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.
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.
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.
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.
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
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.
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.
Make sure we only make the buffer for the follower larger when we
downsample because then we need to ask for more data from the follower
to fill up a quantum.
Never try to make the follower buffer smaller than the quantum limit.
The reason is that the graph rate could be decreased dynamically and
then we would end up with too small buffers.
See #4490
Load multiple graphs with audioconvert.filter-graph.N where N is the
order where the graph is inserted/replaced. Run the graphs before the
channelmixer.
Graphs can be added and removed at runtime.
Instead of recalculating what to do every cycle, we can prepare a
static schedule and just run that. We only need to reevaluate it when
something changes.
For input streams, first run the resampler and then the channelmix. This
ensures that the channelmix is run with the rate of the graph instead
of the rate of the input. This is nicer because rate and quantum align
with the graph and the sample accurate volume ramps will work as
intended.
For output streams, leave the resampler after the channelmix for the same
reasons.
Use the helper instead of duplicating the same code.
Also add some helpers to parse a json array of uint32_t
Move some functions to convert between type name and id.
Add spa_json_begin_array/object to replace
spa_json_init+spa_json_begin_array/object
This function is better because it does not waste a useless spa_json
structure as an iterator. The relaxed versions also error out when the
container is mismatched because parsing a mismatched container is not
going to give any results anyway.
The IO_Buffers is used in the data thread to check if the port should be
scheduled or not. Make sure it is only set after we set buffers on the
port and cleared before the buffers are cleared.
Make sure we sync the port->io with the data thread.
See #4094
Can be used to group ports together. Mostly because they are all from
the same stream and split into multiple ports by audioconvert/adapter.
Also useful for the alsa sequence to group client ports together.
Also interesting when pw-filter would be able to handle streams in the
future to find out what ports belong to what streams.
Add a monitor.passthrough option. This will pass all latency information
directly between the port and its monitor ports.
This is interesting when the adapter (and audioconvert) is used with a
null-audio-sink that simply forwards the data to a real sink/souce. In
that case, we want the sink/source latency to be passed unmodified.
Set the monitor.passthrough on the pulseaudio null-sink because
a passthrough virtual sink is the most likely use case for this.
Add some monitor.passthrough default config where it makes sense.
Fixes#3888
Keep track of the valid ports and don't emit port info for
invalid ports. When a listener is added while the ports are being
created, it is possible that the ports are still NULL or invalid.
When starting the converter, calculate the initial size needed by
the resampler to fill one quantum.
This makes it possible to get the requested amount of samples before
the first process call is made.
