According to the alsa-info.txt in the pipewire issues of #747 and #1206,
the Front Playback Volume is shared by Headphone and Lineout or
Headphone and Speaker, But Headphone, Lineout or Speaker they all have
independent Playback Switch, change to only use switch to mute the
Lineout or Speaker. This could resolve the issues of #747 and #1206.
See #1206 and #747
Check if the port has latency param and only try to set the
latency param when it appears available. This avoids sending unknown
latency params to an old client and erroring out.
RLIMIT_RTTIME is Linux specific, there is no equivalent replacement
for FreeBSD. Save trouble and build module-rt for Linux only, rtkit
WIP fork should be fine on FreeBSD for now.
If `struct pw_map::free_list` is not initialized to `SPA_ID_INVALID`,
then `pw_map_insert()` will try to read `map->items.data[0]` when
inserting the first element, but `data` is a `NULL`
pointer after initialization.
Emitting events using the listener_list can not be done from multiple
threads at the same time. For this reason, make a copy of the events
with the process event and call it explicitly from the data thread.
See #1122
If the message was too long, then the `vsnprintf()` call would
fill up `location`, leaving no space for the color escape sequence
and the newline, causing a stack buffer overrun here:
size += snprintf(p + size, len - size, "%s\n", impl->colors ? suffix : "");
Fix that by reserving the last 24 bytes of the message buffer.
Add support for listening on IPv6 addresses.
The following address formats are supported:
* tcp:[<ipv6-addr>]:<port>,
* tcp:<ipv4-addr>:<port>,
* tcp:<port>, and
* unix:<path>.
The IP addresses are parsed using `inet_pton()`,
only the formats supported by that function
are accepted.
The IPv6 address must be surrounded by square brackets,
they do not mean "optional" here. Specifying only the
port is equivalent to the following two addresses:
* [::]:<port>, and
* 0.0.0.0:<port>.
Address parsing has been made stricter: the port
must always be specified explicitly.
Fixes#1216.
Keep both input and output latency on the ports.
When setting the port latency only apply the rate latency values
and assume the node latency does not depend on buffer-size or time.
This is likely true because jack only knows about latency relative
to the rate.
It combines all latency from one direction and sets it as the latency
for the other direction.
Also keep only one latency value per port, in the object.
Rework the param handling a little.
Keep properties for ports. Use this in the client and port info.
Trigger the latency callback when the port latency changed.
Update the port latency when it changed.
Trigger a recalculation of the port latency when the peer port
latency changed.
This makes the latency update from a port propagate through the
pipeline.
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.
Add a new latency param that contains a latency object.
The latency object contains the min and max delay from a port to
the terminal sink/source. It is also possible to express this
delay as a fraction of the quantum to avoid having to recalculate
the latency every time the quantum changes.
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.
