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Remove initializations of spinlock and mutexes which
are done earlier in snd_wss_new(). The snd_wss_new()
is called from snd_wss_create().
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
snd-opti92x-ad1848 mistakingly passes WSS_HW_OPTI93X currently. This
fixes it as tested with a OPTi 82C929A/AD1848 card.
Signed-off-by: Rene Herman <rene.herman@gmail.com>
Acked-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Limit opti93x cards capture formats to only linear ones.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Reviewed-by: Rene Herman <rene.herman@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Use the wss detection code and kill the ad1848 library.
The library is fully assimilated into the new wss library.
This required reworking of the AD1848 family code
so the code is changed to correctly detect chips from
the AD1848 and CS4231 families.
I have tested it on following cards:
Gallant SC-6600 (codec: AD1848, driver: snd-sc6600)
SoundScape VIVO/90 (codec: AD1845, driver: snd-sscape)
SG Waverider (codec: CS4231A, driver: Rene Herman's snd-galaxy)
Opti930 (codec: built-in - CS4231 compatible, driver: snd-opti93x)
Opti931 (codec: built-in - CS4231 compatible, driver: snd-opti93x)
Gallant SC-70P (chip/codec: CS4237B, driver: snd-cs4236)
Audio Plus 3D (chip/codec: CMI8330A, driver: snd-cmi8330)
Dell Latitude CP (chip/codec: cs4236, driver snd-cs4232)
Sound playback and recording works on all these cards.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Reviewed-by: Rene Herman <rene.herman@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Use the wss pcm code and kill the ad1848 pcm code.
The AD1848 chip is much slower than CS4231 chips
so the waiting loop was increased 100x (10x is not
enough).
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Reviewed-by: Rene Herman <rene.herman@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Use the wss mixer code and kill the ad1848 mixer code.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Reviewed-by: Rene Herman <rene.herman@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Use CS4231P instead of AD1848P (kill the AD1848P).
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Reviewed-by: Rene Herman <rene.herman@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Use the wss macros instead of ad1848 ones.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Reviewed-by: Rene Herman <rene.herman@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Use wss constants for mode.
Move ad1848 hardware constants to the wss.h.
Move mixer tlv macros into the ad1848_lib.c from the ad1848.h.
Drop the MODE_RUNNING spurious IRQ guard on AD1848 as it doesn not seem
to be needed.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Reviewed-by: Rene Herman <rene.herman@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The snd_wss is superset of the snd_ad1848 so kill
the latter and replace it with the snd_wss.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Reviewed-by: Rene Herman <rene.herman@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Rename functions and structures from the former
cs4321_lib to names more corresponding with the
new name: wss_lib.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Reviewed-by: Rene Herman <rene.herman@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Rename file include/sound/cs4231.h
into include/sound/wss.h
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Reviewed-by: Rene Herman <rene.herman@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Move the file sound/isa/cs423x/cs4231_lib.c
into sound/isa/cs423x/wss_lib.c
This is the first step toward merging all libraries
for Windows Sound System compatible chips
into a single library.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Reviewed-by: Rene Herman <rene.herman@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
snd_seq_oss_synth_make_info() incorrectly reports information
to userspace without first checking for the validity of the
device number, leading to possible information leak (CVE-2008-3272).
Reported-By: Tobias Klein <tk@trapkit.de>
Acked-and-tested-by: Takashi Iwai <tiwai@suse.de>
Cc: stable@kernel.org
Signed-off-by: Willy Tarreau <w@1wt.eu>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
If an OSS application calls SNDCTL_DSP_SYNC, then ALSA will call the driver's
_hw_params and _prepare functions again. On the Freescale MPC8610 DMA ASoC
driver, this caused the DMA controller to be unneccessarily re-programmed, and
apparently it doesn't like that. The DMA will then not operate when
instructed. This patch relocates much of the DMA programming to
fsl_dma_open(), which is called only once.
Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: ASoC: Export dapm_reg_event() fully
ALSA: ASoC: Update Poodle to current ASoC API
ALSA: asoc: restrict sample rate and size in Freescale MPC8610 sound drivers
ALSA: sound/soc/pxa/tosa.c: removed duplicated include
When compiled with CONFIG_SND_DYNAMIC_MINORS the ALSA core is fine
to have more than 8 PCM devices per card, except one place - the
SNDRV_CTL_IOCTL_PCM_NEXT_DEVICE ioctl, which will not enumerate
devices > 7. This patch fixes the issue, changing the devices list
organisation.
