The error message for snd_pcm_hw_params_set_period_wakeup was
printing "ret", but "ret" wasn't being set.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
This is where the actual changes happen.
Some additional checks would be required to make sure the
rate is actually supported
Tested with both PCM and passthrough streams
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
The new module argument can be used to provide a custom
directory for loading alsa path configuration files. This is
useful for testing: no need to be root to create test
configuration files.
Watermark level and latency values are not restored when
resuming, the values used prior to suspending are reused.
This leads to side effects when underruns happen and buffer
sizes are updated, PulseAudio can never meet lower latency
requirements.
Solution: keep track of watermark and latency values on sink or
source creation, and reapply them on resume to start with
a clean slate.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Instead of relying on the snd_mixer_* functions failing, we check for
POLLERR and POLLNVAL first. After this, any errors in handling the mixer
events are deemed fatal (that is we cause the ALSA source/sink thread to
terminate).
The case where POLLERR is set but POLLNVAL is not does not actually
occur, but we're making this a soft failure (stop polling the mixer, but
don't kill the I/O thread). If other conditions where POLLERR occurs
turn up, we need to handle them explicitly.
Thanks to Linus Torvalds for helping get this right.
This improves the error handling in the mixer rtpoll callback. It avoids
a crash if an error occurs (the rtpoll_item is freed but still
referenced), and specifically makes sure we don't continue trying to
poll the device if the card is disconnected.
This makes set_formats() put PCM formats lower down the list than
compressed formats since we negotiate by picking the first format in
this list that is also in the client-provided list of possible formats
during sink input creation.
This will be incorrect if we ever decide to do encoding in PA (for
things like AC3/DTS encoding for multichannel output over S/PDIF).
Sometimes the ALSA mixer can be modified during a point at shutdown
which causes a race condition trying to update the volume of an
unlinked sink.
Includes typo fix by our Chief Typo Spotter, Colin, and a clarifying
comment by me.
BugLink: http://bugs.launchpad.net/bugs/841968
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
This just covers Lennart's concern over the terminology used.
The majority of this change is simply the following command:
grep -rli sync[-_]volume . | xargs sed -i 's/sync_volume/deferred_volume/g;s/PA_SINK_SYNC_VOLUME/PA_SINK_DEFERRED_VOLUME/g;s/PA_SOURCE_SYNC_VOLUME/PA_SOURCE_DEFERRED_VOLUME/g;s/sync-volume/deferred-volume/g'
Some minor tweaks were added on top to tidy up formatting and
a couple of phrases were clarified too.
The Kinect shows up as a UAC device after the firmware has been loaded,
but in order to be detected by pulseaudio a 4-channels input only
mapping is needed. Provide a new profile for that and set it with a udev
rule.
fdo#39664
I've seen more than one system where the volume control named
"Front" is a part of audio path for headphones. This is somewhat
of a compromise: While we don't merge it into the path, as that
would be regressing machines where "Front" isn't a part of the
audio path, it would still enable sound on these machines.
BugLink: http://bugs.launchpad.net/bugs/804178
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
The priorities should be set in the approximately likeliness that someone
would like to use them. This is hard to guess of course, but this is
a better guess compared to the existing configuration.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
This is required for E-AC3 streams, as well as to let receivers we're
sending non-PCM data (which avoids playing noise if the data is
incorrect for some reason).
The "Line HP Swap" element needs to be set correctly for some Dove
boards to work correctly. Thanks to Daniel T Chen for the patch.
BugLink: http://bugs.launchpad.net/bugs/451635
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
This switch needs to be off for analog output to work on several
Audigy cards.
Thanks to Daniel T Chen for the patch.
BugLink: http://bugs.launchpad.net/bugs/408370
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
This implements the sink get_formats() and set_formats() API in
alsa-sink. Modules can use this to allow users to specify what formats
their receivers support.
revents marked as POLLOUT|POLLERR|POLLWRNORM in "underrun" case that will
trigger unexpected log "ALSA woke us up to write new data to the device, but
there was acturally nothing to write...".
This patch avoids this log message.
