Olaf Hering [Thu, 7 Dec 2006 07:25:01 +0000 (08:25 +0100)]
[ALSA] create driver symlink in snd-aoa /sys/bus/aoa-soundbus/devices/*/
create sysfs driver symlink for snd-aoa in /sys/bus/aoa-soundbus/devices/*/ Acked-by: Johannes Berg <johannes@sipsolutions.net> Signed-off-by: Olaf Hering <olaf@aepfle.de> Signed-off-by: Andrew Morton <akpm@osdl.org> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Olaf Hering [Thu, 7 Dec 2006 07:24:12 +0000 (08:24 +0100)]
[ALSA] create device symlink in snd-aoa
create sysfs device symlinks for snd-aoa in /sys/class/sound/controlC0 This
allows hald to recognize the device as sound device. Furthermore it allows
the desktop user to actually access the sound device nodes. hald and
related packages will modify the acl attributes.
Fixes https://bugzilla.novell.com/show_bug.cgi?id=106294 Acked-by: Johannes Berg <johannes@sipsolutions.net> Signed-off-by: Olaf Hering <olaf@aepfle.de> Signed-off-by: Andrew Morton <akpm@osdl.org> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
[ALSA] emu10k1: Rename the digital optical capture control for the Audigy 2 ZS
Notebook.
Digital playback and capture now works, but it is not bit accurate because it
passes through a resampler.
Bit accurate playback and capture will be implemented later via the p17v.
Signed-off-by: James Courtier-Dutton <James@superbug.co.uk> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
[ALSA] Current driver does not utilize 44.1kHz high quality sampling rate converter.
Following patch will make the driver to use the 44.1kHz SRC automatically
if the pcm source is 44.1kHz signed 16bit stereo.
The SRC is available in YMF754 only.
Signed-off-by: Teru KAMOGASHIRA <teru@sodan.ecc.u-tokyo.ac.jp> Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Adrian Bunk [Tue, 28 Nov 2006 11:10:09 +0000 (12:10 +0100)]
[ALSA] sound/soc/soc-dapm.c: make 4 functions static
Make the following needlessly global functions static:
- dapm_power_widgets()
- dapm_mux_update_power()
- dapm_mixer_update_power()
- dapm_free_widgets()
Signed-off-by: Adrian Bunk <bunk@stusta.de> Signed-off-by: Andrew Morton <akpm@osdl.org> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Jonathan Woithe [Tue, 28 Nov 2006 10:35:52 +0000 (11:35 +0100)]
[ALSA] hda-codec - Make internal speaker work on Acer C20x tablets
The following patch creates a new 'Mono speaker' control in alsamixer
when the Realtek 'acer' model is used with hda_intel. This is needed so
the internal mono speaker (when present) can be controlled.
This new control won't do anything in Acer laptops which are not fitted with
a mono speaker. Acer models which are known to have a mono speaker are the
C20x tablet series but there may be others. I guess we could define a new
model specifically for Acers with mono speakers but this seems a bit silly
given that such a model will be identical to the normal 'acer' model except
for this added control.
This patch also adds the C20x tablets to the list of PCI ids associated with
the 'acer' model. This means that owners of C20x machines will no longer
have to supply 'model=acer' when loading hda_intel.
Signed-off-by: Jonathan Woithe <jwoithe@physics.adelaide.edu.au> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Frank Mandarino [Fri, 24 Nov 2006 14:49:39 +0000 (15:49 +0100)]
[ALSA] Update AT91 ASoC driver for 2.6.19 kernel.
Changes were required to support latest AT91 header files.
Also updated to remove AT91RM9200-specific code in the ASoC
platform drivers to support the AT91SAM9260 and AT91SAM9261
chips, but no testing was performed on these chips.
Signed-off-by: Frank Mandarino <fmandarino@endrelia.com> Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Takashi Iwai [Fri, 24 Nov 2006 14:42:07 +0000 (15:42 +0100)]
[ALSA] intel8x0 - Add spdif_aclink option
Added spdif_aclink module option to specify whether the board
has SPDIF over AC-link or a direct connection from the controller
chip.
NForce and ICH4 (or newer) boards may be equipped with SPDIF
through AC97 codec. In such a case, SPDIF should be handled
as if the old ICH style (the same slot for analog and digital).
A quirk list is added to detect this automatically for known
hardwares.
Corresponds to ALSA bug#2637.
Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Giuliano Pochini [Fri, 24 Nov 2006 12:03:58 +0000 (13:03 +0100)]
[ALSA] echoaudio, add TLV support
This patch adds TLV support to the echoaudio driver.