Instead of adding new device to the tail, the list is now kept always
ordered (by card number, then device number). Thus, during enumeration,
it is easy to discover the fact that there is no more given card's
devices.
Additionally the device field of struct snd_pcm had to be changed to int,
as its "unsignednity" caused a lot of problems when comparing it to
potentially negative signed values. (-1 is 0xffffffff or even more then ;-)
Signed-off-by: Pawel Moll <pawel.moll@st.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Added a new US122L usb-audio driver. This driver works together with a
dedicated alsa-lib plugin.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This reverts commit fb3d6f2b77bdec75d45aa9d4464287ed87927866.
New, updated patch with same subject replaces this commit.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
When compiled with CONFIG_SND_DYNAMIC_MINORS the ALSA core is fine
to have more than 8 PCM devices per card, except one place - the
SNDRV_CTL_IOCTL_PCM_NEXT_DEVICE ioctl, which will not enumerate
devices > 7. This patch fixes the issue, changing the devices list
organisation.
Instead of adding new device to the tail, the list is now kept always
ordered (by card number, then device number). Thus, during enumeration,
it is easy to discover the fact that there is no more given card's
devices. The same limit was present in OSS emulation code. It has
been fixed as well.
Additionally the device field of struct snd_pcm is now int, instead of
unsigned int, as there is no obvious reason for keeping it unsigned.
This caused a lot of problems with comparing this value with other
(almost always signed) variables. There is just one more place where
device number is unsigned - in struct snd_pcm_info, which should be
also sorted out in future.
Signed-off-by: Pawel MOLL <pawel.moll@st.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
ASOC: convert use of uint to unsigned int
Signed-off-by: Jon Smirl <jonsmirl@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Allow all the CODEC drivers to be built without a machine driver in order
to facilitate testing of subsystem-wide changes and gain better coverage
from automated testing efforts. This also helps things like the generic
OpenFirmware machine driver for PowerPC.
Currently AC97 CODECs are not included since the current setup relies
on having a controller driver available.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Grant Likely <grant.likely@secretlab.ca>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This fixes sparse warnings and allows non-OpenFirmware systems to attempt
to bind to the device.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Grant Likely <grant.likely@secretlab.ca>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
ASoC Codec driver for the TLV320AIC26 device. As it stands, this driver
doesn't support all the modes and clocking options of the AIC16, but it
is a start.
Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This is an I2S bus driver for the MPC5200 PSC device. It depends on the
soc-of helper functions to match a PSC device with a codec based on data
in the device tree.
Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Simple utility layer for creating ASoC machine instances based on data
in the OpenFirmware device tree. OF aware platform drivers and codec
drivers register themselves with this framework and the framework
automatically instantiates a machine driver. At the moment, the driver
is not very capable and it is expected to be extended as more features
are needed for specifying the configuration in the device tree.
This is most likely temporary glue code to work around limitations in
the ASoC v1 framework. When v2 is merged, most of this driver will
need to be reworked.
Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Most of the ASoC controls refer to the maximum value that can be set for
a control as mask but there is no actual requirement for all bits to be
set at the highest possible value making the name mask misleading.
Change the code to use max instead.
Signed-off-by: Jon Smirl <jonsmirl@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Convert bitfields in ASoC into full int width. This is a
simple mechanical conversion. Two places in the DAPM code
were fixed to properly use mask.
Signed-off-by: Jon Smirl <jonsmirl@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Some codecs have unusual features in their register maps such as very
large registers representing arrays of coefficients. Support these
codecs in the register cache sysfs file by allowing them to provide a
function register_display() overriding the default output for register
contents.
Also ensure that we don't overflow PAGE_SIZE while writing out the
register dump.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Use input_free_devce() correctly instead of kfree() at error path.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Since jack detection requires the input subsystem which may not be
desired on small systems it is not built unless required by a driver
that is being built. Drivers using jack detection should use a pattern
like this:
config SND_FOO
tristate "..."
...
select SND_JACK if INPUT=y || INPUT=SND
to ensure that the jack detection API is enabled if the input subsystem
is. If the input subsystem is not enabled then a stub version of the
API is provided.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Currently very few systems provide information about jack status to user
space, even though many have hardware facilities to do detection. Those
systems that do use an input device with the existing SW_HEADPHONE_INSERT
switch type to do so, often independently of ALSA.
