Use the fm_port and mpu_port variables directly in a probe function.
This completely eliminates a need to copy the fm_port value to
the snd_miro structure.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Update control names to be more closer to their meaning.
Change the "Mono" name to the "Beep" as this line is usually
used to forward the PC beeper signal to sound card's output.
Update names for both cs423x and wss.
Clean up cs4235 controls according to the cs4235 doc. Rename
some of the cs4235 controls to be consistent with the cs4236's
ones.
Also, delete one misnamed cs4231 register define.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
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: ice1724 - make some bitfields unsigned
ALSA: hda - Dell Studio 1557 hd-audio quirk
ALSA: ice1724 - Fix section mismatch in prodigy_hd2_resume()
ALSA: hda - Add another Nvidia HDMI codec id (10de:0005)
ALSA: hda: Use model=mb5 for MacBookPro 5,2
* 'hostprogs-wmissing-prototypes' of git://git.kernel.org/pub/scm/linux/kernel/git/josh/linux-misc:
Makefile: Add -Wmising-prototypes to HOSTCFLAGS
oss: Mark loadhex static in hex2hex.c
dtc: Mark various internal functions static
dtc: Set "noinput" in the lexer to avoid an unused function
drm: radeon: Mark several functions static in mkregtable
arch/sparc/boot/*.c: Mark various internal functions static
arch/powerpc/boot/addRamDisk.c: Mark several internal functions static
arch/alpha/boot/tools/objstrip.c: Mark "usage" static
Documentation/vm/page-types.c: Declare checked_open static
genksyms: Mark is_reserved_word static
kconfig: Mark various internal functions static
kconfig: Make zconf.y work with current bison
The ALC262 has a quirk entry matching with all Sony Vaio laptops
to use model=sony-assamd as default. But, model=auto works much better
for new models in the recent driver versions, thus it's safer to disable
that default quirk entry.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Vaio type G laptop doesn't work with the current quirk setup.
After some tests, it turned out that it should be model=auto as default.
Reported-by: Mattia Dongili <malattia@linux.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Disable the master volume control in the PCM2702 chipset.
The datasheet documents two independent channel volume controls, one
master mute control and one master volume control. All controls are
fully functional except for the master volume control, which returns
USB stalls on all GET requests.
Signed-off-by: Javier Kohen <jkohen@users.sourceforge.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the {orig,midi}_dev equals num_midis, that's one too
large already.
Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mike Rapoport <mike@compulab.co.il>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Acked-by: Steve Sakoman <steve@sakoman.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The beep_mode option value was wrongly defined: it must be 0 = off and
1 = on.
Also, evaluate the beep_mode value at snd_hda_attach_beep_device()
properly so that no device is created when beep_mode=0 is given.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds support for dynamically created controls to proc codec file
(Control: lines).
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is an initial patch to show universal control<->NID assigment in
proc codec file. The change helps to debug codec related problems.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The beep_mode parameter for snd-hda-intel module allows to choose among
different digital beep device registation to the input layer.
0 = do not register to the input layer
1 = register to the input layer all time
2 = use "Beep Switch" control exported to user space mixer applications
Also, introduce CONFIG_SND_HDA_INPUT_BEEP_MODE for default value.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The unregister work should be also canceled in snd_hda_detach_beep_device()
function.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The massive register/unregister calls for input device layer might be
overkill. Delay unregister call by one HZ as workaround.
Also, as benefit, beep->enabled variable is changed immediately now
(not from workqueue).
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Original implementation was keeping registered input device for SND_BEEP
and SND_TONE events all time. This patch changes this behaviour:
If digital PC Beep is turned off using universal control switch,
the input device is unregistered.
Explanation: The kd_mksound() send SND_BEEP and SND_TONE only to last
registered device acceping those events. It means that the HDA Intel
audio driver blocks also the internal PC Speaker device (pcspkr.c
driver) even if the HDA Beep is muted. The user can easy disable
all beeps using 'setterm -blength 0' or 'xset b off' command.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is a clean up and doesn't change the behavior.
