When setting the Latency parameter on one side of the converter, set
it also on the other size. We should actually implement propagating
the latency through all the elements of the converter later.
Implement latency handling on fmtconvert.
merger and splitter change latency on all ports when on port changes.
All this makes the configured and exposed latencies visible on all
ports from adapter.
Implement a port recalculate latency method that takes the min
and max latency of all peer ports and sets that as the new port
latency.
When a link is made, let the output and input port recalculate
latencies.
Pass latency param in audioconvert.
When we add a new listener to an object, it will emit the full state
of the object. For this it temporarily sets the change_mask to all
changes. Restore the previous state after this or else we might not
emit the right change_mask for the next listener.
Consider the case where one there are two listeners on an object.
The object emits a change and the first listener wants to enumerate the
changed params. For this is adds a new listener and then triggers the
enumeration. If we set the change_mask to 0 after adding the listener,
the second listener would get a 0 change_mask and fail to update
its state.
SPA_MEMBER is misleading, all we're doing here is pointer+offset and a
type-casting the result. Rename to SPA_PTROFF which is more expressive (and
has the same number of characters so we don't need to re-indent).
This is needed for example for Clang compiler which uses different
annotations than GCC. It will make WebRTC to happily use PipeWire
since the spa library is header-only and WebRTC defaults to use
Clang with -Wimplicit-fallthrough.
Make a new DRAINED status.
Place the DRAINED status on an input IO when a stream is out of
buffers and draining.
All nodes that don't have HAVE_DATA on the input io need to copy
it to the output io and return the status. This makes sure the
DRAINED is forwarded and nodes return DRAINED from _process()
DRAINED on the resampler flushes out the last queued samples and then
forwards the DRAINED in the next iteration.
Emit a new drained signal from the context when a node returns
DRAINED. Use this to trigger the drained signal in the stream.
If the target node is set to 0, remove the autoconnect flag. This makes
the session manager disable stream autoconnect and some other program
needs to connect the stream to a sink or node.
Use the channelmap from the file, if available.
Add option to specify/override the channel map for playback.
This is more in line with wayland and it allows us to create new
interfaces in modules without having to add anything to the type
enum. It also removes some lookups to map type_id to readable
name in debug.
Add a new PortConfig parameter to configure ports of elements that
are marked with the SPA_NODE_FLAG_*_PORT_CONFIG. This is used to
configure the operation of the audioconver/audioadapter nodes and
how it should convert the internal format. We want to use the
Profile parameter only for cases where there is an enumeration of
values, like with device configuration.
Add unit tests for audioconvert and adapter to check if they handle
PortConfig correctly.
Make the media session use the PortConfig to dynamically configure
the device nodes.
Remove audio-dsp, it is not used anymore and can/should be implemented
with a simple audioconvert spa node now and some PortConfig.
This allows picking F32LE as the default format on links that have
no restriction and it avoids failing negotiation when the restricted
end cannot handle S16/F32/F32P
For instance this pipeline would previously fail:
audio-dsp mode=merge ! audio-dsp mode=convert ! alsa-sink
old negotiation: S16LE S24_32LE
new negotiation: F32LE S24_32LE
The link between the audio-dsp nodes has no restriction, so previously
it would negotiate S16LE, which would then fail to negotiate with alsa-sink
because fmtconvert does not know how to convert S16LE to S24_32LE directly.
With this change, the middle link negotiates to F32LE, which can be
converted to anything.
