Parse TMAP / GMAP features from MediaEndpoint:SupportedFeatures and pass
them onto the codec in SelectProperties, so it can determine which
mandatory features the device supports.
Add configuration option for specifying which TMAP / GMAP feature bits
we advertise to remote side.
Although some of these could be determined automatically, for production
systems it's better to have explicit option to specify which ones should
be advertised as this may depend on HW capabilities.
When the dynamic data flag is set on the buffer data, it means the
consumer can deal with any data pointer set on the buffer and we can
simply pass the one from upstream to downstream. If the flag is not set,
we need to copy the buffer data.
See #5009
When we recalculate the headroom we also update the latency in frames.
We should express this latency in the graph rate. This is usually the
rate that is suggested in the target_rate but when we are forcing our
own rate (mostly when using IRQ or when in DSD/IEC mode) we should
ignore that value and use our own rate that we will force instead.
Fixes#4977
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.
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.
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.
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.
Reverts alsa-mixer.c changes from 32a3ffc74. The comprehensive ACP
layer fix (bdb82be4e) handles all scenarios including pro-audio
profiles that bypass mixer paths, making this approach redundant.
IEC958 (S/PDIF, HDMI, DisplayPort) switches default to muted in ALSA
drivers, causing no audio output on digital devices.
While UCM configurations and mixer paths can handle IEC958 unmuting,
several scenarios lack coverage:
- Pro-audio profiles (bypass UCM and mixer paths by design)
- Devices without UCM configurations
- Devices with incomplete mixer path definitions
- Cards with multiple HDMI/DP outputs (indexed switches)
This ensures IEC958 switches are enabled during device activation and
port changes. The implementation uses the device mixer when available,
falls back to the card mixer for pro-audio profiles, and enables all
IEC958 switches regardless of index.
Safe for all configurations: the operation is idempotent and provides
defense-in-depth even when UCM or mixer paths handle it correctly.
Tested on AMD Rembrandt GPU with 3 HDMI outputs in pro-audio mode.
When selecting an HDMI/DisplayPort (IEC958) output path, the hardware
mute switch remains in kernel default state (muted), causing no audio
output despite correct software routing.
Root cause: pa_alsa_path_select() only sets mute switches when
mute_during_activation is enabled. No mixer paths enable this setting,
making the switch configuration code unreachable for IEC958 paths.
Solution: Always set mute switches to match device mute status after
path activation, regardless of mute_during_activation setting.
Testing: Added test-alsa-path-select tool to verify the fix.
- Loads mixer path and calls pa_alsa_path_select()
- Verifies switch states match expected values
- Tested on AMD Radeon HDMI and Realtek ALC257 analog
Manual verification:
- Before: IEC958 switch OFF, no audio
- After: IEC958 switch set correctly, audio works
This bug was inherited from PulseAudio's ALSA mixer path code where
HDMI path configurations lack IEC958 unmute sections.
Fixes: https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/3261
See-Also: https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/562
See-Also: 33be660e4b
See-Also: https://bugs.launchpad.net/hundredpapercuts/+bug/681996
