Avoid resampling or use integer resampling when supported by the
sinks/sources
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
pa_{sink,source}_volume_change_apply were being called by the ALSA I/O
thread on every iteration, causing a pa_rtclock_now() call, which can
sometimes be heavy. We avoid this call by making sure there actually are
changes to apply before proceeding into the function.
While we're at it, also dropping a redundant check on s->write_volume.
This makes sure that when we're traversing the device chain for sources
and sinks with shared volume, we handle the case that a sink-input or
source-output of one of these might be unlinked (while unloading a
module, for example).
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 handles the case where a virtual sink/source and it's master have
different channel counts. The solution is not ideal because if the
former has fewer channels and the master has channel volumes that are
not all at the same level, it will lose this information and have all
channels at the same level.
This is not just a theoretical problem, since module-echo-cancel
prefers a mono virtual source/sink and will usually be sitting on top of
a stereo ALSA source/sink.
That said, I don't really see a good solution to this problem, so the
idea is to make volume sharing optional (on by default) in
module-echo-cancel, so that the few people who care can then disable it
if they so desire.
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 callback should also be reset in reset_calbacks().
The extra check in _volume_change_apply() is needed because when the sink is unlinked the callbacks are reset,
but there still may be pending volume changes.
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 was added to ensure symmetry between playback and recording streams
code, but in reality this makes little sense practically speaking and thus
it is removed.
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)
In most cases it is expected that clients cannot consume compressed
data from monitor sources, so we suspend the monitor source when the
sink goes into passthrough mode.
Eventually, when the extended API includes client notifications for
changed formats, we should emit a notification on the monitor so that
clients can decide what they want to do when this happens (disconnect or
consume the data anyway).
This allows modules to know when certain ports are changed.
This will allow e.g. a filter module (or LADSAP) to only load
when a certain port is used on the device (e.g. to only filter
headphones and not normal speakers).
(Comment from Colin Guthrie: This may also have use in UCM)
This change doesn't add any functionality in itself, but it will be useful in
the future for operating on chains of sinks or sources that are piggy-backing
on each other.
For example, the PA_PROP_DEVICE_MASTER_DEVICE property could
be handled in the core so that each virtual device doesn't have to maintain it
separately. By using the origin_sink and destination_source pointers the core
is able to see at stream creation time that the stream is created by a virtual
device, and then update that device's property list using the name of the
master device that the stream is being connected to. The same thing can be done
also when the stream is being moved from a device to another, in which case the
_MASTER_DEVICE property needs updating.
When looping through the streams on a given device checking to see if the
stream is 'active' there should be no assert if the stream is not linked, it
should simply be ignored.
This assert can be hit if a sink and a sink input are both created and setup
but the final put calls are left to the end as is done in module-ladspa-sink.
While the order of the calls in module-ladspa-sink could be altered, we should
deal gracefully with the way it is now and not complain about ending up
in this state.
A trigger case was trivial:
1. Load a ladspa-sink.
2. Play a stream and move it to it.
3. Unload the module, then reload it.
4. Due to module-stream-restore and module-suspend-on-idle, the hook callbacks
will ultimately hit this assert.
Thanks to Kim Therkelsen for highlighting this issue.
Instead of using string contents for type identification use the address
of a constant string array. This should speed up type verifications a
little sind we only need to compare one machine word instead of a full
string. Also, this saves a few strings.
To make clear that types must be compared via address and not string
contents 'type_name' is now called 'type_id'.
This also simplifies the macros for declaring and defining public and
private subclasses.
- drop the 'virtual_' prefix from s->virtual_volume since we don't
distuingish between reference and real volumes for sources
- introduce an accuracy for source volumes: if the hardware can control
the volume "close enough" don't necessarily adjust the rest in
software unless it is beyond a certain threshold. This should save a
little bit of CPU at the expensive of a bit of accuracy in volume
handling.
- other minor cleanups
This of course makes the name 'fixed' a bit of a misnomer. However the
definitions are now like this:
fixed latency: the latency may change during runtime, but is solely
controlled by the backend, the client has no influence.
dynamic latency: the latency may change during runtime, influenced by
the requests of the clients.
i.e. fixed vs. dynamic is from the perspective of the client.
This adds pa_assert_io_context() and pa_assert_ctl_context() in addition
to a few related macros. When called they will fail when the current execution
context is not IO resp. not control context. (aka 'thread' context vs.
'main' context)
In some situations a rewind request travelling downstream might be
optimized away on its way and an upstream rewind processing might never
come back. Hence, call _process_rewind() before each _render()just to
make sure we processed them all.
Completely rework mixer logic. This now allows controlling a full set of
elements from a single sink's volume slider/mute button.
This also introduces sink and source "ports" that can be used to choose
different input or output ports with the UI. (i.e. "mic"/"line-in" or
"speaker"/"headphones".
The mixer paths and device maps are now configered in external
configuration files and can be tweaked as necessary.
While flags should generally be initialized by passing them to
pa_{sink|source}_new() we make an exception for the volume related flags
which may be initilized afterwards, but before _put().
