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>
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>
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>
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>
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>
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>
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>
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>
* '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()
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 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>
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>
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>
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>
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>
The S3C64XX DMA implementation will work a lot better with the ability
to enqueue circular buffers as the hardware can do it's own linked-list
management.
Add a function s3c_dma_has_circular() to show that the system can do this
and a flag for the channel.
Update the s3c24xx/s3c64xx I2S DMA code to deal with this.
Signed-off-by: Ben Dooks <ben@simtec.co.uk>
Signed-off-by: Ben Dooks <ben-linux@fluff.org>
Acked-by: Mark Brown <broonie@@opensource.wolfsonmicro.com>
We can replace PPC32 || PPC64 as a dependancy with just PPC as all
powerpc platforms (32-bit and 64-bit) define PPC now.
Signed-off-by: Kumar Gala <galak@kernel.crashing.org>
Signed-off-by: Benjamin Herrenschmidt <benh@kernel.crashing.org>
I thought it could be usefull to add some information on how to get the device
fully supported by loading a line discipline on the modem line.
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some codec DAIs like stac9766, wm9712, wm9713, ad1980 don't register themselves
then it loses to the chance to be given a null_dai_ops in snd_soc_register_dai
if they have no ops. When functions like soc_pcm_open, soc_pcm_hw_params etc.
access the ops field in these DAIs, panic will happen.
Signed-off-by: Barry Song <21cnbao@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The ALC662/663 parser calls wrongly alc880_auto_create_input_ctls()
to check the capture source selections. This should be alc882, instead.
Reference: Novell bnc#546918
http://bugzilla.novell.com/show_bug.cgi?id=546918
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the commit f0613d5752
ALSA: hda - Add full rates/formats support for Nvidia HDMI
the flag LIMITIED_RATE_FMT_SUPPORT was set as default, as I forgot
to clear before commit.
Let's enable all formats/rates as default.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
After a reboot on an ARM1176 which amounts to a softreset, it has been
noted that the ALSA driver does not get registered and the probe fails
with the error "aaci-pl041 fpga:04: ac97 read back fail". In the process
of reading from a register the SL1TxBusy bit is set indicating that the
transceiver is busy and remains so until the default timeout occurs.
Set the Power down register 0x26 to an arbitrary value as specified in
the PL041 manual (page: 3-18) so that AACISL1TX/AACISL2TX registers take
their default state.
Signed-off-by: Philby John <pjohn@in.mvista.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The volume-knob widget needs to be set with 0x7f instead of 0xff
for Dell laptops with STAC9228 codec, too, like the previous commit.
Reference: Novell bnc#545013
http://bugzilla.novell.com/show_bug.cgi?id=545013
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On FSC laptops, the sound gets muted gradually when the volume is chnaged.
This is due to the wrong volume-knob widget setup. The delta bit (bit 7)
shouldn't be set for these devices.
This patch adds a new quirk to set the value 0x7f to the widget 0x24
instead of 0xff.
Reference: Novell bnc#546006
http://bugzilla.novell.com/show_bug.cgi?id=546006
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Access to damp_power_widgets() is assumed to be single-threaded.
Concurrent accesses to dapm_power_widgets() may result in
unpredictable behavior.
Calls from:
close_delayed_work()
soc_codec_close()
soc_pcm_prepare()
soc_suspend()
soc_resume_deferred()
to snd_soc_dapm_stream_event() do not have the codec->mutex
taken to cover the call to dapm_power_widgets(). Thus, take
the mutex in these paths also to assure single-threaded use
of dapm_power_widgets().
Signed-off-by: Eero Nurkkala <ext-eero.nurkkala@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
snd-usb-us122l: corrent error number for not probing US-144 on ehci-hcd
This is the correct error number for telling the USB system that this
driver is not for the device.
Signed-off-by: Tobias Hansen <Tobias.Hansen@physik.uni-hamburg.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALC861-VD lenovo model causes overflow of spec->init_verbs entries due to
the recent changes. Simply increase the array size to avoid the overflow.
