Fixes Kconfig dependency of TWL4030 audio codec driver
with TWL4030 core driver on both overo and omap2evm
boards
Signed-off-by: Arun KS <arunks@mistralsolutions.com>
Acked-by: David Brownell <dbrownell@users.sourceforge.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Patch adds support for mono audio links so that McBSP DAI can operate with
real mono codecs. In I2S, the signalling remains the same but only first
frame (left channel) is transmitting audio data and second frame having null
data. In DSP_A, only first frame is transmitted.
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Prepare for upcoming McBSP DAI update adding support for mono links by
restricting number of channels to 2 in N810. This is due tlv320aic3x which
claims channels_min = 1 and playing pure mono audio over I2S would cause
it to be played only from left channel if both cpu and codec DAI's claim to
support mono.
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Check the model type instead of PCI SSID for detection of the mic types
on Dell laptops with IDT 92HD73xx codecs. In this way, a new laptop
can be tested via model module option.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixed the quirk string for Dell studio 1535 (the product name wasn't
published at the time the patch was made).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
STAC/IDT driver creates "Headphone as Line-Out" switch even if there
is no line-out pins on the machine. For devices only with headpohnes
and speaker-outs, this switch shouldn't be created.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- add support for pcxhr stereo cards mixer controls
- adjust tlv db scales to real dBu values
- fix bug with monitoring volume control pcxhr_monitor_vol_put
- do some cleanup
Signed-off-by: Markus Bollinger <bollinger@digigram.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- Add support for pcxhr stereo cards and their firmware
- autorize sound cards without analog IO
- do some cleanup
Signed-off-by: Markus Bollinger <bollinger@digigram.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- Add support for pcxhr stereo cards
- minor bugfixes : period and buffer size consraints
- fix PLL register values
- do some clean up
Signed-off-by: Markus Bollinger <bollinger@digigram.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- Add support for pcxhr stereo cards
- do some clean up
Signed-off-by: Markus Bollinger <bollinger@digigram.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since we need to handle many unsolicited events assigned to different
widgets, allocate the event dynamically using the existing events
array, and use the tag appropriately instead of combination of fixed
number and widget nid. (Note that widget nid can be over 4 bits!)
Also, replaced the call of unsol_event handler with a dedicated
function to be more readable.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The AFG pin power-mapping isn't properly set for the fixed I/O pins
on IDT 92HD* codecs. This resulted in the low power mode after the
boot until any jack detection is executed, thus no output from the
speaker.
This patch fixes the power mapping for the fixed pins, and also fixes
the GPIO bits and digital I/O pin settings properly in stac92xx_ini().
Reference: Novell bnc#446025
https://bugzilla.novell.com/show_bug.cgi?id=446025
Signed-off-by: Takashi Iwai <tiwai@suse.de>
SPDIF status bits controls are written via snd_hda_codec_write()
without caching. This causes a regression at resume that the bits
are lost.
Simply replacing it with the cached version fixes the problem.
Reference:
http://lkml.org/lkml/2008/11/24/324
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now that the ASoC resume has been punted to a workqueue for a release
cycle without attracting bug reports it should be safe to make the
log messages associated with it debug level, reducing noise and kernel
size in production configurations.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Special handling is required for suspend and resume of AC97 codecs
due to the control path going over the data bus.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
DAI type information is only ever used within ASoC in order to special
case AC97 and for diagnostic purposes. Since modern CPUs and codecs
support multi function DAIs which can be configured for several modes
it is more trouble than it's worth to maintain anything other than a
flag identifying AC97 DAIs so remove the type field and replace it with
an ac97_control flag.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Medion MD96630 has ALC268 codec on slot#2 although it's not used
for any purpose. This codec conflicts with the primiary codec ALC888
on slot#0, and gives mixer errors.
This patch adds a corresponding entry to probe_mask blacklist.
Reference: Novell bnc#412528
https://bugzilla.novell.com/show_bug.cgi?id=412528
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some of the gain controls in TWL (mostly those which are associated with
the outputs) are implemented in an interesting way:
0x0 : Power down (mute)
0x1 : 6dB
0x2 : 0 dB
0x3 : -6 dB
Inverting not going to help with these.
Custom volsw and volsw_2r get/put functions to handle these gains.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add CGAIN (Coarse gain control) to TWL4030 codec.
The range of the CGAIN is:
0 dB to 12 dB in 6 dB steps.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
TWL4030 FGAIN volume control has a range:
-62 to 0 dB in 1 dB steps, 0 in the FGAIN means mute.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Keep Soft-volume disabled for now, since if it is enabled
the FGAIN volume controls are not working in the current
configuration:
CODEC_MODE:OPT_MODE = 1
OPTION:ARXR2_EN = 1
OPTION:ARXL2_EN = 1
OPTION:ARXR1_EN = 0
OPTION:ARXL1_VRX_EN = 0
RX_PATH_SEL:RXL1_SEL = 0x0 (or 0x1)
RX_PATH_SEL:RXR1_SEL = 0x0 (or 0x1)
After the patch, FGAIN volume control works.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch fixes the bug 0004240: ALC888 - Intel HDA - Headphone Controlling.
It is made against the 2008-11-23 snapshot.
