This does a couple of things: first, we implement revents demangling,
which seems to be required (although hw: devices work fine without it).
The second is to actually read the ctl events so we can tell when
elements we care about have changed, instead of reading everything and
trying to do a diff.
The latter is also required from a correctness perspective, as otherwise
the ctl might keep triggering wakeups while the fd is ready to be read.
The output buffer size is always the current cycle buffer_size.
The size that is give by the JACK API is only used to restrict the
number of mixdown samples or midi offsets.
Fixes#3892
Remove some of the hardcoded values (the absolute smallest buffer size,
independent on the sample rate) to a config option. Set it to the
default value of 4, like what it was before.
Change the hardcoded absolute lowest limit to 1 because anything else
does not make sense.
Enforce the quantum-floor when calculating the final graph quantum.
Fixes#3908
Add a monitor.passthrough option. This will pass all latency information
directly between the port and its monitor ports.
This is interesting when the adapter (and audioconvert) is used with a
null-audio-sink that simply forwards the data to a real sink/souce. In
that case, we want the sink/source latency to be passed unmodified.
Set the monitor.passthrough on the pulseaudio null-sink because
a passthrough virtual sink is the most likely use case for this.
Add some monitor.passthrough default config where it makes sense.
Fixes#3888
When the midi message does not fit in the current cycle, save it in a
buffer and send it in the next cycle.
Incomplete midi messages will be received in the next cycle and need to
be sent out at timestamp 0 of the cycle.
Detect dropped and incomplete messages and give a warning.
Try to close/open the ffado device when streaming_start() fails. This
seems to be the case after suspend/resume or a cable disconnect.
Unfortunately, ffado seems to be stuck in an unrecoverable state,
though, to make it really work...
Clean up port buffers when the filter is destroyed.
The FFADO source needs to be the driver and the sink the follower so
that captured data can flow to playback without delay.
Instead of starting a new thread for FFADO, use a timer and the data
loop to wait for FFADO. This is not so nice because we do blocking waits
on the data thread but then we can schedule the source and sink without
a context switch from FFADO. We use a timer so that we can set a timeout
period before starting the graph and catch xruns. The timer will be
restarted immediately when the graph completes and we can go back into
the FFADO wait.
FFADO Xrun should result in a new wait() call.
Handle channels better, use AUX for the channels when they don't match
the given positions.
Silence playback when we don't have a sink or sink data.
Stop and start FFADO when the sink/source pause/resume.
PIPEWIRE_CONFIG_NAME=client-rt.conf pw-cli -m load-module
libpipewire-module-ffado-driver '{ ffado.period-size=32 ffado.period-num=2 }
now gives 4.722ms latency measured with jack_iodelay, equal to JACK.
See #3558
Make a method to get the current time to compare agains the pw_time-now
field. This is currently CLOCK_MONOTONIC but make this into a method
so that we can more easily change it later.
On the midi input ports, do the same trick as on the output ports:
first convert the midi to JACK and then copy the whole buffer to the
port specific storage.
This makes it possible to have a different midi buffer per port and
allow multiple threads to get the buffer concurrently.
Fixes#3901
If ncurses is not initialized, then the global `LINES` variable stays 0.
This will cause problems because there is an unconditional `if (y > LINES)`
check when printing the driven nodes for a given driver node, resulting
in only the first one being printed.
Commit 71653e04d2 ("pw-top: add 'batch-mode' and iterations known from top")
that introduced batch mode missed this one condition, so fix that
by only checking the for terminal overflow in non-batch mode as
it is done a couple lines above.
Fixes#3899
gst-play uses autoaudiosink by default when playing audio, which
iterates over all sinks sorting them by rank. By default,
pipewiresink sets the rank to 0, but it can be overridden
by setting the GST_PLUGIN_FEATURE_RANK env. var. like this:
`GST_PLUGIN_FEATURE_RANK=pipewiresink:268 gst-play-1.0 /usr/share/sounds/alsa/test.wav`
But that doesn't work either because the autoaudiosink plugin also
filters the available options, testing for "Sink" and "Audio" to
appear in the classification metadata
(in the strstr comparison in
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/blob/main/subprojects/gst-plugins-good/gst/autodetect/gstautodetect.c?ref_type=heads#L220
klass is what's set by pipewire as classification,
self->type_klass is "Sink" and self->media_klass is "Audio")
Just adding the word Audio to the classification metadata fixes
this and allows pipewiresink to be selected by autoaudiosink.
