Only update the resampler rate when we ask for more data, when we have
more input data, use the previously configured rate to calculate how
many samples we will consume.
Fixes resync errors with multiple sources. One source would do rate
matching, audioconvert would ask it to produce X samples, the source is
scheduled to produce the samples, the rate match is updated with the new
rate correction, audioconvert is scheduled again. It should now use the
X samples it asked to produce and apply the new rate correction for the
next iteration.
Only simple upmixing would replicate the FC channel into REAR/SIDE.
The PSD method would take the diff between FL/FR (which would
be 0 if only FC is available) and not generate output.
Always use the DSP rate on DSP ports for format conversion, not the
previous used rate.
This avoids some resampler reconfiguration as it negotiates a non-passthrough
rate conversion and then switches to passthrough when the rate correction is
done to match the graph rate.
See #2614
By: Kevin Yin
R->A calculation removed: it wasn't valid anyway. No behavior change.
Placed existing A in there directly.
cosh window -> strangely tweaked exp window: remove the discontinuity
at the border, which is wrong for a window function. If A changes in the
future, this window will be better. With the current A, you will not be
able to tell the difference on any graph. (Of course, it's not a cosh
window anymore.)
Fixes#2574
As currently implemented, input format convert channel remap is no-op.
This is because although the out_datas array is permuted, the original
pointer array is not referred to later on, so the only effect is that
the temporary data array is stored in permuted order.
Fix the permutation by permuting the pointers only for the conversion
step.
Make a new noise method called PATTERN and use it to add a slow (every
1024 samples) repeating pattern of -1, 0.
Only use this method when we don't already use triangular dither.
See #2540
Also clamp the amount of input samples we push when flushing. do several
rounds of zero pushing until we have flushed enough.
Handle the cases where no input is needed or no output is generated.
Fixes crashes when downsampling from 96000 to 1000 Hz or so.
When the audioconverter needs more data, let it return NEED_DATA. This
can happen before the ports actually have consumed all the input data.
For example, then the next cycle would require 1024 samples but there
are currently only 16 samples queued, the next cycle will consume the
16 samples and then need another buffer to produce output.
For rt streams, this is not a problem because a new buffer will be
fetched in the next cycle synchronously.
When the stream is async, we can use this NEED_DATA to prefetch a
new buffer so that we have one in the next cycle.
This fixes hickups with async streams that provide random sized
buffers.
Move the setup of the output buffers first.
Then figure out how many samples we need to produce and consume.
Make sure we use the resampler to only convert the input samples that
are needed to produce the output samples.
Fixes some muddled sound with mpv when upmixing.