Added Realtek ALC888 model entry for the Fujitsu-Siemens Amilo Xa3530
laptop. It has 4 jacks: HP out, Mic-in, Line-in and Line-out/Side/SPDIF
(this one is on the laptop side, the other ones are on the rear).
Model detection works.
Headphone jack sense works now.
Front mic works now, was same as Acer Aspire 4930G.
Added channel mode from 2 to 8 channels.
In 2ch and 4ch modes, the front is also sent to the Line-out/side jack
for convenience instead of just muting the Line-out/side jack like other
models do.
When using the Mic-in jack as CLFE, the sound is very low (bug?). To
work it around, in 6ch mode the CLFE channel is duplicated to the
Line-out/side jack because this one has a better amp.
Cc: manu@frogged.de
Signed-off-by: Vincent Petry <PVince81@yahoo.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
STAC/IDT driver creates "Headphone as Line-Out" switch even if there
is no line-out pins on the machine. For devices only with headpohnes
and speaker-outs, this switch shouldn't be created.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix "defined but not used" build warning by moving eld_versoin_names[]
and cea_edid_version_names[] into hdmi_print_eld_info().
Signed-off-by: Wu Fengguang <wfg@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
DisplayPort is a digital display interface standard put forth by
the Video Electronics Standards Association (VESA). It defines a
new license-free, royalty-free, digital audio/video interconnect,
intended to be used primarily between a computer and its display monitor,
or a computer and a home-theater system.
- From Wikipedia, the free encyclopedia
Signed-off-by: Wu Fengguang <wfg@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- rename ELD proc write routine to hdmi_write_eld_info()
- support modifying WMAPro's profile
Write to some ELD fields (monitor_name, manufacture_id, product_id,
eld_version, edid_version) are deliberately not supported, since that
won't correct wrong behaviors and only leads to confusions.
Signed-off-by: Wu Fengguang <wfg@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- make some messages more user friendly
- add message prefix "HDMI:" to indicate the problem's domain
(also easier to do `dmesg | grep HDMI` ;-)
Signed-off-by: Wu Fengguang <wfg@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Print some CA selecting info, which could be valuable for debugging when
something goes wrong.
Signed-off-by: Wu Fengguang <wfg@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_print_pcm_rates() and snd_print_pcm_bits() are used by both
hda_proc.c and hda_eld.c, thus they have to be defined in the common
place.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixed the quirk string for Dell studio 1535 (the product name wasn't
published at the time the patch was made).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the following sparse warning:
sound/pci/hda/patch_nvhdmi.c:161:25: warning: symbol
'snd_hda_preset_nvhdmi' was not declared. Should it be static?
Signed-off-by: Hannes Eder <hannes@hanneseder.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Implement support for the Marvell Zylonite PXA3xx reference platform,
supporting standard AC97 stereo and AUX interfaces together with the
auxiliary I2S interface of the WM9713.
The board has two options for the MCLK of the WM9713: either the standard
AC97 system clock can be used or the 13MHz CLK_POUT output of the PXA3xx
can be used, selected via SW15 on the board. Currently only the AC97
system clock is supported by this driver.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Liam Girdwood's ASoC v2 work avoids having two different ops structures
for DAIs by merging the members of struct snd_soc_ops into struct
snd_soc_dai_ops, allowing per DAI configuration for everything.
Backport this change.
This paves the way for future work allowing any combination of DAIs to
be connected rather than having fixed purpose CODEC and CPU DAIs and
only allowing CODEC<->CPU interconnections.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Clean up our record of the active streams in shutdown(), fixing
subsequent failures of snd_pcm_hw_constraints_complete after closure of
a stream.
NOTE:
- The ssm2602 allows pairs of non-matching PB/REC rates.
- This is a fix for less evil:
The logic is flawed (e.g. the slave might startup before the
master's rate and sample_bits are set).
Signed-off-by: Karl Beldan <karl.beldan@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
One of the issues with the ASoC v1 API which has been addressed in the
ASoC v2 work that Liam Girdwood has done is that the ALSA card provided
by ASoC is distributed around the ASoC structures. For example, machine
wide data such as the struct snd_card are maintained as part of the
CODEC data structure, preventing the use of multiple codecs. This has
been addressed by refactoring the data structures so that all the data
for the ALSA card is contained in a single structure snd_soc_card which
replaces the existing snd_soc_machine and snd_soc_device.
Begin the process of backporting this by renaming struct snd_soc_machine
to struct snd_soc_card, better reflecting its function and bringing it
closer to standard ALSA terminology.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Make the codec re-configuration feature selectable via Kconfig,
CONFIG_SND_HDA_RECONFIG.
Also mark it as experimental (as it really is).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The jack instances created in patch_sigmatel.c may be double-freed.
The device management code checks the invalid element, and thus there
is no real breakage, but it spews annoying warning messages.
But, we can't simply remove the release calls of these jack instances
because they have to be freed when the codec is re-configured.
Now, a new flag, bus->shutdown is introduced to indicate that the bus
is really being unloaded, i.e. the objects managed by the device
manager will be automatically deleted. We release these objects only
when this flag isn't set.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Allow users to fix quicks of ELD ROMs by writing new values to the ELD proc
interface. The format is one or more lines of "name hex_value".
Users can add/remove/modify up to 32 SAD(Short Audio Descriptor) entries.
Signed-off-by: Wu Fengguang <wfg@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Rename "monitor name" to "monitor_name" to conform with the keyword style.
