For hdmi and iec958 devices, enumerate the iec958 formats and
codecs. Initially only PCM is supported as a codec but with
a property or an init option, the list of codecs can be dynamically
configured.
Add Format subtype and codec property.
Add codecs to include in the format codec property.
Add property to configure elements with supported codecs.
Add type info.
When in passthrough mode, use the position io to update the
io_rate_match fields for the follower. This makes it possible for the
follower to also provide the right amount of data when the converter
is not selected in passthrough.
Add an option to configure the converter in None port config where it
removes all the ports. We can use this when removing the converter to
make sure all it's ports are removed.
When we remove the converter, make sure we expose the follower ports
directly so we can use them for passthrough.
We only need an info update to restore the stream so check for that
in addition to a param update.
The info update can be used to restore the stream and the param update
to update the stored value.
When switching back from the recording mode, save currently selected
profile as the default headset profile to use in future.
This allows users to select a specific profile they want to use for
input, and to avoid autoswitching to HFP/HSP if they want to use a
different microphone, without needing to edit configuration files.
There doesn't seem to be a way to control the A2DP duplex microphone
HW volume gain, and devices sometimes have very low mic volumes.
Work around this by boosting the software volume scale by +20 dB. If it
causes clipping, the user can just reduce the volume to bring SW gain
below 1.0.
aptX-LL sink devices may send back mSBC encoded data corresponding to
microphone input. It appears to be enabled when the bidirectional link
is set in the caps, and the device also supports this.
Implement mSBC decoding in the duplex channel.
Tested to be working on Avantree Aria Pro.
For unknown reason the BT socket when working with A2DP "duplex" stream,
sometimes stops waking up poll when data arrives. Regardless, recv() can
read data packets from it.
To work around this, when A2DP source is in duplex mode, instead of
polling on data, we poll on a timer.
Some non-standard A2DP codecs (FastStream/aptX-LL) have "voice duplex
channel" that can be used to provide an A2DP duplex mode.
Add support for duplex channels, accounting for the fact that the two
directions may be encoded with different actual codecs.
This pushes the latency to 256 samples, which still seems to work
Latency of 128 seems to be too small. (NB. the number of input samples
is also the size of packets sent in bytes)
Since pipewire clients usually use powers of two, this may be the
smallest value that makes sense.
Support the low-latency variant of the aptx codec.
The magic mostly seems to be on the device side, since the stream is the
same as standard aptx, but latency is smaller even if stream/packet
sizes are the same.
Sound output latency is noticeably less than with the standard aptx.
Tested on Sennheiser HD 250 / Avantree Aria Pro.
The codec in principle also supports bidirectional duplex streams,
but that is not implemented here.
The codec IDs are user-visible properties.
Some codecs can have multiple endpoints (e.g. different caps struct, or
multiple possible vendor ids), so this detail should not leak to the
user.
This property is exposed on the device Route and forwarded to the
nodes. It then configured the process_latency.ns field, which
influences the reported port latency.
This makes it possible to change the internal port latency on the
sink and source with pavucontrol and tweak the synchronization to
compensate for internal latencies in the device.
Disable flush polling when we don't have data ready to write to the
socket (or socket send failed). This avoids entering into a poll busy
loop, which may result to rtkit killing the process.