Bit fields should always be unsigned. Otherwise pm_suspend_enabled will
be -1 when you want it to be 1. The other bad thing is that the sparse
checker will complain 36 times if they aren't unsigned.
The other bitfields in that struct are unsigned already.
Signed-off-by: Dan Carpenter <error27@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add proper suspend/resume code for Juli@ cards. Based on ice1724
suspend/resume work of Igor Chernyshev.
Fixes bug https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4413
Tested on linux-2.6.31.6
Signed-off-by: Aleksey Kunitskiy <alexey.kv@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
sound/pci/hda/patch_via.c: In function 'via_hp_bind_automute':
sound/pci/hda/patch_via.c:2074: internal compiler error: in do_SUBST, at combine.c:462
Please submit a full bug report,
with preprocessed source if appropriate.
See <URL:http://gcc.gnu.org/bugs.html> for instructions.
[added a comment by tiwai]
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The DAPM widgets are now insntantiated by the core when creating the card
so there is no need for the individual CODEC drivers to do so.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The jack_status_check callback function is the interface to check the
status of the jack. Some target provides the method to distinguish what
is the jack inserted - headphone jack, microphone jack, tvout jack, etc,
so we can implement it using the jack_status_check function.
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
While probing, AC97 codec drivers and soc-core generically execute the
following sequence:
snd_soc_new_ac97_codec -> snd_soc_new_pcms -> reset ac-link/read AC97 ID
to detect ->... -> set platform_data to ac97 by soc-core
commit 474828a40f adds platform_data to
snd_ac97 instance. But ac97 platform data hasn't given to snd_ac97
before actual ac97 operations. Then while ac97_read access platform_data
of snd_ac97 for detecting, NULL pointer oops will fire. That means old
platform_data patch doesn't work in real-life cases.
This patch moves the operation of setting ac97 platform_data earlier
than ac97 reading/writing operations. Then it makes platform_data of
AC97 become practically useful.
Signed-off-by: Barry Song <21cnbao@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Corrected the order of 'source' and 'pll_id' arguments.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Many devices need to calculate the bit clock rate desired to
work out the clock configuration required for the device.
Provide utility functions to do this using both hw_params
structures and raw numbers.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
CONFIG_SND_HDA_POWER_SAVE is independent from CONFIG_SND_HDA_HWDEP.
Thus snd_hda_hwdep_add_power_sysfs() needs the check of both kconfigs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit e330323520
"ALSA: hda - proc - show which I/O NID is associated to PCM device"
introduces the access to substream pointer. But, PCMs may have no
substreams in one or both directions, and this results in NULL
dereference. Also, print the first substream number doesn't make
sense.
This patch removes the access to the substream pointer, and reformat
to fit to the standard coding style.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://bugs.launchpad.net/bugs/462098
Until we can look closer at the verbs, let's use ALC885_MB5 for
codec SSID 0x106b4600 to enable playback and capture for MacBookPro
5,2s.
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the power on/off counter and expose via sysfs files.
The sysfs files, power_on_acct and power_off_acct, are created under
each codec hwdep sysfs directory (e.g. /sys/class/sound/hwC0D0).
The files show the msec length of the codec power-on and power-off,
respectively.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The test of index `i' is after the read - too late - and
unsafe: if snd_hda_get_connections() fails in the last
iteration a read beyond the array is possible.
Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Record the pid of the task that opened a RawMIDI substream.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Record the pid of the task that opened a PCM substream. For sound
cards with hardware mixing, this allows determining which process
is associated with a specific substream's volume control.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The substream_opened field is to count the number of opened substreams,
not the number of times that any substreams have been opened.
Furthermore, all substreams being opened does not imply that the next
open would fail, due to the possibility of O_APPEND. With this wrong
check, opening a substream multiple times would succeed only if the
device had more unopened substreams.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit 9a1b64caac in 2.6.30 broke the
error handling code in rawmidi_open_priv().
