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.
This replaces the manual check for "true" and some (inconsistent) return value
of atoi. All those instances now require either "true" or "1" to parse as
true, any other value (including NULL) is boolean false.
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).
When we have a soft Mute or Volume, use the soft volume.
When we get a volume update with only a channel Mute/Volume, use
the channel volumes.
See #1140
Only run the lowpass filter on the LFE channel when we are upmixing
and there is a valid cutoff frequency defined.
Otherwise we might filter away a valid LFE channel.
Fixes: rhbz#1941366
Make a new softVolume property that contains only the soft volume
to apply.
In the case of HW/SW volume, we pass the real volume in the
channelVolume and the leftover volume in softVolume. We don't
use the monitorVolume for this anymore because it is a completely
separate volume handled by the merger node.
This way, channelVolume always represents the effective volume
set on routes, channelmix and merger and only the softVolume (when
available) is applied as software volume by channelmix.
This makes things map a bit better to what is actually happening with
the real volume and leftover software volumes after applying the
hardware volumes in the device.
With this change, the volume on the monitor is not affected by the
sink volume anymore and we can use the monitorVolume for this later.
This also means that the monitor volume in pavucontrol of the sinks
does not change when the sink volume changes. PulseAudio is inconsistent
here: If the volume is HW, the monitor volume is not affected, if the
volume is SW, it is. In PipeWire there is an option in merger to
let the volume affect the monitor with monitor.channel-volumes = true.
The channelmatrix was limited to the 19 known channels, leaving
only the first 19 channels usable.
Use the full 64x64 matrix instead and make sure to fill all 64
channels with the identity matrix in the case we need to copy or
distribute the matrix to unknown positions.
See #57
Just leave them be. Something else is supposed to changed the volumes
if needed. In the usual case where nothing changes, we don't want to
override the volumes.
Remove the soft mute/volume events, add a new method to get the soft
volume and use the volume_changed event to emit the changed soft and
monitor (real) volumes event for the node.
Make sure the monitor ports always uses the monitor volume, which is the
real volume unaffected by the mixer volumes.
This configures the soft and real volume on the sink/source in all
cases and makes the monitor port follow the real volume of the sink.
See #897
Make info parsing a bit easier to read by assigning the key and
value to temporary variables.
Improve the parsing of channelmap using json parser to make it
support more cases.
Add a unit test for channelmap parsing options.
Avoid calling memset on a large piece of memory when draining
the resampler because it might use up all the allocated time in
our realtime thread. Instead use a prealloced empty buffer.
Keep all the children param flags around and use those to decide
if something changed. Also don't change the param flag serial when
we are simply adding a listener.
This should reduce the number of param updates, most notably the
PropInfo that was updated along with the Props on volume changes.
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
