Add feedback and feedforward controls to the delay. This makes it
possible to make comb and allpass filters with the delay to build
custom reverb effects.
built:
* es_builder: create stream with state variables and counters
* acmp: do not use the stream list, go through the descriptor to find
the index
* stream: do not store redundant information such as the index and
descriptor
* internal: removing the stream server and function associated to it
module-avb: internal, stream: removing server_find_stream
Otherwise we might end up with partial channels when code doesn't
check the unpositioned flag. It's better to set everything to unknown
when there is a mismatch between channel count and layout.
The "+BIND: <a>,<state>" reply to AT+BIND? should be sent for every
supported indicator.
"AT+BIEV= <assigned number>,<value>" should only be provided for
enabled indicators. The AG shall respond with an ERROR response code
if it receives updates for disabled or unknown HF indicators or values
that are out of bounds.
This allows to pass PTs tests:
- HFP/AG/HFI/BV-02-C
AG receives an updated HF Indicator value
- HFP/AG/HFI/BI-03-C
AG receives invalid updated HF Indicator values
When releasing multiple transports, call Release() simultaneously
instead of serializing the calls.
This operations still needs to be blocking currently, as all releases
have to finish before we do other state-modifying ops.
This works around broken firmware on Creative Zen Hybrid Pro with BAP,
whose Disable command misbehaves when shutting down sink + source CIS
otherwise. It's also anyway better to shut down everything at once.
The dont-inhibit-auto-suspend flag does not do anything when using
direct-on-input-idx (capturing from a stream) in pulseaudio, so also
make it do nothing on pulse-server.
See #4991
Increase the cutoff frequency for the lower quality modes. This should
give significantly better high frequency preservation at the expense of
more (but likely inaudible) aliasing.
Reduce the cutoff for the higher qualities in blackman and exp to
compensate the the wider transition band in those windows.
Increase cutoff for kaiser because of the sharper rolloff.
Automatically make an empty PeerCapability param when we receive a
Latency event without a PeerCapability. This makes new client always
receive a PeerCapability param, even when the other side did not provide
anything or when the server is too old to collect them.
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.
to attach ressource to the descriptors instead of having them splitted.
It is the case for the avb-streams which in a seperated list. Instead they
should be encapsulated within the descriptor itself, as one cannot leave
without the other.
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.