Signed-off-by: Wu Fengguang <wfg@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added STAC_DELL_M4_3 quirk for Dell systems, also reorganized the
board config switch to assign number of digital muxes, microphones,
and SPDIF muxes via the PCI quirk defined.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the following sparse warnings:
sound/sound_core.c:460:2: warning: returning void-valued expression
sound/sound_core.c:477:2: warning: returning void-valued expression
sound/sound_core.c:510:5: warning: symbol 'soundcore_open' was not
declared. Should it be static?
Signed-off-by: Hannes Eder <hannes@hanneseder.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The changes specific for Samsung laptops seem unapplicable to other
hardware models like ASUS. The mic inputs are lost on such hardware
by the change 5d5d5f43f1.
This patch adds back the old laptop-eapd model, and create a new
model "samsung" for the new one specific to Samsung laptops with
automatic mic selection feature.
Reference: kernel bugzilla #12070http://bugzilla.kernel.org/show_bug.cgi?id=12070
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add support for EAPD on system suspend and disabling EAPD on headphone jack
detection for STAC_DELL_M6 laptops.
This patch fixes the regressions, the silent output on HP of some Dell
laptops (see Novell bnc#446025):
https://bugzilla.novell.com/show_bug.cgi?id=446025
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added Realtek ALC888 model entry for the Acer Aspire 4930G laptop that
fixes the following features:
- internal microphone
- heaphone jack sense
- channel mode
Signed-off-by: Vincent Petry <PVince81@yahoo.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Replace 5 free-and-return-err blocks with goto-out-free ones.
This makes the main logic more outstanding.
Signed-off-by: Wu Fengguang <wfg@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds twl4030 audio support on omap2evm
Signed-off-by: Arun KS <arunks@mistralsolutions.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The PCM3008 is used on the Lyrtech SFFSDR board, in conjunction with an
FPGA that generates the bit clock and the master clock
[Downgraded the rate debug print to pr_debug() in hw_params, converted
asm/gpio.h to linux/gpio.h -- broonie]
Signed-off-by: Hugo Villeneuve <hugo@hugovil.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The PCM3008 is a 16-bit stereo audio codec. It accepts
left-justified format for ADC, and right-justified format
for DAC. Independent power-down modes for ADC and DAC are
provided, as well as a digital de-emphasis filter (4 modes).
[Merged Makefile & Kconfig, changed asm/gpio.h to linux/gpio.h -- broonie]
Signed-off-by: Hugo Villeneuve <hugo@hugovil.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Strip out some ELD printk messages that end user won't care,
and make the output compact.
Signed-off-by: Wu Fengguang <wfg@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Introduce a global function snd_print_pcm_bits() and use it in the ELD code.
Signed-off-by: Wu Fengguang <wfg@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Moved support for EAPD mute on suspend from stac92hd71xx_suspend
to the generic stac92xx_suspend function.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Refactor the channel mapping code for consistent naming
and make it more informed about channel allocations.
Signed-off-by: Wu Fengguang <wfg@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
To play a 3+ channels LPCM/DSD stream via HDMI,
- HDMI sink must tell HDMI source about its speaker placements
(via ELD, speaker-allocation field)
- HDMI source must tell the HDMI sink about channel allocation
(via audio infoframe, channel-allocation field)
(related docs: HDMI 1.3a spec section 7.4, CEA-861-D section 7.5.3 and 6.6)
This patch attempts to set the CA(channel-allocation) byte in the audio infoframe
according to
- the number of channels in the current stream
- the speakers attached to the HDMI sink
A channel_allocations[] line must meet the following two criteria to be
considered as a valid candidate for CA:
1) its number of allocated channels = substream->runtime->channels
2) its speakers are a subset of the available ones on the sink side
If there are multiple candidates, the first one is selected. This simple
policy shall cheat the sink into playing music, but may direct data to the
wrong speakers.
Signed-off-by: Wu Fengguang <wfg@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
code refactor: make a global function snd_print_channel_allocation().
Signed-off-by: Wu Fengguang <wfg@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
code refactor: make a standalone function hdmi_fill_audio_infoframe().
Signed-off-by: Wu Fengguang <wfg@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Do not ignore the return of 'device_create_file' in
'snd_card_register' and thereby fixing the following warnings:
sound/core/init.c: In function 'snd_card_register':
sound/core/init.c:640: warning: ignoring return value of
'device_create_file', declared with attribute warn_unused_result
sound/core/init.c:641: warning: ignoring return value of
'device_create_file', declared with attribute warn_unused_result
Signed-off-by: Hannes Eder <hannes@hanneseder.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Mike Frysinger <vapier.adi@gmail.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Mike Frysinger <vapier.adi@gmail.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
A probe function should have a clean return 0 path.
Cc: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Michael Hennerich <michael.hennerich@analog.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
clean up redudent code and correct building problem in non-mmap mode
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch provides a option for users to enable multi-channel function support
in Blackfin ASoC driver. Because Blackfin is without MMU, it is easy for us and
the user to enable this function at compiling stage not dynamically on the fly.
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We added multi-channel function to this codec driver and Blackfin ASoC driver as well.
It was tested on Blackfin hardware.
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
tweak SPORT range for non-BF54x so we get proper behavior for BF52x parts
Signed-off-by: Mike Frysinger <vapier.adi@gmail.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix concurrent capture/playback issue.
