When we enable a device, the node will be created and its software
volume will be set to 100%. Update the device volume with this as
well so that changing the volume has an effect.
Fixes#1198
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
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.
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.
Motu M4 has four inputs (two line-in inputs, and two complete ones
with gain and XLR and whatnot), as well as four outputs (two monitor
pairs, and an unnaccounted headphone).
Sadly, like a few other interfaces, it wasn't being given an input
profile, since the matching code goes through default.conf testing
each config, and ends up selecting 'analog-surround-40', which does
not have input mapping. The inputs would fallback to 'multichannel-
input', which also doesn't have input paths.
Add input paths to all analog-surround-* mappings, and remove their
'direction=output' fields since they handle both out and in.
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.
udev's ID_MODEL_ID and ID_VENDOR_ID are inconsistent: always 4-digit hex but
sound devices are prefixed with 0x, v4l devices are not. Depending on the
actual ID, the value will look like decimal (1234) or hex (a234).
pw-dump will then print those as either decimal integers (i.e. 0x1234 becomes
decimal 1234) or double (i.e. a234 becomes 41524.00).
Make this consistent by converting the string from hex do decimal where we
get it.
This way we can reset the dbus connection when we got a disconnect
and signal the event. This can then be used by the client to
do a new connection_get().
av_codec_next() was deprecated on 2018-02-06,
in lavc 58.10.100 (36c85d6e77). It was then later
removed on 2021-04-27 (4b2be3f8d). `av_codec_iterate()`
was added in lavc 58.10.100, so use that when available.
Fixes#1138.
`av_register_all()` is the only thing used from libavformat,
but that doesn't seem to be needed for codec enumeration,
so remove it, and call `avcodec_register_all()` instead
when libavcodec < 58.10.100.
ffmpeg.c had 'Spa V4l2 Support' in it, fix that to say
FFmpeg. Furthermore, change 'FFMpeg' to 'FFmpeg' in
ffmpeg-{enc,dec}.c as that is how it's referred to
in official texts.
Reset the software volume when initializing the device and when the
hardware volume is updated.
The software volume was set to 0 by default and if the session manager
did not change the volume enough to cause a software volume change,
it would remain 0 and there would be silence.
Also improve the debug a little.
See #1160#1167#1164#1049#1117
When we start freewheeling, pause the device and resume when we
finish freewheel.
In freewheel mode, just discard samples in the sink and produce
silence in the source.
