Based on HFP specs, the audio connection is independent of the active
call status, which should be managed by the ModemManager part of the
plugin.
But when using HFP AG without modem attached, e.g. during zoom meeting,
the connection will be closed after a while unless call status has been
forced to active,
cf. https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/1744.
Currently and for HFP AG PTS tests requesting to get an audio connection
in 3 seconds after a call activates, this prevent to start audio
connection before starting a call.
This commit prevents to force the call status during audio (dis)connection
if a modem is available.
Get the ModemManager interfaces when the ModemManager starts after native
HFP has started.
Also add log topic to be able to select log level for modemmanager plugin
part.
Do not parse, and do not report the negotiated frame rate. It is
completely ignored. Until support for the `FrameDurationLimits`
libcamera control is implemented, ignore any and all frame rates.
This field is only used during the setup of buffers, so it can be
transformed into a local variable, but even that is not needed
because the data type can be chosen on a per-plane basis.
Make the params introspectable, use the names to configure them in
audioconvert.
Only use precomputed filters when using the defaults.
Tweak the kaiser filter alpha and n_taps based on values calculated from
stopband attenuation and transition bandwidth.
This was introduced before for some threading implementation to have
definitions for 'tss_t' and 'tss_set()' and similar, but these were
later removed, while the header was still kept.
This header doesn't exist on older systems with older glibc and is the
reason for the only patch we need to have in Chromium to make PipeWire
to build.
Refactor the IR loading code, Make some generic open/load/close
functions and handle the different filenames in the open call.
This makes it possible to reuse some of the delay and alloc code.
It also makes it easier to add new IRs loading code.
Support /IR:<rate>,<value1>,<value2>,... as an inline IR definition.
Flush errqueue for iso-io also from the media-source handler, to avoid
dropping packet tx reports.
This applies to bidirectional streams. The sink/source handlers poll on
different fd, one being dup'd, and epoll does not trigger them in any
specific interleaved order.
`libusb_free_device_list()` should be passed `true` as its second
argument to unreference every single device in the list.
Fixes: 5e0b63b149 ("bluez5: backend-native: use quirks + usb adapter caps for checking msbc")
`dbus_connection_register_object_path()` does not take ownership of the
path passed to it, so currently the callers `telephony_{ag,call}_register()`
leak a copy of the path string. Fix that by not making an unnecessary copy.
Fixes: b28399ac57 ("bluez5: add telephony D-Bus service implementation")
Add a new thread.reset-on-fork property for the thread creator. when set
to false, it will clear the default SCHED_RESET_ON_FORK flag and new RT
threads will be able to fork and inherit the rt policy and priority.
When creating a thread make sure we set SCHED_RESET_ON_FORK when the
thread.reset-on-fork property is not explicitly false;
module-rt needs to preserve the SCHED_RESET_ON_FORK flag when changing
the policy.
Set thread.reset-on-fork=false explicitly for JACK clients to restore
the JACK behaviour where implementations can fork and inherit the RT
policy and priority by default.
Fixes#4966
We need at least a default value and the default value as a possible
value.
For enum type in the PropInfo we usually list just the basic type and
then use labels to list alternatives.
SPA_POD_CHOICE_ENUM_Int must always take at least 2 values as first one
is default. (Just 1 value no longer works on current master, and it's
anyway incorrect.)
Replace with just SPA_POD_Int, as there's just one choice.
In the unknown form-factor case (which I saw on a set of headphones
here, Creative Zen Hybrid Pro), let's avoid an apocryphal acronym in
favour of a term that might be more familiar to the user.
Unknown preprocessor symbols are replaced with 0 in preprocessor
conditionals, so if `__has_attribute()` is not supported, then
the condition will be the following:
if 0 && 0(__cleanup__)
which is syntactically incorrect.
So split the condition and only check `__has_attribute()` if
it exists.
Fixes: 65d949558b ("spa: utils: add scope based resource cleanup")
Fixes#4962
Check if the pod size is at least big enough to hold 1 item when getting
array or choice items.
Check that the number of choice values is at least enough to handle the
given choice type. Remove some redundant checks.
Instead of always renegotiating a new format and buffers when the
driver changes, only renegotiate when the converter notifies us
of an EnumFormat change. It's possible that changing the driver does
not actually change the result of EnumFormat and then we would simply
renegotiate to the same format and buffers for nothing.
In the audioconvert, check if the target_rate of the new driver has
changed. We keep the last value in our own clock so that we don't
renegotiate when we are removed from a driver (which sets our own
clock as the position). The target rate influences the result of the
EnumFormat params when we are resampling and so we need to emit a
EnumFormat change.
Makes it possible to define audio channels and position with a
predefined layout string.
It is easier and less error prone to say "5.1" than to spell out
[ FL FR FC LFE RL RR ].
AUX channels have a special syntax. AUX<N> will make <N> AUX
channels. Easier to say AUX128 than to write an array with the
128 AUX channels.
spa_json_init assumes that we start in an object and always requires a
key/value pair. If the last part is a key, it returns and error and does
not want to return the key value.
This causes problems when parsing AUX0,AUX1,AUX2 or any relaxed array
withand odd number of elements.
Make a new spa_json_init_relax that takes the type of the container
we're assuming we're in and set the state of the parser to array when we
are parsing a relaxed array.
Fixes#4944
The behavior before b8eeb2db45 was that spa_audio_info_raw_update()
always sets audio.channels when audio.position is updated. The new
behavior does not set audio.channels when parsing audio.position.
This breaks things e.g. when combine-sink and loopback nodes are created
with only audio.position specified.
Restore the previous behavior.