The issue is caused by re-initialization of control registers used specifically
for capture or playback in both capture and playback operations.
Signed-off-by: Steve Chen <schen@mvista.com>
Signed-off-by: Naresh Medisetty <naresh@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Allow the multiple imux instances for matrix-type mixers like ALC882.
So far, only ALC260 used this feature, but other codecs may need a
similar stuff.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixed the GPIO mask and co initialization in patch_stac92hd71bxx()
so that the gpio_maks for HP_M4 model is set properly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixed the restore of pin configs at resume for some STAC/IDT codec
models. These models set explicitly the pin configs after the default
init configs, and these aren't restored properly at resume.
This patch introduces two changes:
- Allocate always pin_configs array in stac_spec so that the driver
can overwrite the value freely
- Introduce stac_change_pin_config() to change the pin config value
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Call the init callback and remove duplicated codes in stac92xx_resume().
This also fixes the missing initialization such as digital I/O pins.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The jack detection input elements should be created in build_controls
callback instead of init callback because init can be called multiple
times by suspend/resume and power-saving.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Create /proc/asound/card<card_no>/eld#<codec_no> to reflect the audio
configurations and capabilities of the attached HDMI sink.
Some notes:
- Shall we show an empty file if the ELD content is not valid?
Well it's not that simple. There could be partially populated ELD,
and there may be malformed ELD provided by buggy drivers/monitors.
So expose ELD as it is.
- The ELD retrieval routines rely on the Intel HDA interface,
others are/could be universal and independent ones.
- How do we name the proc file?
If there are going to be two HDMI pins per codec, then the current naming
scheme (eld#<codec no>) will fail. Luckily the user space dependencies should
be minimal, so it would be trivial to do the rename if that happens.
- The ELD proc file content is designed to be easy for scripts and human reading.
Its lines all have the pattern:
<item_name>\t[\t]*<item_value>
where <item_name> is a keyword in c language, while <item_value> could be any
contents, including white spaces. <item_value> could also be a null value.
Signed-off-by: Wu Fengguang <wfg@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ELD handling routines can be shared by all HDMI codecs,
and they are large enough to make a standalone source file.
Signed-off-by: Wu Fengguang <wfg@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We want to share some code with print_pcm_rates(),
so extract a common routine snd_print_pcm_rates() from it.
Signed-off-by: Wu Fengguang <wfg@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove get_amp_nid(): it duplicates the one defined in hda_local.h
Signed-off-by: Wu Fengguang <wfg@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A small additional power saving can be achieved for the WM8990 by
maintaining VMID using a 2*250k divider when in standby mode.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Enable a hardware workaround which avoids problems with the clocking of
the ADCs in certain configurations.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Only fully documented registers are cached in the WM8990 but additional
registers exist.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
FGAIN for playback is in range of 0-0x3f, while for capture GAIN it
is in the range of 0-0x1f.
The original value of 128 (0x7f) would modify the CGAIN also for
playback.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Steve Sakoman <steve@sakoman.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Reorder HDMI audio enabling sequence so that
1) the sink knows about the coming audio stream
2) unmute
3) start transferring audio samples
The theory is that in the path A=>B=>C, we first make C ready, and then
enable B, and lastly allow A to send audio samples.
Signed-off-by: Wu Fengguang <wfg@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add support for EAPD on system suspend and disabling EAPD on headphone jack
detection for STAC_DELL_M6 laptops.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a new alc883 model ALC1200_ASUS_P5Q for ASUS P5Q-EM boards.
It is the same as ALC883_6ST_DIG except that the SPDIF digital
output nid is 0x10.
Tested-by: Andrei Tanas <andrei@tanas.ca>
Signed-off-by: Wu Fengguang <wfg@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Check board preset model instead of codec->subsystem_id in
patch_92hd71bxx() so that other hardwares configured via the model
option work like the given model.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Error handling code following a kzalloc should free the allocated data.
The error handling code is adjusted to call pci_disable_device(pci); as
well, as done later in the function
The semantic match that finds the problem is as follows:
(http://www.emn.fr/x-info/coccinelle/)
// <smpl>
@r exists@
local idexpression x;
statement S;
expression E;
identifier f,l;
position p1,p2;
expression *ptr != NULL;
@@
(
if ((x@p1 = \(kmalloc\|kzalloc\|kcalloc\)(...)) == NULL) S
|
x@p1 = \(kmalloc\|kzalloc\|kcalloc\)(...);
...
if (x == NULL) S
)
<... when != x
when != if (...) { <+...x...+> }
x->f = E
...>
(
return \(0\|<+...x...+>\|ptr\);
|
return@p2 ...;
)
@script:python@
p1 << r.p1;
p2 << r.p2;
@@
print "* file: %s kmalloc %s return %s" % (p1[0].file,p1[0].line,p2[0].line)
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Changed value for STAC_VREF_EVENT from 0x40 to 0x00 because the
unsol response value is only 6-bits width and the former value
was 1<<6 which is an overrun.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The WM8728 is a high performance stereo DAC designed for applications
such as DVD, home theatre and digital TV.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This reverts commit 8dc840f88d. Christian
Pellegrin <chripell@gmail.com> reported that on some systems the patch
caused DMA to fail which is much more serious than the original skipped
audio issue. Further investigation by Dave shows that the behaviour
depends on the clock speed of the SoC - a better fix is neeeded.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Split the bound Master control to individual Headphone and Speaker
volume controls for VAIO with STAC982x codecs.
