Make a new variable to iterate the other ports so that we can use the
original port to emit notifications.
Fixes Latency and other params set on DSP ports.
When the follower updates EnumFormat, it probably wants to renegotiate
to a new format, so clear the current format so that we do that when
starting the next time.
EnumFormat should also not only return the current format in case we
are negotiated but it should return all possible formats.
See #2832
After the ports are reconfigured, we need to perform the setup again so
that buffers and processing can happen with the right settings.
This fixes an issue when autoswitching between A2DP and HFP with
bluetooth headsets when there is also a stereo capture device available.
The input stream of the browser is quickly reconfigured between stereo
and mono with only a Pause command in between, clearing the setup state
is enough to redo the setup when going back to Playing.
Fixes#2764
The parsing functions expect float values in the default locale so use
the spa_dtoa function to generate such a float.
Fixes setting params with floating point values when the locale is not
the default locale.
When we were using the resampler and then bypass it when the configured
rate == 1.0, we create a pop because we don't process the queued data in
the resampler anymore.
Avoid this by keeping the resampler active as soon as the rate property
is set on the audioconvert. The resampler itself will use a more
efficient copy method in that case anyway and it is expected that the
rate will change again later when we need to reactivate the resampler.
uint32_t i;
for (i = 0; i < SPA_N_ELEMENTS(some_array); i++)
.. stuff with some_array[i].foo ...
becomes:
SPA_FOR_EACH_ELEMENT_VAR(some_array, p)
.. stuff with p->foo ..
A flush command is not supposed to stop playback but just clear the
current state. Normally, to avoid complications, an application will
Pause, Flush and optionally Start to do things smoothly without
interfering with the process loop, but things should not crash if that's
not the case.
Fixes#2726
A Suspend should clear all the negotiated state and start a new
negotiation in Start. Use a flag to control this.
This avoids recalculation of state for each pause/play state change.
See #2701
Add a resampler option to prefill the resampler with 0. This then
results in the resampler always outputing and consuming the same
amount of data instead of a short buffer in the beginning.
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.
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
