Keep per graph latency. Sum all the graph latencies together and keep
this around as the process-latency.
Refactor the port latency setter. Make a function to recalculate the
latency of all other ports. Take into account the graph latencies.
Update the port latencies when the total graph latency changes.
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.
Move the param enumeration code out of the main enum function.
Emit node events after completion of the set_param functions to ensure
we only emit things once.
Make the converter format a higher priority in all cases. The converter
has been negotiated first and is able to make a better suggestion for
the ideal format in all cases.
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.
Because we advertize on out ports that we support DYNAMIC data, we need
to read the data pointer directly from the buffer and only fall back to
our cache (mmaped) pointer when it is NULL.
With DYNAMIC data, the peer element (mixer-dsp) directly copies the
input data pointer into the buffer data in the processing loop in order
to avoid a memcpy when there is no mixing needed.
When there is no data and the buffer is mmapable, try to mmap it. Unmap
again when clearing the buffers.
Use the mmaped data pointer of the buffer when processing.
Prefer to let the follower allocate buffers. If we are allocating
buffers, first do use_buffers on the allocating node or else the
non-allocating node just ends up with NULL buffers.
The audioconverter starts in Convert mode, so make sure it goes to the
None mode before we attempt to reconfigure ourselves.
Also remove the ports on audioconvert when going to None mode. This used
to somewhat work because we configured it in DSP mode without any
params, which is like None without ports.
Try to avoid conversions by taking the output port format and using that
as a filter for the input port format.
Because filtering pods prefer the values of the filter, this will prefer
the output format values and thus avoid conversions.
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.
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.
The delay is always expressed in samples at the output rate of the
resampler. For input streams we need to convert this to the expected
input rate.
Make the delay reporting in playback streams more accurate.
Resampler delay for N taps is N/2-1 input samples.
Add test that checks this.
When input rate is varying, the resampler also accumulates additional
sub-sample delay. The resampler does not currently have API to report
the instantaneous sub-sample delay. Add knownfail test for it.
