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>
Adding 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>
The pop-removal specific values are configured for TWL4030 codec
for OMAP3EVM through this patch.
Signed-off-by: Anuj Aggarwal <anuj.aggarwal@ti.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Capture path also need the APLL enabled, adding DAPM_SUPPLY
for the Virtual ADCs.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
It seams that certain part of the twl4030 codec needs the APLL
enabled before they are enabled.
Paths which has any digital processing needs need the APLL
enabled before they can function.
For example the vibra output will have some random data after
it is enabled and before the APLL also enabled.
If only analog components are in use (analog bypass), than it
seams, that the APLL does not need to be enabled. This lowers
the power consumption with around ~0.005A.
Adding DAPM_SUPPLY to the Digital playback route and also
to the capture route to enable and disable the APLL.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch fixes vibrator playing incoherently, when driven
with audio. There is something wrong in switch 3 at
H-bridge and VIBRA_SET still affects PWM generator.
Slowest value fixes things.
Signed-off-by: Jari Vanhala <ext-jari.vanhala@nokia.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The CS4270 codec features an de-emphasis filter for compensation of
audio material filtered by an 50/15 uS algorithm. Not sure whether we
should always enable it for 44100Hz sampling frequency, but it should at
least be configurable by the user.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Acked-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix up some comments, remove all enable_pin() calls (edge widgets
are all enabled by default) and mark the microphone as disabled by
default since it requires a resistor fit to connect it.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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>
codec_muted is missleading, change it to apll_enabled,
which is what it is doing: enabing and disabling the APLL.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since ASoC core now handling the codec bias differently
there is no need to do the tracking of bypass switch states
anymore.
Handling of the common bit for analog loopbacks is done with
DAPM_SUPPLY for the bypass paths.
Now this bit is only enabled when there is a complete analog
bypass path, compared to the previous implementation, when the
global switch was enabled if there were any of the analog
bypass switch was on (regardless if there were complete path or
not)
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-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>
Change the way how the twl4030 soc codec driver is
loaded/probed.
Use the device probing via tlw4030_codec MFD device.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Remove the register descriptions from the twl4030.h file and use
the linux/mfd/twl4030-codec.h instead, which has the codec
related register descriptions also.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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>
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>
Fix the definition of DAC33_LTM field, the LTM bits in
FIFO_IRQ_MODE_B register are starting at bit 6.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Hi Mark,
Here is a patch that corrects small omissions I have found in my code.
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Move the remaining headers under plat-omap/include/mach
to plat-omap/include/plat. Also search and replace the
files using these headers to include using the right path.
This was done with:
#!/bin/bash
mach_dir_old="arch/arm/plat-omap/include/mach"
plat_dir_new="arch/arm/plat-omap/include/plat"
headers=$(cd $mach_dir_old && ls *.h)
omap_dirs="arch/arm/*omap*/ \
drivers/video/omap \
sound/soc/omap"
other_files="drivers/leds/leds-ams-delta.c \
drivers/mfd/menelaus.c \
drivers/mfd/twl4030-core.c \
drivers/mtd/nand/ams-delta.c"
for header in $headers; do
old="#include <mach\/$header"
new="#include <plat\/$header"
for dir in $omap_dirs; do
find $dir -type f -name \*.[chS] | \
xargs sed -i "s/$old/$new/"
done
find drivers/ -type f -name \*omap*.[chS] | \
xargs sed -i "s/$old/$new/"
for file in $other_files; do
sed -i "s/$old/$new/" $file
done
done
for header in $(ls $mach_dir_old/*.h); do
git mv $header $plat_dir_new/
done
Signed-off-by: Tony Lindgren <tony@atomide.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>
If the NULL test on jack is needed, then the derefernce should be after the
NULL test.
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: Mark Brown <broonie@opensource.wolfsonmicro.com>
Codec read/write functions: wait 21us between the pokings of hardware.
Add timeouts to unbounded loops waiting for bits to change.
Signed-off-by: Manuel Lauss <manuel.lauss@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Verify that the correct register has been received from the codec.
Signed-off-by: Manuel Lauss <manuel.lauss@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Do not rewrite the whole register, but only update the needed
bits in set_dai_sysclk functions.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Driver for Texas Instruments TLV320DAC33 (SLAS546) low power stereo
audio DAC.
TLV320DAC33 is a stereo audio codec with integrated 24KB FIFO for low
power audio playback.
The digital interface can use I2S, DSP (A or B), Right and Left
justified formats.
DAC33 has stereo analog input, which can be bypassed to the analog
outputs.
