This saves some proplist allocations and a couple of code lines. Also,
logging is better, because the set_property() functions work with
string values, while the update_proplist() functions assume opaque
binary data, and therefore can't log the property values.
device-manager reroutes all streams whenever a new device appears.
When filter-apply has loaded a filter for some stream, the filter
device may not be what device-manager considers the best device for
the stream, which means that when an unrelated device appears,
device-manager may break the filtering that filter-apply had set up.
This patch changes filter-apply so that it saves the filter device
name to the stream proplist when it sets up a filter. device-manager
can then check the proplist when it does rerouting, and skip the
rerouting for streams that have a filter applied to them.
The proplist isn't cleaned up when the stream moves away from the
filter device, so before doing any decisions based on the
filter_device property, it should be checked that the stream is
currently routed to the filter device. It seemed simpler to do it this
way compared to setting up stream move monitoring in filter-apply and
removing the property when the stream moves away from the filter
device.
When autoloaded, module-echo-cancel doesn't support moving the sink
input and source output that it creates, but the move prevention was
implemented by manually requesting module unloading in the middle of
the stream move procedure, rather than by just setting the DONT_MOVE
flags. This patch removes the module unloading code from the moving()
callbacks and adds the DONT_MOVE flags. In addition to saving some
code, this also prevents problems related to trying to move streams
connected to the echo cancel sink or source while the echo cancel sink
or source is in the middle of a move too (a crash will happen in such
situation, as demonstrated in
https://bugs.freedesktop.org/show_bug.cgi?id=93443).
In case of invalid argument for volume, the crash occurs in pa_stream_interaction_done().
pa_xnew() is replaced with pa_xnew0() to fix it.
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>
Now, trigger_roles, ducking_roles and volume can be divided into several groups by slash.
That means each group can be affected by its own volume policy.
If we need to apply ducking volume level differently that is triggered from
each trigger role(s), this feature would be useful for this purpose.
For example, let's assume that tts should take music and video's volume down to 40%
whereas voice_recognition should take those and tts's volume down to 20%.
In this case, the configuration can be written as below.
trigger_roles=tts/voice_recognition ducking_roles=music,video/music,video,tts volume=40%/20%
If one of ducking role is affected by more than two trigger roles simultaneously,
volume of the ducking role will be applied by method of multiplication.
And it works in the same way as before without any slash.
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>
`Mic` is now detected as `Mic-In/Mic Array` (there are 2 microphones physically, nice to se this being understood).
`Line` is now detected as `Line In`.
Removed all output modes except officially supported stereo, 5.1 and stereo S/PDIF.
Also microphone/line in now might be used simultaneously with either output mode, yay!
The code was mixing sink and sink input domain rate updates, and that
only works if the rate of the RTP stream is the same as the rate of the
sink. This changes all the calcuations to be on the sink-input rate,
since that's the rate we are trying to guess (and resample for).
Now that we have the necessary infrastructure to memexport and
mempimport a memfd memblock, extend that support higher up in the
chain with pstreams.
A PA endpoint can now _transparently_ send a memfd memblock to the
other end by simply calling pa_pstream_send_memblock() – provided
the block's memfd pool was earlier registered with the pstream.
If the pipe does not support memfd transfers, we fall back to
sending the block's full data instead of just its reference.
** Further details:
A single pstream connection usually transfers blocks from multiple
pools including the server's srbchannel mempool, the client's
audio data mempool, and the server's global core mempool.
If these mempools are memfd-backed, we now require registering
them with the pstream before sending any blocks they cover. This
is done to minimize fd passing overhead and avoid fd leaks.
Moreover, to support all these pools without hard-coding their
number or nature in the Pulse communication protocol itself, a new
REGISTER_MEMFD_SHMID command is introduced. That command can be
sent _anytime_ during the pstream's lifetime and is used for
creating on demand SHM ID to memfd mappings.
Suggested-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Ahmed S. Darwish <darwish.07@gmail.com>
This patch deals with the case that applications start new streams corked.