All gains are in the range -127dB to +6dB with steps of 1dB, and -128 is
mute. VU-meters levels go from -128 to 0dB. The input gain of the Layla20
ranges from -25dB to +25dB in steps of 0.5dB.
Takashi Iwai [Wed, 22 Nov 2006 10:52:52 +0000 (11:52 +0100)]
[ALSA] hda-codec - Add asus-laptop model for ALC861 (ALC660)
Added a new model 'asus-laptop' for ASUS F2*/F3* laptops
with ALC861 (equivalent with ALC660) codec chip.
Also fixed the model for PCI SSID 1043:1338.
Corresponding to ALSA bug#2480.
Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Takashi Iwai [Tue, 14 Nov 2006 11:30:52 +0000 (12:30 +0100)]
[ALSA] hda-codec - Add support for Sony UX-90s
Added the model entry (model=hippo) for Sony UX-90s with ALC262 codec.
Although the device has no SPDIF output, the hippo model adds a
PCM output, but it must be harmless.
Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Tobin Davis [Tue, 14 Nov 2006 11:13:39 +0000 (12:13 +0100)]
[ALSA] Add Conexant audio support to the HD Audio driver
This driver adds limited support for the Conexant 5045 and 5047 HD Audio
codecs. Some issues still need to be resolved. The code is based
primarily on code from the Analog Devices AD1981 support and the Realtek
ALC260 support. Some code came from the original code developed by Alex
Pototskiy (see alsa bugtracker 2485).
Signed-off-by: Tobin Davis <tdavis@dsl-only.net> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Takashi Iwai [Thu, 9 Nov 2006 15:47:26 +0000 (16:47 +0100)]
[ALSA] ice1724 - Add support of M-Audio Audiophile 192
Added the (experimental) support of M-Audio Audiophile 192 board.
Currently, the analog and the digital playbacks seem working fine.
The inputs seem not working as far as I've tested yet.
Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Liam Girdwood [Thu, 9 Nov 2006 15:35:01 +0000 (16:35 +0100)]
[ALSA] ASoC - mixer name changes for older OSS app support
This patch suggested by Richard Purdie changes the names of some WM8731
and WM8750 mixers so that they will be recognised by some older OSS
mixer apps.
Changes:-
o WM8731 Playback changed to Master Playback
o WM8750 Out1 changed to Headphone
o WM8750 Out2 changed to Speaker
Takashi Iwai [Mon, 6 Nov 2006 14:38:55 +0000 (15:38 +0100)]
[ALSA] hdspm - Fix printk warnings
sound/pci/rme9652/hdspm.c: In function 'snd_hdspm_hw_params':
sound/pci/rme9652/hdspm.c:3681: warning: format '%08X' expects type 'unsigned int', but argument 4 has type 'unsigned char *'
sound/pci/rme9652/hdspm.c:3692: warning: format '%08X' expects type 'unsigned int', but argument 4 has type 'unsigned char *'
Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Instead of using a somewhat algorithmic approach of initializing the
YSS225's registers, just use a simple series of port/value pairs.
This makes it easier to later replace or entirely remove the register
data blob.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Hubert Kahlert [Tue, 31 Oct 2006 14:31:27 +0000 (15:31 +0100)]
[ALSA] Fix mask to stop AT91 SSC clock on shutdown
This patch by Frank Mandarino and Hubert Kahlert fixes a bug in the AT91
SSC (i2s) shutdown code that would erroneously disable other AT91
peripheral clocks.
Signed-off-by: Hubert Kahlert <hkahlert@hk-datentechnik.de> Signed-off-by: Frank Mandarino <fmandarino@endrelia.com> Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Takashi Iwai [Tue, 24 Oct 2006 16:25:29 +0000 (18:25 +0200)]
[ALSA] ac97 - Suppress power-saving mode on non-supporting drivers
Don't enable power-saving mode on drivers that don't support
it. The supporting drivers set AC97_SCAP_POWER_SAVE to scaps
at creation of ac97 instance.
Currently enable on the following drivers: intel8x0, intel8x0m,
atiixp, atiixp-modem, via82xx and via82xx-modem.
Also, a bit clean up of power-saving stuff:
- Don't create an own workq
- Remove superfluous ifdefs
Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Liam Girdwood [Thu, 19 Oct 2006 18:35:56 +0000 (20:35 +0200)]
[ALSA] ASoC: Add support for BCLK based on (Rate * Chn * Word Size)
This patch adds support for the DAI BCLK to be generated by multiplying
Rate * Channels * Word Size (RCW).
This now gives 3 options for BCLK clocking and synchronisation :-
1. BCLK = Rate * x
2. BCLK = MCLK / x
3. BCLK = Rate * Chn * Word Size. (New)
Changes:-
o Add support for RCW generation of BCLK
o Update Documentation to include RCW.
o Update DAI documentation for label = value DAI modes.
o Add RCW support to wm8731, wm8750 and pxa2xx-i2s drivers.
