When using a filter, it makes more sense to use the default value
of the filter as a first attempt.
One case is in adapter when we try to find a passthrough format first. The
audioconverter suggests a default rate of the graph rate but the follower
filters this out for another unrelated default value and passthrough is not
possible (altough it would be because the default value of the filter is
in the supported follower range).
Fixes#4619
Use a simple free/active linked list for the filter-graphs and insert
the new filters in the right position in the list. Then simply copy the
list to an array for the processing thread.
when reconfiguring, set up all the filters again because the number of
channels might have changed.
SysEx in UMP can span multiple packets. In MIDI1 we can't split them up
into multiple events so we need to collect the complete sysex and then
write out the event.
Fixes SysEx writes to ALSA seq by running the event encoder until a
valid packet is completed.
Also fixes split MIDI1 packets in the JACK API when going through the
tunnel or via netjack.
When parsing the graph, parse the input and output port names into
a separate string array. This was we can keep them around when
setting up the graph.
Instead of setting up the graph right after loading it, do the graph
setup when we activate the graph. This makes it possible to pass the
input channels to the filter-graph and let it create the right amount of
plugins and ouput channels.
When setting up the graphs in the audioconverter, pass the current
number of channels as the input to the graph and keep track of the
channels that each filter produces.
This way we can also load a custom upmix or downmix graph, for example.
GoXLR Mini has different numbers of channels actually available (21, 23,
or 25) depending on its firmware/etc, but its UCM profile specifies
always 23. The count can then be bigger or smaller than what is actually
available.
Fail a bit more gracefully in the case of too few channels: create all
the split devices specified by the profile. The channels that aren't
actually available in HW just won't get routed anywhere.
ALSA upstream IIUC is saying that the channel counts should be fixed, so
spew warnings that say the UCM profiles are wrong if they look wrong.
The volume synchronization could be done even if there's no audio link
and so no transport opened.
This patch allows to send the Speaker (AT+VGS) and Microphone (AT+VGM)
commands at the end of the SLC. And to exchange volume updates using the
telephony DBus interface, even without a transport.
Depending on the direction of the conversion, we run the resampler
before or after the channelmix. This means we need to use the channel
count before or after the channelmixer instead of always using the
channels after channelmixing.
Fixes#4595
Although the two structs have same initial sequence, it's not really
correct to cast between their pointers. Alsa-lib also does this only
internally, but not in API.
Support also non-UMP IO with ALSA seq, in case either alsa-lib or the
kernel does not have UMP enabled.
Add configuration option "api.alsa.seq.ump" for optionally turning UMP
I/O off, for easier debugging.
Generally ALSA UCM profiles should all work as they're supposed to be
device-specific, so be more noisy when the profile fails to be supported
due to the PCM device failing to open.
Some logging on the probe outcome in failure case also makes
spa-acp-tool etc. log output easier to read.
In SplitPCM mode, Focusrite Scarlett Gen 4 (USB 1235:8218) UCM profile
specifies "CaptureChannels 2" for the Mic1/2 inputs, but
snd_pcm_hw_params_set_channels(2) fails for the HW device.
Fix by not requiring the channel count to be exact for SplitPCM, but
also allow larger numbers of channels than what UCM profile specifies.
HFP/HF/TWC/BV-03-C test, which setup an active and a held calls,
expects to receive AT+CHLD=1 (release and swap calls) instead of
AT+CHUP on active call hang up request.
As this changes the active call to disconnected and held call to
being active, the call states should be managed in hfp_hf_hangup
instead of waiting for +CIEV (callheld=0) event which will drop
the previously held call before AT+CLCC reply can inform this call
is now active.
HFP/HF/TWC/BV-01-C test creates an incoming call as soon as the SLC is
completed, i.e. a +CIEV: <callsetup>,1 event just after AT+CHLD=? reply
has been received. This try to parse the rfcomm->telephony_ag->call_list
which has not yet been created.
This commit move the telephony_ag creation to the SLC completed event.
When the pod to filter is in the target builder memory and reallocation
is needed, make sure we refer to the filter in the reallocated memory
instead of the old freed memory.
Fixes#4445
As AG, set node.rate for output streams that originate from remote
source, so that graph switches rate as needed. This follows what
pipewire-pulse etc. do.
The patch is by Wim.
In this error condition, execution is supposed to return immediately
because rfcomm is no longer valid. However, the code was incorrectly
changed to jump to the done label, which would try to use rfcomm
again to process pending commands.
To be sure that the AG reply correspond to the command sent, this
postpone the new command if previous reply (OK, ERROR or +CME) has
not yet been received.
The postponed command is sent on reception of the reply.
This allows implementing UI mechanisms to transfer the audio of a call
to the HF (pipewire) only when the user explicitly asks/allows it.
Normally, when a call is connected, the phone initiates a SCO connection
and the HF accepts it, transfering audio automatically. In order to
allow for user interaction, this patch enables the UI to set the RejectSCO
property to 'true' in order to automatically reject the SCO connection.
Later on, at the UI's discression, the audio may be reconnected by calling
the Activate() method, which sends AT+BCC to re-initialize the SCO channel.
A configuration file option is also added to configure the default value
of the RejectSCO property. By setting this to 'true' in the config file,
it is possible to implement rejecting the audio of a call that is already
active at the time the Bluetooth connection to the phone initializes.
