Use alsa:acp: as the object prefix to make it different from the
alsa:pcm prefix when we are not using ACP.
Replace the card index in the object.path with the card_name, which
we try to construct from the alsa.id when it is available. This brings
the object.path in line with the alsa-pcm-device naming and adds the
user configurable id in the object.path to make it possible to
differentiate between identical devices.
Place the acp device index after the card_name instead of the unstable
device name (which depends on the alsa card number, which depends on the
kernel probing order).
This should make the object.path stable accross reboots for ACP and user
configurable with udev. With such a stable id, the other fields can be
made stable as well with custom rules.
See #3912
Use snd_ctl_card_info to set some more card properties such as the
alsa.id, alsa.mixer_name and alsa.components.
alsa.id is interesting because it is possible to use udev rules to set a
custom id, which is handy when you have two identical cards in the
system and want to assign unique ids to them.
See #3912
As part of LibreSSL's cleanup of removed ENGINE support, <openssl/x509.h> is no longer included in <openssl/engine.h>. <openssl/evp.h> was transiently included in <openssl/x509.h>, compared to <openssl/rand.h> for OpenSSL.
The v4l2 build unsets _FILE_OFFSET_BITS, which is not allowed when setting
_TIME_BITS=64. Having verified that nothing in this module is sensitive to
64-bit time_t (none of the functions it intercepts handle time), we also
unset _TIME_BITS to allow this to build as before.
Signed-off-by: Steve Langasek <steve.langasek@canonical.com>
Reviewed-by: Dylan Aïssi <dylan.aissi@collabora.com>
Parse the audio rate from the caps and set this as the stream node.rate
property. This will try to switch the graph to this rate and avoids
resampling.
Rework some of the property handling.
Add pw_conf_section_update_props_rules() that will not only update the
properties of a section but wil also apply rules in section.rules and
match against the context properties.
Use this by default when using pw_context_conf_update_props().
Add a new method to get a string name of the VM type. Place the
cpu.vm.name in the context properties.
This makes it possible to deprecate the vm.overrides with something more
flexible based on rules. Update the conf files and docs to refect this.
For capture streams, place the time of the cycle when we received this
buffer in the new buffer time field.
Without this (and without timestamps in an optional header metadata)
it's impossible to determine when a buffer was created. With this field
and the pw_stream_get_time_n() and pw_stream_get_nsec() we can derive
the mapping to ticks and delay in the graph.
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.
Always copy the timing info when capturing, not only when there is a
buffer to dequeue.
We can update the rate_queued and rate_size with defaults when we don't
have rate matching. We can then also directly use the rate_size field
to update the buffer requested size.
Add a new extension that can create a server on a user provided socket
with user provided security properties.
This is mainly used in flatpaks that want to create and bind a pipewire
socket with specific permissions for the flatpak app.
The flatpak will also provide an fd that will be closed when the server
can be removed.
nframes in the midi buffer should be set to the current cycle
buffer_size and it should restrict the timestamps that can be set on the
midi events.
Keep the last max_frames around in a globals so that we can use it to
set the midi buffer to the default size.
Return NULL when we do jack_port_get_buffer() with larger nframes than
the current cycle buffer_size, just like JACK. Otherwise this could
result in a crash when we try to mix more than the available buffer
space.
Check and reset the midi buffer better. Check if the MAGIC is still ok.
jack_midi_reset_buffer() should restore the MAGIC and other values.
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
Add options to change the 'prefix' and 'sysconfdir' values shown in
documentation, e.g. on config file man pages.
Update CI to set them, so that its produced output doesn't show
/builds/pipewire/... on man pages
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
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.
When timer is not using monotonic clock, apply clock offset to translate
the time values to the monotonic clock when putting them to spa_io_clock
nsec fields.
Get appropriate clock offset by smoothed filtering. The parameters here
keep the offset jitter < 10ns or so.
As monotonic/boottime/realtime all contain adjtime(), there generally is
no drift in the offset here, so just averaging should be fine.
Also fix using wrong timer clock when freewheeling.
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
Variable declarations after a label are not allowed,
and clang does not accept them. Fix the build failure
by removing the variable.
Fixes b3fbd0e607 ("alsa-pcm: add_bind_ctl_param: add support for array")
`gst_caps_make_writable()` may create a copy which we have to keep
using afterwards. The return value was meant to be used for that,
but was promptly forgotten for the initial user.
Avoid such errors in the future by using an in-out parameter instead.
While on it, add a type check and remove a check for an impossible
condition.
Fixes: 8a271a87b ("gst: Sanitize caps before translating")
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.
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
