This was added in error (88873e295b) but
we discussed it in !2180 and it turned out CFLAGS="... -fstrict-aliasing"
was set in the environment, overriding the earlier -fno-strict-aliasing
already set in meson.build (which we're keeping here).
That is, when -fno-strict-aliasing is in-effect, there's already no
need for -Wno-error=strict-aliasing (as the warnings aren't emitted), but
if there *is* a need for a -Wno-error=strict-aliasing for a user, it
means they're (likely unintentionally) enabling strict-aliasing and we
certainly want to make them aware of that.
This allows our a2dp-source (sink endpoint) to report delay to the
remote source endpoint. Setting the "Delay" property is already
implemented, but by failing to configure DelayReporting, bluez rejects
the property change.
Idle the source when no packets are received and resume when new packets
arrive.
Add a stream.may-pause property to pause the stream when no packets are
received during the timeout window.
Make sure the rtp.streaming property is updated correctly and as soon as
we get the first packet.
Fixes#4456
Reorganize some code to separate the creation and sending of the SAP
message.
Check if when the node changed, we have an actual change in the SDP
before we send BYE and the new SAP message. It's possible that nothing
changed, for example when the node simply changed state or an unrelated
property.
Partially revert 86af9de739
The PortParam does not give enough information to derive the direction
of the converter. If the converter is configured in convert/convert
there is just no way to know when to output a quantum or not.
Fix this by doing a quick probe of the follower and then pass the
direction to the converter.
See !2227
Report the "fractional" part of the resampler delay in
spa_io_rate_match::delay_frac, in nanosamples (1/1e9 sample) at node
rate.
The delay values are best reported in units where it is clear what the
clock domain is, so report the value in fractional samples instead of
nanoseconds. Conversion to ns is also just dividision by the appropriate
rate.
Resampler without prefill was sometime outputting with different delay
than with prefill. Adjust initial history by 1 which seems to bring it
more in line.
The resampler phase also appears to depend on how many samples remain in
history which leads to possibly unexpected +-1 variation. Take this into
account in reported phase.
These changes make the resampler delay tests pass.
Both changes are sort of empirical --- I don't fully understand why
these would fix things but they seem to be needed to make the delay
calculations agree with what the resampler outputs.
When we do any other blocking in the render function, we should unblock
and call _wait_preroll() when we go to PAUSED.
We can have this situation when all the buffers are queued in the
pw_stream and we get a new _render() call. We can't get more buffers
from the pool and so we must block and wait. When we go to PAUSED we
need to unlock and go to _wait_preroll(). Implement this by setting a
pool paused flag that is set when the sink goes to paused, we can then
return a special value that does the wait_preroll().
See !2248
Setting bufferpool to flushing state in PAUSED state is preventing the
buffer flow if there is a seek/flush event.
Instead, set the bufferpool to flushing during the `flush-start` event
and clear it during the `flush-stop`
Deactivate/activate the stream during flush event only if the sink is
in the PLAYING state. In the PAUSED or READY state, the stream would be
inactive and we do not want to alter that
When we write samples, check if we make a jump in the ringbuffer and
clear the samples we jumped over.
If we don't do this, the reader side might pick up old samples that we
didn't write or clear but that are now available for reading after we
made a jump in the ringbuffer.
This migh not be exactly what pulseaudio does but it is good for now.
Fixes#4464
flush the pw buffers to the stream's queue during a FLUSH_START event
and return the unqueued pw buffers, if they are dropped/released without
being rendered, so they can be available to be dequeued for the subsequent
`acquire` calls
The delay is always expressed in samples at the output rate of the
resampler. For input streams we need to convert this to the expected
input rate.
Make the delay reporting in playback streams more accurate.
We now automatically move non-rt clients into non-rt threads so the
client-rt.conf is obsolete.
Move the module-rt in client.conf and add conditions to disable modules.
Transparently load client.conf in case applications still specify
client-rt.conf.
Custon configuration in the client-rt.conf.d/ should be moved to
client.conf.d/
The resampler delay was off by one sample, so remove the corresponding
fudge factor here. This matters for BAP output synchronization.
The resampler has also some fractional delay, so there can still be
sub-sample offset between the original and resampled timelines. This is
not currently taken into account.
Resampler delay for N taps is N/2-1 input samples.
Add test that checks this.
When input rate is varying, the resampler also accumulates additional
sub-sample delay. The resampler does not currently have API to report
the instantaneous sub-sample delay. Add knownfail test for it.
Wireplumber and other system services use local real time timestamps in
logging, so it's more convenient if also PW uses them.
Add env var for selecting the timestamp type, default to "local".
In timestamps, support different clocks and local time as formats.
Local real time timestamps are useful when trying to correlate logs from
different sources.
