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>
Instead of using the hctl interface, we can find controls belonging
to other iface types than "mixer". We do this by introducing a new
mixer class "SND_MIXER_ELEM_PULSEAUDIO" and create snd_mixer_elem's
for all PCM and CARD iface types (as Jacks are of the CARD type and
ELD controls are of the PCM type).
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
hwparams_copy needs to be reset (as it is also reset for the third and
fourth try) before the second try.
If the reset is not done and the first try fails:
D: [lt-pulseaudio] alsa-util.c: Maximum hw buffer size is 743 ms
I: [lt-pulseaudio] alsa-util.c: snd_pcm_hw_params_set_buffer_size_near() failed: Invalid argument
I: [lt-pulseaudio] alsa-util.c: snd_pcm_hw_params_set_period_size_near() failed: Invalid argument
I: [lt-pulseaudio] alsa-util.c: snd_pcm_hw_params_set_buffer_size_near() failed: Invalid argument
D: [lt-pulseaudio] alsa-util.c: Set only period size (to 1102 samples).
We have three failures and finally the fourth (only period size) succeed.
With this patch:
D: [lt-pulseaudio] alsa-util.c: Maximum hw buffer size is 743 ms
I: [lt-pulseaudio] alsa-util.c: snd_pcm_hw_params_set_buffer_size_near() failed: Invalid argument
D: [lt-pulseaudio] alsa-util.c: Set period size first (to 1102 samples), buffer size second (to 4408 samples).
We only fail with the first try, the second (period followed by buffer) is
fine.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
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".
PCM Devices which have the BATCH flag set update the PCM pointer only with
period size granularity. Using timer based scheduling does not have any
advantage in this mode. For one devices which have that flag set usually update
the position pointer in software after getting the period interrupt. So
disabling the period interrupt is not possible for this kind of devices.
Furthermore writing to or reading from the buffer slice for the current period
is not possible since the position inside the buffer is not known. On the other
hand the tsched algorithm seems to get easily confused for this kind of
hardware, which results in garbled audio output. This typically means that timer
based scheduling needs to be manually disabled on systems with such devices.
Auto disabling tsched in this case allows these systems to run with the default
configuration.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
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.
Currently, this function only reads the monitor name, but could
be extended to read e g supported formats as well.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
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>
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>
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 patch reflects a new capability that Lennart was wishing
for. Wish granted...
Re-submitting it now that alsa-lib 1.0.24
provides additional entry points to disable period
wakeups in timer-scheduling mode if hardware can
work without it (HDAudio, oxygen and Intel SST).
Example with standard playback on HDAudio output
Before change:
Top causes for wakeups:
3.8% ( 5.4) [hda_intel] <interrupt>
2.8% ( 4.0) alsa-sink
After change:
Top causes for wakeups:
2.3% ( 3.0) alsa-sink
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@intel.com>
The PCM handle is already opened with the SND_PCM_NONBLOCK flag.
This additional call is useless.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@intel.com>
Use snd_pcm_avail_delay() in pa_alsa_safe_delay() so that we can check the delay
value against the avail value and patch it up when it looks invalid. Only do
this for capture.
In virtual machines sound card clocks and OS scheduling tend to become
unreliable, adding various 'uneven' latencies. The adaptive algorithm
that handles drop-outs does not handle it this well: in contrast to
drop-outs on real machines that are evenly distributed, small and can
easily be encountered via the adpative algorithms, drop-outs in VMs tend
to happen abruptly, and massively, which is not easy to counter.
This patch simply disables timer based scheduling in VMs reverting to
classic IO based scheduling. This should help make PA perform better in
VMs.
https://bugzilla.redhat.com/show_bug.cgi?id=532775
- As discussed on alsa-devel it's probably better to initialize the
buffer size first, followed by the period size. If that fails try the
other way round. If that fails try to configure only buffer size. If
that fails try to configure only period size. Finally, try to
configure neither.
- Don't require integral periods anymore.
Both of these changes should help improving compatibility with various
weirder sound devices, such as TV cards.
Apparently some ALSA drivers aren't happy with getting the buffer size
configured first followed the period size. So swap the order again and
document this for future reference so that we don't turn that around
again.
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.
This has the benefit that we can properly support ALSA devices where
only the raw 'hw' device exists but no 'front' although it's a proper
2ch stereo device.