If only the output substream of a RawMIDI device has been opened and
if this device is then opened with O_RDWR | O_APPEND and if the
initialization of the input substream fails (either because of low
memory or because the device driver's open callback fails), then the
runtime structure of the already open output substream will be freed
and all following writes through the first handle will cause
snd_rawmidi_write() to use the NULL runtime pointer.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit 9a1b64caac in 2.6.30 dropped the
check that a substream must already have been opened with O_APPEND to be
able to open it a second time.
This would make it possible for a substream to be switched to append
mode, which would mean that non-atomic writes would fail unexpectedly.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit 9a1b64caac in 2.6.30 moved the
substream initialization code to where it would be executed every time
the substream is opened.
This had the consequence that any further opening would drop and leak
the data in the existing buffer, and that the device driver's open
callback would be called multiple times, unexpectedly.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The present quirk for HP dc5750 seems broken and maps the pins wrongly.
Since the auto-parser works well for this device, set the default entry
to use model=auto.
Reference: Novell bnc#552154
https://bugzilla.novell.com/show_bug.cgi?id=552154
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add reboot notifier to each codec so that it can do some workarounds
needed for reboot.
So far, patch_sigmatel.c calls its shutup routine for avoiding noises
at reboot on some HP machines.
References: Novell bnc#544779
http://bugzilla.novell.com/show_bug.cgi?id=544779
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The TX and RX irq handlers are identical. Merge them
Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
APLL_CTL register is configured by the twl4030-codec MFD
driver.
Remove code, which makes changes in the APLL_CTL register,
and replace those with checks against the configured
audio_mclk configuration done in the MFD driver.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch increases the number of supported audio channels from 4
to 16 and has been sponsored by Shotspotter Inc. It also fixes a
FSYNC rate calculation bug when McBSP is FSYNC master.
Signed-off-by: Graeme Gregory <gg@slimlogic.co.uk>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Tested-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Our contacts at Conexant suggested that we reduce the external
microphone bias to 50% in order to center the input signal with
the DC input range of the codec. This is because the microphone
port is DC coupled for potential use with sensors.
Signed-off-by: Daniel Drake <dsd@laptop.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://bugs.launchpad.net/bugs/478309
The internal microphone on this VAIO model does not work unless the
"auto" quirk is used.
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Upcoming change to omap-mcbsp.c require that machine drivers using OMAP
as a DAI master to pass sample rate generator input clock frequency to
the omap-mcbsp.c DAI driver.
Pandora is using 256*Fs output from the TWL4030 codec as an input clock to
the McBSP sample rate generator.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Tested-by: Grazvydas Ignotas <notasas@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Convert PCMCIA drivers to use the dynamic debug infrastructure, instead of
requiring manual settings of PCMCIA_DEBUG.
Also, remove all usages of the CS_CHECK macro and replace them with proper
Linux style calling and return value checking. The extra error reporting may
be dropped, as the PCMCIA core already complains about any (non-driver-author)
errors.
CC: Jaroslav Kysela <perex@perex.cz>
CC: alsa-devel@alsa-project.org
Signed-off-by: Dominik Brodowski <linux@dominikbrodowski.net>
1. Set the third argument of the snd_device_new to not NULL, so there is
no warning about bug during chip detection. The third argument is not
used in this driver. It was changed in my previous patch.
2. Remove the fm_port and mpu_port fields from the snd_es18xx structure.
They can be converted to function arguments.
3. Remove the dmaN_size fields from the snd_es18xx structure. These
values are used only in pointer functions and can be easily calculated.
4. Remove the ctrl_lock spinlock which is used only in one read function
which is called once during chip initialization. There are many
writes to the same register and they are not protected on purpose
(see the comment ina the snd_es18xx_config_write()).
5. Use the first part of the text5Sources string table as the text4Soruces
table (they are the same).
6. Merge the same cases for the ES1887 and ES1888 when setting chip's caps.
7. Move the snd_es18xx_reset() to __devinit section.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The MPC5200 AC97 driver is disabling the slots when a stop
trigger is received, but not reenabling them if the stream
is started again without processing the hw_params again.