Reported-by: Luca Tettamanti <kronos.it@gmail.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>
If two streams are started immediately after one another (such as a
playback and a recording stream), the call to set hw params fails with
EBUSY. This patch makes the call succeed, so playback and recording will
work properly.
Signed-off-by: David Henningsson <launchpad.web@epost.diwic.se>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The chipsets with the isa_dma_bridge_buggy set do not stop DMA during
DMA counter reads. The DMA counter is read in two 8-bit read steps
on x86 platform. Sometimes, such reads happen during higher byte
change so the lower byte is already decremented (rolled over) but
the higher byte is not. It introduces an error that position is
moved 256 bytes ahead of the true position. Thus, the next DMA
position read can return a lower value then the previous read.
If the DMA position is decreased (reversed) the ALSA subsystem is
tricked into the playback underrun error and resets the playback.
It results in a "pop" during a playback.
Work around the issue by reading the counter twice and choosing a higher
value.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The C4231 control set is a superset of the AD1848 control
set so reuse the CS4231 controls definitions for the AD1848.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Convert CS4231 mixer to dB scale after AD1848 mixer.
Also, add missing microphone boost control for the AD1848
and correct wrong bits for loopback volume on the AD1848.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix coding style errors in the driver.
Also, add missing argument for CMD_XXX_MIDI_VOL command.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the num_total_dacs setting for Chaintech AV710. The existing comment
that only PSDOUT0 is connected is correct, but since the card is using
packed AC97 mode to send 6 channels to the codec, num_total_dacs should be
set to 6 and not 2. This allows 6-channel surround to work. Also clarify
a comment regarding the additional WM8728 codec on this card (it's connected
to the SPDIF output and always receives the same data).
Signed-off-by: Robert Hancock <hancockrwd@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Allow Nvidia HDMI to support more possible sample rates and formats.
At best, the really supported rates and formats should be determined
together with the negotiation with the HDMI receiver, but it's currently
not implemented yet (Nvidia stuff seems incompatible with HDMI 1.3
standard in this regard). As a compromise, we enable all bits, assuming
that all recent devices do support such rates/formats.
Tested-by: Alan Alan <alanwww1@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Increase the default and maximum PCM buffer prellocation size for ice1724's
SPDIF and independent stereo pair outputs to 256K, which is the hardware's
maximum supported size. This allows a reduction in interrupt rate and
potentially power usage when an application is not latency-critical.
Signed-off-by: Robert Hancock <hancockrwd@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* PLEASE NOTE - this change requires the corresponding update of
envy24control for ice1712 - kind of an ABI change.
* The "Multi Track Peak" control is read-only level meters indicator.
* The control is VERY confusing to most users since it is currently displayed
in regular mixers. E.g. alsamixer ignores its read-only status
and allows changing the levels with keys which makes no sense.
Signed-off-by: Pavel Hofman <pavel.hofman@ivitera.com>
Acked-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since patch_alc268() doesn't call set_capture_mixer() (due to its h/w
design different from other siblings), it needs to call fixup_automic_adc()
explicitly to set up the auto-mic routing. Otherwise the indices for
int/ext mics aren't set properly.
Reference: Novell bnc#544899
http://bugzilla.novell.com/show_bug.cgi?id=544899
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The "VIA DXS" controls are actually volume controls that apply to the
four PCM substreams, so we better indicate this connection by moving the
controls to the PCM interface.
Commit b452e08e73 in 2.6.30 broke the
restoring of these volumes by "alsactl restore" that most distributions
use; the renaming in this patch cures that regression by preventing
alsactl from applying the old, wrong volume levels to the new controls.
http://bugzilla.kernel.org/show_bug.cgi?id=14151http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=532613
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
alc_subsystem_id() tries to pick up a headphone pin if not configured,
but this caused side-effects as the problem in commit
15870f05e9.
This patch fixes the driver behavior to pick up invalid HP pins; at least,
the pins that are listed as the primary outputs aren't taken any more.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ASUS A7K needs additional GPIO1 bit setup; it has to be cleared.
