Move some of the tracking code for the DLL to where it is used.
Add resync.ms (default 10) option at which we give up rate adjusting
and instead do a hard resync. This results in a jump in the position
of the graph clock.
When we get a node info about our own state, we can use the active state
of the node to decide if we are running or not.
Otherwise, we will try to hide ports from JACK clients that suspend
(while still active).
See #3794
We can increase the MAX_LATENCY again if we increase the amount of
buffers when we are using a small buffer.
Normally we ask for 4 * quantum-limit as the buffer. This should be good
to use 1 buffer and quantum-limit as the quantum with enough headroom
to not run out of buffers.
If we are however using less buffer-frames we need to be careful and
allocate an extra buffer. Imagine using a buffer of 4096 frames, we can
support a quantum of up to 2048 frames if we use 2 buffers.
See #3744
Half of the buffersize is not enough to support as a max-quantum, we
need to divide by (4 * frame_scale) to allow some headroom and account
for the DSD scaling. We do the same calculation to suggest a buffer size
using the quantum-limit.
See #3744
When the filter has no format property, just enumerate all possible
framerates. Handle error case where the filter has the wrong type.
Makes gst-launch gstpipewiresrc ! video/x-raw ! fakesink work.
See #1793
Actually count the number of frame fractions we add. If we added 0, we
don't have any supported framerate that intersects with the filter and we
try the next frame size.
See #1793
When we are notifying the application of changed params, don't emit any
changes applied to the params from within the callback with
pw_stream_update_params(). This will be done after we complete the
callback.
This also avoids reseting the change counter so that we don't
accidentally think we updated the formats param when we simply changed
some other params.
Unknown transports visible in DBus usually belong to a different
sound server instance that is talking to BlueZ.
Explain this in the warning message that we log, so that people can more
easily understand why things are not working.
When freewheeling we will immediately schedule a new graph cycle when we
get a process call because the graph completed.
When the process call is not done, because of some xrun or
because some node was removed that causes the graph to fail completion,
The next cycle will happen after a timeout.
This timeout was calculated as the ideal wakeup time (after a quantum of
time) and would accumulate for each timeout. The result is that the
timeout ended up far in the future and would stall the freewheel driver
for a long time.
Fix this by always setting the next timeout to wakeup time + freewheel.timeout
seconds. Also add a config property for the timeout (10 seconds, like
jack2 by default).
Primark True Wireless earbud doesn't support sbc-xq. Having it
enabled causes bluez to enter into a loop enabling/disabling
the device dozens of times per minute, making it unusable.
GStreamer uses a time stamp for the decoding time, but PipeWire uses an
offset to the presentation time. Thus, the pipewiresink must not use the
DTS as dts_offset, but has to calculate the offset.
If the buffer's DTS is invalid, assume that dts is pts.
get_device_info() requires us to call update_object_info() in the added
and updated events.
Fixes a bug where the properties were invalid in the avahi txt record.
When the session manager sends an error to the client, it typically
also destroys the node after the error, which causes the stream to go
to STATE_UNCONNECTED via proxy_removed(). In that case, make sure
we exit the loop early, otherwise it will take 30 seconds to unblock
gst_element_set_state()
This is a revised version of the fix that was commited via !1763
and then reverted, as it was problematic. Now the code ensures
that it breaks out only if the state was previously CONNECTING
or higher.
The monitor sources also list the port of the sink and so the active
port needs to be collected as well so it doesn't fall back to the first
port (which might not be available).
Keep a list of ports for the node. When the node goes away, clear the
port links to the node. Handle the case where the port no longer has a
node.
This avoids a crash when, for example, the node permission is removed
and the port points to the now freed node_data.
Fixes#3708
As part of the setup for IRQ based scheduling, a period event
was installed. Not only is a timer based polling unecessary for
IRQ scheduling, depending on the state of the system, the timer
could fire far enough from the IRQ, causing alsa wakeup events
with no data in the ring buffer. Pipewire would identify these
events as an "early wakeup", adding an extra quantum of time
to the next_time estimate, skewing the clock and causing issues
with apps that depend on precise timing.
Update the started and ready state after we suspend/pause the node so
that we don't complain if scheduling happens between setting the fields
and actually stopping the follower.
Also only complain when the scheduling happens when the node is not
ready. It is possible that the node is scheduled before we manage to set
the started field.
Move the driver and warned bits after the int field in the struct so
that they are placed in separate memory.
Otherwise, a write from the data thread might race with a write from the
main thread and leave the bits in the wrong state.
This reverts commit 49cdb468c2.
We should not do this, the nsec field should be relatable to the clock
monotonic time. If we use the estimated time, without actually using it
as a timer, we might end up with a wakeup time in the future compared to
the MONOTONIC clock time.
Instead, you can use the estimated current time simply by subtracting
the rate corrected duration from the next_nsec. This is really only
useful for some selected use cases (like in the JACK library).
This fixes some issues where in pro-audio mode, a client would try to
compare the current MONOTONIC time to nsec and find that it is in the
past.
This commit was done in an attempt to fix#3657 but it turned out the
real problem was something else.
We should not use nsec from the clock as the current_usec because
current_usec is supposed to be an idealized time when the wakeup would
have happened and nsec is when it actually happened.
Instead use next_nsec and subtract the rate corrected period from it to
get the idealized current_usec.
We can still use nsec to calculate the elapsed time since the wakeup and
be sure that it is always in the past.
The Pro Link 1 replies with Audio-latency=0, patch that up to
1500ms to make it work again.
Previously it configured 1500ms as the default latency but that seems
unnecessary in the usual case.
Fixes#3698
For params that don't emit change events, the param enumeration does not
start core sync, and its enumeration may be incomplete if a previous
core sync completes first.
Fix by always starting a core sync if we are updating params.
This fixes dev_info.active_port being sporadically SPA_ID_INVALID if
there is event pressure, which causes do_set_volume randomly set the
volume on the node instead of setting it on the device. This can be
triggered e.g. by shaking the volume slider in Pavucontrol with mouse.
The alsa sequencer rate matching was not actually working correctly.
It would compare the previous queue time with the current time and
compare that to the quantum. This would include uncorrected errors from
jitter and would result in the timeouts being scaled in the wrong
direction forever.
Instead, calculate an ideal queue time and compare our current queue
time against that. We then use the correction to scale the timeout or
the next queue time prediction.
Also use the predicted time as the base time for the event timestamps.
this results in less jitter.
Fixes#3657
