| Commit message (Collapse) | Author | Age | Files | Lines |
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This is added so that module-udev-detect can load multiple module-alsa-card
instances with the same card name - forcing namereg_fail to false allows the
name registry to mangle the card names to be unique.
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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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Second version after Tanu's feedback
TODO:
- notify client that volume control is disabled
- change sink rate in passthrough mode if needed
- automatic detection of passthrough mode instead of hard
coded profile names
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@intel.com>
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Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@intel.com>
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This is required to when playing on a52: device, rewind is broken
in those plugins.
Credits to Michael Rans <mcarans@yahoo.co.uk> for finding this
workaround, and Tanu Kaskinen <tanuk@iki.fi> for providing
valuable feedback.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@intel.com>
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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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PA_ALSA_ENUMERATION_IGNORE.
This fix doesn't have any concrete effect, because the two constants have the
same value.
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BugLink: https://launchpad.net/bugs/533877
Some laptops have 'Digital Mic' exposed as an 'Input Source', e.g., Dell
XPS 1330, so handle these, too.
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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.
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http://pulseaudio.org/ticket/778
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instead of coming up with pointless aliases, reuse the already established
names, for second headphones, and second speakers.
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https://bugzilla.redhat.com/show_bug.cgi?id=562216
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https://bugzilla.redhat.com/show_bug.cgi?id=558638
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As exposed by really old Microsoft USB sound systems
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http://pulseaudio.org/ticket/740
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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
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As seen on some HDA chips (e.g. Fujitsu Siemens S6410)
Refs http://pulseaudio.org/ticket/772
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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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http://pulseaudio.org/ticket/702
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On some cards line-out is independant of Sepaker and it is a good idea
to cover that so that they can independantly be activated.
https://bugzilla.redhat.com/show_bug.cgi?id=520884
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As seen on some HDA chips:
https://bugzilla.redhat.com/attachment.cgi?id=359804
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As seen on some drivers:
https://bugzilla.redhat.com/show_bug.cgi?id=498612
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As used by some HDA devices:
https://bugzilla.redhat.com/attachment.cgi?id=365290
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As used by some HDA chips:
https://bugzilla.redhat.com/attachment.cgi?id=366816
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users.
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s/Sourround/Surround/
Spotted by Colin Guthrie
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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.
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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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broken drivers apparently need that
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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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