In case of module-role-cork it will only mute the stream because corking is
removed later by the application.
When a trigger stream changes mute or cork state, the cork streams should
react to this. The same applies if a stream changes its role to or from the
trigger role.
When corking do not ignore streams without media.role. Instead treat
them as if media.role="no_role", so that you can specify "no_role" as
trigger or cork role.
If u->save_time_event is non-NULL when the module is being unloaded,
it means that there are some changes to the database that haven't
yet been flushed to the disk.
Acked-by: David Henningsson <david.henningsson@canonical.com>
By refactoring volume probing into its own function, we can reduce
indentation a lot. Also, if an error occurs during the volume probe,
that volume element is now always skipped (instead of taking down
the entire path with it).
Also, a bug for elements with more than two channels is fixed, as
previously, the volume parsing code was continuing, potentially
referencing somewhere outside the array (which has max two channels).
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
If you have headphones plugged in and plug in HDMI; you want sound
to stay on headphones.
If you have HDMI plugged in and you plug in headphones; you want sound
to switch to headphones.
Hence we need to take priority into account as well when determining
whether to switch to a new profile or not.
BugLink: https://bugs.freedesktop.org/show_bug.cgi?id=93903
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
It is expected that the underlying AGC mechanism will likely provide a
single volume for the source rather than a per-channel volume. Dealing
with per-channel volumes just adds complexity with regards to the
actual volume setting (depending on whether volume sharing is enabled or
not, we would set the volume on the source output of the virtual source,
and their sample specs may be different).
Using a single volume allows us to sidestep this problem entirely.
This is required to have unequal channel counts on capture in and out
streams, which is needed for beamforming to work. The deinterleaved API
only works with floating point samples.
The calculations around how many samples were sent to the canceller
engine was not updated when we started supporting different channel
counts for playback and capture.
The AGC code no longer seems to honour the analog volume limits we set,
and internally uses 0-255 as the volume range. So we switch to use that
(keeping the old API usage as is in case this gets fixed upstream).
This allows us to inherit the sample spec parameters from the sink and
source master (rather than forcing 32 kHz / mono). It is still possible
to override some of the parameters for the source side with modargs.
My original testing showed that these parameters provided a decent
perf/quality trade-off on lower end hardware (which I no longer have
access to). I figure it makes sense to continue with that for now, and
in the future this can be relaxed (use_master_format=yes could be the
default, and resource-constrained systems can disable it).
In the refactoring, I'm expressing the constraints in what I see to be a
more natural way -- rec_ss expresses what we're feeding the canceller,
so it makes sense to apply the constraints on what the canceller accepts
there. This then propagates to the output spec.
This also exposes the range of sample rates that the library actually
supports (8, 16, 32 and 48 kHz).
The original intention was to configure low enough parameters to keep
CPU consumption down. Prior to this change, we assumed that the EC
backend would override the sink parameters based on the source
parameters to achieve this goal, and with this change we remove that
assumption by forcing the default parameters for the sink to be low
enough.
It's not possible to enable the intelligibility enhancer at the
moment, because the feature would require modifying the audio that we
play to speakers, which we don't do currently. All audio processing is
done at the source side, and it's not easy to change that.
This patch is based on Arun Raghavan's code, I just reordered things
a bit and reworded the FIXME comment.
This creates a longer filter that is more complex and less sensitive to
incorrect delay reporting from the hardware. There is also a
delay-agnostic mode that can eventually be enabled if required.
In some very quick testing, not enabling this seems to provide better
results during double-talk.
modules/module-stream-restore.c: In function 'clean_up_db':
modules/module-stream-restore.c:2344:74: warning: comparison of constant '0' with boolean expression is always true [-Wbool-compare]
pa_assert_se(entry_write(u, item->entry_name, item->entry, true) >= 0);
reported by Ubuntu gcc-6
Signed-off-by: Peter Meerwald <pmeerw@pmeerw.net>