Added a new fixup hook for this laptop so that it works as is.
Refernece: Novell bnc#494309
http://bugzilla.novell.com/show_bug.cgi?id=494309
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Module parameters shouldn't be marked as __devinitdata since they can be
referred via sysfs even after probing.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Old Soundscape cards (pre PnP) work only with AD1848 codecs.
If the CS4231 codec is installed it must be used in AD1848
compatible mode.
Also, add gameport support and remove an unused mpu field.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent auto-parser doesn't work for machines with a single output
with ALC861, such as Toshiba laptops, because alc_subsystem_id() sets
the hp_pins[0] while it's listed in line_outs[0].
This ends up with the doubled initialization of the same mixer widget,
and it mutes the DAC route because hp_pins has no DAC assigned.
To fix this problem, just check spec->autocfg.hp_outs and speaker_outs
so that they are really detected pins.
Reference: Novell bnc#544161
http://bugzilla.novell.com/show_bug.cgi?id=544161
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There is no sense to limit open MIDI connections with limit
as high as ULONG_MAX.
Also, convert more messages to use the snd_printk.
Correct few old and misleading comments as well.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix for typo in commit 8d50e447d1
ASoC: Factor out I/O for Wolfson 8 bit data 16 bit register CODECs
Signed-off-by: Jonathan Cameron <jic23@cam.ac.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Don't pass the advanced position to strlcat() but just gives the buffer
head position so that the max size limit can be checked correctly.
Introduced a new helper function to standaralize strlcat() calls.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since the SND_SOC_DAPM_LINE can be input or output, additional check is
needed in order to determine if the widget is connected as input or
output.
When checking for connected outputs, if the widget is line, than check
if the sources list is not empty (line is connected as output)
For input endpoint check, when the widget is line, also check if the
sinks list is not empty (line is connected as input).
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The remove callback has to be marked as __devexit, as the dynamic unbind
is possible.
Reported-by: Uwe Kleine-König <u.kleine-koenig@pengutronix.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The auto-parser for ALC662/663/272 codecs doesn't work properly when
a speaker is connected to mono NID 0x17, and doesn't handle the dynamic
DAC assignment properly.
This patch fixes the issues and also improves the assignment of DACs
so that HP and speakers can have independent volume controls.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On Soundblaster X-FI Titenium with emu20k2 the SIDE and SURROUND mute
functions are swapped.
It was checked with 'speaker-test -c 8 -s 3' and (un)mute surround or
'speaker-test -c 8 -s 7' and (un)mute side. The volume seems not
to be affected and works as expected.
Signed-off-by: Sven Eckelmann <sven.eckelmann@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
I can't see any reason for struct i2c_driver keywest_driver to not be
static.
Signed-off-by: Jean Delvare <khali@linux-fr.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://bugs.launchpad.net/bugs/410933
This Sony VAIO model also needs External Amplifier unmuted 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>
When the auto-mic switching between an analog and a digital mic is
needed with IDT codecs, the current driver doesn't reset the connection
of the digital mux.
This patch fixes the behavior by checking both mux connections properly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some of the Blackfin options don't directly follow the kconfig options
they depend on, so kconfig is unable to display the proper tree. So sort
the options such they expand/collapse properly.
Signed-off-by: Barry Song <barry.song@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Patch was tested on Toshiba NB200 and is found to enable sound.
Signed-off-by: Manoj Iyer <manoj.iyer@canonical.com>
Cc: stable@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The conversion solves the problem that firmware size was set to 64KB
while non PnP cards have 128KB firmware files.
An additional firmware initialization code has been moved from the OSS
driver.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The client->driver pointer can be NULL when i2c-device probing fails
in i2c_new_device(). This patch adds the NULL checks for client->driver
and return the error instead of blind assumption of driver availability.
Reported-by: Tim Shepard <shep@alum.mit.edu>
Cc: Jean Delvare <khali@linux-fr.org>
Cc: Johannes Berg <johannes@sipsolutions.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
