The audioconverter starts in Convert mode, so make sure it goes to the
None mode before we attempt to reconfigure ourselves.
Also remove the ports on audioconvert when going to None mode. This used
to somewhat work because we configured it in DSP mode without any
params, which is like None without ports.
Setting the default size to 0 and outside of the min/max range now means
that there is no suggestion for the size and it should use the
suggestion of the peer.
Counter-intuitive as it seems, when we are driving the clock, we can't
also provide a clock from PipeWire to the pipeline -- we need the
pipeline to drive the graph.
So we make the mode control whether we provide a clock or not.
When using PW source, one might want to dynamically link PW source to
a different source. Setting possible_caps to NULL prevents the caps
intersect from returning a successful result on format change. Do not
set possible_caps to NULL as we get that from peer caps which should
stay the same ideally for the duration of pipeline run. That allows
re-linking PW source any number of times with a pipeline like below.
gst-launch-1.0 pipewiresrc autoconnect=false ! queue ! video/x-raw,format=YUY2 ! videoconvert ! xvimagesink
The above pipeline can be made to switch between a camera source and a
screen capture source like wf-recorder.
Note that this fix only improves the status quo and won't work if the
peer caps change due to a re-negotiation.
We might end up in a situation where depending on the pipeline,
intersect might not give us fixated caps.
Possible example of such a pipeline can be below.
gst-launch-1.0 -e pipewiresrc target-object=<path> ! audioconvert !
audio/x-raw,format=S16LE,rate=48000,channels=2 ! lamemp3enc !
filesink location=test.mp3
This results in non-fixated caps like below when intersecting caps from
format param and possible_caps which depends on what we have downstream
in the pipeline.
audio/x-raw, layout=(string)interleaved, format=(string)S16LE, rate=(int)48000, channels=(int)2, channel-mask=(bitmask)0x0000000000000003;
audio/x-raw, layout=(string)interleaved, format=(string)S16LE, rate=(int)48000, channels=(int)2
To fix this, fixate the caps explicitly.
The device passed to gst_device_provider_device_add() is transfer:floating, so
we need increase its ref, otherwise the pointer we keep internally will be a
dangling ref.
Also gst_device_provider_device_remove() doesn't actually release the device, so
we have to do it ourselves.
Fixes#4616
When we construct an Enum, check if we only added 1 value and remove the
duplicate default value we added. If we added more values, promote the
choice to an enum.
In case negotiation is first attempted with unfixed caps, bufferpool support was
unconditionally disabled. Then at a second caps negotiation attempt it wasn't
restored according to the property value.
Try to avoid conversions by taking the output port format and using that
as a filter for the input port format.
Because filtering pods prefer the values of the filter, this will prefer
the output format values and thus avoid conversions.
Use per port allocated memory so that we can easily increase the size
and add more buffers. This is necessary when we add filter-graphs that
require more ports.
Check midi client version after setting it, to see if it was really
successfully set. Old kernels without UMP don't know about the midi
version fields, so snd_seq_set_client_midi_version() appears to fail
silently there.
Solution suggested by Xi Ruoyao.
The dbus user service is required for various features - the summary says:
'dbus (Bluetooth, rt, portal, pw-reserve)'
On session logout the dbus service gets shut down while the Pipewire one
relies on a timeout. If a user logs in again before PW timed out, the
later stays alive but doesn't handle re-connecting to the dbus service
of the new session, breaking the camera portal and potentially other
features.
Thus hard-depend on the dbus service (if enabled at build time) and thus
shut down together with it.
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
The midi events have their large data offsets relative to the start of
the buffer and the large data is at the end of the buffer. Because we
copied it down, right after the events, but we didn't adjust the
offsets, calculate a correction offset when unpacking the events.
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.
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.