This patch fixes the problem by caching the slot enable bit
settings calculated at hw_params time so that they can be
reapplied every time the start trigger is received.
Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Move the resolving of the psc_dma_stream pointer to a helper function
to reduce duplicate code
Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Sound drivers PCM DMA is supposed to free-run until told to stop
by the trigger callback. The current code tries to track appl_ptr,
to avoid stale buffer data getting played out at the end of the
data stream. Unfortunately it also results in race conditions
which can cause the audio to stall.
Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
All DMA blocks are lined up to period boundaries, but the DMA
handling code tracks bytes instead. This patch reworks the code
to track the period index into the DMA buffer instead of the
physical address pointer. Doing so makes the code simpler and
easier to understand.
Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
So far, CORB/RIRB still remains even if the driver is switched to the
single_cmd mode. The specification says that this should be disabled,
but I hoped this isn't the case; indeed most devices worked together with
CORB/RIRB.
However, Poulsbo (US15W) seems problematic with this setup, and it
requires to disable CORB/RIRB when single_cmd is used.
Now this patch disables CORB/RIRB initialization when the single_cmd
mode is used. Also the unsolicited event is disabled because it can't
work without RIRB.
Reported-and-tested-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix combine_word problem where first octet is not
read properly. The only affected place seems to be the
INPUT_TERMINAL type. Before now, sound controls can be created
with the output terminal's name which is a fallback mechanism
used only for unknown input terminal types. For example,
Line can wrongly appear as Speaker. After the change it
should appear as Line.
The side effect of this change can be that users
can expect the wrong control name in their scripts or
programs while now we return the correct one.
Probably, these defines should use get_unaligned_le16 and
friends.
Signed-off-by: Julian Anastasov <ja@ssi.bg>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
SPIN_LOCK_UNLOCKED is deprecated. Use __SPIN_LOCK_UNLOCKED instead.
Signed-off-by: Thomas Gleixner <tglx@linutronix.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some laptops cause annoying clicks or noises at shutdown/reboot since
the speaker pin is set still high. Apply the same procedure used for
the suspend to avoid such clicks/noises for freeing the codec, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the missing clk_enable after acquiring the 'audio-bus' clock.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
After DMA burst mode has been introduced in sound/soc/omap/omap-pcm.c,
omap_pcm_prepare() unconditionally calls:
omap_set_dma_src_burst_mode(prtd->dma_ch, OMAP_DMA_DATA_BURST_16);
omap_set_dma_dest_burst_mode(prtd->dma_ch, OMAP_DMA_DATA_BURST_16);
Current implementation of those two functions found in
arch/arm/plat-ompa/dma.c doesn't support OMAP_DMA_DATA_BURST_16 on OMAP1 at
all, so they both end with BUG() on that machine. That results in
ASoC being completely unusable, at least on my OMAP5910 based Amstrad Delta.
The patch corrects the problem by not calling those two functions when run on
OMAP1 class based machines.
Created against linux-2.6.32-rc5.
Tested on Amstrad Delta.
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Instead of storing the PID number, take a reference to the task's pid
structure. This protects against duplicates due to PID overflows, and
using pid_vnr() ensures that the PID returned by snd_ctl_elem_info() is
correct as seen from the current namespace.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We do not need to save the ID of the process that locked a control
because that information is already available in the owner's file data.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Function hp_bseries_system() is always used, outside of
CONFIG_ boundaries/controls, so move it.
sound/pci/hda/patch_sigmatel.c:5458: error: implicit declaration of function 'hp_bseries_system'
Signed-off-by: Randy Dunlap <randy.dunlap@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The cs4236 was two step detection with call to the snd_wss_free()
between two steps. The snd_wss_free() did not free a sound device
created in the snd_wss_create(). This caused an OOPS during module
removal as the same sound device was released twice. The same OOPS
happened if the cs4236 module loading failed.