Regarding to the internal 24KB FIFO the driver implements 'FIFO bypass'
mode (default) and nSample mode (FIFO is in use).
a) In 'FIFO bypass' mode the internal FIFO is not in use, the codec is
working synchronously as a normal codec (it needs constant stream of
data on the digital interface).
b) The nSample mode implementation uses one interrupt line from DAC33 to
the host:
Alarm threshold is set to 10ms of audio data (limit by the driver
implementation).
DAC33 will signal an interrupt, when the FIFO level goes under the
Alarm threshold.
The host will write to nSample register a value (number of stereo
samples), to tell DAC33 how many samples it should read in a burst from
the host. When the DAC33 received the number of samples, it disables the
clocks on the I2S bus. When the FIFO use again goes under the Alarm
threshold, DAC33 signals the host with an interrupt, and the process is
repeated.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Igor Grinberg <grinberg@compulab.co.il>
Signed-off-by: Mike Rapoport <mike@compulab.co.il>
CC: Mark Brown <broonie@opensource.wolfsonmicro.com>
CC: alsa-devel@alsa-project.org
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The PM core will grow pm_link infrastructure in 2.6.33 which can be
used to implement the intended functionality of the ASoC-specific
device suspend and resume callbacks so drop them.
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>
48 kHz limit is for slightly better stability, and sample rates other
than 48k (like 96k/192k) are for better sound quality.
We choose better quality, so remove the 48k limit.
Signed-off-by: Logan Li <loganli@viatech.com.cn>
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>
Remove the <plat/audio.h> include from arch/arm/plat-s3c/include/plat/audio.h
as it provides nothing to the current kernel and is not in any future plans
for the system.
Signed-off-by: Ben Dooks <ben@simtec.co.uk>
Signed-off-by: Simtec Linux Team <linux@simtec.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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>
The power for the amplifier should be handled internally
by the tpa6130a2 driver.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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>
Fixes:
sound/pci/hda/patch_via.c: In function 'patch_vt1718S':
sound/pci/hda/patch_via.c:4951: error: expected expression before 'return'
sound/pci/hda/patch_via.c: In function 'patch_vt1716S':
sound/pci/hda/patch_via.c:5441: error: expected expression before 'return'
sound/pci/hda/patch_via.c: In function 'patch_vt2002P':
sound/pci/hda/patch_via.c:5794: error: expected expression before 'return'
sound/pci/hda/patch_via.c: In function 'patch_vt1812':
sound/pci/hda/patch_via.c:6148: error: expected expression before 'return'
Signed-off-by: Stephen Rothwell <sfr@canb.auug.org.au>
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>
According to customer request, hp should be default to redirected mode,
i.e. PW4 connect select default to to MW0.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Logan Li <loganli@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
To via_control_templates[].
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Logan Li <loganli@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As init verbs, vt17xx_volume_init_verb is a better place to hold them.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Logan Li <loganli@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
With snd_hda_override_amp_caps.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Logan Li <loganli@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Rewrite nid_vol/mute assignment for clearity, and check line connection
before adding control for it.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Logan Li <loganli@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Rewrite nid_vol/mute assignment for clearity, and check line connection
before adding control for it.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Logan Li <loganli@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Replaced with via_playback_multi_pcm_prepare/cleanup to support
multi-stream operations
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Logan Li <loganli@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
like seqassoc 0xff, seqassoc 0xf0 of vt1708 should override Port
Connectivity field into 'AC_JACK_PORT_COMPLEX'
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Logan Li <loganli@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
VT1708 does not support unsolicited response, but we need hp detect to
automute speaker. Implemented in workqueue.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Logan Li <loganli@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Smart 5.1 is for 3-jacks model, to reuse input pins as outputs.
While off, they act as "line out" / "line in" / "mic in".
While on, they acts as "line out" / "back left/right" / "center/lfe".
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Logan Li <loganli@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use hp_independent_mode_index to store hp index, and simplify function
via_independent_hp_put with it.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Logan Li <loganli@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For VT1708S and VT1702, deactivate "Headphone Playback Volume" and
"Headphone Playback Mute" control if "Independent HP" mode is OFF.
and rename VT1702 "Independent HP" text.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Logan Li <loganli@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For VT1708B, VT1708S and VT1702, enter low current mode if no analog
stream is opened and all aa path mute.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Logan Li <loganli@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Enter low power state if AA-Path volume is muted.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Logan Li <loganli@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
according to customer request, VT1702 AA-Path max volume (12 dB) is too
high, so limit to 0 dB.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Logan Li <loganli@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
IS_VT17*_VENDORID macros are used nowhere, so clean them up.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Logan Li <loganli@viatech.com.cn>
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>
- Staticise ttpa6130a2_client.
