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
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.
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.
Remove the redundant remap array.
First set up the array with remapped output pointers in case we
can do passthrough output. Make sure stages write to the remapped
array in passthrough.
Remap the input array after unpack/convert.
This avoid an input and output memcpy in the common case of
remapping.
When we are actively driving the stream and the converter needs more
data, call the stream process function again to get it so that we
don't underrun.
Fixes#2494
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.
