When the search path is /usr/lib/, /usr/lib/foo.so fails to load because
there is no / after the search path. Fix this by requiring that either
the search path end with / or the following char is a /.
When we add a Format property after we dereffed all the other params in
the builder, we might relocate the builder memory and invalidate all
previously dereffed params, causing corruption.
Instead, first add all the params to the builder and then deref the
params.
There is a special case when we have both a capture and playback
stream. The capture stream will receive all filter params and the
playback stream will just receive its Format param.
Fixes#5202
If we pass a path /usr/libevil/mycode.so, it might have a prefix of
/usr/lib but we should still reject it. Do thi by checking that after
the prefix match, we start a new directory.
Check that the number of fds for the message does not exceed the number
of received fds with SCM_RIGHTS.
The check was simply doing an array bounds check. This could still lead
to out-of-sync fds or usage of uninitialized/invalid fds when the
message header claims more fds than there were passed with SCM_RIGHTS.
Found by Claude Code.
They are emited from the streaming thread and therefore can be emitted
concurrently with the events on the main thread. This can cause crashes
when the hook list is iterated.
Instead, make those events into callbacks that are more efficient,
and threadsafe.
Add a control.ump port property. When true, the port wants UMP and the
mixer will convert to it. When false, the port supports both UMP and
Midi1 and no conversions will happen. When unset, the mixer will always
convert UMP to midi1.
Remove the CONTROL_types property from the filter. This causes problems
because this is the format negotiated with peers, which might not
support the types but can still be linked because the mixer will
convert.
The control.ump port property is supposed to be a temporary fix until we
can negotiate the mixer ports properly with the CONTROL_types.
Remove UMP handling from bluetooth midi, just use the raw Midi1 events
now that the mixer will give those and we are supposed to output our
unconverted format.
Fix midi events in-place in netjack because we can.
Update docs and pw-mididump to note that we are back to midi1 as the
default format.
With this, most of the midi<->UMP conversion should be gone again and we
should be able to avoid conversion problems in ALSA and PipeWire.
Fixes#5183
Avoid doing conversions in the nodes between Midi formats, just assume
the imput is what we expect and output what we naturally produce.
For ALSA this means we produce and consume Midi1 or Midi2 depending on the
configurtation.
All of the other modules (ffado, RTP, netjack and VBAN) really only
produce and consume MIDI1.
Set the default MIDI format to MIDI1 in ALSA.
Whith this change, almost everything now produces and consumes MIDI1
again (previously the buffer format was forced to MIDI2).
The problem is that MIDI2 to and from MIDI1 conversion has problems in
some cases in PipeWire and ALSA and breaks compatibility with some
hardware.
The idea is to let elements produce their prefered format and that the
control mixer also negotiates and converts to the node prefered format.
There is then a mix of MIDI2 and MIDI1 on ports but with the control
port adapting, this should not be a problem.
There is one remaining problem to make this work, the port format is
taken from the node port and not the mixer port, which would then expose
the prefered format on the port and force negotiation to it with the
peer instead of in the mixer.
See #5183
Don't accept absolute library paths that are not in the search path,
skip the ../ in paths to avoid opening arbitrary libraries from
unexpected places.
When the driver changes, the clock position can also change and there
would be a discont in the rtp_timestamp.
This is not usually a problem except in RAOP mode where the base rtp
timestamp is negotiated and anything that deviates too much is to be
discarded.
If we are not using direct_timestamp for the sender, make sure we always
keep the rtp_time aligned to avoid this problem.
See #5167
Don't close an -1 fd in clear_data.
If we let the client allocate buffer, set our fd and data to invalid
values. If the client decides to renegotiate before we get the buffer
data we might otherwise try to clear the mem_id (default 0) or
close the fd (also default 0).
Fixes#5162
Sink/Source pairs should not have the same link-group otherwise the
session manager will not be able to autoconnect them with a loopback or
some other internally linked stream.
