| Commit message (Collapse) | Author | Age | Files | Lines |
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Move the code to start the capture and the smoother closer together to improve
smoother accuracy.
Rework things to look more like the alsa sink where the device is started in
only one place.
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40ms for the smoother window is too small. Increase the size to 4 seconds, like
we do for the sinks.
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This allows the name registry to mangle the names of auto-detected sinks and
sources to be unique, which makes it possible to load multiple identical sound
cards using module-udev-detect.
At least for now the module argument can only be passed through
module-alsa-card.
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The smoother is paused when the device is suspended but never resumed on
unsuspend. Pass the paused = FALSE flag to the pa_smoother_reset() call to make
it unpause when unsuspending. This patch improves source timings quite a bit.
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Positive base volume can happen, if the alsa volume range has been limited. For
example, in an embedded environment it may be known that the sound device is
capable of louder output than what the speakers can handle, so setting the max
volume below 0 dB makes sense.
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period settings we had before
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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
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metrics so that we don't accidently set a buffer size that is suitable for tsched where we don't use tsched
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way we do it for initial opening
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- 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.
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smart
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sources, following the scheme for sinks
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- 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
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if we didn't call snd_pcm_avail immediately before
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Merge commit 'e4d914c945c13d23b131d7ba75fbdd03cb6d0043'
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finally fixed for granularity
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in one place
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to allow profiles mit multiple sinks or multiple sources
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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.
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This should close rhbz #494851, mandriva bz #51234.
Probably the same as our own #572, launchpad #352732.
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becomes unused
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This allows us to forward the fixed latency directly from the sink to
the monitor source withut having to wait for pa_sink_put().
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This allows us to easily use different mixer controls for analog and
spdif output.
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Issue pointed out by Jaroslav Kysela
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fixups' on resuming
The primary reason for this change is to allow time graphs that do not
go through the origin and hence smoothing starting from the origin is
not desired. This change will allow passing time data into the smoother
while paused and then abruptly use that data without smoothing using the
'quick fixup' flag when resuming.
Primary use case is allowing recording time graphs where the data
recorded originates from a time before the stream was created. The
resulting graft will be shifted and should not be smoothened to go
through the origin.
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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
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