Fix this by adapting the snd_cs4236_create() to correctly work with
chips less capable then cs4236. The snd_cs4236_create() behaves the
same as the snd_wss_create() if the chip is less capable than the cs4236.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
To unify control names, rename "PC Speaker" to "Speaker" for PPC ALSA drivers.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
To avoid confusion in control names for the standard analog PC Beep generator
using a small Internal PC Speaker, rename all related "PC Speaker" and "PC
Beep" controls to "Beep" only. This name is more universal and can be also
used on more platforms without confusion.
Introduce also "Internal Speaker" in ControlNames.txt for systems with
full-featured build-in internal speaker.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://bugs.launchpad.net/bugs/474972
This Sony model needs External Amplifier muted for audible playback, so
make sure we set the inv_eapd quirk.
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds support for the TI ADS117x family of multichannel ADCs
and was sponsored by Shotspotter Inc.
Signed-off-by: Graeme Gregory <gg@slimlogic.co.uk>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The XO-1.5 laptop now has a unique subvendor/subproduct ID, which can
be used to automatically select the correct CXT5066 configuration.
Signed-off-by: Daniel Drake <dsd@laptop.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is a port of the sound/oss/sh_dac_audio.c driver.
The driver uses an on-chip 8-bit D/A converter, which has a speaker connected
to one of its channels, found in several ancient HP machines.
For interrupts it uses a high-resolution timer (hrtimer).
Tested on SH7709 based hp6xx (HP Jornada 680/690 and HP Palmtop 620lx/660lx).
Also, since OSS Emulation works, the old OSS sound/oss/sh_dac_audio.c driver
would be obsolete soon, and it could be removed.
Signed-off-by: Rafael Ignacio Zurita <rizurita@yahoo.com>
Acked-by: Paul Mundt <lethal@linux-sh.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch enables GPIO to control mute LED indicator on the HP systems
with the special string in BIOS and applies it with the correct polarity on
HP B-series systems.
It also restores configuration of the pin intended as the second Headphone
on HP B-series systems but configured as something else in the BIOS to
pass MS DTM.
Signed-off-by: Vitaliy Kulikov <Vitaliy.Kulikov@idt.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_soc_init_card() is always called as the last part of the CODEC probe
function so we can factor it out into the core card setup rather than
have each CODEC replicate the code to do the initialiastation. This will
be required to support multiple CODECs per card.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
A fairly hefty change in diff terms but no actual code changes, will be
used by the next commit.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* 'fixes-s3c-2632-rc5' of git://git.fluff.org/bjdooks/linux:
ARM: S3C2410: Fix sparse warnings in arch/arm/mach-s3c2410/gpio.c
ARM: S3C2440: mini2440: Fix spare warnings
ARM: S3C24XX: Fix warnings in arch/arm/plat-s3c24xx/gpio.c
ARM: S3C2440: mini2440: Fix missing CONFIG_S3C_DEV_USB_HOST
ARM: S3C24XX: arch/arm/plat-s3c24xx: Move dereference after NULL test
ARM: S3C: Fix adc function exports
ARM: S3C2410: Fix link if CONFIG_S3C2410_IOTIMING is not set
ARM: S3C24XX: Introduce S3C2442B CPU
ARM: S3C24XX: Define a macro to avoid compilation error
ARM: S3C: Add info for supporting circular DMA buffers
ARM: S3C64XX: Set rate of crystal mux
ARM: S3C64XX: Fix S3C64XX_CLKDIV0_ARM_MASK value
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: hda - Don't check invalid HP pin
ALSA: dummy - Fix descriptions of pcm_substreams parameter
ALSA: pcmcia: use dynamic debug infrastructure, deprecate CS_CHECK (sound)
ALSA: hda: Use quirk mask for Dell Inspiron Mini9/Vostro A90 using ALC268
sound: via82xx: deactivate DXS controls of inactive streams
ALSA: snd-usb-caiaq: Bump version number to 1.3.20
ALSA: snd-usb-caiaq: Lock on stream start/unpause
ALSA: snd-usb-caiaq: Missing lock around use of buffer positions
ALSA: sound/parisc: Move dereference after NULL test
ALSA: sound: Move dereference after NULL test and drop unnecessary NULL tests
ALSA: hda_intel: Add the Linux device ID for NVIDIA HDA controller
ALSA: pcsp - Fix nforce workaround
ALSA: SND_CS5535AUDIO: Remove the X86 platform dependency
ASoC: Amstrad Delta: add info about the line discipline requirement to Kconfig help text
ASoC: Fix possible codec_dai->ops NULL pointer problems
ALSA: hda - Fix capture source checks for ALC662/663 codecs
ASoC: Serialize access to dapm_power_widgets()
Set the codec->bias_level to SND_SOC_BIAS_OFF before changing
the initial bias level to STANDBY.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add support for the Wolfson Microelectronics WM8727 DAC, this is a simple
non-configurable DAC.