I also set it in the source plugin since looking at the code,
autoaudiosrc works exactly the same.
Currently the HDMI output paths have jack mixers named "HDMI/DP" and
with append-pcm-to-name=true. However, most of the SOC audio drivers
are just named "HDMI" and don't add the ",pcm=N". Add these alternate
jack names to the HDMI audio path files so that jack detection will work
on these SOCs.
When we disable autoconnect, disable the timeouts as well. Otherwise the
user has to connect the stream within the 30 second timeout or get a
failure. With autoconnect we can reasonably assume there is a problem
when the stream is not connected after 30 seconds.
Fixes#3884
Keep track of the valid ports and don't emit port info for
invalid ports. When a listener is added while the ports are being
created, it is possible that the ports are still NULL or invalid.
The conf argument was unused before, but it is correctly populated
using the context's conf in pw_context_new(), so the code behaves
the same effectively.
This allows parsing standard conf sections from other configuration
sources, as long as the sections are placed in a dictionary.
Keep track of the active number of mixer ports and update the global mix
io in sync with the data thread because that is where we will check the
state of the global mix io.
This is mostly important for output ports. When removing all links from
an output port, we first will clear all the mixer io and then remove the
global mixer with client_node_port_set_mix_info(). If we don't clear the
io before that, the data thread will be using that buffers as they are
cleared.
See !1915
When we clear the port io, pause the core until the invoke call
completed. This way we don't start processing other messages until we
have safely removed the port io.
Normally, when clearing a link on a port, first the mix io will be set
to NULL and then the format will be cleared, which clears the buffers
as well. By delaying the processing of the format clear until the io
is removed from the data thread we avoid taking away the buffer memory
from the processing thread prematurely.
When creating a link, first the format and buffers are configured and
then the io is set, which should be safe in all cases.
See !1915
Propagate the error if spa_system_eventfd_create() fails. Also copy
errno before calling spa_log_debug() in spa_system_eventfd_create() to
make sure it is not overwritten.
Pick relevant doc updates.
doc: fix DoxygenLayout.xml for doxygen 1.10
doc: add reference docs + index for config and env variables
doc: add some more coverage
doc: move config man pages under config/
doc: add sections to index, clarify discussion of properties
doc: disable dot graphs in API docs again
doc: some updates
doc: fix up example dependencies in meson
The rate we get from dlls can have a subsample precision. However,
the check for using process_copy is in sample precision. This means
that an adaptive stream will oscillate rather then lock into the
exact rate.
We need to load and apply the overrides in the order:
If absolute config path, use only that.
If environment variable, use only that.
Else
/usr/share/pipewire/*.conf.d/
/etc/pipewire/*.conf.d/
$HOME/.config/pipewire/*.conf.d/
Before this patch we would first apply $HOME and then /etc and /usr,
which is not expected.
Update doc/ from master branch.
tutorial: fix s16 scale and add some docs
doc: add 'Configuration' page
doc: disable deprecated list
doc: fix some doxygen warnings
doc: put new pulse modules to right place
doc: filter some constructs that confuse doxygen
doc: Fix typo 'statis' -> 'static'
doc: include pipewire-pulse modules explanations also on man page
doc: add pw-v4l2.1 and spa-*.1
doc: add pw-reserve.1
doc: internals/access: update documentation vs current state
Make a socket of the same address family as the address we're going to
bind it to.
When the source.ip is not specified, use the default in the same address
family as the sap.ip.
Probe the interface for addresses of the same family as the sap.ip.
Makes pactl load-module module-rtp-recv sap_address=:: work.
See #3851
Make a rtprio-server and rtprio-client option. Leave the server
priority by default to 88 but lower client priority to 83. JACK
does something similar by setting clients to rtprio-server - 5.
Make module-rt use the client priority by default and bump the server
priority explicitly in the config file.
Leave the pulse-server to the default rtprio-client, there is no reason
to lower this any further because it is really just a regular client.
Bump the ffado packetizer thread to rtprio-server + 5 because that is
also what JACK does.
88 is still much higher than the value of 60 that JACK uses in
Fedora but now this is at least configurable.
We would timestamp within an unlikely block, which would introduce
additional jitter to current_time, which would have an impact on
the performance of the timer sensitive code.
impl-metadata would not monitor for globals being removed.
This would cause stale metadata to remain in the
store, causing future objects on the same ids to have invalid data.
