The errors at buffer preallocations aren't fatal and safe to ignore.
The buffer will be allocated dynamically when opened.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Check the continuity of allocated pages to reduce the BDL size as much
as possible so that it can use more than 1MB buffers.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Most hardwares have limited buffer-descriptor table length. This
also restricts the max buffer size of the sound driver.
For example, snd-hda-intel has 1MB buffer size limit, and this is
because it can have at most 256 BDL entries. For supporting larger
buffers, we need to allocate larger pages even for sg-buffers.
This patch changes the sgbuf allocation code to try to allocate
larger pages first. At each head of the allocated pages, the
number of allocated pages is stored in the lowest bits of the
corresponding entry of the table addr field. This change isn't
visible as long as the driver uses snd_sgbuf_get_addr() helper.
Also, the patch adds a new function, snd_pcm_sgbuf_get_chunk_size().
This returns the size of the chunk on continuous pages starting at
the given position offset. If the chunk reaches to a non-continuous
page, it returns the size to the boundary.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
snd_dma_alloc_pages_fallback() always tries to reduce the size in a half,
but it's not good when the given size isn't a power-of-two.
Check it first then try to align.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Clean up SG-buffer helper functions and macros. Helpers take substream
as arguments now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
It is ALSA bug #1754.
D6 bit in 02 register is PW4.
Signed-off-by: Alexander Beregalov <a.beregalov@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
ALC269 codec has a beep, but it was not used, so far.
Create a beep control appropriately.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Replace snd_opl3sa2_info_xxx() with snd_wss_info_xxx().
Drop check of card->private_data which cannot be NULL
if card is not NULL (spotted by Rene Herman).
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 specs-derived register name enums instead of open-coded numeric
values, for better understanding of things
- fix naming confusion ("gcr" was _NOT_ the GCR register stuff, but
instead the io _base_ which has multiplexed _access_ to GCR config
space, at _sub_ registers 0x08 and 0x0c)
- add FIXME comments about a race condition and MPU401 features
Signed-off-by: Andreas Mohr <andi@lisas.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
With the restructering of the indy button handling the old OSS HAL2 driver
got broken. Since there is a new ALSA driver for HAL2, the experimental
OSS driver is obsolete and will be removed by this patch.
Signed-off-by: Thomas Bogendoerfer <tsbogend@alpha.franken.de>
Cc: Ralf Baechle <ralf@linux-mips.org>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
When bdl_pos_adj=0 is given, disable the position-check and the delayed
period update mechanism. Usually bdl_pos_adj=0 is set only for the
debugging purpose on really broken hardwares. It's better to disable
the extra complexity in such a case.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Mute sound by setting mute bit without
setting volume to 0. It makes both source code
and binary shorter.
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>
Added the basic support of AD1882A codec chip.
It's almost compatible with AD1882, but with a digital mic and some
differences in connections.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Remove card pointer from the snd_opl3sa2
structure and break circular reference
snd_card->snd_opl3sa2->snd_card.
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>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Set up channel information for HDMI widgets. This will allow LPCM
with multiple channels supported on some HDMI devices.
TODO: It still doesn't check ELD and doesn't change PCM parameters
dynamically.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The dummy driver uses runtime->private_free but still frees
its pcm structures on error paths.
This is esoteric because the error paths in question are
unreachable. Thus the bug is only a problem when someone
copies this code into other drivers.
Signed-off-by: Daniel R Thompson <daniel.thompson@st.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Fixed several noise issues with DACs and ADCs on some 92HD75xxx based codecs
with certain revision id's.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Dynamically create capture mux volume controls when a output amp is detected.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Added the ignore_ctl_error parameter to enable/disable the control-error
handling for mixer interfaces. It was a hard-coded ifdef, and now you
can change it more easily.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Export HDA codec subvendor ID and revision ID to user space via the
components variable. Our alsactl utility requires these values for
the perfect hardware identification.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The widget node 0x21 should be initialized as unmuted/full (0dB)
as default. This will reduce additional manual work by user at the
first time use.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The SPDIF output is toggled at each time any SPDIF status bits are changed
because of the known problems on some codecs. But, this also results in
loosing the sync, and the problem is more obvious on HDMI output over
SPDIF. Since the toggle is necessary only for some codecs, we should
check whether this workaround is needed and skip if unnecessary.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Kill snd_assert() in other places, either removed or replaced with
if () with snd_BUG_ON().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Kill snd_assert() in sound/pci/*, either removed or replaced with
if () with snd_BUG_ON().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Kill snd_assert() in sound/isa/*, either removed or replaced with
if () with snd_BUG_ON().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Kill snd_assert() in sound/core/*, either removed or replaced with
if () with snd_BUG_ON().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
On the Freescale MPC8610, SSI1 is hard-coded to use DMA channels 0 and 1
for playback and capture, and SSI2 is hard-coded to use DMA channels 2 and 3.
