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>
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>
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>
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>
FreeSurround needs more buffering from its input, so increase buffering
of previous node to 100ms.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Visualization now buffers in the audio output pipeline, and uses a
container system to delay multiple buffer chains from emitting
visualization data over top of each other. This should stabilize
display output significantly, while introducing minimal lag before
DSP configuration changes take effect.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
The last of the built-in processors is now in the threaded processing
chain, and all DSPs are marked high priority and with short buffers.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
This class can more flexibly process and emit varying chunk sizes than
the previous code could, solving the problem of wide tempo changes.
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>
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>
The output now uses AVSampleBufferAudioRenderer to play all formats, and
uses that to resample. It also supports Spatial Audio on macOS 12.0 or
newer. Note that there are some outstanding bugs with Spatial Audio
support. Namely that it appears to be limited to only 192 kHz at mono or
stereo, or 352800 Hz at surround configurations. This breaks DSD64
playback at stereo formats, as well as possibly other things. This is
entirely an Apple bug. I have reported it to Apple with reference code
FB10441301 for reference, in case anyone else wants to complain that it
isn't fixed.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Ensure that dynamic info updates, even on static files, only update the
exact track they apply to, by atomically assigning the userInfo property
before opening the decoder, so that callbacks to the player indicate the
correct track and don't assume it's the one that's currently visibly
playing. Fixes start of track metadata notifications from overwriting
the previously playing track.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
When decoder is redirected to the internal silence decoder, show an icon
on the playlist indicating a playback error.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Now the output is restarted on the current file at the current position
if the output format has changed. This should resolve the issue finally.
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>
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>
After all this rewriting, down or upmixing the audio is now handled with
the lowest latency possible, meaning that toggling the HRIR option now
takes effect immediately.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
