Forward existing impl-node peer_added and peer_removed signals on the
stream.
Because the stream is not on the server, there is no impl_node in the
node target. Add the node id to the peer_added/removed signal and use
that for the stream event argument. Implementations can then look up the
details on the global.
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
Some downstream elements trigger re-negotiations when the stream is
running, e.g. due to output window resizing for the below pipeline.
gst-launch-1.0 pipewiresrc ! videoconvert ! video/x-raw ! gtk4paintablesink
This fails due to a side effect introduced by commit 77143e54 with
handle_format_change() setting `is_video` to `false` when param was
NULL. Fix this by explicitly tracking media type and not just video
using a boolean.
GST_SECOND * t.rate.num can turn into a negative gint, resulting in
assertions like:
_gst_util_uint64_scale_int: assertion 'num >= 0' failed
Just use the 64bit version instead.
Add IGMP recovery mechanism that monitors RTP packet reception and
triggers multicast group refresh when no packets are received if
a deadline is reached. The deadline is configurable via a new stream
property "igmp.deadline.sec" (in seconds), with the default value
being 30 seconds (and a minimum of 5 seconds).
A timer checks regularly if the deadline was reached. That timer's
interval is set by the igmp.check.interval.sec property (in seconds),
with the default value being 5 seconds (and a minimum of 1 second).
When the deadline is reached, the mechanism performs IGMP leave/rejoin
operations to refresh multicast group membership. This ensures RTP
data continues to be received when network conditions cause IGMP
membership to expire or become stale due to router timeouts or
network issues.
Add IGMP recovery mechanism that monitors SAP packet reception and
triggers multicast group refresh when no packets are received if
a deadline is reached. The deadline is set to half of the cleanup
interval, with a minimum of 1 second.
When the deadline is reached, the mechanism performs IGMP leave/rejoin
operations to refresh multicast group membership. This ensures SAP
announcements continue to be received when network conditions cause
IGMP membership to expire or become stale due to router timeouts or
network issues.
Keep the samples in the ringbuffer that are needed the next cycle to
avoid discontinuity when the aec blocksize is not equal to or divisible
by quantum.
Call process() when capture and sink ringbuffers contain data from the
same graph cycle and only process the latest block from them to avoid
adding latency that can accumulate if one of the streams gets more than
one buffer before the other gets its first buffer when starting up.
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.
Move the code to recalculate the hash and version into make_sdp. Add a
boolean argument to the make_sdp function. Recalculate the hash and
version when we are making a new SDP and leave the old values if we
are making an SDP to compare against the current one.
Add support for rlimit.<resource> = <value> in the context.properties to
adjust the rlimits of the process. A value of -1 sets the max limit.
This can be used to increase the number of file descriptors in a
pipewire process when select() is not used, for example. Other resource
limits might be interesting as well maybe.
Fixes#4047
We need exclusive port use if we negotiated SyncTimeline because there
can only be one consumer of the syncobj.
We also need to enable reliable transport if synctimeline is supported
but the release flag isn't.
Add some more logging to the port when the exclusive and reliable states
changed.
Fixes#4885
Add a PW_KEY_NODE_RELIABLE and PW_KEY_PORT_RELIABLE property. the port
property value is inherited from the parent when not explicitly set.
Setting the property on a port will activate a more reliable tee, that
actually only recycles buffers that were consumed. It will also activate
a mode in stream that gives out new buffers only when the previous one
was recycled and nothing else is queued. This is necessary to avoid
queuing in the stream when the other side is not consuming.
When a link is async but the output node is a driver of the input, we
can avoid async io. This also removes a potential out-of-order buffer
recycling when the node resumes at a different cycle.
See #4885
Add an example producer and consumer using the SyncTimeline metadata.
The syncobj are just eventfd for the purpose of the example.
Also demonstrate the RELEASE feature when negotiated.
Add some switches to tweak the SyncTimeline and features support.
See #4885
Avoid shadowing some variables from the parent block.
The node of a target can be NULL when the target is running in another
instance. We already do some checks for this but make sure we never
deref the NULL pointer.
Fixes#4922
Add an example of a filter sink that requires MemFd memory on the
input port. Show that it gets automatically mapped and that it contains
MemFd memory.
Fixes#4918
Don't just blindly mmap the buffer but only when the data pointer
is NULL. If it was mapped already by the peer or the adapter or the
buffer allocation, we don't want to mmap it again and override the buffer
data pointer.
Also mmap with the permissions on the data. There is not much point in
limiting the permissions for an input port (to read only). We could do
this but then we would not be allowed to modify the existing data
pointer. The problem is that when the stream mmaps the data as READ only
and set the data pointer, if it is then handed to the mixer, it would
assume it is mapped with the permissions and then segfault when it
tries to write to the memory. It's just better to only mmap when the
data is NULL.