Signed-off-by: Neil Jones <neil.jones@imgtec.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
alc_automute_pin() might be called even if any HP pin is defined, and
it will result in verbs with NID=0.
This patch adds a check for the validity of HP widget before issuing
any verbs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Convert psc-ac97,i2s to platform drivers similar to the davinci ones.
Signed-off-by: Manuel Lauss <manuel.lauss@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Convert PCMCIA drivers to use the dynamic debug infrastructure, instead of
requiring manual settings of PCMCIA_DEBUG.
Also, remove all usages of the CS_CHECK macro and replace them with proper
Linux style calling and return value checking. The extra error reporting may
be dropped, as the PCMCIA core already complains about any (non-driver-author)
errors.
Signed-off-by: Dominik Brodowski <linux@dominikbrodowski.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://bugs.launchpad.net/bugs/368629
We should use a quirk mask for these Dell Inspiron Mini9s and Vostro
A90s, as the model=dell quirk appears to enable audio on them.
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently, if the high-res timers are unavailable, snd-pcsp does not
initialize. People who choose it over pcspkr, loose their console beeps
in that case and get annoyed.
With this patch, the console beeps remain regardless of the high-res
timers. Additionally, the "nopcm" modparam is added to forcibly
disable the PCM capabilities of the driver.
Signed-off-by: Stas Sergeev <stsp@aknet.ru>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Introduce a wrapper call snd_soc_update_bits_locked()
that will take the codec mutex. This call is used
when the codec mutex is not already taken.
Drivers calling snd_soc_update_bits() may wish to
make sure the codec mutex is taken from the driver.
Signed-off-by: Eero Nurkkala <ext-eero.nurkkala@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Remove the io_mutex. It has a drawback of serializing
all accesses to snd_soc_update_bits() even when multiple
codecs are in use. In addition, it fails to actually do
its task - during snd_soc_update_bits(), dapm_update_bits()
may also be accessing the same register which may result in
an outdated register value.
Signed-off-by: Eero Nurkkala <ext-eero.nurkkala@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When any codec communication error happens, try to switch to the polling
mode first before turning off MSI. MSI gets more stable nowadays, thus
we should keep it on as much as possible.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove intermediate snd_audiodrive structure between
snd_card structure and snd_es18xx. This reduces size of
source code and binary driver.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The snd_card pointer is redundant and code can be easily
changed to work without it.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The OSS driver for Ensoniq SoundScape cards is broken after conversion
to mutexes and a new ALSA snd-sscape driver handles all devices handled
by the OSS one.
The ALSA driver was tested with these cards:
Spea V7 MediaFX
Ensoniq Soundscape Elite
Ensoniq Soundscape VIVO (this card is not handled by the OSS driver)
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Activate the DXS volume controls only when the corresponding stream is
being used. This makes the behaviour consistent with the other drivers
that have per-stream volume controls.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added a proper ifdef CONFIG_SND_DEBUG_VERBOSE to avoid a compile warning:
sound/pci/hda/patch_intelhdmi.c:406: warning: ‘hdmi_get_channel_count’ defined but not used
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix a bug which can result in white noise from the driver after stream
start or unpause.
