Do the channel remapping to the cannonical format when we
deinterleave/interleave instead. Otherwise we would completely skip
the remapping when we have interleaved input.
Fixes#2502, #2490
When we get something else that a drain status as input, bring us back
to the non-drained state.
When we are draining, don't remove the drained flag on the input
io status. This needs to be cleared by the host when the draining is
finished.
Fixes speaker-test
Ensure that our temporary buffers can hold at least quantum_limit
samples. When no output or input is connected, we can generate up
to a quantum_limit of silence, which requires all the buffers to
be scaled correctly.
Fixes a segfault in mpv.
Make dither noise as a value between -0.5 and 0.5 and add this
to the scaled samples.
For this, we first need to do the scaling and then the CLAMP to
the target depth. This optimizes to the same code but allows us
to avoid under and overflows when we add the dither noise.
Add more dithering methods.
Expose a dither.method property on audioconvert. Disable dither when
the target depth > 16.
We need to do dithering and noise when converting f32 to the
target format. This is more natural because we can work in 32 bits
integers instead of floats.
This will also make it possible to actually calculate the error between
source and target values and implement some sort of feedback and
noise shaping later.
Move the noise setting in the dither struct so that it can be
handled separately.
Setup dither separately.
Set used cpu_flags in structures after setup.
The quantize is the amount of bits we want to keep from the original
signal, subtract the amount of bits for noise. Clamp this to 0 (all
noise).
Calculate the scale factor better with powf() and avoid overflows.
Fixes#2479
Rename empty.noise -> dither.noise and always add this amount of noise
when > 0. This also adds the noise to silent sounds, not only when
nothing is connected because that would also be a problem when an amp
needs to be kept alive with an non-0 signal.
Rename noise -> dither because we can use this also for dithering later.
See #705
Add an empty.noise option that specifies the number of bits to
use for noise when the input signal is pure silence.
Some amplifiers can go into suspend mode pretty easily when they
get pure silence. With empty.noise = 1, audioconvert will now generate
a bitpattern that can keep those amplifiers alive, together with
disabling suspend in the session manager.
Fixes#705
Add an EMPTY chunk flag to mark a piece of memory as 'empty'. For audio
this means silence.
Use the empty flag to avoid mixing 0 samples.
Set the empty flag in output buffers on audioconvert.
this->monitor enabled adds an additional port in reconfigure_mode. If
there was already the maximum 64, this will crash.
Make maximum number of ports one larger than max channels to avoid
problems.
Only advance the in_offset with the number of samples that were consumed
by the resampler. In case when the resampler is filling up an old
buffer, this can be less than n_samples.
Fixes a2dp source and possibly others.
Use the offset to skip entries in the sequence array.
Use one loop to handle intermediate and trailing samples.
Fixes an issue where the last chunk of a sequence would be ignored.
Ignore latency params on the monitor ports of merger. They interfere
with the latency reporting set by the adapter.
The adapter will set the latency param from the follower on port 0 in
the converter, so pass this on to the merger as the latency and ignore
all other updates.
Fixes a case where the latency of a sink would become 0 when a monitor
port was recorded from.
Make the alignment parameter optional when negotiating buffers.
Default to a 16 bytes alignment and adjust for the max cpu
alignment.
Remove the useless align buffer parameter in plugins, we always
set it to 16 anyway.
When adding a listener, don't add a listener for the resampler because
it does not contribute to the result, we only listen for results from
the resampler.
When in passthrough mode, use the position io to update the
io_rate_match fields for the follower. This makes it possible for the
follower to also provide the right amount of data when the converter
is not selected in passthrough.
Add an option to configure the converter in None port config where it
removes all the ports. We can use this when removing the converter to
make sure all it's ports are removed.
When we remove the converter, make sure we expose the follower ports
directly so we can use them for passthrough.
Mark some structures, arrays static/const at various places.
In some cases this prevents unnecessary initialization
when a function is entered.
All in all, the text segments across all shared
libraries are reduced by about 2 KiB. However,
the total size increases by about 2 KiB as well.
When setting the Latency parameter on one side of the converter, set
it also on the other size. We should actually implement propagating
the latency through all the elements of the converter later.
Implement latency handling on fmtconvert.
merger and splitter change latency on all ports when on port changes.
All this makes the configured and exposed latencies visible on all
ports from adapter.
Implement a port recalculate latency method that takes the min
and max latency of all peer ports and sets that as the new port
latency.
When a link is made, let the output and input port recalculate
latencies.
Pass latency param in audioconvert.