- Remove unneeded cast from void.
- Use explict NULL rather than 0.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Driver for Texas Instruments TPA6130A2 stereo headphone
amplifier.
The driver provides playback gain control and also pre-defined
DAPM_HP widgets and DAPM routings for power management.
The DAPM_HP widget names are:
"TPA6130A2 Headphone Left"
"TPA6130A2 Headphone Right"
From soc machine drivers to use with the tpa6130a2 amplifier,
the tpa6130a2_add_controls has to be called, which adds the alsa
controls and the DAPM routing needed for the tpa6130a2.
After that the machine driver can connect the codec's output
with 'TPA6130A2 Left' and 'TPA6130A2 Right':
{"TPA6130A2 Left", NULL, "CODEC LEFT OUT"},
{"TPA6130A2 Right", NULL, "CODEC RIGHT OUT"},
Internally the left and right channels are powered separately.
When none of the channels are needed the amplifier is powered
down:
hard power: valid GPIO number is passed within platform data
soft power: Using the software shutdown of the amplifier
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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>
The modified revision of at91sam9g20 Evaluation Kit rev. C and onwards share
with previous ones its audio connexion to Wolfson wm8731. Modify the SoC file
to extend the machine ID checking.
Signed-off-by: Nicolas Ferre <nicolas.ferre@atmel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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>
Sometimes it is desirable to have a mux which does not reflect any
direct register configuration but which will instead only have an
effect implicitly (for example, as a result of changing which parts
of the device are powered up). Provide a virtual mux for this purpose.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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>
We don't need to check for an event callback since we also check for
an appropriate event flag when applying mux status changes.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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>
Change the way the debugfs entries are created:
If the codec->dev is valid, than use:
debugfs/asoc/{codec->name}.{dev_name(codec->dev)}/
if the codec->dev is NULL:
debugfs/asoc/{codec->name}/
as root for the debugfs entries.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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>
In order to support multiple codecs on the same system in the debugfs
the directory hierarchy need to be changed by adding directory per codec
under the asoc direcorty:
debugfs/asoc/{dev_name(socdev->dev)}-{codec->name}/codec_reg
/dapm_pop_time
/dapm/{widgets}
With the original implementation only the debugfs files are only
created for the first codec, other codecs loaded later would fail to
create the debugfs files (since they are already exist).
Furthermore in this situation any of the codecs has been removed, would
cause the debugfs entries to disappear, regardless if the codec, which
created them are still loaded (the one which loaded first).
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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>
Fix the missing argument of snd_soc_dai_set_pll() in neo1973_*.c,
which was forgotten in the commit 85488037bb.
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>
BugLink: https://bugs.launchpad.net/bugs/410933
This Sony VAIO model 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>
Fix a couple of typos and a missing header file inclusion to build wm8711.c
properly with CONFIG_SPI_MASTER.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Don't use a static for WM8974 PLL factors - we don't support more than
one device so it won't happen but no sense in leaving the race condition
hanging around. Also, pre_div is a single bit and it's a bit simpler if
we move the handling of the factor of 4 in the output into the
coefficient setup.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The DMA params for McASP with FIFO has been updated so that it works for
various FIFO levels. A member- 'fifo_level' has been added to the DMA
params data structure. The fifo_level can be adjusted by the tx[rx]_numevt
platform data. This is relevant only for DA8xx/OMAP-L1xx platforms. This
implementation has been tested for numevt values 1, 2, 4, 8.
Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Mia has an undocumented line-out control, and it has to be exposed.
Signed-off-by: Giuliano Pochini <pochini@shiny.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the commit fdbc66266c, I mistakenly
replaced the capture mixer array for ALC268_ACER to nosrc version
although this should be kept to alt_mixer. Now fixed back.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Reference: ALSA bug #0004614https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4614
port-A (0x11) - front hp-out
port-D (0x12) - rear line out
port-E (0x1c) - front mic-in
port-F (0x16) - Internal speakers
digital-mic (0x17) - Internal mic
init verbs, mixers, jack sensing and PCI_QUIRK to support this hardware
Signed-off-by: Miguel de Barros <miguel.de.barros@bluewin.ch>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since the recent kernel can handle MSI properly on non-Intel platforms,
let's enable MSI as default.
