WirePlumber recently added a mechanism to force mono mixdown on audio
outputs, which is a useful feature for accessibility. Let's also expose
that setting via libpulse for existing audio settings UIs to be able to
use.
The dont-inhibit-auto-suspend flag does not do anything when using
direct-on-input-idx (capturing from a stream) in pulseaudio, so also
make it do nothing on pulse-server.
See #4991
When a device profile changes (e.g., Bluetooth headset switching from
a2dp-sink to headset-head-unit), the active port information changes
but PulseAudio compatibility layer clients don't receive the expected
PA_SUBSCRIPTION_EVENT_SOURCE or PA_SUBSCRIPTION_EVENT_SINK change events.
Root cause:
The collect_device_info() function updates the active_port index from
SPA_PARAM_Route parameters, but doesn't update the corresponding
active_port_name field. When update_device_info() uses memcmp() to
detect changes in the device_info structure, it compares the entire
structure including active_port_name. If the pointer value doesn't
change (even though the actual port changed), no change is detected,
and the change_mask flag (PW_MANAGER_OBJECT_FLAG_SOURCE/SINK) is not
set, preventing subscription events from being sent.
Solution:
After setting active_port in collect_device_info(), look up the
corresponding port name from SPA_PARAM_EnumRoute parameters by
matching both the port index and direction. Initialize active_port_name
to NULL at the start to ensure it's always recalculated.
This fix applies to all device types (Bluetooth, USB, PCI sound cards)
and all profile switching scenarios, ensuring applications using the
PulseAudio compatibility layer receive proper device change notifications.
Tested with:
- Bluetooth headset profile switching (a2dp-sink ↔ headset-head-unit)
- Applications subscribing to PA_SUBSCRIPTION_MASK_SOURCE/SINK events
- Verified no regression in audio initialization
Add a function that accepts the size of the position array when reading
the audio positions. This makes it possible to decouple the position
array size from SPA_AUDIO_MAX_CHANNELS.
Also use SPA_N_ELEMENTS to pass the number of array elements to
functions instead of a fixed constant. This makes it easier to change
the array size later to a different constant without having to patch up
all the places where the size is used.
Instead of using timerfd, use the context timer-queue to schedule
timeouts. This saves fds and removes some redundant code.
Make the rtp-source timeout and standby code a bit better by using
atomic operations.
When clients connect with IP, add the peer IP address to properties. We
might use this later to make a better stream node.name than a copy of the
client application name.
Use the timer queue for scheduling stream and object data timeouts.
This avoids allocating timerfds for these timeouts and the timer queue
can handle many timeouts more efficiently.
If we don't get a link on a stream, we might never send a create stream
reply. The client handles this fine by timing out after 30s and dropping
the stream, but the server holds on to the pw_stream forever (or until
the client quits).
Let's add a timer to clean up such streams on the server.
Fixes: https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/4901
Only send out SUSPENDED event when there is a change in the suspended
state. This avoids sending out unsuspend events when we simply uncork.
Implement the fail_on_suspend flag for capture and playback streams.
Instead of suspending those streams, we need to kill them.
The parser does not check that POD arrays have the correct size for
their type, so the calling code must do that.
This also enumerates some of the code that cannot handle the size of the
values of an array not being the exact expected size for its type.
There is a lot of it.
config.h needs to be consistently included before any standard headers
if we ever want to set feature test macros (like _GNU_SOURCE or whatever)
inside. It can lead to hard-to-debug issues without that.
It can also be problematic just for our own HAVE_* that it may define
if it's not consistently made available before our own headers. Just
always include it first, before everything.
We already did this in many files, just not consistently.
When we simply need to change some state for the code executed in the
loop, we can use locked() instead of invoke(). This is more efficient
and avoids some context switches in the normal case.
Including C headers inside of `extern "C"` breaks use from C++. Hoist
the includes of standard C headers above the block so we don't try
to mangle the stdlib.
I initially tried to scope this with a targeted change but it's too
hard to do correctly that way. This way, we avoid whack-a-mole.
Firefox is working around this in their e21461b7b8b39cc31ba53c47d4f6f310c673ff2f
commit.
Bug: https://bugzilla.mozilla.org/1953080
Make the state_changed event and _get_state() function set errno with
the current error value if the state is in error, so that application
can use this to give more detailed error reporting.
Use this in alsa, v4l2 and pulse to give some other error codes than
EIO.
Fixes#4574
We use the remote name as a suffix for the default server address and so
it should not contain any slashes. Take everything after the last slash
if there is one.
When we write samples, check if we make a jump in the ringbuffer and
clear the samples we jumped over.
If we don't do this, the reader side might pick up old samples that we
didn't write or clear but that are now available for reading after we
made a jump in the ringbuffer.
This migh not be exactly what pulseaudio does but it is good for now.
Fixes#4464
