Valid channel id range is from 0 to SND_MIXER_SCHN_LAST,
inclusive, so the size of the masks array in pa_alsa_element
has to be SND_MIXER_SCHN_LAST + 1. Similar "too small"
arrays were also in alsa-sink's and alsa-source's userdata,
but actually those arrays were not used at all so they were
removed.
element_is_subset() in alsa-mixer.c skipped the last channel
id when iterating the element masks array; that's now fixed
as well.
Thanks to David Henningsson for spotting the too small
arrays in alsa-sink and alsa-source and the
element_is_subset() problem.
Support the new jack detection interface implemented in Linux 3.3
(and Ubuntu's 3.2 kernel).
Jacks are probed and detected using the snd_hctl_* commands, which
means we need to listen to them using fdlists. As this detection
needs to be active even if there is currently no sink for the jack,
so this polling is done on the card level.
Also add configuration support in paths, like this:
[Jack Headphone]
required-any = any
...where 'Jack Headphone' should match 'Headphone Jack' as given by
ALSA (as seen in e g 'amixer controls').
"Required", "required-any" and "required-absent" is supported. Using
required-any, one can have several ports even though there is no
other indication in the mixer that this path exists.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
To be able to add ports to all profiles, we need to probe all
profiles at startup. To speed this up, we now have a cache of
probes paths which is owned by the profile set. Since paths
are now owned by the profile set, the path set must now have
a hashmap of paths instead of a linked list.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
The recommended way of setting available status is to call
pa_device_port_set_available, which will send a subscription event
to the relevant card. It will also fire a hook.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
This adds a boolean module parameter to disable automatic dynamic
latency readjustments on underruns, but leaves automatic dynamic
watermark readjustments untouched.
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>
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.
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).
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.
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).
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.
Since we currently have two mechanisms to signal a passthrough
connection (non-PCM format or PA_SINK_INPUT_PASSTHROUGH flag), we move
all the related checks into functions and use those everywhere.
This makes things more consistent, and should we decide to get rid of
the flag, we only need to change pa_sink_input_*_is_passthrough()
accordingly.
When a passthrough sink-input is added, we need to reconfigure the
sink's sample rate since no resampling occurs. We revert to the original
rate when the passthrough sink-input is removed.
This removes the passthrough flag from sinks since we will drop
exclusively passthrough sinks in favour of providing a list of formats
supported by each sink. We can still determine whether a sink is in
passthrough mode by checking if any non-PCM streams are attached to it.
When logging a suppression message do so on the same log level as the
suppressed messages.
Cherry picked by Colin Guthrie from ec5a785712
with a couple of additional changes due to extra limiting in master
that was not present in stable-queue.
Currently if sink base volume differs from 0dB and sync-volume is used,
wrong volume values are written to hw. This patch fixes that.
Signed-off-by: Juho Hämäläinen <ext-juho.hamalainen@nokia.com>
How about this? There are a couple of bugs in sink_write_volume_cb,
by the way. Another patch will be sent once this dB value printing
patch is accepted.
-- 8< --
Currently when rewinding alsa, a fixed value of 256 bytes is used,
which represents 1.33ms @ 48kHz (2ch, 16bit). This is typically fine
and due to DMA constraints we would not want to rewind less than this.
However with more demanding sample specs, (e.g. 8ch 192kHz 32bit)
256 bytes is likely not sufficient, so calculate what 1.33ms would
be and use which ever value is bigger.
Discussed with David Henningsson and Pierre-Louis Bossart here:
http://thread.gmane.org/gmane.comp.audio.pulseaudio.general/7286
