed in a
timely manner. Prior to this, buffer objects were freed, but not being
released, and thus accumulating in memory indefinitely, as the original
threads and functions had autorelease pools that scoped the entire
thread, rather than individual function blocks that utilized the new
buffering system. This fixes memory growth caused by playback.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
This implements the basic output and mixing support for channel config
bits, optionally set by the input plugin.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Rewrite attempt number two. Now using array lists of audio chunks, with
each chunk having its format and optionally losslessness stashed along
with it. This replaces the old virtual ring buffer method. As a result
of this, the HRIR toggle now works instantaneously.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
