| Commit message (Collapse) | Author | Age | Files | Lines |
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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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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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Take into account the delay between taking the snapshot from the source and the
sink. Improves the quality of the timings.
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Make a new echo-cancel module that exposes a new sink and source. All data sent
to the sink is matched against the data captured from the source and
echo-canceled using the speex echo canceler.
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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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asynchronous subscription events.
Using the subscription events caused an assertion crash sometimes when a sink
was removed and a new sink was created (i.e. card profile change) and a stream
was moved from the removed sink to the new sink. The stream dbus object's
subscription callback got a change event before the core dbus object's
subscription callback got the sink remove/creation events. The stream's
subscription callback then queried the core for the object path of the new
sink, and since the core was not yet aware of the new sink, an assertion was
hit in pa_dbusiface_device_get_path().
Now that the core uses synchronous hooks to keep the sink and source lists up
to date, this particular problem can't occur anymore.
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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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alive.
Instead, watch for org.freedesktop.DBus.Disconnected signals.
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immediately.
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Previously we used libdbus's memory as keys in listening_signals, which caused
that the memory of the hashmap keys got overwritten, which led to that signals
weren't sent properly.
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dbus_entry pointer instead of a userdata pointer.
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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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All API calls are now consolidated in AudioObject* calls, the old model
has been deprecated in 10.6. Follow that change.
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The syntactically correct error meant that the timestamp was always
marked as found and only the first header was checked.
In the case where the timestamp was the first header, things
would have worked as expected.
Thanks to pino for reporting via bug refs #818
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We fail to detect when people disable IPv6 in there kernels. This patch
makes sure we don't ignore this error condition.
http://pulseaudio.org/ticket/752
https://bugzilla.mozilla.org/show_bug.cgi?id=533470
http://bugs.freedesktop.org/show_bug.cgi?id=25742
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device selection
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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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We need to use pa_memblockq_pop_missing() for all request handling,
including the initial request, because otherwise the counters will be
stay off during the entire runtime.
This should fix:
https://bugzilla.redhat.com/show_bug.cgi?id=559467
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This should make it unlikely that we loop on SIGHUP indefinitely.
Also, this makes it possible for callbacks not to process all events and
still not busy loop.
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We need to resume audio devices even for streams that are created in
corked stat, so that the latency ranges of the audio device are known
during the initial latency negotiation. If we don't the latency
negotiation will be based on placeholder data and changed later on which
clients do not expect.
This should fix issues with Skype.
https://bugzilla.redhat.com/show_bug.cgi?id=554929
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