This patch fixes the fabric driver so that it uses the right channels.
Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The WM8580 is an audio CODEC designed for DVD and surround sound
applications, offering three stereo DACs and a stereo ADC.
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>
Make snd_ad1848_probe() easier to follow. With an exception for only
trying once as soon as the codec is out of init (which should be all
that's needed) the detection logic should be unchanged.
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>
Use pin 37 for mono-out as default on ALC203.
Reported-by: george pee <georgepee@gmail.com>
Tested-by: george pee <georgepee@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The support for the SB Extigy's remote seems to be broken in all
recent ALSA versions, including 1.0.17. The driver detects the event
correctly, then submits a URB to query the RC code. On the Extigy, the
URB is submitted with a length of 2 bytes. My hardware, however, only
replies with 1 byte, containing the correct RC button code. The
current implementation discards this as being too short. (line 1784 of
usbmixer.c)
This patch specifies a "minimum packet length" in the remote control
configuration. I've left the values for the Audigy 2/Live! the same as
the packet length, as I'm assuming the existing code works with them.
(I don't have the hardware to confirm) This fixes the Extigy RC
support, e.g. for use with Lirc.
Signed-off-by: Phillip Michael Jordan <phil@philjordan.eu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The WM8900 is designed for portable multimedia applications requiring
low power consumption, high performance audio and a compact form factor
providing:
- 24 bit stereo ADC and DAC
- Microphone and line inputs
- Line outputs
- Class G headphone amplifier
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>
Thanks to Sistema Fenix (http://www.sistemafenix.com.br/) and CDI Brasil
(www.cdibrasil.com.br/) for sponsoring this development.
Signed-off-by: Gilberto <gilberto@sistemafenix.com.br>
Signed-off-by: Mauro Carvalho Chehab <mchehab@infradead.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This patch adds sound support for NEC Versa S9100
With it, we get sound on the internal speaker and headphone (with
automute working) while there is no sound by default.
External mic also works fine but I don't know if there is an internal
one (if there is an internal mic it does not work currently), and I
had to send back the hardware.
Signed-off-by: Pascal Terjan <pterjan@mandriva.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
After the transition from cs4321_lib to wss_lib, azt2320 probe visits
snd_ad1848_probe during detection. It expects register 0 (LEFT_INPUT)
to be able to retain the value 0xaa during detection but AZT2320 does
not support MIC gain meaning it reads back as 0x8a instead.
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>
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>
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>
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>
Added digital pc-beep support using linear tone generation for hd-codecs along
with initial support for several IDT codecs.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Correct headphone detection for 1st generation iMac G3 Slot-loading (Screamer).
This patch fixes the regression in the recent snd-powermac which
doesn't support some G3/G4 PowerMacs:
http://lkml.org/lkml/2008/10/1/220
Signed-off-by: Risto Suominen <Risto.Suominen@gmail.com>
Tested-by: Mariusz Kozlowski <m.kozlowski@tuxland.pl>
Cc: stable@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add mixer controls for PowerMac G4 AGP (Screamer).
This patch fixes the regression in the recent snd-powermac which
doesn't support some G3/G4 PowerMacs:
http://lkml.org/lkml/2008/10/1/220
Signed-off-by: Risto Suominen <Risto.Suominen@gmail.com>
Tested-by: Mariusz Kozlowski <m.kozlowski@tuxland.pl>
Cc: stable@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Rob Sims wrote:
"I can't seem to turn on register 0x17, bit 3 in the sound chip, except
by codec_reg_write; the mixer lacks direct or indirect control. It
seems there are two names for the output of the rec mixer:
Capture ST Mixer
Playback Mixer
Would the following do the trick?"
I confirm that this solves the audio problems I was having.
Signed-off-by: Jonas Bonn <jonas.bonn@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Dell Inspiron 1525 seems to have a buggy BIOS setup and screws up
the recent codec parser, as reported by Oleksandr Natalenko:
http://lkml.org/lkml/2008/9/12/203
This patch adds the working model, dell-3stack, statically.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
The error path in cs4270_probe/cs4270_remove is pretty broken:
* If cs4270_probe fails, codec is leaked.
* If snd_soc_register_card fails, cs4270_i2c_driver stays registered.
* If I2C support is enabled but no I2C device is found, i2c_del_driver
is never called (neither in cs4270_probe nor in cs4270_remove.
Fix all 3 problems by implementing a clean error path in cs4270_probe
and jumping to its labels as needed.
Signed-off-by: Jean Delvare <khali@linux-fr.org>
Acked-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Conversion to new-style i2c driver missed the error path of the
probe function. Fix it.
