Since some digital mics have the phase-inversion problem in one channel,
adding both channels for mono stream results in the noise.
Use route_policy copy to avoid that situation.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The change that removed the Master controls for Xonar D2(X) cards was
reverted, so we don't need to use the softvol plugin anymore ...
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
The hd-audio driver may have no analog stream but only a digital one
which has usually non-zero device number. To avoid the bogus namehint,
set hint.device 0 for pcm.default (it wasn't handled properly due to
asym).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The system-wide default PCM config works now (by the fix of plug
mmap_emul support), thus no card-specific hackish config is needed
for GUS.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Master Volume controls were removed from Xonar D2/D2X cards; add the
softvol plugin so that we have at least PCM volume.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
The ice1724-based cards can handle only 32bit while the apps almost
expet 16bit format for SPDIF I/O. This prevents the default config
working on many apps like mplayer, xine, etc.
This patch simply adds the least automatic conversion by linear plugin.
Note that "plug" isn't used here. Otherwise we get a problem of the
routing (plug over plug is buggy).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Addeed a new option "truncate" to indicate the behavior of creating
the output file. When it's true (the default), the file is overwritten
and truncated at creation. When false, the plugin tries to open a
unique file with a number suffix.
The global behavior of "file" and "tee" PCMs is defined via
defaults.pcm.file_truncate option. You can overwrite it in ~/.asoundrc.
"./configure" options for selecting ALSA configuration (default
/usr/share/alsa) and plugin (/usr/lib/alsa-lib) directories introduced
by alsa-hg/alsa-lib changeset 2284 cause problems with cross-compilation
and packaging - there is no way to redefine them in runtime, during
installation phase.
This patch adds a level of indirection between constants and their
usage - alsaconfigdir for ALSA_CONFIG_DIR and alsaplugindir for
ALSA_PLUGIN_DIR - which can be redefined during "make install" stage.
Signed-off-by: Pawel MOLL <pawel.moll@st.com>
The direct plugins have the automatic format-detection feature but it
wasn't enabled properly in the interface. Now you can pass the format
"unchanged" to make the plugin detect a proper format.
This will change the default format of some drivers, such as, HD-audio.
reasons:
- rapid development
- class-like code structure
- more readable code
features:
- hcontrol binding is managed from python (opportunity to create
virtual mixer without driver or join multiple cards to behave as one)
This patch allows for gain in the softvol plugin, in addition to attenuation.
The plugin now has a "max_dB" parameter (up to 50 dB) as well as the
original "min_dB" parameter (down to -51 dB). max_dB defaults to 0 dB, so
unless max_dB is specified in a device conf, the behavior of the plugin will
be the same as before (attenuation only).
HDA-Intel.conf is also modified to use softvol for its default capture.
So now, capture is filtered through softvol (range -30 to +30 dB) before
being passed on to dsnoop as before.
The softvol plugin allows a range of -51 to +50 dB, so max_dB could be
increased to 50. But eventually samples are going to get clipped. At 40
dB I was beginning to get clipping when recording a sample sound at a
"reasonably soft" volume using a digital mic on the stac9205 HDA codec.
The motivation for this work is that some HDA codecs have no hardware gain
control for some paths. For instance, the stac9205 has support for digital
mics, but there is no gain control widget for this signal before it is placed
on the Azalia link (only a mute). Therefore gain can only be accomplished
via software.
Signed-off-by: Steve Longerbeam <stevel@embeddedalley.com>
Added --with-configdir and --with-plugindir options to configure
which specify the directories for config files and plugin objects
respectively. The default paths when these options are not
specified are unchanged.
This replaces all occurences of alsa-devel@lists.s[ource]f[orge].net
that a simple recursive grep found in the current HG ALSA repos by
alsa-devel@alsa-project.org.
Signed-off-by: Rene Herman <rene.herman@gmail.com>
Revert dmix.conf and dsnoop.conf.
The ipc key offset had been already modified to be unique for
each card, stream, device and subdevice interanally in dmix &
co plugins.
- add long card name to device description
- create empty PCM plugin to allow right hint description parsing
- reorder devices in alsa.conf
- make namehint more configurable (using default.namehint.showall switch)
- add two levels basic and exteded for hints to default configuration files
- do not show direct device aliases
- removed all known memory leaks
- add snd_device_name_hint() and snd_device_name_free_hint() functions
- add snd_ctl_iface_conf_name() functions
- do not accept parameters for the plugin definition without @args section
- add defaults.pcm.dmix.card/device and dsnoop.card/device definitions
- add hints for HDA-Intel.conf, pcm/dmix.conf, pcm/dsnoop.conf and alsa.conf
- add test/namehint test utility
- doxygen related cleanups
Use dmix/dsnoop for maestro3 boards. Although maestro3 has multiple
playback capability, it supports only two streams (with the currently
available firmware).