If any borken device is found, we can add the quirk entry to the list,
which is currently empty.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Insert "Playback" into the input monitor control names to prevent
alsa-lib from treating these controls as global controls.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a control that allows disabling the high-pass filter of the WM8785 ADC.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a control to select between sharp and slow roll-of filter responses
of the DACs.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a control to increase the oversampling factor to 128x on cards with
PCM1796 or PCM1792A DACs.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a callback that allows model drivers to modify the default I2S MCLK
rate.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a mixer control to adjust the headphone amplifier output for
headphones with different impedances.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Keep a cache of codec registers to avoid unnecessary writes.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add the possibility to route a mix of the two channels of stereo data to
the center and LFE outputs. This is implemented only for models where
the DACs support this, i.e., for the Xonar D1 and DX.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On card models with two-channel outputs, the base driver can
automatically disable the upmixing control so that the model
drivers do not need to do this.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Essence ST uses the CS2000 chip to generate the DAC master clock, so
we better initialize and program it appropriately.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The virtuoso.c file has become rather big. This patch splits it up so
that only code for very similar card models is in one file.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If the card is used with a Pericom PI7C9X110 PCI-E/PCI bridge,
reconfigure the latter's PCI buffering to fix an unknown problem.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On cards where the EEPROM was deliberately omitted, we do not need to
try to restore the EEPROM's contents.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move code from the OSS sscape driver in order to support old Soundscape OEM models.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* mark struct vm_area_struct::vm_ops as const
* mark vm_ops in AGP code
But leave TTM code alone, something is fishy there with global vm_ops
being used.
Signed-off-by: Alexey Dobriyan <adobriyan@gmail.com>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
When running in TDM mode there can be more than 2 channels used. Datasheet has
figures for upto 8 channels so increase max_channels on all SSP interfaces to
this figure.
Signed-off-by: Graeme Gregory <dp@xora.org.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
After puting a cd-audio inside my laptop there was no sound out here,
so I decided to install alsa-driver with debug level and setup a
model=test, it didn't help, but then I look at source code and added
this few lines, now cd-audio is working both when playback/recording.
[Additional minor fixes of mixer element/item names by tiwai]
Signed-off-by: Lukasz Marcinowski <nowymarluk@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
McASP write FIFO registers should be modified for playback and read FIFO
registers for capture. Check the PCM mode before manipulating the
FIFO registers. Currently, irrespective of playback/capture both the
FIFOs are enabled or disbaled. This resulted in errors in audio loopback
mode.
Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch removes references to cpu_dai->dma_data.
It makes struct davinci_pcm_dma_params part of
struct davinci_mcbsp_dev or struct davinci_audio_dev.
It removes the unused name variable from davinci_pcm_dma_params.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When both playback and capture stream were open
davinci_i2s_hw_params was setting parameters for
the wrong stream. The fix for davinci_i2s_hw_params
is sufficient, but it looks like a race still happens
in davici_pcm_open. This patch also makes the race smaller
but the next patch provides a better fix.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* 'fix/asoc' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ASoC: wm8753: fix mapping when MONOMIX is set to Stereo
ASoC: some minor changes for AD1836 and AD1938 codec drivers
ASoC: DaVinci: Fixes to McASP configuration
ASoC: Blackfin I2S: fix resuming when device hasn't been used
ASoC: Blackfin I2S: add lost platform_device parameter to resume function
ASoC: fix typos in Blackfin headers
ASoC: bf5xx-sport: the irq save/restore funcs take an unsigned long
ASoC: Blackfin AC97: add a few missing multichannel define handling
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
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>
Since the active field of the dai already tells us the stream activity,
the local counter variable is redundant and can be replaced.
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>
Adds support for US-144 when attached on USB1.1.
Unlike the US-122L it uses both USB interfaces 0 and 1.
Signed-off-by: Tobias Hansen <Tobias.Hansen@physik.uni-hamburg.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When MONOMIX is set to Stereo, Left PGA was not powered on but should be.
Add a mapping from Capture Left Mux to Capture Left Mixer to fix the issue.
Signed-off-by: Phil Vandry <vandry@TZoNE.ORG>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
1. delete redundant assignment to bus field in spi_driver structure
2. fix lost assignment to set_bias_level entry in ad1938 codec dai
3. change spi driver name of ad1836 from "ad1836-spi" to "ad1836"
Signed-off-by: Barry Song <barry.song@analog.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* Support for customization of the external clock names
* Adding hooks to playback_pro_open and capture_pro_open, allowing e.g.
limiting available stream rates to a single value when the external
clock rate is detected
Signed-off-by: Pavel Hofman <pavel.hofman@ivitera.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* pro-rate-locking applies to internal clock mode only
* required rate and current rate are compared for internal clock mode only
Signed-off-by: Pavel Hofman <pavel.hofman@ivitera.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Julian Anastasov