Signed-off-by: Jean Delvare <khali@linux-fr.org>
Cc: Timur Tabi <timur@freescale.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Update the CS4270 ALSA device driver to use the new-style I2C interface.
Starting with the 2.6.27 PowerPC kernel, I2C devices that have entries in the
device trees can no longer be probed by old-style I2C drivers. The device
tree for Freescale MPC8610 HPCD has included an entry for the CS4270 since
2.6.25, but that entry was previously ignored by the PowerPC I2C subsystem.
Since that's no longer the case, the best solution is to update the CS4270
driver to a new-style interface, rather than try to revert the behavior of
new PowerPC I2C subsystem.
Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The power_mutex lock in snd_pcm_drop may cause a possible deadlock
chain, and above all, it's unneeded. Let's get rid of it.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The PCM and rawmidi open callbacks have a lock against card->controls_list
but it takes a wrong one, card->controls_rwsem, instead of a right one
card->ctl_files_rwlock. This patch fixes them.
This change also fixes automatically the potential deadlocks due to
mm->mmap_sem in munmap and copy_from/to_user, reported by Sitsofe
Wheeler:
A: snd_ctl_elem_user_tlv(): card->controls_rwsem => mm->mmap_sem
B: snd_pcm_open(): card->open_mutex => card->controls_rwsem
C: munmap: mm->mmap_sem => snd_pcm_release(): card->open_mutex
The patch breaks the chain.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
CONFIG_AC97_BUS is used from both sound and ucb1400 drivers.
The recent change in Kconfig introduced the exclusive dependency on
CONFIG_SOUND, and disabled the ucb1400 build without sound.
This patch makes CONFIG_AC97_BUS independent.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Tested-by: Randy Dunlap <randy.dunlap@oracle.com>
pxa2xx-i2s: probe actual device and use it for clk_get call
thus fixing error during startup hook
Signed-off-by: Dmitry Baryshkov <dbaryshkov@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the EQ distortion fix to the dell_m6_core_init.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When changing the sample rate, the CMI8788's master clock output becomes
unstable for a short time. The AK4396 needs the master clock to do SPI
writes, so writing to an AK4396 control register directly after a sample
rate change will garble the value. In our case, this leads to the DACs
being misconfigured to I2S sample format, which results in a wrong
output level and horrible distortions on samples louder than -6 dB.
To fix this, we need to wait until the new master clock signal has
become stable before doing SPI writes.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This reverts commit 3e0e469fa2.
The patch introduced a wrong detection of other intel Macs with
ALC88* codec because they share the same PCI SSID (but have different
codec subsystem-IDs). See http://lkml.org/lkml/2008/8/24/143
Reported-and-tested-by: Guillaume Chazarain <guichaz@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Thanks to Felipe Balbi <felipe.balbi@nokia.com> by noticing that if clk_get
to sys_clkout2_src fails, then n810_snd_device is never released.
Add also sys_clkout2_src release into error path, error code return and
release the clocks at exit.
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Xonar DX does not have CD Capture controls, so we have to check that
a control actually exists before muting it.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@kernel.org>
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: Fix double free and memory leak in many codec drivers
ALSA: CA0106 on MSI K8N Diamond PLUS Motherboard
Many SoC audio codec drivers have improper freeing of memory in error
paths.
* codec is allocated in the platform device probe function, but is not
freed there in case of error. Instead it is freed in the i2c device
probe function's error path. However the success or failure of both
functions is not linked, so this could result in a double free (if
the platform device is successfully probed, the i2c device probing
fails and then the platform driver is unregistered.)
* codec->private_data is allocated in many platform device probe
functions but not freed in their error paths.
This patch hopefully solves all these problems.
Signed-off-by: Jean Delvare <khali@linux-fr.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Correct a previous patch for the ca0106 onboard the MSI K8N Diamond PLUS
motherboard. Confirmed to have Line/Mic/Aux working for input, and sound
output working as expected.
Signed-off-by: Travis Place <wishie@wishie.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch lets the files using linux/version.h match the files that
#include it.
Signed-off-by: Adrian Bunk <bunk@kernel.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
Using init_hook to call alc888_coef_init() is problematic for configurations
that already set another init_hook. Better to put it in alc_init() as is
(although it looks a bit hackish).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Enable the snoop for nvidia hda controller to avoid data coherence issue.
Signed-off-by: Peer Chen <peerchen@gmail.com>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On some Realtek codecs like ALC882 or ALC883, the capture source is
no mux but sum widget. We have to initialize all channels properly
for this type, otherwise noises may come in from the unused route.
The patch assures to mute unused routes, and unmute the currently
selected route.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Tested-by: Daniel J Blueman <daniel.blueman@gmail.com>
The latest revisions of the WM8990 provide a programmable gain amplifier
for the speaker - configure the register cache and implement controls
for this. Older revisions of the device ignore writes to these controls.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add support for the Asus Xonar D1. It is the same as the DX, but
without the external power detection.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add missing output VREF. After a65f0568f6
it's critical, since it makes chip routing initialisation to fail.
