These headers are designed for including in the project. So the user doesn't
need to install valgrind-devel and we don't have to worry about whether the
headers are available or not.
Set this once during setup so we don't have to remember to call fflush() after
each logging operation.
Signed-off-by: Peter Hutterer <peter.hutterer@who-t.net>
In the interested of making the logs narrower, let's drop some digits from the
clock_gettime() seconds value. Clamping to 5 digigts, this gives us just under
28h before we wrap which is likely good enough for debugging.
Write the timestamp and location into a temporary buffer, then include them in
the message print. This makes bugs involving size vs length less likely and
provides a fixed limit for how much space the filename can take in the
message.
The two are functionally equivalent, but spa_snprintf never returns a value
higher than the size, preventing memory corruption where our input string
exceeds the target buffer size (see c851349f1).
Niche case: we can no longer differ between real overflow and fitting an
N-byte string into an N+1 sized buffer, we now get a "...truncated" message
now for log messages of exactly 999 bytes long.
bluez5 nodes will always be removed & created again during profile changing, hence
node volume & mute will always be reset. This is OK if profile did changed, because
session manager would carries volume & mute to bluez5 route param. But if profile
was not changed after setting profile (a2dp-sink-sbc -> a2dp-sink -> a2dp-sink-sbc),
session manager would think node volume & mute are not changed and no route
setting is performed, causing route volume out of sync with node volume.
To fix this, we emit node volume and mute right after bluez5 node is created.
Fixes#1254
Use _alibpref to check if a device needs a UCM local config. Mark
the device as such and use this to set the OPEN_UCM property on
the device.
Open the UCM for a card when the device has the property set. Use the
same logic for loading the UCM as the acp code.
See #1251
UCM devices can require local data from use_case_mgr_open() but since
we do that in a separate process, make sure we reopen the use case
manager in case we are passed a UCM device so that the config is
available.
See #1251
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.
