| Commit message (Collapse) | Author | Age | Files | Lines |
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This new audio interface from Native Instruments has 2 stereo channels
for both input and output direction. This patch adds mappings for them.
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Make new defines for the smoother window size and adjust time constants instead
of reusing some unrelated constant.
Increase the smoother window size even more because the bigger it is, the
better. Since we have a 200ms max update interval and the max smoother history
is 64 entries, 10seconds is a good default.
Decrease the smoother adjust time to 1 second. The previous value of 4 seconds
was too much to adapt quickly after a resume.
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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.
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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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Currently when rewinding alsa, a fixed value of 256 bytes is used,
which represents 1.33ms @ 48kHz (2ch, 16bit). This is typically fine
and due to DMA constraints we would not want to rewind less than this.
However with more demanding sample specs, (e.g. 8ch 192kHz 32bit)
256 bytes is likely not sufficient, so calculate what 1.33ms would
be and use which ever value is bigger.
Discussed with David Henningsson and Pierre-Louis Bossart here:
http://thread.gmane.org/gmane.comp.audio.pulseaudio.general/7286
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GCC gave a warning, because the pointer given to pa_modargs_get_value_u32() had
type size_t instead of uint32_t.
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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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40ms for the smoother window is too small. Increase the size to 4 seconds, like
we do for the sinks.
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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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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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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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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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