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.
Add jack.connect-audio and jack.connect-midi to specify an array of port
names to link to instead of the default phyisical ports.
Also actually fixes linking the midi ports correctly.
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.
have_avx = cc.has_argument(avx_args)
gcc generates an error when AVX is unsupported, whereas clang only
produces a warning.
Thus, using Clang on the RISC-V platform incorrectly enables AVX file
compilation, leading to build failures.
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.
The `access(2)` based multi-user mediation mechanism doesn't quite work
for the root user, which may cause it to conflict with a running
foreground user session. Prevent this by not running the user service at
all for the root user, which nobody should be doing anyway.
Commit b160a72018 introduced this change
before, but it was omitted in e1e0a886d5.
This makes again sure we don't call process callback while disconnecting
stream.
Fixes#3314
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.
In case of the video, if the buffer to be rendered is from upstream and
not from the pipewirepool, map the memory into video frames and copy the
frames instead of doing a buffer copy.
Avoid splitting of buffers in the case of video, because that might break
the frame layout, especially planar formats, for the applications which
use pipewiresink as a camera source to capture video.
Add an option to add the MIDI2 flag on ports. This is disabled by
default because most JACK apps don't know about the flag and then
refuse to show the MIDI ports.
Fixes#4584
Make the state_changed event and _get_state() function set errno with
the current error value if the state is in error, so that application
can use this to give more detailed error reporting.
Use this in alsa, v4l2 and pulse to give some other error codes than
EIO.
Fixes#4574
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