This patch introduces a standard method for representing jacks to user
space into ALSA. It allows drivers to register jacks for a sound card with
the input subsystem, binding the input device to the card to help user
space associate the input devices with their sound cards. The created
input devices are named in the form "card longname jack" where jack is
provided by the driver when allocating a jack. By default the parent for
the input device is the sound card but this can be overridden by the
card driver.
The existing user space API with SW_HEADPHONE_INSERT is preserved.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
dapm_reg_event() is used by devices using SND_SOC_DAPM_REG() so needs to
be exported to support building them as modules and prototyped to avoid
sparse warnings and potential build issues.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Freescale MPC8610 SSI device has the option of using one clock for both
transmit and receive (synchronous mode), or independent clocks (asynchronous).
The SSI driver, however, programs the SSI into synchronous mode and then
tries to program the clock registers independently. The result is that the wrong
sample size is usually generated during recording.
This patch fixes the discrepancy by restricting the sample rate and sample size
of the playback and capture streams. The SSI driver remembers which stream
is opened first. When a second stream is opened, that stream is constrained
to the same sample rate and size as the first stream.
A future version of this driver will lift the sample size restriction.
Supporting independent sample rates is more difficult, because only certain
codecs provide dual independent clocks.
Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Removed duplicated include <asm/arch/tosa.h> in
sound/soc/pxa/tosa.c.
Signed-off-by: Huang Weiyi <weiyi.huang@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This follows the sparc changes a439fe51a1f8eb087c22dd24d69cebae4a3addac.
Most of the moving about was done with Sam's directions at:
http://marc.info/?l=linux-sh&m=121724823706062&w=2
with subsequent hacking and fixups entirely my fault.
Signed-off-by: Sam Ravnborg <sam@ravnborg.org>
Signed-off-by: Paul Mundt <lethal@linux-sh.org>
Added support for 92HD81/83 family of codecs.
This also includes a pwr_mapping array for pins that have more than
one amp to power down.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: Allow to force model to intel-mac-v3 in snd_hda_intel (sigmatel).
ALSA: cs4232: fix crash during chip PNP detection
ALSA: hda - Add automatic model setting for the Acer Aspire 5920G laptop
ALSA: make snd_ac97_add_vmaster() static
ALSA: sound/pci/azt3328.h: no variables for enums
ALSA: soc - wm9712 mono mixer
ALSA: hda - Add support of ASUS Eeepc P90*
ALSA: opti9xx: no isapnp param for !CONFIG_PNP
ALSA: opti93x - Fix NULL dereference
ALSA: hda - Added support for Asus V1Sn
ALSA: ASoC: Factor PGA DAPM handling into main
ALSA: ASoC: Refactor DAPM event handler
ALSA: ALSA: ens1370: communicate PCI device to AC97
ALSA: ens1370: SRC stands for Sample Rate Converter
ALSA: hda - Align BDL position adjustment parameter
ALSA: Au1xpsc: psc not disabled when TX is idle
ALSA: add TriTech 28023 AC97 codec ID and Wolfson 9701 name.
The type and type2 fields were unused and so could be removed.
Instead add a vfl_type field that contains the type of the video
device.
Signed-off-by: Hans Verkuil <hverkuil@xs4all.nl>
Signed-off-by: Mauro Carvalho Chehab <mchehab@infradead.org>
Currently, even if you pass model=intel-mac-v3 as a module parameter to
snd_hda_intel, the function patch_stac922x (patch_sigmatel.c) will still
try to auto-detect the model type. This is a problem on my MacBook Pro 1st
generation, which needs intel-mac-v3, but sometimes incorrectly reports
0x00000100 as subsystem id, which causes the switch in patch_stac922x to
select intel-mac-v4.
To fix this, I added a new model called intel-mac-auto, so in case no
module parameter is passed, and an Intel Mac board is detected, the
model will be automatically detected, while no detection will be done
if the model is forced to intel-mac-v3.
This problem has been around for quite a while, and I used to fix it
by moving the case statement for 0x00000100 in patch_stac922x so that
intel-mac-v3 is chosen.
Another way to fix the problem would be to check if a module parameter
was set directly in patch_stac922x, using something like this:
if (spec->board_config == STAC_INTEL_MAC_V3 &&
!codec->bus->modelname) {
But I think it is less elegant (if you prefer that way, I can prepare a
patch).
Signed-off-by: Nicolas Boichat <nicolas@boichat.ch>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The acard->wss pointer is uninitialized in this function
which leads to crash during chip PNP detection.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Acked-by: Rene Herman <rene.herman@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
