If we have in and out rates with a very small GCD, we might end up with
a lot of phases. Limit the number of phases to 1024 and switch to
interpolating mode. 1024 phases is enough to accurately interpolate
from.
Together with the MAX_TAPS limit we will never create a filter
size that overflows 32 bits.
Fixes#5073
Our AVX optimizations are really AVX2 so rename the files and functions
and use the right HAVE_AVX2 and cpu flags to compile and select the
right functions.
Fixes#5072
Force filter graph reconfiguration in setup_convert.
When adding/removing filter-graphs, only perform setup when we were
already setup, otherwise we will do this in setup_convert later.
Don't do channelmix_init when we were not setup.
Deactivate the filter-graphs when we suspend.
Fixes#4866
First do the essential properties to set up the node, then set up the
node and then parse the params. The params might do some setup that
relies on a completely configured node, such as emit events.
Use per port allocated memory so that we can easily increase the size
and add more buffers. This is necessary when we add filter-graphs that
require more ports.
Use a simple free/active linked list for the filter-graphs and insert
the new filters in the right position in the list. Then simply copy the
list to an array for the processing thread.
when reconfiguring, set up all the filters again because the number of
channels might have changed.
When parsing the graph, parse the input and output port names into
a separate string array. This was we can keep them around when
setting up the graph.
Instead of setting up the graph right after loading it, do the graph
setup when we activate the graph. This makes it possible to pass the
input channels to the filter-graph and let it create the right amount of
plugins and ouput channels.
When setting up the graphs in the audioconverter, pass the current
number of channels as the input to the graph and keep track of the
channels that each filter produces.
This way we can also load a custom upmix or downmix graph, for example.
Commit cbbf37c3b8 changed the logic of the
Start command. Before this commit, when there was no converter, the
follower would always get the Start command. After the commit, the
follower would only get Start when previously Paused.
This however breaks when we set a format or buffers on the follower
without a converter because those actions might change the state of the
follower to Paused implicitly.
We should simply remove the started check here and always call Start on
the converter and follower, the implementations themselves will keep track
if anything needs to be done.
Fixes#4911
We don't actually need to calculate the GCD for each resampler rate
update. The GCD is only used to scale the in/out rates when using the
full resampler and this we can cache and reuse when we did the setup.
The interpolating resampler can work perfectly fine with a GCD of 1 and
so we can just assume that.
Ensure we have at least a `.` after `audioconvert.filter-graph`, so we
don't try to read past the end if it does not exist.
Cherry-picked from a328e0ae28, dropping
the param doc update as that doesn't exist.
Before fixating the format, we try to add as many of the PortConfig
fields as we can as defaults. However, when we already have a property
that intersects, prefer to use the original negotiated one prefered by
the follower.
This got changed when the default for the pod filter changed to the
filter value.
The effect is, for example, when the follower asks for FL FR FC LFE SL
SR and the sink is using FL FR SL SR FC LFE, the PortConfig (and the
default) would then be FL FR SL SR FC LFE. After negotiation, we would
get FL FR FC LFE SL SR as the format but then with the filter and the
wrong default we would fixate to FL FR SL SR FC LFE, which does not
match what the follower wanted and either results in wrong channels
or an error of the follower.
See #4722
When the follower has no buffer suggestion, it can return -ENOENT, which
should not generate an error but simply use the converter buffer
suggestion instead.
Since 3abda54d80 we prefer the format
of the filter. This reverses the selection of the default value when
negotiating buffers from the target to the follower.
If the follower does not select a reasonable value for the buffer size,
for example, this then results in wrongly sized buffers.
Fix this by reversing the order of allocation from target to follower
where we let the target (converter) select a default value, which is
more likely to be correct.
See #4713, #4619
There is no need to encode the potential format in the format.dsp of
control ports, this is just for legacy compatibility with JACK apps. The
actual format can be negotiated with the types field.
Fixes midi port visibility with apps compiled against 1.2, such as JACK
apps in flatpaks.
Parameter values read into a 512 byte long buffer, which is insufficient
for medium to long filter-graph parameters.
Increase the buffer to 4096 bytes to give some wiggle-room.
Depending on the direction of the conversion, we run the resampler
before or after the channelmix. This means we need to use the channel
count before or after the channelmixer instead of always using the
channels after channelmixing.
Fixes#4595
Setting a format does not automatically mean making a wav file.
Instead, check for the extension on the output file name and when we
can't find any good containet, use the same as the input file.
This makes more sense than doing `spa-resample test.flac -r 44100 test.wav`
and ending up with a FLAC file named test.wav. Or
`spa-resample test.flac -r 44100 -f s16 test2.flac` and ending up with a
wav file named test2.flac.
We only need to switch between the converter ports and the follower port
when we switch between passthrough. Avoid emiting remove of the (non
existing) follower ports when the mode changes from none to DSP at startup.
Partially revert 86af9de739
The PortParam does not give enough information to derive the direction
of the converter. If the converter is configured in convert/convert
there is just no way to know when to output a quantum or not.
Fix this by doing a quick probe of the follower and then pass the
direction to the converter.
See !2227
Report the "fractional" part of the resampler delay in
spa_io_rate_match::delay_frac, in nanosamples (1/1e9 sample) at node
rate.
The delay values are best reported in units where it is clear what the
clock domain is, so report the value in fractional samples instead of
nanoseconds. Conversion to ns is also just dividision by the appropriate
rate.
Resampler without prefill was sometime outputting with different delay
than with prefill. Adjust initial history by 1 which seems to bring it
more in line.
The resampler phase also appears to depend on how many samples remain in
history which leads to possibly unexpected +-1 variation. Take this into
account in reported phase.
These changes make the resampler delay tests pass.
Both changes are sort of empirical --- I don't fully understand why
these would fix things but they seem to be needed to make the delay
calculations agree with what the resampler outputs.