Frank Mandarino [Thu, 19 Oct 2006 16:22:53 +0000 (18:22 +0200)]
[ALSA] ASoC AT91 DAI modes update
This patch by Frank Mandarino updates the AT91RM9200 I2S DAI audio modes
as follows:-
o fixes a typo in the 16k mode
o removes experimental 24k mode
o adds a 32k mode.
Signed-off-by: Frank Mandarino <fmandarino@endrelia.com> Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Remy Bruno [Tue, 17 Oct 2006 10:41:56 +0000 (12:41 +0200)]
[ALSA] hdsp - Add DDS register support for RME9632 rev >= 152
Add DDS register support for RME9632 rev >= 152.
This register sets the sample rate for these cards and is required
in addition to the standard control register. It corresponds to a
quartz divisor.
Signed-off-by: Remy Bruno <remy.bruno@trinnov.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Kailang Yang [Tue, 17 Oct 2006 10:32:26 +0000 (12:32 +0200)]
[ALSA] hda-codec - Add new modesl for Realtek codecs
Changes from Realtek driver:
- New models hippo and hippo_1 for ALC262
- New models tagra-dig and tagra-2ch-dig for ALC883
- New id for ALC660 codec chip
Signed-off-by: Kailang Yang <kailang@realtek.com.tw> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Liam Girdwood [Mon, 16 Oct 2006 19:19:48 +0000 (21:19 +0200)]
[ALSA] ASoC - Fix build warnings in soc-core.c
This patch fixes some build warnings in soc-core.c
Changes:-
o Check the return value of soc_ac97_dev_register()
o Check return value of calls to device_create_file()
Remy Bruno [Mon, 16 Oct 2006 10:46:32 +0000 (12:46 +0200)]
[ALSA] hdspm: Add support for AES32
Add support for AES32. Difference between MADI and AES32 is done
through revision. Master support is not finished for now (RME so-called DDS
feature is not supported yet)
Signed-off-by: Remy Bruno <remy.bruno@trinnov.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Liam Girdwood [Fri, 13 Oct 2006 10:33:56 +0000 (12:33 +0200)]
[ALSA] ASoC DAI capabilities labelling
This patch suggested by Takashi changes the DAI capabilities definitions
in pxa-i2s.c, at91rm9200-i2s.c, wm8731.c, wm8750.c and wm9712.c to use a
label = value style.
Liam Girdwood [Thu, 12 Oct 2006 12:29:03 +0000 (14:29 +0200)]
[ALSA] ASoC pxa2xx AC97 support
This patch adds pxa2xx AC97 ASoC audio support. It's based on
sound/arm/pxa-ac97 by Nicolas Pitre with the following differences.
o Modified driver structure to use ASoC core PCM callbacks.
o Removed AC97 configuration function (all handled in ASoC core)
o Added and exported ASoC DAI configuration table.
o Added DMA support for AUX DAC and Mic ADC
o Separated out AC97 reset into cold and warm reset functions.
From: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Nicolas Pitre <nico@cam.org> Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Liam Girdwood [Thu, 12 Oct 2006 12:26:55 +0000 (14:26 +0200)]
[ALSA] ASoC pxa2xx DMA support
This patch adds pxa2xx ASoC DMA audio support. It's based on
sound/arm/pxa-pcm.c by Nicolas Pitre with the following differences.
o Modified driver structure to use ASoC core PCM callbacks and data
structures.
o Registration with ASoC core.
From: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Nicolas Pitre <nico@cam.org> Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Clemens Ladisch [Mon, 9 Oct 2006 06:17:48 +0000 (08:17 +0200)]
[ALSA] pci: select FW_LOADER instead of depending on it
Let the AudioScience, Echoaudio and Riptide drivers select FW_LOADER
instead of depending on it so that they can be configured without having
to enable FW_LOADER manually.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Frank Mandarino [Fri, 6 Oct 2006 16:40:25 +0000 (18:40 +0200)]
[ALSA] ASoC AT91RM92000 I2S support
This patch adds I2S support to the Atmel AT91RM9200 CPU.
Features:-
o Playback/Capture supported.
o 16 Bit data size.
o 8k - 48k sample rates.
o ssc0, ssc1 and ssc2 supported as I2S ports.
Signed-off-by: Frank Mandarino <fmandarino@endrelia.com> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Richard Purdie [Fri, 6 Oct 2006 16:37:32 +0000 (18:37 +0200)]
[ALSA] ASoC codecs: WM9712 support
This patch adds ASoC support for the WM9712 codec.