The Master controls is still created as a vmaster.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Without this patch it is possible to select drivers which require
bestcomm support without bestcomm support being selected. This
patch reworks the bestcomm dependencies to ensure the correct
bestcomm tasks are always enabled.
Reported-by: Hans Lehmann <hans.lehmann@ritter-elektronik.de>
Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
The digital beep widget may have no mute control, and always enabling
the beep is ofen pretty annoying, especially on laptops.
This patch adds a mixer control "PC Beep Playback Switch" when there
is no mixer amp mute is found, and controls it on software.
Reference: Novell bnc#444572
https://bugzilla.novell.com/show_bug.cgi?id=444572
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Originally it was put too tight limits to support only 44.1 kHz and 48 kHz
sample rates in McBSP DAI driver. Extend it now to 8 kHz - 96 kHz. With
96 kHz and 2*16 bits, bit clock is 3.072 MHz < 3.125 MHz (I2S max?).
Tested on Nokia N810 with TVL320AIC33 from rates 8 - 96 kHz and on Texas
Instruments Beagle with TWL4030 from rates 8 - 48 kHz.
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Acked-by: Steve Sakoman <steve@sakoman.com>
Acked-by: Arun KS <arunks@mistralsolutions.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
TWL4030 currently supports rates between 8 kHz and 48 kHz and sets the codec
mode register accordingly in twl4030_hw_params. Expose this info so that
ASoC can match other rates than 44.1 kHz or 48 kHz as well.
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Acked-by: Steve Sakoman <steve@sakoman.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The patch uses HD Audio PCI class code to detect AMD HD Audio cards.
Signed-off-by: Libin Yang <libin.yang@amd.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move the handling of SiI1392 HDMI codec from patch_atihdmi.c to
patch_intelhdmi.c, which makes our ASUS P5E-VM HDMI board work.
Signed-off-by: Wu Fengguang <wfg@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The card->id (card text identification) can be changed at runtime.
It might be confusing to have old text identification in device name.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For udev, we need a way to rename soundcard names. The soundcard numbers
(indexes) are hardwired but we have a text identification which can be
changed at run-time. The ALSA user space tools already allow using of
this text identification.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixes swapping of channels at start of stereo playback.
Channel swap can be observed while playing left-only or right-only audio data. The channel
swap is fixed by handling the XSYNCERR condition.
Signed-off-by: Naresh Medisetty <naresh@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Due to the recent change for multiple HP as line-out switch, only
one of the multiple headphons (usually a wrong one) is toggled
and the other pins are still disabled. This causes the silent output
problem on some Dell laptops.
Also, the hp_switch check is screwed up when a line-in or a mic-in
jack exists. This is added as an additional output, but hp_switch
check doesn't take it into account.
This patch fixes these issues: simplify hp_switch check by using
the NID instead of bool, and clean up / fix the toggle of HP pins
in unsol event handler code.
Reference: Novell bnc#443267
https://bugzilla.novell.com/show_bug.cgi?id=443267
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The input pins are sometimes not initialized properly because
of the optimization check of the current pinctl code.
Force to initialize the mic input pins so that they can be set up
properly even if they were in a weird state. But keep other input
pins if already set up as input, since this could be an extra mic
pin.
Reference: Novell bnc#443738
https://bugzilla.novell.com/show_bug.cgi?id=443738
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The creation of analog-mux mixer element is missing in
patch_stac9200() due to the dynamic allocation patch.
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: hda - Make the HP EliteBook 8530p use AD1884A model laptop
ALSA: gusextreme: Fix build errors
ALSA: hdsp: check for iobox and upload firmware during ioctl
ALSA: HDSP: check for io box before uploading firmware
ALSA: hda - Add another HP model (6730s) for AD1884A
alsa: fix snd_BUG_on() and friends
ALSA: hda - Add a quirk for MEDION MD96630
ALSA: hda - Limit the number of GPIOs show in proc
Added a QUIRK to patch_analog.c for the HP Elitebook 8530p
(IDs 0x103c:0x30e7) to use AD1884A model 'laptop' by default.
Playback and Capture confirmed working.
Signed-off-by: Travis Place <wishie@wishie.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The TI DVEVM board uses the SND_SOC_DAIFMT_CBM_CFM & I2S formats, but the
Lyrtech SFFSDR board uses the SND_SOC_DAIFMT_CBM_CFS & RIGHT-JUSTIFIED formats.
Signed-off-by: Hugo Villeneuve <hugo.villeneuve@lyrtech.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
fixes playing/recording of 8 bit audio files.
Generated on 20081108 against v2.6.27
Signed-off-by: Christian Pellegrin <chripell@fsfe.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
currently, the error message when trying to run hdspmixer or hdspconf
if the breakout box is not connected is somehow misleading, since it
asks the user to upload the firmware.
this patch adds a test, whether the breakout box is connected and
tries to upload the firmware in the case, that it is not present, e.g.
because of power failures of the breakout box.
