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.
On Fri, Mar 27, 2009 at 7:21 AM, Lennart Poettering <lennart@poettering.net> wrote:
>> I tried installing the latest git sources on my Ubuntu Jaunty box but
>> it just broke sound in all my applications. For my own purposes, I'm
>> going to need to start with the Ubuntu-patched 0.9.14. However, if
>> you are willing to accept this patch I will forward port it so that it
>> applies to the latest sources. It's a completely harmless change, so
>> why not apply it?
>
> Yes, I am happy to apply it. Could you please update it for current git?
>
Great. An updated patch is attached. For symmetry, I added this
option to the alsa source module as well.
The Ubuntu folks have customized pulse so much that it is difficult
for me to get this version working on my system. For this patch I
have only made sure that it compiles. But it does pretty much the
same thing as the one for 0.9.14, which is working great for me.
Thanks,
Kyle
