Take the queued input samples into account when calculating the
required input size. This can be 0 when there is still enough
data queued in the input for another period.
Handle 0 read_size in alsa-source and make it push out a 0 buffer,
this will then drain the resampler and make it ask for a new buffer
size. This makes the transition from one period to another more
seamless for the resampler.
Fixes#805
Instead of requiring the upstream node to resubmit the delayed
samples, keep the samples ourselves. The benefit is probably too
small to measure but it simplifies things a lot.
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.
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.
Send suspend to the node when suspending. This is usually the same
as puse for all nodes.
Implement negotiation when we Start audioadapter. This makes it
easier that to track the ports that are negotiated for now.
Use Suspend to clear the audioadapter negotiation.
Add flags to the rate match io area
Add flag to activate/deactivate rate match
Set active flag in rate match when slaved
Update rate before starting resample
Move some things around. Move the duration of the current cycle
to the clock. Also add the estimated next timeout to the clock.
Add a generic media specific counter to the clock.
Clean up the position_bar info. We can do with only a double beat
value and make the signature in floats.
Flesh out the io_position info. This has now the information needed
to convert a raw clock time into a stream time. It basically has
the same kind of features as GStreamer segments such as looping,
variable rate playback etc.. It also contains the state of the
timeline (paused/playing) and it can be used to update the position
and state from clients.
There is also extended information in the position field that
clients can update when they can.
Plugins basically only update the clock info they get (and use
the position info to check if they are slaved or not).
Before each cycle, check if there is a pending position update and
apply it.
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.
Remove the CAN_USE_BUFFERS flag, it is redundant. We can know this
because of the IO params and buffer params.
Add flags to the port_use_buffer call. We also want this call to
replace port_alloc_buffer. Together with a new result event we can
ask the node to (a)synchronously fill up the buffer data for us. This
is part of a plan to let remote nodes provide buffer data.
Use a new rate_match io area to exhange rate matching info between
sink/source and resampler.
Compensate for the rate match delay when scheduling timeouts.
Let the resampler notify the source of how many samples it needs to
produce the desired quantum. Make sure we keep an extra buffer in
the device to be able to make this possible.
Let the adapter directly call the slave node process function.
Define a set of standard factory names and document what they
contain. This makes it possible to change the implementation by
mapping the factory-name to a different shared library.
The interface struct has the type,version and methods of the
interface.
Make spa interfaces extend from spa_interface and make a
separate structure for the methods.
Pass a generic void* as the first argument of methods, like
we don in PipeWire.
Bundle the methods + implementation in a versioned inteface
and use that to invoke methods. This way we can do version
checks on the methods.
Make resource and proxy interfaces that we can can call. We
can then make the core interfaces independent on proxy/resource and
hide them in the lower layers.
Add add_listener method to methods of core interfaces, just
like SPA.
Also emit port info the the merger monitor ports.
Proxy params and buffers to the right monitor ports.
fmtconvert does not always have dsp ports
Increase resampler output buffer sizes so that we can up and downsample.
Fix little off by one in native resampler.
Fix passthrough in speex resampler.
Add a sinc based resampler that, unlike speex, avoids memcpy and
works directly on the source data. It also allows for ssse3
optimizations and aligned loads. It will later switch to table
interpolation when doing variable rate.
