Add a new SPA_TYPE_OBJECT_ParamDict object that contains a struct with
key/value pairs. We're using something similar for Tags but this is a
more generic version.
Make a new Capability param that uses the ParamDict object. This is
meant to be used to describe capabilities with the generic key/value
struct.
Make a new PeerParam object where the keys are generic ids of the peer
objects and the values any Pod. The idea is to use this object to store
a peer objects. Make some helpers to iterate the peers and their
objects.
Add a new PeerCapability param that uses the PeerParam object with
Capability objects. This can be used to send the collection of
Capabilities from all peers to a port. This is a bit like the tags but
in a more generic way. The tags could also be implemented in this new
generic way later.
Make the PeerFormats use the same PeerParam of Format objects.
The Capability param is set on ports. impl-link will collect all
Capability objects into a PeerCapability object and send this to the
peer. The difference with the Tag param is that these Capability params
are not in any way forwared on the node automatically (like what is done
in the loopback module) because they represent the capabilities of the
ports betweem the link.
Some devices refuse to enable microphone if Streaming Context metadata
is just Unspecified.
Set some reasonable values for the stream context we create along TMAP,
and try follow CAP rules for selecting the PAC.
With BAP codec configuration selection goes via multiple functions,
which will need to maintain some private state.
Adjust media_codec to allow for that.
Use it for get_qos().
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")
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.
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.
Parse the audio.position spec completely so that we have the right
number of channels but only store the first max_position channels.
Also rename some field to make it clear that this is about the max
number of channel positions.
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.