Signed-off-by: Mark Hills <mark@pogo.org.uk>
Acked-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix a race which causes snd_pcm_update_hw_ptr_pos() to report a bug.
Signed-off-by: Mark Hills <mark@pogo.org.uk>
Acked-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The variables are unsigned so the test `>= 0' is always true,
the `< 0' test always fails. In these cases the other part of
the test catches wrapped values.
In dac_audio_write() there does not occur a test for wrapped
values, but the test appears redundant.
Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add support for VT1818S codec, which is similiar with VT1708S.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If the NULL test on h is needed in snd_harmony_mixer_init, then the
dereference should be after the NULL test.
Actually, there is a sequence of calls: snd_harmony_create, then
snd_harmony_pcm_init, and then snd_harmony_mixer_init. snd_harmony_create
initializes h, but may indeed leave it as NULL. There was no NULL test at
the beginning of snd_harmony_pcm_init, so I have added one. The NULL test
in snd_harmony_mixer_init is then not necessary, but in case the ordering
of the calls changes, I have left it, and moved the dereference after it.
A simplified version of the semantic match that detects this problem is as
follows (http://coccinelle.lip6.fr/):
// <smpl>
@match exists@
expression x, E;
identifier fld;
@@
* x->fld
... when != \(x = E\|&x\)
* x == NULL
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In pcm.c, if the NULL test on pcm is needed, then the dereference should be
after the NULL test.
In dummy.c and ali5451.c, the context of the calls to
snd_card_dummy_new_mixer and snd_ali_free_voice show that dummy and pvoice,
respectively cannot be NULL.
A simplified version of the semantic match that detects this problem is as
follows (http://coccinelle.lip6.fr/):
// <smpl>
@match exists@
expression x, E;
identifier fld;
@@
* x->fld
... when != \(x = E\|&x\)
* x == NULL
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The attached patch fixes the problems introduced in this commit:
http://git.kernel.org/?p=linux/kernel/git/torvalds/linux-2.6.git;a=commitdiff;h=eea0579fc85e64e9f05361d5aacf496fe7a151aa
- Fix nForce workaround by honouring the pointer_update var
- Revert "ns" to u64, as per the hrtimer API
- Revert to the zero-delay timer startup, since I can't reproduce any
problem with it (please, give me the hint!)
Signed-off-by: Stas Sergeev <stsp@aknet.ru>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Intel IbexPeak HDMI codec supports 2 converters and 3 pins,
which requires converting the cvt_nid/pin_nid to arrays.
The active pin number (the one connected with a live HDMI monitor/sink)
will be dynamically identified on hotplug events.
It exports two HDMI devices, so that user space can choose the A/V pipe
for sending the audio samples.
It's still undefined behavior when there are two active monitors
connected and routed to the same audio converter.
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The new Intel HDMI codec supports 2 independant HDMI/DisplayPort pipes.
We'll be exporting them as 2 pcm devices. So bump up the allowed number
of HDMI devices.
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This unifies the code and data structure,
and makes it easy to add more HDMI devices.
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
SuperH FSI device have the hardware limitation to use DMA.
If DMA is used, LCD output will be broken.
Maybe there are some solution. But I don't know how to do it now.
This patch remove DMA support and use soft transfer.
Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
An earlier patch merely adds id for 0x80862804.
It has 2/3 cvt/pin nodes and shall be tied to the IbexPeak handler.
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
SND_CS5535AUDIO is available on Loongson(MIPS compatible) family
machines, and checked it with ARCH=x86_64, no relative compiling
warnings & errors, so, remove the platform dependency directly.
Reported-by: rixed@happyleptic.org
Acked-by: Andres Salomon <dilinger@collabora.co.uk>
Signed-off-by: Wu Zhangjin <wuzhangjin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Modifying the Kconfig and Makefile in sound/soc/omap folder
to add support for OMAP3517 / AM3517 EVM in Alsa SoC.
Signed-off-by: Anuj Aggarwal <anuj.aggarwal@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