Signed-off-by: Dmitry Baryshkov <dbaryshkov@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Correct route name to be MONO1 instead of MONO to follow
recent fix in wm8750.
Signed-off-by: Dmitry Baryshkov <dbaryshkov@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since first commit wm8750 contained output named MONO, but
all routes mentioned MONO1. Correct MONO to be MONO1.
Signed-off-by: Dmitry Baryshkov <dbaryshkov@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.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: fix SNDCTL_DSP_SYNC support in Freescale 8610 sound drivers
Remove includes of asm/hardware.h in addition to asm/arch/hardware.h.
Then, since asm/hardware.h only exists to include asm/arch/hardware.h,
update everything to directly include asm/arch/hardware.h and remove
asm/hardware.h.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
There are 43 includes of asm/mach-types.h by files that don't
reference anything from that file. Remove these unnecessary
includes.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
This reverts commit 680db0136e. The label
is actually used, but hidden behind CONFIG_SND_DEBUG and the horrible
snd_assert() macro.
That macro could probably be improved to be along the lines of
#define snd_assert(expr, args...) do { if ((void)(expr),0) { args; } } while (0)
or similar to make sure that we always both evaluate 'expr' and parse
'args', but while gcc should optimize it all away, I'm too lazy to
really verify that. So I'll just admit defeat and will continue to live
with the annoying warning.
Noted-by: Robert P. J. Day <rpjday@crashcourse.ca>
Signed-off-by: Linus "Grr.." Torvalds
This fixes the warning
sound/core/pcm_native.c: In function 'snd_pcm_fasync':
sound/core/pcm_native.c:3262: warning: label 'out' defined but not used
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
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
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 a439fe51a1.
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>
* '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>
Make the Acer Aspire 5920G (1025:0121) select ALC883_ACER_ASPIRE
by default.
Signed-off-by: Travis Place <wishie@wishie.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch makes the needlessly global snd_ac97_add_vmaster() static.
Signed-off-by: Adrian Bunk <bunk@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
AZF_FREQUENCIES and AZF_GAME_CONFIGS were variables, and this doesn't
seem to have been intended.
Signed-off-by: Adrian Bunk <bunk@kernel.org>
Acked-by: Andreas Mohr <andi@lisas.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The driver is gone for a long time.
Reported-by: Robert P. J. Day <rpjday@crashcourse.ca>
Signed-off-by: Adrian Bunk <bunk@kernel.org>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
SOUND_TRIDENT was the last PCI OSS driver, and since there's already an
ALSA driver for the same hardware we can remove it.
[muli@il.ibm.com: update CREDITS]
Signed-off-by: Adrian Bunk <bunk@kernel.org>
Signed-off-by: Muli Ben-Yehuda <muli@il.ibm.com>
Signed-off-by: Muli Ben-Yehuda <muli@il.ibm.com>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
On 32-bit architectures PAGE_ALIGN() truncates 64-bit values to the 32-bit
boundary. For example:
u64 val = PAGE_ALIGN(size);
always returns a value < 4GB even if size is greater than 4GB.
The problem resides in PAGE_MASK definition (from include/asm-x86/page.h for
example):
#define PAGE_SHIFT 12
#define PAGE_SIZE (_AC(1,UL) << PAGE_SHIFT)
#define PAGE_MASK (~(PAGE_SIZE-1))
...
#define PAGE_ALIGN(addr) (((addr)+PAGE_SIZE-1)&PAGE_MASK)
The "~" is performed on a 32-bit value, so everything in "and" with
PAGE_MASK greater than 4GB will be truncated to the 32-bit boundary.
Using the ALIGN() macro seems to be the right way, because it uses
typeof(addr) for the mask.
Also move the PAGE_ALIGN() definitions out of include/asm-*/page.h in
include/linux/mm.h.
See also lkml discussion: http://lkml.org/lkml/2008/6/11/237
[akpm@linux-foundation.org: fix drivers/media/video/uvc/uvc_queue.c]
[akpm@linux-foundation.org: fix v850]
[akpm@linux-foundation.org: fix powerpc]
[akpm@linux-foundation.org: fix arm]
[akpm@linux-foundation.org: fix mips]
[akpm@linux-foundation.org: fix drivers/media/video/pvrusb2/pvrusb2-dvb.c]
[akpm@linux-foundation.org: fix drivers/mtd/maps/uclinux.c]
[akpm@linux-foundation.org: fix powerpc]
Signed-off-by: Andrea Righi <righi.andrea@gmail.com>
Cc: <linux-arch@vger.kernel.org>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
