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).
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.
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.
Fix the check of nonblock option for all hw layer.
Instead of passing in asound.conf, check the option in snd_pcm_hw_open()
so that the nonblock option is referred in the case of "type hw ..." style
definition, too.
- Support multi-card/device for dmix/dsnoop/dshare plugins
The unique ipc key is calculated based on card/device/sub index
- Clean up and share the code among all d* plugins
- Refer the defaults.pcm.* configuration
The base ipc_key number, ipc_gid and ipc_perm are referred.
Added a new "nonblock" option for hw layer. This controls the non-blocking
"open" mode as default.
This option is set to TRUE as the default configuration. If the old behavior
is preferred, set "defaults.pcm.nonblock" to 0 in /etc/asound.conf.
Example failure error message before fix:
speaker-test 0.0.8
Playback device is spdif
Stream parameters are 48000Hz, S16_LE, 2 channels
Using 16 octaves of pink noise
ALSA lib setup.c:549:(add_elem) Cannot obtain info for CTL elem (MIXER,'SPDIF Front',0,0,0): No such file or directory
Playback open error: -2,No such file or directory
Routes front channels via fxbus 8 and 9 instead of fxbus 0 and 1. This
makes the front and default devices independent of each other, and
"Wave Surround", "Wave Center" and "Wave LFE" don't have to be muted
during 5.1 playback.
Signed-off-by: Mikael Magnusson <mikma@users.sourceforge.net>
This defines two modem related pcm devices: "phoneline" is driver's pcm
reference, with "modem" the line will be "Hook-Off"ed automatically.
Signed-off-by: Sasha Khapyorsky <sashak@smlink.com>
- Fix surround40 channel numbers
- Fix rear/center_lfe channel assignment
This configuration is based on the channel order on Revo 7.1,
Front/CLFE/Surr/Side
If the other ice1724-based boards have a different order, we need
to rename the driver's id to distinguish the difference.