[Minor coding-style fixes by tiwai]
Signed-off-by: Tim Blechmann <tim@klingt.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
currently the hdsp driver tries to upload the firmware, even if the
io box is not connected. this patch adds a check for the io box
before trying to upload the firmware.
thus instead of messages complaining about the fifo status and firmware
loading failure, the driver gives a message that no multiface or
digiface is connected.
[A minor coding-style fix by tiwai]
Signed-off-by: Tim Blechmann <tim@klingt.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added model=laptop for another HP machine (103c:3614) with AD1884A
codec.
Signed-off-by: Michel Marti <mma@objectxp.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BIOS on Dell Studio Desktop tells wrong codec probe masks.
This patch gives the preset mask value to avoid invalid access.
Reference: Novell bug#440907
https://bugzilla.novell.com/show_bug.cgi?id=440907
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The master switch doesn't influence on NID 0x15, the headphone jack
on HP 3013 model because alc260_hp_master_update() ignores the passed
arguments.
Also, corrected the wrong arguments of hp3013 (0x10 and 0x15) although
this doesn't change any behavior.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add support for more sample rates, different crystals
and split playback/capture rates.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Acked-by: Arun KS <arunks@mistralsolutions.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The widget layout of the Fujitsu Lifebook S6420 (which is ICH9M-based
and uses an ALC269) is similar but not identical to the Lifebook
S6410/E8410 (which are ICH8M-based and use an ALC262).
It is named lifebook as fujitsu is in use for Amilo machines. This builds
on the Quanta FL1 work and supports all analog inputs & outputs that I am
aware of. Microphone autoswitch is implemented. The laptop mic port takes
precedence over the dock mic port if both happen to have a jack plugged in.
This made sense to me as a design decision (imagine a presentation
environment with the dock fully wired in and the presenter quickly wanting
to override the mic with a headset).
There is mention of a digital audio path on the codec graph, so perhaps
the headphone socket is dual-function analog/digital. I will follow up
with another patch if I can acquire equipment to test this.
Signed-off-by: Tony Vroon <tony@linx.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When an error occurs during the codec probing, typically accessing to an
non-existing codec slot, the controller chip gets often screwed up and
can no longer communicate with the codecs.
This patch adds a preparation phase just to probe codec addresses before
actually creating codec instances. If any error occurs during this
probing phase, the driver resets the controller to recover.
This will (hopefully) fix the famous "single_cmd" errors.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The hda_bus ops callback take struct hda_bus pointer.
Also, the command callback takes the composed command word, instead of
each small bits in arguments.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
According to TRM, 256*Fs clock output should be enabled
when TWL4030 is in slave mode, not master.
This allows sound to work on OMAP3 Pandora, which uses
256*Fs clock.
Signed-off-by: Grazvydas Ignotas <notasas@gmail.com>
Acked-by: Steve Sakoman <steve@sakoman.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The S3C24xx dma does not allow more than one buffer to be enqueue prior to
the dma transfers starting. This patch adds an additional parameter to
s3c24xx_pcm_enqueue() to allow for passing an initial dma maximum load
value.
Signed-off-by: David Anders <danders at amltd.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Rather than try to remember to keep the core version number updated
(which hasn't been happening) just remove it. It was much more useful
when ASoC was out of tree.
Signed-off-by: Mark brown <broonie@opensource.wolfsonmicro.com>
this patch adds asoc audio driver for pxa27x based Palm PDAs. I tested it for
palmtx, t5 and ld, it should work with palmz72 as well (slapin, please test).
I sent it here some time ago, but now I got to fixing bugs in it. It should
be somehow mostly ok and ready for applying.
[Converted to use snd_soc_dapm_nc_pin() and bool Kconfig -- broonie]
Signed-off-by: Marek Vasut <marek.vasut@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There are actually two variants of ALC268 Acer implementation, one
with an analog built-in mic (pin 0x19) and another with a digital
mic (pin 0x12). Created a new model, acer-dmic, for the latter case
now.
So far, all known models are assigned to be analog-mic, according to
the BIOS setup. If this doesn't match with the actual case, one needs
to try model=acer-dmic, and fix the entry to point ALC268_ACER_DMIC
if it works.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Asus V1s series laptops have an ALC660VD with PCI id: 0x1043, 0x1633.
1.) remove the previous behaviour of mapping that to the ALC861VD_LENOVO
device.
2.) add a new ALC660VD_V1S device based on ALC861VD_LENOVO, with an
added digital out.
Signed-off-by: Tristan Aston <astrotris@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a proper ifdef to shut out a compile warning:
CC [M] sound/pci/hda/patch_intelhdmi.o
sound/pci/hda/patch_intelhdmi.c:286: warning: ‘hdmi_get_dip_index’ defined but \
not used
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The file(s) below do not use LINUX_VERSION_CODE nor KERNEL_VERSION.
sound/soc/codecs/ad73311.c
This patch removes the said #include <version.h>.
Signed-off-by: Huang Weiyi <weiyi.huang@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Due to the hda-reconfiguration patches, the check of empty stream
is gone, and this results in an error with the codec setup with empty
streams.
