G722 codec id for ASHA is chosen arbitrarily to be different from
A2DP and BAP codec IDs. ASHA spec does not specify a codec ID like
A2DP/BAP.
In places where codec_id comparisons are done, ensure that the check
is done against the codec of the right type viz. A2DP/BAP/ASHA.
Make sure we only make the buffer for the follower larger when we
downsample because then we need to ask for more data from the follower
to fill up a quantum.
Never try to make the follower buffer smaller than the quantum limit.
The reason is that the graph rate could be decreased dynamically and
then we would end up with too small buffers.
See #4490
Load multiple graphs with audioconvert.filter-graph.N where N is the
order where the graph is inserted/replaced. Run the graphs before the
channelmixer.
Graphs can be added and removed at runtime.
Instead of recalculating what to do every cycle, we can prepare a
static schedule and just run that. We only need to reevaluate it when
something changes.
For input streams, first run the resampler and then the channelmix. This
ensures that the channelmix is run with the rate of the graph instead
of the rate of the input. This is nicer because rate and quantum align
with the graph and the sample accurate volume ramps will work as
intended.
For output streams, leave the resampler after the channelmix for the same
reasons.
Remove the chunk and add separate arrays with data and n_samples. This aligns
better with other methods and makes it possible to more easily reuse
arrays of pointers as input and output.
An input port has at most 1 link so we can avoid the loop and the
default port data. We can then also improve the logging and log
something when we set silence.
When api.alsa.split-enable=true for ACP device, instruct UCM to not
use alsa-lib plugins for SplitPCM devices.
Grab the information from UCM for the intended channel remapping, and
add the splitting information to the nodes emitted.
Session manager can then look at that, and load nodes to do the channel
splitting.
The current biquad calculations are based on RBJ's cookbook [1],
except for low-/highpass. Since the filter configuration is also
based on using the definition of Q, it makes sense to also align
the remaining calculations to use the same filter cookbook instead
of using resonance which doesn't result in the same coefficients
as when using Q.
[1] = https://www.w3.org/TR/audio-eq-cookbook/
Currently, the PipeWire daemon registers BlueZ LE Media Endpoints
with audio capabilities covering all settings defined in the BAP spec.
However, some scenarios might require the capabilities to be restricted
to specific configurations.
This adds a method to read LC3 codec specific capabilities from the
Wireplumber config file, and provide those settings when registering
Media Endpoint objects with BlueZ. If the values are not present in
the config file, all settings will be used by default.
Below is an example of how to set the LC3 capabilities in the config
file, to support the 16_2 setting from the BAP spec:
bluez5.bap-server-capabilities.rates = [16000]
bluez5.bap-server-capabilities.durations = [10]
bluez5.bap-server-capabilities.channels = [1, 2]
bluez5.bap-server-capabilities.framelen_min = 40
bluez5.bap-server-capabilities.framelen_max = 40
bluez5.bap-server-capabilities.max_frames = 2
The ladspa plugin uses `dlopen()`, etc. directly,
so add the `dl_lib` dependency. This is not necessary
in a new enough environment since newer glibc versions
have merged most things into libc.
For a BAP Broadcast Source endpoint, the QoS sync_factor enables the user
to adjust the Periodic Advertising interval based on the ISO interval
configured for the stream:
PA_Interval = sync_factor * ISO_Interval
Currently, this value is hardcoded to 2. This commit makes the sync_factor
configurable in the Wireplumber config file, along with the other config
parameters for BIGs.
The EBU R128 filter measures the signal and generates LUFS control
notifications for further processing.
It also adds a plugin that can convert LUFS to a gain (based on a target
LUFS).
Also add an example filter-chain to enable the EBU R128 measurement and
how to use the results to adjust the volume dynamically.
See #2286#222#2210
Don't activate the nodes while linking but make a last stage where all
the nodes are activated. This makes it possible to better set up the
nodes based on the port data.
While initializing ALSA cards with UCM, we call pa_alsa_ucm_add_port()
for each UCM device for each UCM verb. This checks if a port has been
already added by name and skips port initialization if it is already
done. Different UCM verbs can have devices with the same name, which
means their port names end up being the same. So, this port creation
step currently is only done for one UCM verb for both UCM devices.
Volume control setup is also part of this process. The UCM devices only
know about the volume mixer information from the UCM verb that they are
defined in, so the volume control setup is done for one UCM verb at a
time. Skipping this setup when a same-named port exists means only the
profiles belonging to the first initialized UCM verb have working
hardware volume control.
Move the volume control setup out of the port initialization block so
that we try to do it every time, therefore for every UCM verb. However,
check that we don't run it twice for the same UCM verb for a port.
In theory, the PlaybackVolume etc. value can be different per-verb
for the same device, so we can't simplify this code to a single volume
setup per port.
Link: https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/840
Signed-off-by: Alper Nebi Yasak <alpernebiyasak@gmail.com>
Encoders and some decoders have additional internal latency that needs
to be accounted for.
This mostly matters for AAC (~40ms), as the other BT codecs have much
lower delays (~5ms).
Airpods don't follow the specification and set multiple bits in AAC
object type, including the ELD bit, but actually want AAC-LC. So check
the AOT in the right order.
When in A2DP sink role and remote end switches codec, BlueZ nowadays
appears sometimes emit first SetConfiguration (creating new transport),
and then ClearConfiguration (removing old transport).
Handle this case: emit profiles_changed event always when transports
come/go.
Redefine profiles_changed() to take bitmask of profiles whose connection
status changed, so we don't need to emit two remove+add events.
Commit 88bb0bd7cc ("alsa: Allow to augment ucm port properties") adds
icon properties for some common port names, so that GUIs can show a
relevant icon to help disambiguate devices. However, these still do not
show up in pavucontrol, because it shows icons based on mappings'
properties. Add the relevant property to mappings as well.
Link: https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/839
Signed-off-by: Alper Nebi Yasak <alpernebiyasak@gmail.com>
While trying to figure out device subsets that have aren't internally
contain conflicting devices, we walk through all possible subsets and
check each set if it satisfies ConflictingDevices/SupportedDevices
listed in UCM configuration. For a better user experience, we want to
skip subsets that are fully included in another valid subset we will
also generate.
The iterate_device_subsets() function that achieves the former is
intentionally in iterative form to avoid a stack overflow, since it will
walk through 2^n sets. However, the iterate_maximal_device_subsets()
function that skips incomplete sets is in recursive form, as I had
assumed tail-call optimization would take care of the potential problem.
Convert iterate_maximal_device_subsets() to an iterative form, because
the recursion seems to trigger a segfault with more than 16 devices on
PulseAudio. It doesn't seem to happen on PipeWire, but it's better to
not leave it to luck.
Link: https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/838
Signed-off-by: Alper Nebi Yasak <alpernebiyasak@gmail.com>
Because the max plugin is marked as SUPPORTS_NULL, the input and output
pointers can be NULL. Handle these cases. Also reindent with tabs (not 7
spaces).