Supported features:-
o Capture/Playback/Sidetone/Bypass.
o Aux DAC.
o 8k - 48k sample rates.
o DAPM.
Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Richard Purdie [Fri, 6 Oct 2006 16:36:39 +0000 (18:36 +0200)]
[ALSA] ASoC codecs: WM8750 support
This patch adds ASoC support for the WM8750 codec.
Supported features:-
o Capture/Playback/Sidetone/Bypass.
o 16 & 24 bit audio.
o 8k - 96k sample rates.
o DAPM.
Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Richard Purdie [Fri, 6 Oct 2006 16:36:07 +0000 (18:36 +0200)]
[ALSA] ASoC codecs: WM8731 support
This patch adds ASoC support for the WM8731 codec.
Supported features:-
o Capture/Playback/Sidetone/Bypass.
o 16 & 24 bit audio.
o 8k - 96k sample rates.
o DAPM.
Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Liam Girdwood [Fri, 6 Oct 2006 16:34:51 +0000 (18:34 +0200)]
[ALSA] ASoC: documentation & maintainer
This patch adds documentation describing the ASoC architecture and a
maintainer entry for ASoC.
The documentation includes the following files:-
codec.txt: Codec driver internals.
DAI.txt: Description of Digital Audio Interface standards and how to
configure a DAI within your codec and CPU DAI drivers.
dapm.txt: Dynamic Audio Power Management.
platform.txt: Platform audio DMA and DAI.
machine.txt: Machine driver internals.
pop_clicks.txt: How to minimise audio artifacts.
clocking.txt: ASoC clocking for best power performance.
Richard Purdie [Fri, 6 Oct 2006 16:32:18 +0000 (18:32 +0200)]
[ALSA] ASoC: dynamic audio power management (DAPM)
This patch adds Dynamic Audio Power Management (DAPM) to ASoC.
Dynamic Audio Power Management (DAPM) is designed to allow portable and
handheld Linux devices to use the minimum amount of power within the
audio subsystem at all times. It is independent of other kernel PM and
as such, can easily co-exist with the other PM systems.
DAPM is also completely transparent to all user space applications as
all power switching is done within the ASoC core. No code changes or
recompiling are required for user space applications. DAPM makes power
switching decisions based upon any audio stream (capture/playback)
activity and audio mixer settings within the device.
DAPM spans the whole machine. It covers power control within the entire
audio subsystem, this includes internal codec power blocks and machine
level power systems.
There are 4 power domains within DAPM:-
1. Codec domain - VREF, VMID (core codec and audio power)
Usually controlled at codec probe/remove and suspend/resume, although
can be set at stream time if power is not needed for sidetone, etc.
2. Platform/Machine domain - physically connected inputs and outputs
Is platform/machine and user action specific, is configured by the
machine driver and responds to asynchronous events e.g when HP are
inserted
3. Path domain - audio subsystem signal paths
Automatically set when mixer and mux settings are changed by the user.
e.g. alsamixer, amixer.
4. Stream domain - DAC's and ADC's.
Enabled and disabled when stream playback/capture is started and stopped
respectively. e.g. aplay, arecord.
All DAPM power switching decisions are made automatically by consulting
an audio routing map of the whole machine. This map is specific to each
machine and consists of the interconnections between every audio
component (including internal codec components).
Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Frank Mandarino [Fri, 6 Oct 2006 16:31:09 +0000 (18:31 +0200)]
[ALSA] ASoC: core code
This patch is the core of ASoC functionality.
The ASoC core is designed to provide the following features :-
o Codec independence. Allows reuse of codec drivers on other platforms
and machines.
o Platform driver code reuse. Reuse of platform specific audio DMA and
DAI drivers on different machines.
o Easy I2S/PCM digital audio interface configuration between codec and
SoC. Each SoC interface and codec registers their audio interface
capabilities with the core at initialisation. The capabilities are
subsequently matched and configured at run time for best power and
performance when the application hw params are known.
o Machine specific controls/operations: Allow machines to add controls
and operations to the audio subsystem. e.g. volume control for speaker
amp.
To achieve all this, ASoC splits an embedded audio system into 3
components :-
1. Codec driver: The codec driver is platform independent and contains
audio controls, audio interface capabilities, codec dapm and codec IO
functions.
2. Platform driver: The platform driver contains the audio dma engine
and audio interface drivers (e.g. I2S, AC97, PCM) for that platform.
3. Machine driver: The machine driver handles any machine specific
controls and audio events. i.e. turning on an amp at start of playback.
Signed-off-by: Frank Mandarino <fmandarino@endrelia.com> Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.Girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
projects / linux-2.6 / log