| Commit message (Collapse) | Author | Age | Files | Lines |
|
|
|
| |
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
|
|
|
|
| |
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
| |
Unfortunately some more testing revealed some issues with it,
specifically if pulse is running your complete config is replaced the bits in
the on_pulse_is_running directive. Which might not be what one actually wants :)
I couldn't find a proper solution for this. So i've changed the code to
optionally load config files. Just like the load hook does. Actually i just
optionally call the snd_config_hook_load function, but that's not actually in
the alsa API....
Also it now decides pulse is running as soon as the authorizing step begins
(just after the actually connection is setup), which should save some
round-trips and overhead.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
| |
The attached patch extends the alsa pulse plugin set with a alsa
configuration hook. Allowing one to specify some configuration parameters
that only come into effect when pulseaudio is running.
For example a configution file like:
@hooks [ {
func on_pulse_is_running
pcm.!default { type pulse }
ctl.!default { type pulse }
}
]
will redirect the default alsa pcm and ctl to pulse iff pulse is running.
(Assuming you defined the hook function correctly ofcourse)
This is usefull for distributions that don't want to force their users to
switch completely to pulseaudio, but have things a bit more dynamic :)
The solutions isn't optimal though. It will mean that every program loading
accessing alsa will try to make an (extra) connection to pulse to decide what
to do. But i think it's the best we can do for now (or at least that i can do
with my minimal knowledge of alsa).
A nicer solution would be a way to always specify the pulse plugin as default
and have a sort of fallback for when that fails.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
|
|
|
|
|
|
|
| |
ALSA bug#3035:
https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3035
Use dbus_connection_unref() instead of the deprecated dbus_connection_close().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
|
|
|
|
|
|
|
|
|
| |
ALSA bug#3860:
https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3860
The Maemo DSP plugin checks for D-Bus in configure.in and then makes a bold assumption that this means it should use a proprietary resource manager available only on a specific proprietary platform.
Attaching a patch to add --enable-maemo-resource-manager configure flag that enables the resource manager if set and if D-Bus is present.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
|
|
|
|
|
|
|
| |
Ignore hint sections defined by hand.
Those are heplful to get listed in various places, such as aplay -L
ALSA bug#3834:
https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3834
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
|
|
|
|
| |
Remove another assert that results in an unexpected crash.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
|
|
|
|
|
|
|
| |
If stream connection failes, don't assume that stream is connected upon closing.
ALSA bug#3831:
https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3831
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
| |
|
| |
|
| |
|
|
|
|
| |
Patch-level: Merged
|
| |
|
|
|
|
| |
Patch-level: Merged
|
|
|
|
|
|
|
|
|
|
|
| |
From: Lennart Poettering <mznyfn@0pointer.de>
It adds support to report back XRUN to the application if one
happens. This is required to make some applications work on top of the
pulse plugin. One being XMMS, which checks if a song finished to play
by waiting for an XRUN (yes, I don't argue that XMMS shouldn't do
that, but nonetheless it is a good thing if XRUNs are reported
properly.)
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
| |
It increases the "pre-buffering level" (i.e. start threshold) to the
full buffer size minus one period. This makes PA work a little bit
more like normal audio devices, and makes a few drop outs go away for
software which uses very small period sizes.
It also increases the initial maximum buffer size, which allows a
small overcommit. That's not really an issue, but cleaner nonetheless
so I smuggled it into this patch.
Also reported in the ALSA BTS:
https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3578
From: Lennart Poettering <mznyfn@0pointer.de>
|
|
|
|
|
|
|
| |
Added the minmax conditions for period_bytes and periods to pulse plugin.
This fixes ALSA bug#2601.
Patch from Mike Gorse <mgorse@mgorse.dhs.org>
|
|
|
|
|
|
| |
assert(!pcm->stream) shouldn't be checked when the PCM state is
SETUP, too (ALSA bug#3470).
The original patch by Mike Gorse <mgorse@mgorse.dhs.org>
|
|
|
|
|
|
|
|
|
|
| |
This patch fixes the unexpected assert call at calling snd_pcm_hw_params
in PREPARED state. Since multiple hw_params calls are allowed, the pulse
plugin shouldn't call assert.
Handled in ALSA bug#3470.
From: Sean McNamara <smcnam@gmail.com>
|
| |
|
|
|
|
| |
Patch-level: Merged
|
| |
|
|
|
|
| |
Patch-level: Merged
|
|
|
|
|
| |
Take speex rate converter code from speex SVN tree, which includes the
fix for the noises with simple conversion (signed / unsigned mismatch).
|
|
|
|
|
| |
The direct sinc table can be noisy in some conditions (e.g. up-conversion
from 11025 to 44100Hz). Disable it as a temporary solution for now.
|
|
|
|
|
| |
A (temporary) fix for the pop noise at the beginning of playback
with samplerate plugin.
|
|
|
|
| |
Added -no-undefined option to LDFLAGS to make linking sure.
|
| |
|
|
|
|
| |
Patch-level: Merged
|
|
|
|
|
|
|
|
| |
- Add --with-speex configure option to specify the build of speex rate
plugin. As default, it's linked to external library. If not available,
defaults to builtin code.
- Show build conditions at the end of configure script
- Use AS_HELP_TEXT()
|
|
|
|
|
|
|
| |
Added the missing call of avcodec_init() to avoid setfault of a52
plugin with the latest svn revision of ffmpeg.
From: Fabian van der Werf <fvanderwerf@gmail.com>
|
|
|
|
|
| |
Fixed plugindir config setting when no option was given.
Also fixed an obvious typo.
|
|
|
|
|
| |
Added --with-plugindir configure option to specify the directory
for plugin objects.
|
| |
|
| |
|
| |
|
|
|
|
| |
Patch-level: Merged
|
|
|
|
| |
Added missing files for pph speex resampler plugin.
|
|
|
|
|
|
|
|
| |
From: Jean-Marc Valin <jean-marc.valin@usherbrooke.ca>
I'm attaching an updated version of my resampler plugin. It fixes a few
minor issues and it adds support for fixed-point processing (just add
-DFIXED_POINT to the build). Let me know if there's any problem.
|
|
|
|
| |
Added missing gcd.h to rate-lavc/Makefile.am.
|
|
|
|
|
|
| |
Added the documentation for speex rate plugin.
From: Jean-Marc Valin <jean-marc.valin@usherbrooke.ca>
|
|
|
|
|
|
|
|
| |
Remove parameter constraints where we actually have none. Also, restrict
total buffer size to 4 MB as current versions of the PulseAudio server
will refuse streams larger than that.
Signed-off-by: Pierre Ossman <ossman@cendio.se>
|
| |
|
| |
|
| |
|
| |
|
|
|
|
| |
Patch-level: Merged
|
|
|
|
|
|
|
| |
Added another rate resampler plugin based on speex code.
Light weight and much better quality.
From: Jean-Marc Valin <jean-marc.valin@usherbrooke.ca>
|
|
|
|
|
|
|
| |
The libsamplerate rate plugin has wrong implementations of input_frames
and output_frames callbacks. They have to be swapped.
From: Jean-Marc Valin <jean-marc.valin@usherbrooke.ca>
|