As reported in http://kpaste.net/04f1f3f
it is possible to call enumeration_is_subset with null pointers.
Handle that case instead of crashing. (It is also possible that
Tanuk's pending element_is_subset patch solves the issue, but this
nevertheless gives some extra security.)
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Now that subset mixer paths are removed, this workaround is no longer needed.
This effectively reverts 1c38b5d478 but due
to me forgetting to add files and adding a couple extra workarounds after,
it's easier to just do this manually rather than run git-revert.
In order to try and avoid 'spamming' the user with port choices,
attempt to detect and remove any pointless paths in a path set. That is
any paths which are subsets of other paths.
This should solve a problem case with some USB Headsets which result in
two paths both involving the 'Speaker' element. When no 'Master' element
exists (which is quite common on head/handsets), then the first path
(analog-output) will contain the 'Speaker' in a way that completely fits
with in the use of the 'Speaker' element in the other path
(analog-output-speaker).
Unification is really just a 'lowest common denominator' system. If any
paths do not support volume, mute or decibels, then mark them all as not
having them.
This was originally done this way because the flags set on sinks that
dictate if it supports h/w volume, mute etc. could not be changed after
the sink was created.
The fact that these flags could not change has now been change in the
previous commits, and thus there is now no need to use this 'lowest
common denominator' approach as we can fully support the various
different combinations, even if they change after initial creation
of the sinks/source.
Some sink flags are really just a product of what callbacks
are set on the device. We still enforce a degree of sanity
that the flags match the callbacks set, but we also set the
flags automatically in our callback setter functions to
help ensure that a) people use them and b) flags & callbacks
are kept in sync.
This is not currently useful but future commits will make further
changes concerning automatic setting of flags and event delivery
that makes this structure necessary.
This is a workaround - these usb headsets have one output volume
control only, labeled "Speaker". This causes the default profile
set to not control the volume at all, which is a bug. Workaround
that by creating a separate profile set.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
The previous logic in ade0a6f884
does not work with for input volumes.
This was discussed on the mailing list:
https://tango.0pointer.de/pipermail/pulseaudio-discuss/2011-May/010091.html
This approach can introduce a problem when setting the volumes
for sources. What follows is Tanu Kaskinen's analysis:
[quote]
I'll quote the log:
D: protocol-native.c: Client pavucontrol changes volume of source alsa_input.pci-0000_00_1b.0.analog-stereo.
D: alsa-source.c: Requested volume: 0: 45% 1: 45%
D: alsa-source.c: in dB: 0: -20.71 dB 1: -20.71 dB
D: alsa-source.c: Got hardware volume: 0: 45% 1: 45%
D: alsa-source.c: in dB: 0: -21.00 dB 1: -21.00 dB
D: alsa-source.c: Calculated software volume: 0: 101% 1: 101% (accurate-enough=no)
D: alsa-source.c: in dB: 0: 0.29 dB 1: 0.29 dB
D: source.c: Volume going up to 29273 at 270475970821
D: source.c: Volume change to 29273 at 270475970821 was written 34 usec late
D: alsa-source.c: Written HW volume did not match with the request: 0: 45% 1: 45% (request) != 0: 42% 1: 42%
D: alsa-source.c: in dB: 0: -21.00 dB 1: -21.00 dB (request) != 0: -22.50 dB 1: -22.50 dB
Looking at the last line, the requested volume seems to hit exactly the
right step (-21.00dB), but for some reason Alsa decides to choose
something else. I'm pretty sure that this happens because of rounding
errors. In the first phase we ask Alsa what dB value we should set, and
it returns -21.00 dB. The value is given as a long int, but we convert
that to pa_cvolume. Then when we set the volume, we convert the
pa_cvolume value back to a long integer. At this point I believe it gets
converted to -2101. This is not visible in the debug message for some
reason - the rounding algorithm must be different from what was used
with the pa_cvolume -> long conversion.
[/quote]
The commit after this contains a patch that addresses this issue.
This piggy backs onto the previous changes for protocol 22 and
thus does not bump the version. This and the previous commits should be
seen as mostly atomic. Apologies for any bisecting issues this causes
(although I would expect these to be minimal)
