Upstream functions which return empty chunks on error do not return nil,
so the caller should check for an empty duration instead.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Impulses should be gain scaled roughly based on the sample ratio
relative to the original impulses. Lower target sample rate means less
impulses means gain goes up, higher target sample rate means more
impulses so gain goes down. Somewhat simple, seems to work.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
This prepares the filter to be the same as the rest of the filters, in
that they support flexible sample rates to match the output device.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
And disable it by default in new installations, otherwise leave the
setting alone. The disablement setting is shared with the engine
setting, so the default should not really change anything, except for
new installs.
Also, the time/pitch shifting dialog disables itself and displays an
obvious notice button, which opens the Rubber Band settings.
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>
Attempt to completely fill the input buffer of the Rubber Band library
between each call to the process function, instead of processing in
as small an increment as the source node provides. May reduce processing
power required.
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>
FreeSurround, like the Equalizer, which attempt to coalesce Audio Chunks
into larger blocks of 4096 samples, must check if the audio format has
changed between blocks, and stop stacking chunks together when a new
format is detected. They will continue processing with less sample data
than expected, as necessary.
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>
Change one remaining semaphore wait to 500us, and change the buffering
so that it can always overflow the requested duration by one chunk, so
that at least one chunk will always fit in the buffer. This also allows
the DSP nodes to flush at the end of the stream without losing their
output.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
The end of stream flushing should only request remaining samples once,
as should the rest of the process. The problem with the Rubber Band code
in this case is that it will wrap the remaining samples pointer after it
has been flushed, and emit a really huge number.
Also, add code to try to equalize the samples output with the samples
input, relative to the tempo stretching, as Rubber Band seems to flush
entirely too much data at end of stream, which can create noticeable
gaps in the output. This solves that as well.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
This should be guarded, so that no other thread tries to free the DSP
while it is potentially writing to the Rubber Band instance.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
This comment was copied by accident when duplicating the original
Converter Node class for the new DSP base.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
DSP threads, such as the Rubber Band processing, and planned moves of
other processing to buffer threads, such as the Equalizer, FreeSurround,
HRTF, and Downmixing for output, because they all have small output
buffers. Since these buffers drain and fill fast, they should be
processed at a high priority. Hopefully, App Store doesn't complain
about the use of these APIs.
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>
I will implement the more complex setup of providing options for
most of the configuration that Rubber Band provides, at a later
date, when I feel like creating a complex configuration dialog
for it, and asking for help translating every option and setting.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Implements a simple speed control using a resampler
designed for real time changes. A rubberband speed
control will be implemented at a later date.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Volume scaling would potentially crash when handling
unaligned blocks of samples, and also handled them
completely wrong. It should be counting up single
samples until the buffer is aligned to a multiple of 16
bytes, and it should not exceed the intended count.
BUG: It was not only counting the unaligned samples
backwards, it was ignoring the real sample count.
Fixes#380
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Shuffle around @autoreleasepool blocks, and also add one
to the audio processing code in the playback callback, so
audio memory is released during playback instead of
accumulating.
Fixes#379
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Buffer up to 20 seconds per stage, and buffer only up
to 2 seconds before starting the next stage.
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>
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>
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>
This new method should cause all stops to default to immediate stoppage,
and only stops that occur after an end of track signal should indicate
to play out the entire buffer.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
This magically fixes the stupid header maps that were pulling the system
semaphore.h into Swift projects, when they shouldn't have been doing
that in the first place. This is the same reason that the FLAC library
has its assert.h renamed to FLAC_assert.h.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
These imports needed to be changed so that Swift bridging didn't import
the system's semaphore.h instead of CogAudio's Semaphore.h, which is a
completely different thing.
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>
Sample rate changes will now occur on exact sample boundaries, like they
are supposed to. Also, FreeSurround accounts for its output latency.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
Apparently, all these new changes with FreeSurround have pushed the
default 512KB thread stack size to the limit. And I'm not even using
stack variables, really, except for maybe the autoreleasepools.
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 should also seal up any potential hole for problems if there's an
audio format change and no audio buffered.
Signed-off-by: Christopher Snowhill <kode54@gmail.com>
