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Fix GUS.conf to work for default PCM.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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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>
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The softvol must be inside the plug. Otherwise it gets stuck.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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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>
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Added the iec958 PCM definition for PS3.
Since it's a new feature, the definition is marked as optional.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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apply softvol before plug as softvol doesn't support U8 as of now.
This also improves the sound quality.
Signed-off-by: Stas Sergeev <stsp@aknet.ru>
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Use "defaults.pcm.file_format" for the default file format of
file plugin. It's set to "raw" as default for compatibility.
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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.
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ALSA_PLUGIN_DIR
"./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>
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Added the new PCM "hdmi" for HDA-Intel.
It's still experimental.
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Now the board with ALC850 can work with 8-channel outputs.
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See ALSA bug#3735
https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3735
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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.
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Remove the softvol plugin from all other CMI8788 devices.
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Add mixer controls to manage the S/PDIF channel status bits.
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Remove the now superfluous softvol plugin from the CMI8788
configuration, use 24-bit samples for dmix, and add an alias for the
AV200 driver.
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Add a whitespace to make the ctl.hw definition better readable.
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The attached patch updates the PC-Speaker.conf for the use of softvol.
Signed-off-by: Stas Sergeev <stsp@aknet.ru>
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The dmix and dsnoop plugins need a fixed substream number instead of
the next-available one (-1) as the default number. Now it's set to 0.
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Add a .conf file to enable dmix/dsnoop and softvol for CMI8788.
Using dmix helps mask the bug that all audio is forced to 48 kHz. :-)
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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)
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Added PS3 configuration.
No iec958 PCM at this stage since it doesn't support passthru yet.
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PPC drivers should use S16_BE as the base format of dmix/dsnoop.
This avoid confusion of format endianess via CPU emulation.
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The subdevice number of a dmix slave PCM has to be specified explicitly
for the device with multiple substreams such as Maestro3.
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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>
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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.
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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>
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add a configuration file for USB audio devices
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Fix the capture slave to hw for CS46xx default PCM since dsnoop
seems not working with this hardware well.
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fix a typo introduced in changeset d14ade7ede2a
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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.
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- remove device 0/subdevice 0 from configuration files (it's default)
- name_hint
- fixed parsing slaves
- obtain device numbers directly from 'type hw' configurations to
avoid poluting of configurations scripts with hint.device lines
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- 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
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See ALSA bug#1573
Also add card_inum, iadd, imul functions to configuration files.
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- 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
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Use dmix/dsnoop for maestro3 boards. Although maestro3 has multiple
playback capability, it supports only two streams (with the currently
available firmware).
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Added --disable-alisp configure option to disable alsip support.
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Add a new config for new snd-aoa driver, aliased to PMacToonie.
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Fixed driver alias of Aureon 7.1 Universe. Aureon71Universe was too long
as the driver name. The corrected name is Aureon71Univ.
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Automatically turn on iec958 capture of iec1724 boards
with spdif PCM via hooks plugin.
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Added defaults.pcm.dmix_format and dmix_rate definitions to
alsa.conf. They are referred as the default values of standard
dmix/dsnoop PCM.
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Fixed surround40 config for ens1370. Added missing interface for
the hook control.
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Fix device number of control emenets in ICH4 iec958 PCM hooks.
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Add PCM definitions "dpl" and "dpl2" in pcm/dpl.conf.
Include the file via
<confdir:pcm/dpl.conf>
for use.
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Introduce "max_periods" option to specify the max number of periods
per buffer to each plugin.
- When max_periods = -1, the fixed buffer size as the slave size is
used (old behavior).
- When max_periods = 0 (or 1), the number of periods is variable
between 2 and the slave buffer size.
- When max_periods greater than 2 is given, it specifies the max
periods of that pcm explicitly.
When no option is given in the PCM defintion, the value
"defaults.pcm.dmix_max_periods" is referred as default.
The default value is 0, as defined in alsa.conf.
You can override this in ~/.asoundrc or /etc/asound.conf as you like.
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With the patch, dmix allows apps to use more flexible buffer sizes.
The max buffer size is unlimited, and the minimal buffer size is
(period size * 2). The buffer size is aligned to period size.
The period size is still bound to the period size of slave PCM.
To back to the old behavior (the fixed buffer size), you can set
defaults.pcm.dmix_variable_buffer false
in your configuration.
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