Fix path comparison in is_socket_unix() and don't unset LISTEN_FDS since
the function that uses it is called more than once and it was not unset
when sd_listen_fds() was used.
Fixes#5140
Roc-toolkit log records are captured via a callback and
written to PipeWire log with corresponding verbosity level.
The log.level config parameter limits record verbosity at
the roc-toolkit level.
Patch by Lairton Lelis da Fonseca Junior (@lairton)
Remove the hard skip for IPv4 link-local addresses and add an interface
binding (matching the existing IPv6 link-local behavior).
The host needs a link-local address on the interface (ip addr add
169.254.x.x/16 dev wlan0 or via NetworkManager +ipv4.addresses).
Fixes#4830
Socket activation uses sd_listen_fds from libsystemd, and can only be
compiled on systems with systemd.
This is an issue for Alpine / postmarketOS, where upstream has no
systemd package, but downstream depends on upstream's pipewire package
and wants to rely on socket activation. This also prevents using
socket-activation on other non-systemd distributions, including
non-Linux.
Implement equivalent functionality without a dependency on libsystemd.
This can easily be overlooked if the RTP rate equals the clock rate, which
is fairly common (for example, RTP rate and clock rate both being 48 kHz).
And, if an ASRC is active, and converting the output of the RTP source
node, the resampler's delay need to be taken into the account as well.
Clear the ringbuffer in stream_stop() when processing stops to prevent old invalid packets
from being sent when processing resumes via rtp_audio_flush_packets().
This ensures a clean state when the stream restarts.
Clearing the ring buffer is important not only in the direct timestamp
mode, but also in the constant delay mode, since missed packets can lead
to gaps in the ring buffer. These gaps may have stale data inside if the
ringbuffer is not cleared after reading from it.
In corner cases where the read and write pointers are very close, it may
not be possible to read out all the wanted samples. This can for example
happen when there is a jump in the graph driver position. Currently, the
code reads the wanted number of samples out of the ring buffer regardless
of the write and read pointer positions. It does so even when the read
pointer is ahead of the write pointer (that is, an underrun occurs).
Fix this by checking the fill level and reading only the available amount
of samples if that amount is less than the wanted amount. The remaining
space in the target buffer is then filled with nullbytes.
This reverts commit dcdc19238b.
Reverting this because it caused big sync errors of ~62 ms in test setups.
Further discussions about this can be found here:
https://gitlab.freedesktop.org/pipewire/pipewire/-/merge_requests/2666
Followup commits modify the device delay application (by scaling it),
which is another reason why this needs to be reverted.
Add 35 sec timeout for PLAY_SAMPLE streams to start streaming, similar
to what we do with normal streams, and fail playback if they don't
start.
This avoids pending sample playback using up resources indefinitely if
streams fail to start for some reason, e.g. session manager is not
linking them.
If we get EPROTO, we likely have missed on some messages from the
server, and our state is now out of sync.
It's likely we can't recover (e.g. if error is due to fd limit hit), so
just drop the server connection in this case, similarly as if we got
EPIPE.
When the graph has no inputs and the channels is set to 0, don't create
a capture stream. Likewise, don't create a playback stream when there
are no graph outputs and the output channels is 0.
You can use this to make a sine source or a null sink.
There is no reason to fail when there is no input or output port.
We can simply run the graph with what there is. Even if there is no
input or output at all, running one instance of the plugins is
possible.
Add a busy builtin plugin that has no ports and keeps the CPU IDLE or
busy for the give percent.
The filter graph has, after parsing, a default number of input and
output ports. This is based on the description or the first/last element
input and output ports. Pass this information in the properties when
we emit the info.
Don't use the number of configured input/output ports as the default
number of channels in filter-chain because this is only determined after
activating the graph. Instead, use the default input/output channels.
The result is that when you load filter-chain without any channel
layout, it will default to the number of input/outputs of the graph
instead of 0. This allows for the node to be visible in the pulseaudio
API.
Fixes#5084