This patch adds the check again to avoid the error at probing.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add NULL-check of the return value of snd_kctl_new1() before
accessing it. Also, make a sanity NULL check to snd_BUG_ON()
for debugging only.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Call device_create_file only once in snd_soc_dapm_sys_add function.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: hda: make a STAC_DELL_EQ option
ALSA: emu10k1 - Add more invert_shared_spdif flag to Audigy models
ALSA: hda - Add a quirk for another Acer Aspire (1025:0090)
ALSA: remove direct access of dev->bus_id in sound/isa/*
sound: struct device - replace bus_id with dev_name(), dev_set_name()
ALSA: Fix PIT lockup on some chipsets when using the PC-Speaker
ALSA: rawmidi - Add open check in rawmidi callbacks
ALSA: hda - Add digital-mic for ALC269 auto-probe mode
ALSA: hda - Disable broken mic auto-muting in Realtek codes
Add support for explicitly enabling the EQ distortion hack for
systems without software biquad support.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Cc: stable@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added a quirk for another Acer Aspier laptop (1025:0090) with ALC883
codec. Reported in Novell bnc#426935:
https://bugzilla.novell.com/show_bug.cgi?id=426935
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Removed the direct accesses of dev->bus_id in sound/isa/* by replacement
with dev_err() or dev_warn() functions.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
[stripped sound/isa/* changes, replaced with the next patch -- tiwai]
Signed-off-by: Kay Sievers <kay.sievers@vrfy.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The drivers (e.g. mtpav) may call rawmidi functions in irq handlers
even though the streams are not opened. This results in Oops or panic.
This patch adds the rawmidi state check before actually operating the
rawmidi buffers.
Tested-by: Ingo Molnar <mingo@elte.hu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We broke O_NONBLOCK handling in OSS dmasound_core in 2.3.11-pre3 - the
original code copied f_flags to open_mode and then checked for
O_NONBLOCK in there, but that got changed to copying f_mode and
O_NONBLOCK has not reached that field in any kernel version.
Since we do not care for any other bits, the fix is obvious...
Signed-off-by: Al Viro <viro@zeniv.linux.org.uk>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
As it is, all instances of ->release() for files that have ->fasync()
need to remember to evict file from fasync lists; forgetting that
creates a hole and we actually have a bunch that *does* forget.
So let's keep our lives simple - let __fput() check FASYNC in
file->f_flags and call ->fasync() there if it's been set. And lose that
crap in ->release() instances - leaving it there is still valid, but we
don't have to bother anymore.
Signed-off-by: Al Viro <viro@zeniv.linux.org.uk>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
The IRQMASK register has to be set to zero expclitily at the initialization
otherwise you'll get no interrupts properly at later operations.
Also, removed the old commented out codes.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Unified the capture mixer creation in patch_realtek.c.
ALC268 is still an exception since it has no AMP in ADC but in
MUX widget.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Unify the capture callbacks in patch_realtek.c.
The difference of matrix or mux style is checked via spec->is_mix_capture
flag.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The digital mic wasn't detected properly for ALC269 auto-probing mode
because of its widget number. Fixed now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent addition of automatic mic-muting is broken in some cases.
The code assumes that the pin nids <= 0x18, but the digital pins can
be less than 0x18.
Also, it assumes the front-mic being the internal mic, but it depends
on the hardware implementation actually.
Instead of complex case-fixes, better to disable the code as now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* git://git.kernel.org/pub/scm/linux/kernel/git/davem/sparc-2.6:
sparc64: Add missing null terminating entry to bq4802_match[].
sparc: use the new byteorder headers
rtc-m48t59: shift zero year to 1968 on sparc (rev 2)
dbri: check dma_alloc_coherent errors
sparc64: remove byteshifting from out* helpers
Add audio support for the Atmel AT91SAM9G20ek board(uing wolfson 8731).
It is based on the former eti_b1_wm8731.c file, using the atmel scc API.
Signed-off-by: Sedji Gaouaou <sedji.gaouaou@atmel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The Ateml AT91 and AVR32 SoC share common IP for audio and can share the
same driver code using the atmel-ssc API provided for both architectures.
Do this, creating a new unified atmel ASoC architecture to replace the
previous at32 and at91 ones.
[This was contributed as a patch series for reviewability but has been
squashed down to a single commit to help preserve both the history and
bisectability. A small bugfix from Jukka is included.]
Tested-by: Jukka Hynninen <ext-jukka.hynninen@vaisala.com>
Signed-off-by: Sedji Gaouaou <sedji.gaouaou@atmel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
sound/soc/soc-dapm.c: In function 'snd_soc_dapm_sys_add':
sound/soc/soc-dapm.c:828: error: 'ret' undeclared (first use in this function)
Signed-off-by: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: ASoC: Fix WM9713 ALC Decay Time name
ALSA: ASoC: Fix some minor errors in mpc5200 psc i2s driver
ALSA: ASoC: Fix mono controls after conversion to support full int masks
ALSA: sound/ice1712: indentation & braces disagree - add braces
ALSA: usb - Add quirk for Edirol UA-25EX advanced modes
sound: struct device - replace bus_id with dev_name(), dev_set_name()
ALSA: hda - Add reboot notifier
ALSA: Warn when control names are truncated
ALSA: intel8x0 - add Dell Optiplex GX620 (AD1981B) to AC97 clock whitelist
ALSA: hda - Fix SPDIF mute on IDT/STAC codecs
ALSA: hda: Add HDA vendor ID for Wolfson Microelectronics
ALSA: hda - Add another HP model for AD1884A
Disable the automatic volume control feature of the CS4270 audio codec. This
feature, which is enabled by default, causes volume change commands to be
delayed. Sometimes the volume change happens after playback is started.
Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Move the bus dependencies in SND_SOC_ALL_CODECS into the individual
codec options rather than have them centrally. This allows the
inclusion of AC97 codecs when testing on platforms with AC97 support
and will also handle codecs on multi-function devices more gracefully.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM9713 comes out of cold reset in low power mode so always requires
a warm reset to bring up the AC97 link after a cold reset.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The SSP ports PXA series processors can be used to implement a variety of
audio interface formats. This patch implements support for I2S, DSP A and
DSP B modes on these ports.
This patch is based on the previous out of tree pxa2xx-ssp driver (which
was originally written by Liam Girdwood with updates from Philipp Zabel
and Nicola Perrino) and pxa3xx-ssp driver (originally written by Seth
Forsee based on the pxa2xx-ssp driver). Testing coverage is not complete
currently.
Tested-by: Daniel Ribeiro <drwyrm@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
As well as ensuring that UI-relevant parts of control names don't get
truncated in the DAPM code this avoids conflicts in long control names
that differ only at the end of a long string.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since we can query the playback stream power state directly we do not
need to infer if it is powered up from the timer being scheduled. Doing
this avoids problems that previously existed with streams being
incorrectly determined to be powered up caused when the timer is
scheduled when streams are closed after being partially set up.
Reported-by: Nobin Mathew <nobin.mathew@gmail.com>
Reported-by: Jukka Hynninen <ext-jukka.hynninen@vaisala.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
i.e. echo 6 59 >/sys/kernel/debug/soc-audio.0/codec_reg
will set register 0x06 to a value of 0x59.
Also, pop_time debugfs interface setup is moved so that it
is setup in the same function as codec_reg
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The control had an extra space at the end of the name.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix missing unsigned for irqsave flags in psc i2s driver
Make attribute visiblity static
Collect all sysfs errors before checking status
[Word wrapped DEVICE_ATTR() lines for 80 columns -- broonie]
Signed-off-by: Jon Smirl <jonsmirl@gmail.com>
Acked-by: Grant Likely <grant.likely@secretlab.ca>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When ASoC was converted to support full int width masks SOC_SINGLE_VALUE()
omitted the assignment of rshift, causing the control operatins to report
some mono controls as stereo. This happened to work some of the time due
to a confusion between shift and min in snd_soc_info_volsw().
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Neither has any significance currently to the flow
because err is checked for the same condition before
the place of disagreement.
Signed-off-by: Ilpo Järvinen <ilpo.jarvinen@helsinki.fi>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the quirk for UA-25EX advanced modes.
UA-25EX is almost compatible with UA-25.
Tested-by: Serge Perinsky <sergebass@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Needs to check for dma_alloc_coherent() allocation failure.
Signed-off-by: FUJITA Tomonori <fujita.tomonori@lab.ntt.co.jp>
Signed-off-by: David S. Miller <davem@davemloft.net>
The current snd-hda-intel driver seems blocking the power-off on some
devices like eeepc. Although this is likely a BIOS problem, we can add
a workaround by disabling IRQ lines before power-off operation.
This patch adds the reboot notifier to achieve it.
The detailed problem description is found in bug#11889:
http://bugme.linux-foundation.org/show_bug.cgi?id=11889
Tested-by: Luiz Fernando N. Capitulino <lcapitulino@mandriva.com.br>
Cc: stable@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is likely to confuse user interfaces since the end of the control
name is interpreted (eg, "Volume", "Switch").
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The SPDIF mute switch code seems broken. It doesn't set unmute bits
properly. Also it contains the duplicated lines (merge error?) to be
cleaned up.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add Wolfson Microelectronics to the HDA vendor ID table.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Removed the old workaround to avoid the non-existing codec slot.
The current code should work without that workaround. If any,
we can add a quirk table.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: ASoC: Blackfin: update SPORT0 port selector (v2)
ALSA: hda - Restore default pin configs for realtek codecs
sound: use a common working email address
pci: use pci_ioremap_bar() in sound/
- Setting the TFS pin selector for SPORT 0 based on whether the selected
port id F or G. If the port is F then no conflict should exist for the
TFS. When Port G is selected and EMAC then there is a conflict between
the PHY interrupt line and TFS. Current settings prevent the conflict
by ignoring the TFS pin when Port G is selected. This allows both
ssm2602 using Port G and EMAC concurrently.
- some code cleanup
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some machines have broken BIOS resume that doesn't restore the default
pin configuration properly, which results in a wrong detection of HP
pin. This causes a silent speaker output due to missing HP detection.
Related bug: Novell bug#406101
https://bugzilla.novell.com/show_bug.cgi?id=406101
This patch fixes the issue by saving/restoring the default pin configs
by the driver itself.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds support to the IDT codec families to report jack status
to the jack abstraction layer. This required some reorganization in the
stac92xx_unsol_event function in which the index value is changed to
reporting the nid with the event.
Also adds an sigmatel_jack struct to keep track of the nid relation
to the jack abstraction layer instance. Also adds functions to set and
retrieve data values for each nid, this is used in stac92xx_unsol_event
to retrieve the GPIO mask for STAC_VREF_EVENT.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds snd_hda_get_jack* functions for reporting jack location,
device, and connectivity type.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
