The suspend cause isn't yet used by any of the callbacks. The alsa sink
and source will use it to sync the mixer when the SESSION suspend cause
is removed. Currently the syncing is done in pa_sink/source_suspend(),
and I want to change that, because pa_sink/source_suspend() shouldn't
have any alsa specific code.
There are no behaviour changes, the code from almost all the SET_STATE
handlers is moved with minimal changes to the newly introduced
set_state_in_io_thread() callback. The only exception is module-tunnel,
which has to call pa_sink_render() after pa_sink.thread_info.state has
been updated. The set_state_in_io_thread() callback is called before
updating that variable, so moving the SET_STATE handler code to the
callback isn't possible.
The purpose of this change is to make it easier to get state change
handling right in modules. Hooking to the SET_STATE messages in modules
required care in calling pa_sink/source_process_msg() at the right time
(or not calling it at all, as was the case on resume failures), and
there were a few bugs (fixed before this patch). Now the core takes care
of ordering things correctly.
Another motivation for this change is that there was some talk about
adding a suspend_cause variable to pa_sink/source.thread_info. The
variable would be updated in the core SET_STATE handler, but that would
not work with the old design, because in case of resume failures modules
didn't call the core message handler.
build_pollfd() isn't likely to fail, but if it does, pa_sink/source_put()
will crash on an assertion failure. I haven't seen such crash happening,
this is just something that I noticed while studying the state change
code.
The suspend cause isn't yet used by any of the callbacks. The alsa sink
and source will use it to sync the mixer when the SESSION suspend cause
is removed. Currently the syncing is done in pa_sink/source_suspend(),
and I want to change that, because pa_sink/source_suspend() shouldn't
have any alsa specific code.
This rejigs the update_rate() logic to encompass changes to the sample
spec as a whole, as well as passthrough status. As a result,
sinks/sources provide a reconfigure() method which allows
reconfiguration as required.
The behaviour itself is currently unchanged -- alsa-sink/-source do not
actually implement anything other than rate updates for now (nor are
they ever requested to). This can be modified in the future, to allow,
for example 24-bit output when incoming media supports it, as well as
channel count changes for passthrough sinks.
Another related change is that passthrough status is now part of
sink/source reconfiguration, and we can stop doing a suspend/unsuspend
when entering/leaving passthrough state. So that part is now divided
in two -- pa_sink_reconfigure() sets the sink in passthrough mode if
required, and pa_sink_enter_passthrough() sets up everything else
(this currently means only volumes, but could disable other processing)
for passthrough mode.
The reported latency of source or sink is based on measured initial conditions.
If the conditions contain an error, the estimated latency values may become negative.
This does not indicate that the latency is indeed negative but can be considered
merely an offset error. The current get_latency_in_thread() calls and the
implementations of the PA_{SINK,SOURCE}_MESSAGE_GET_LATENCY messages truncate negative
latencies because they do not make sense from a physical point of view. In fact,
the values are truncated twice, once in the message handler and a second time in
the pa_{source,sink}_get_latency_within_thread() call itself.
This leads to two problems for the latency controller within module-loopback:
- Truncating leads to discontinuities in the latency reports which then trigger
unwanted end to end latency corrections.
- If a large negative port latency offsets is set, the reported latency is always 0,
making it impossible to control the end to end latency at all.
This patch is a pre-condition for solving these problems.
It adds a new flag to pa_{sink,source}_get_latency_within_thread() to allow
negative return values. Truncating is also removed in all implementations of the
PA_{SINK,SOURCE}_MESSAGE_GET_LATENCY message handlers. The allow_negative flag
is set to false for all calls of pa_{sink,source}_get_latency_within_thread()
except when used within PA_{SINK,SOURCE}_MESSAGE_GET_LATENCY. This means that the
original behavior is not altered in most cases. Only if a positive latency offset
is set and the message returns a negative value, the reported latency is smaller
because the values are not truncated twice.
Additionally let PA_SOURCE_MESSAGE_GET_LATENCY return -pa_sink_get_latency_within_thread()
for monitor sources because the source gets the data before it is played.
Bug 96741 shows a case where an assertion is hit, because
pa_asyncq_new() failed due to running out of file descriptors.
pa_asyncq_new() is used in only one place (not counting the call in
asyncq-test): pa_asyncmsgq_new(). Now pa_asyncmsgq_new() can fail too,
which requires error handling in many places. One of those places is
pa_thread_mq_init(), which can now fail too, and that needs additional
error handling in many more places. Luckily there weren't any places
where adding better error handling wouldn't have been easy, so there are
many changes in this patch, but they are not complicated.
BugLink: https://bugs.freedesktop.org/show_bug.cgi?id=96741
If a card has been hot-plugged after pulseaudio start, alsa-lib still has
old configuration in memory, which doesn't have PCM definitions for the
new card. Thus, this error appears, and the device doesn't work:
I: [pulseaudio] (alsa-lib)confmisc.c: Unable to find definition 'cards.USB-Audio.pcm.front.0:CARD=0'
I: [pulseaudio] (alsa-lib)conf.c: function snd_func_refer returned error: No such file or directory
I: [pulseaudio] (alsa-lib)conf.c: Evaluate error: No such file or directory
I: [pulseaudio] (alsa-lib)pcm.c: Unknown PCM front:0
I: [pulseaudio] alsa-util.c: Error opening PCM device front:0: No such file or directory
The snd_config_update_free_global() function makes alsa-lib forget any
cached configuration and reparse all PCM definitions from scratch next
time it is told to open anything.
The trick has been copied from Phonon.
Fixes: https://bugs.freedesktop.org/show_bug.cgi?id=54029
Signed-off-by: Alexander E. Patrakov <patrakov@gmail.com>
This change doesn't affect behaviour, because accessing boolean fields
in the new data was safe even after the done() call, but it was still
bad style.
FSF addresses used in PA sources are no longer valid and rpmlint
generates numerous warnings during packaging because of this.
This patch changes all FSF addresses to FSF web page according to
the GPL how-to: https://www.gnu.org/licenses/gpl-howto.en.html
Done automatically by sed-ing through sources.
frames_per_block is the mempool's maximum block size in frames
v2 (thanks David Henningson)
* rename max_frames to frames_per_block
Signed-off-by: Peter Meerwald <pmeerw@pmeerw.net>
Now that we have switched to using the mixer handle only,
there is no use for sending hctl handles around.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Forcing all mute changes to go through set_mute() makes it easier to
check where the muted field is changed, and it also allows us to have
only one place where notifications for changed mute are sent.
When given an explicit device.description in card_properties, prefer
this information over other default prefixes (e.g. 'Built-in Audio')
when constructing sink/source descriptions.
For example, if I manually configure the card description to be
"FooBar", I then expect that the sinks and created by the card also
have "FooBar" in their description instead of generic "Built-in
Audio".
module-alsa-{sink,source}.c call pa_alsa_{sink,source}_new with
mapping set to NULL. Guard against this, like the rest of the
function does.
module-alsa-card does not use NULL, so this went unnoticed so far.
This allows mappings to override some or all of the sample_spec used to
open the ALSA device. The intention, to start with, is to use this for
devices in UCM that need to be opened at a specific rate (like modem
devices). This can be extended to allow overrides in profile-sets as
well.
All pa_cvolume_snprint(), pa_volume_snprint(),
pa_sw_cvolume_snprint_dB() and pa_sw_volume_snprint_dB() calls have
been replaced with pa_cvolume_snprint_verbose() and
pa_volume_snprint_verbose() calls, making the log output more
informative and the code sometimes simpler.
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>
