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.
From: Julian Scheel <julian@jusst.de>
Hi all,
attached is an updated ICE1724.conf - I have added channels X parameters to
the .pcm.surroundXX-fields, because otherwise it's for example not possible
to play a 5.1-stream on surround51 and a stereo-stream on side simultaneous.
I need this feature to do multiroom-sound through one single sound-card (one
room playing a DVD with 5.1, another room playing some MP3s through channels
7 and 8).
I hope you could follow my explanations and will add this to the tree.
Cheers,
Julian
- separated the "base library" from ac97.c (dynamically loaded)
- added necessary handling of all private structures
- added selector and sid registration functions and handling code
- added basic code for hda.c (yeah, my notebook has this hw)
- exported all necessary functions to create a mixer module outside alsa-lib
- separated simple mixer API from the simple mixer implementation
(using callbacks as usuall)
- src/mixer/simple.c is the core
- src/mixer/simple_none.c is the current (no-abstraction) implementation
based on control names; note that this module does not depend on
internal ALSA structures now
- src/mixer/simple_abst.c is the ongoing abstraction which will use
external dynamic modules; src/conf/smixer.conf will describe which
modules will be used depending on the components from the driver
Call hwsync of slave PCM on snd_pcm_hwsync() over dmix.
This reduces ocasional output noises, typically seen in xmms and bmp with
multi-threading on SMP/HT systems.
Disabled slowptr of default dmix definition again.
This patch adds a DSP patch to fix an spdif_bug on some Audigy2 cards.
It also does detailed identification of the capabilities of the sound
card so that the patch is only applied to the sound cards that really
need it.
Signed-off-by: James Courtier-Dutton
