This patch removes all occurrences of double and triple
newlines.
Command used for this:
find . -type d \( -name ffmpeg \) -prune -o \
-regex '\(.*\.[hc]\|.*\.cc\)' \
-a -not -name 'adrian-aec.*' -a -not \
-name reserve.c -a -not -name 'rtkit.*' \
-exec sed -i -e '/^$/{N;s/^\n$//}' {} \;
Two passes were needed to remove triple newlines.
The excluded files are mirrored files from external sources.
This patch replaces every occurrence of ')\n{' with ') {'.
Command used for this:
find . -type d \( -name ffmpeg \) -prune -o \
-regex '\(.*\.[hc]\|.*\.cc\)' \
-a -not -name core-util.c -a -not \
-name adrian-aec.c -a -not -name g711.c \
-exec sed -i -e '/)$/{N;s/)\n{$/) {/}' {} \;
The excluded files are mirrored files from external sources.
The tsched_watermark is in bytes, not in usecs. Fix this by introducing
a new variable, and also use that variable in some places for optimisation.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Now you can actually see *which* sink/source that sends a specific
message to the log, which is quite useful if you have more than
one sound card.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
I was looking at a log that showed that a suspend happened (at
a strange time), but the log didn't tell me why the suspend was done.
This patch tries to make sure that that won't happen again.
We inadvertantly stopped supporting non-standard rates when the
passthrough work was done. This makes sure that if no standard rates are
supported, we try to fallback to whatever ALSA gives us.
Refactor code to fetch avail, delay and timestamp values
in a single call to snd_pcm_status().
The information reported is exactly the same as before,
however it is extracted in a more atomic manner to
improve timer-based scheduling.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Sometimes the kernel does not schedule us in due time, thus causing
an underrun. Adding a detection and a debug message will be a helpful
step in determining the cause of an underrun.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
In UCM basic functions, we only assign intended roles from modifier
to sink/source, but we don't have a chance to set the ucm modifiers.
Here we amend the functions so that when roled stream starts or
stops, we have the following results:
1. stream will be routed to sink/source specified in modifier by
module-intended-roles
2. After that, modifier will be enabled or disabled.
3. when multiple streams with matched roles of modifier start, only
the first one will enable the modifier, and when they end, the
last one will disable the modifier.
Signed-off-by: Feng Wei <wei.feng@freescale.com>
Signed-off-by: Arun Raghavan <arun.raghavan@collabora.co.uk>
Modifiers may have their own PlaybackPCM/CapturePCM and for these, we
create separate sinks/sources. These are marked with the
device.intended_roles property to let role-based routing take care if
streams are tagged appropriately.
The proplist isn't used by the conventional alsa-mixer code path, but
can be used by UCM to transfer properties from UCM data to the
sinks/sources corresponding to a mapping. These properties could be used
later in policy, etc.
The specific use for which I'm writing this now is for UCM modifiers
that have their own PlaybackPCM/CapturePCM field. These will be
translated to a separate sink/source corresponding to the modifier by
adding an additional mapping per sink/source. These mappings' proplist
will be populated with the name of the modifier and corresponding
"device.intended_role" property. The latter will be used in the usual
routing-by-role way, and the former will be used during sink/source
activation and deactivation to know what UCM modifier is to be enabled
or disabled.
UCM basic functions will provide another way to handle the alsa mixer
and controls. That means alsa card module will make use of alsa ucm
configurations provided by various audio systems instead of mixer and
paths configurations provided by PA. PA profiles come from UCM verb, PA
sinks/sources and ports come from UCM devices.
In case the proper UCM configurations are found, ucm branches are
activated, or we will still fall through to the original way.
Signed-off-by: Feng Wei <wei.feng@freescale.com>
As these functions are called together and are related, we might merge
them and call setting_select from pa_alsa_path_select by passing
optional pa_alsa_setting argument.
Make also the setting_select static as it is not called outside of
alsa-mixer.c after this change.
[Additional note from Tanu Kaskinen: this change improves the
mute-during-activation feature, because now the mixer changes related
to selecting the setting happen while the hw is muted.]
Move pa_alsa_setting_select call just after the pa_alsa_path_select in
[sink | source]_set_port_cb functions as there is no dependency to volume
calculations that are done between these two calls. Idea here is to make
possible to merge these two functions since they are called together from
other places too.
Compilation with -DDEBUG_TIMING fails due to a missing header:
modules/alsa/alsa-source.c: In function 'check_left_to_record':
modules/alsa/alsa-source.c:426:9: warning: implicit declaration of function 'raise' [-Wimplicit-function-declaration]
modules/alsa/alsa-source.c:426:9: error: 'SIGTRAP' undeclared (first use in this function)
Signed-off-by: Eero Nurkkala <eero.nurkkala@offcode.fi>
Log in as user A, fast user switch to user B, let user B change
port, volume or mute status, then switch back to user A.
At this point we must make sure that the ALSA and PA volumes are
synchronised by writing to the ALSA mixer when the ALSA device
becomes available.
BugLink: https://bugs.launchpad.net/bugs/915035
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
If deferred volumes were activated, set_volume does not really set
the volume, and is probably only meant to be called from the main
thread.
As we're currently really setting the port and the mute here (i e
modifying ALSA), we should really modify the volume as well.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
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>
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.
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.
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)
