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>
Previously a missing key would cause this kind of log output:
D: [pulseaudio] module-device-manager.c: Database contains invalid data for key: sink:auto_null (probably pre-v1.0 data)
D: [pulseaudio] module-device-manager.c: Attempting to load legacy (pre-v1.0) data for key: sink:auto_null
D: [pulseaudio] module-device-manager.c: Size does not match.
D: [pulseaudio] module-device-manager.c: Unable to load legacy (pre-v1.0) data for key: sink:auto_null. Ignoring.
That is now replaced with
D: [pulseaudio] module-device-manager.c: Database contains no data for key: sink:auto_null
If 'pa_modargs_new' returns a NULL, we need to be careful to not call
'pa_modargs_free' in the failure path since it requires that we pass it
a non-null argument. Also updates 'module-bluetooth-policy.c:pa__init'
to follow the standard "goto fail" pattern used everywhere else.
Signed-off-by: Jason Gerecke <killertofu@gmail.com>
This is an OpenPGP/MIME signed message (RFC 4880 and 3156)
The old code fetched the channel name via AudioObjectGetPropertyData()
and accessed the "returned" data as a plain char buffer.
This may or may not have worked at some point according to the Apple
CFString documentation, which warns that the actual data layout is an
implementation detail and subject to change at any time.
On recent OS X versions, this behavior led to "random data" channel
names like >H��{, H��{<.
We need to actually let AudioObjectGetPropertyData() populate a CFString
struct and convert this into a plain char buffer.
The conversion function will not free the CFString, so do that in the
caller.
Signed-off-by: Mihai Moldovan <ionic@ionic.de>
This isn't a great fix, but we need ALSA API to do this right. In the
mean time, USB devices work fine with timer-based scheduling, so there's
no reason to force a large minimum latency by disabling tsched on them.
This forces the canceller engine to be invoked even if playback is not
currently active. We need to do this for cases where the engine provides
additional processing that is independent of playback, such as noise
suppression and AGC.
Fixes: https://bugs.freedesktop.org/show_bug.cgi?id=83557
This allows a configuration scheme where after loading configuration
from "somefile", the parser loads configuration from files in
directory "somefile.d". This feature needs to be enabled on a per-file
basis, though, and this patch doesn't yet enable the feature for any
files.
When synthesized alsa path is freed there is an assert from NULL
proplist. Create empty proplist for the path to fix.
Signed-off-by: Juho Hämäläinen <juho.hamalainen@nomovok.com>
From the NetBSD manual:
The first argument of these functions is of type int, but only a very
restricted subset of values are actually valid. The argument must either
be the value of the macro EOF (which has a negative value), or must be a
non-negative value within the range representable as unsigned char.
Passing invalid values leads to undefined behavior.
-- ctype(3)
pa_module_unload() takes two pointers: pa_module and pa_core.
The pa_core pointer is also available via the pa_module object,
so the pa_core argument is redundant
[David Henningsson: Rebased to git HEAD]
It doesn't work currently (fails and falls back to PCM), due to channel
count mismatch between the sink sample spec and the sample spec required
by IEC61937.
To be reverted when someone implements changing channel count without
switching profiles. This would also be required for HBR passthrough over
HDMI.
Reported-by: Xamindar <junkxamindar@gmail.com>
Signed-off-by: Alexander E. Patrakov <patrakov@gmail.com>
If we always write entries of the latest version, we can simplify
code a little by only handling old versions in the "entry_read"
function and assume we have the latest version everywhere else.
Suggested-by: Tanu Kaskinen <tanuk@iki.fi>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
This makes the routing slightly more aggressive:
* It will try to route to another profile, if such a profile
is preferred by the port.
* It will allow changing profiles on transitions both to
PA_AVAILABLE_YES and PA_AVAILABLE_NO
To accommodate there is also some refactoring.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
In case input or output names are filled in, we can use this to
get a better match in the profile_good_for_input/output functions
instead of guessing based on number of sources and channels.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Adding AGC broke this test, so we hard-disable the volume code in test
mode. This is probably okay for now, since at least with analog AGC, the
source volume changes and the data we get is going to be with AGC
applied, but digital gain won't be encapsulated here.
Long term, we might need to figure out how to deal with this properly.
We encountered an alsa plugin a while ago (not sure if the source
can be shared) which had mixer controls, but no descriptors to
poll for changes.
Quit early to avoid latter assertion failures.
BugLink: https://bugs.launchpad.net/bugs/1092377
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
