Rewinding the ring buffer completely causes audible issues with DMAs.
Previous solution didn't work with tsched=0, and used tsched_watermark
for guardband, which isn't linked to hardware and could become really high
if underflows occurred.
Added separate parameter that can be tuned to hardware limitations and size
of DMA bursts.
This is not 100% ideal as we have not way to tie specific boosts to specific
inputs and this particular chipset (as noted in #772) appears to
support just that.
For the time being incorporate it into the normal boost logic.
See http://pulseaudio.org/ticket/772
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
If an element does not control some channels assume they are 0dB in
comparison to the other elements, i.e. do not influence the volume at
all. Previously we were assuming they were as high as the highest of the
channels we do control.
- 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.
Exponentially increase the amount of time between smoother updates. We start
with a 2ms interval and increase up to 200ms intervals.
Smoother updates and the resulting linear regression take a fair amount of CPU
so we want to reduce the amount of updates.
- We now implement a logic where the sink maintains two distinct
volumes: the 'reference' volume which is shown to the users, and the
'real' volume, which is configured to the hardware. The latter is
configured to the max of all streams. Volume changes on sinks are
propagated back to the streams proportional to the reference volume
change. Volume changes on sink inputs are forwarded to the sink by
'pushing' the volume if necessary.
This renames the old 'virtual_volume' to 'real_volume'. The
'reference_volume' is now the one exposed to users.
By this logic the sink volume visible to the user, will always be the
"upper" boundary for everything that is played. Saved/restored stream
volumes are measured relative to this boundary, the factor here is
always < 1.0.
- introduce accuracy for sink volumes, similar to the accuracy we
already have for source volumes.
- other cleanups.
