Setting the volume control ports with a volume param also initializes
the controls. This ensures the controls are not restored to their
default value when the graph is activated.
Fixes#5192
When we are asked to add noise bits, don't call the clear function.
Make the passthrough and clear-on-empty flags available with a new flag
field to make it more extensible.
Fixes#5260
Make a new library.filter-path for the filter-graph that will filter and
restrict the dlopen filenames (used for the LADSPA plugin only).
By default this is false and so filter-chain can load from absolute
paths without extra checks.
Enable the extra checks for the pulse LADSPA modules and the
audioconvert filter graphs because these allow loading LADSPA plugins
into other processes.
Fixes#5222
If the "debug" log level is not enabled for the "spa.alsa" log topic,
then there is no point in going into the loop and splitting the data
into lines, so skip that.
Do not use `strcspn()` because it assumes a null terminated string,
but the `fopencookie()` write callback receives a (ptr, length) pair.
So use `memchr()` instead to find the `\n`.
The `fopencookie()` write callback should return the number of consumed
bytes, but it currently only ever returns 0, which signals an error
condition according to the documentation.
Fix that by not overwriting `size`.
Fixes: 73073eb33f ("alsa: redirect alsa output to log file")
When the graph rate changes it is possible that the follower node can
renegotiate to the new suggested audioconvert rate without requiring
resampling and so the extra check for the disabled resampler is not
required.
Fixes#4933
File and Resource Handling: Medium
The V4L2 device file descriptor was opened without the O_CLOEXEC flag.
If a child process is subsequently spawned (e.g., via fork+exec), the
video device fd would be inherited, potentially allowing the child
process unauthorized access to the camera device.
Fixed by adding O_CLOEXEC to the open() flags.
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
File and Resource Handling: Medium
Several file and socket operations were missing the close-on-exec flag,
which causes file descriptors to leak to child processes created via
fork+exec. This could allow child processes unintended access to
privileged resources.
- node-driver.c: SOCK_DGRAM socket for SIOCETHTOOL ioctl leaked to
child processes
- pw-container.c: Unix domain listen socket leaked to spawned
container processes
- compress-offload-api.c: ALSA compress-offload device fd leaked to
child processes
Added O_CLOEXEC to open() calls and SOCK_CLOEXEC to socket() calls.
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
It's a terrible idea, doesn't work so well (locks up the data-loop when
read is blocked) and a security mightmare. If you really need to pipe
samples through some program, do that somewhere else, like from the
command line with pw-cat and pw-record.
ACP was re-selecting the “best” port on every port availability event,
even when a port was already explicitly selected by the user. This
differs from PulseAudio’s behavior, where port switching decisions are
left to higher-level policy.
This caused issues on devices where Line Out (speakers) and Headphones
share the same analog interface: when headphones are plugged in, ACP
would immediately switch away from the user-selected Line Out, or end up
in a state where no sound is produced despite selecting speakers explicitly from
clients like pwvucontrol.
Fix this by only re-evaluating and switching ports when:
- no active port is currently selected, or
- the active port has become unavailable
This preserves manual user choices and prevents ACP from fighting client
port selections during route activation.
Additionally, adjust ALSA mixer paths to better separate Line Out and
Headphones behavior:
- Disable Line Out controls in the headphones path
- Add explicit Line Out and Auto-Mute Mode handling in the lineout path
Together, these changes align PipeWire’s behavior more closely with
PulseAudio and fix cases where selecting speakers while headphones are
plugged results in no audio output.
Signed-off-by: John Titor <masumrezarock100@gmail.com>
MT7925 fails to setup a SCO connection that results to working LC3-24kHz
audio. Other controllers (Intel etc) appear to work OK.
Add quirk for disabling this codec, and disable it for this Mediatek
controller.
(cherry picked from commit 84e6845aa6)
When there is a stream without tx_latency enabled, the fill_count ends
with MIN_FILL value. This causes one buffer of silence to be written to
every stream before the actual data in each iteration.
Consequently, more data is written than consumed in each iteration.
After several iterations, spa_bt_send fails, triggering a
group_latency_check failure in few next iterations and leading to
dropped data.
Skip streams without tx_latency enabled in fill level calculations
to prevent these audio glitches.
(cherry picked from commit 42415eadd9)
FDK-AAC encoder uses band pass filter, which is automatically
applied at all bitrates.
For CBR encoding mode, its values are as follows (for stereo):
* 0-12 kb/s: 5 kHz
* 12-20 kb/s: 6.4 kHz
* 20-28 kb/s: 9.6 kHz
* 40-56 kb/s: 13 kHz
* 56-72 kb/s: 16 kHz
* 72-576 kb/s: 17 kHz
VBR uses the following table (stereo):
* Mode 1: 13 kHz
* Mode 2: 13 kHz
* Mode 3: 15.7 kHz
* Mode 4: 16.5 kHz
* Mode 5: 19.3 kHz
17 kHz for CBR is a limiting value for high bitrate.
Assume >110 kbit/s as a "high bitrate" CBR and increase the
band pass cutout up to 19.3 kHz (as in mode 5 VBR).
Link: d8e6b1a3aa/libAACenc/src/bandwidth.cpp (L114-L160)
(cherry picked from commit a35b6b0c4b)
The CHECK_PORT condition in impl_node_port_reuse_buffer was inverted with a negation operator, causing the function to reject valid output ports and accept invalid ones.
Fixes the logic so that valid ports proceed to buffer recycling and invalid ports are properly rejected.
When the port is destroyed we need to remove it from the mix_list or
else the process function will keep trying to use the invalid memory.
This is because the port logic does not want to call any functions on
the port (like clearing the IO or Format) after it emitted the destroy
signal and we need to clean up ourselves.
Fixes#5221
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 /.
The control values are only set in the port control_data after the
filter has been activated and the instances are created.
Property enumerations might happen before that and then we can either
return the current_value (when set in a control section or later with a
param property) or the default value.
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.
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.
Check codec kinds for each direction properly when mapping to profiles
corresponding to it. Being sloppy here masked another bug, so best fix
it.
(cherry picked from commit 22a5fad902)
HFP codecs don't have a direction dependent "target" profile, and this
function was returning false if A2DP is disabled.
Don't check target profile for HFP, leave checks to backend.
Fixes HFP-only configurations, which were missing profiles.
(cherry picked from commit 75c3d3ecf8)
Non-spec compliant devices may set multiple bits in code config, which
we currently reject in validate_config().
enum_config() does work to deal with multiple bits set, but this is
never used, so write the code in a simpler way to return a single
configuration.
(cherry picked from commit d42646e91f)
Non-spec compliant devices may set multiple bits in AAC AOT, which is
invalid.
In this case, we should normalize to MPEG-2 AAC LC which is the
mandatory value in spec, not to MPEG-4 AAC LC. In select_config() we
also prefer MPEG-2 over MPEG-4.
(cherry picked from commit 67b4732c26)
Some non-spec compliant devices (Sony XB100) set multiple bits
in all AAC field, including the frequency & channels.
Although they set multiple bits, these devices appear to intend that the
sender picks some specific format and uses it, and don't work correctly
with the others.
validate_config() already picks one configuration, so use the result in
enum_config(), instead of allowing also other settings.
Assume devices generally want preferably 44.1 kHz stereo.
Note we cannot reject the configuration, as BlueZ does not necessarily
retry, leaving the device connected but with no audio.
(cherry picked from commit 5f8ece7017)
When there is no DBus session bus, creation of the telephony backend
fails, and we later crash on null ptr deref.
In this case, avoid crash trying to create telephony_ag or iterate its
call list.
(cherry picked from commit f9e2b1d8b9)
