Commit Graph

275247 Commits

Author SHA1 Message Date
Eliot Blennerhassett
72868339e4 ALSA: asihpi - Add new function codes.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-12-22 08:13:04 +01:00
Eliot Blennerhassett
d8aefaef1b ALSA: asihpi - Remove unused structs and defs
Structs related to network flash update are not required in kernel.

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-12-22 08:13:03 +01:00
Eliot Blennerhassett
502f271ae3 ALSA: asihpi - Update node types.
Add "Internal" node type.
Remove GPI and GPO node types.

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-12-22 08:13:02 +01:00
Eliot Blennerhassett
09c728aced ALSA: asihpi - Only set sync if card supports hardware stream grouping.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-12-22 08:13:02 +01:00
Eliot Blennerhassett
0be55c453f ALSA: asihpi - Relax drained check for more reliable playback startup.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-12-22 08:13:01 +01:00
Eliot Blennerhassett
8e0874ea72 ALSA: asihpi - Correct stray capital letters in identifier.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-12-22 08:13:00 +01:00
Eliot Blennerhassett
cbd757daf5 ALSA: asihpi - Use snd_pcm_debug_name to get substream name.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-12-22 08:12:59 +01:00
Eliot Blennerhassett
d4b06d23ab ALSA: asihpi - Volumes and meters may have 1 or 2 channels.
The channel count can be queried to determine which.

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-12-22 08:12:58 +01:00
Eliot Blennerhassett
c382a5da5c ALSA: asihpi - Low latency mode stream has fixed channel count.
Unlike other streams which support 1..max channels,

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-12-22 08:12:58 +01:00
Eliot Blennerhassett
40818b6242 ALSA: asihpi - Update copyright to 2011
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-12-22 08:12:57 +01:00
Eliot Blennerhassett
f6baaec2af ALSA: asihpi - Split hpi version info into separate header file.
and update HPI version to 4.10

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-12-22 08:12:56 +01:00
Eliot Blennerhassett
47a74a5d1e ALSA: asihpi - fix pcm dma pointer tracking
Elapsed counter should only count data committed to snd_pcm_period_elapsed,
rather than all data available

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-12-22 08:12:52 +01:00
Takashi Iwai
cde944803d ALSA: Add missing module parameters for als300 and cs5530 drivers
These drviers defined only variables but didn't declare as module
parameters.  Also fix the enable variable to bool type.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-12-19 10:34:44 +01:00
Rusty Russell
a67ff6a540 ALSA: module_param: make bool parameters really bool
module_param(bool) used to counter-intuitively take an int.  In
fddd5201 (mid-2009) we allowed bool or int/unsigned int using a messy
trick.

It's time to remove the int/unsigned int option.  For this version
it'll simply give a warning, but it'll break next kernel version.

Signed-off-by: Rusty Russell <rusty@rustcorp.com.au>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-12-19 10:34:41 +01:00
Sergiusz Urbaniak
1bba160a07 ALSA: snd-usb: added VOX ToneLab ST midi handling
Signed-off-by: Sergiusz Urbaniak <sergiusz.urbaniak@googlemail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-12-12 12:49:02 +01:00
Thomas Meyer
6d2d431369 ALSA: asihp: Use kcalloc instead of kzalloc to allocate array
The advantage of kcalloc is, that will prevent integer overflows which could
result from the multiplication of number of elements and size and it is also
a bit nicer to read.

The semantic patch that makes this change is available
in https://lkml.org/lkml/2011/11/25/107

Signed-off-by: Thomas Meyer <thomas@m3y3r.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-12-06 13:27:32 +01:00
Thomas Meyer
1d5d37f408 ALSA: ctxf: Use kcalloc instead of kzalloc to allocate array
The advantage of kcalloc is, that will prevent integer overflows which could
result from the multiplication of number of elements and size and it is also
a bit nicer to read.

The semantic patch that makes this change is available
in https://lkml.org/lkml/2011/11/25/107

Signed-off-by: Thomas Meyer <thomas@m3y3r.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-12-06 13:27:22 +01:00
David Dillow
705978516f ALSA: sis7019 - convert to dev_*() logging
Signed-off-by: David Dillow <dave@thedillows.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-12-02 10:44:01 +01:00
Takashi Iwai
b1ac29620b Merge branch 'fix/misc' into topic/misc 2011-12-02 10:43:52 +01:00
David Dillow
fc084e0b93 ALSA: sis7019 - give slow codecs more time to reset
There are some AC97 codec and board combinations that have been observed
to take a very long time to respond after the cold reset has completed.
In one case, more than 350 ms was required. To allow users to have sound
on those platforms, we'll wait up to 500ms for the codec to become
ready.

As a board may have multiple codecs, with some faster than others to
reset, we add a module parameter to inform the driver which codecs
should be present.

Reported-by: KotCzarny <tjosko@yahoo.com>
Signed-off-by: David Dillow <dave@thedillows.org>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-12-02 10:43:06 +01:00
Takashi Iwai
cf54d47c13 Merge branch 'fix/asoc' into for-linus 2011-12-01 16:32:18 +01:00
Charles Chin
88d686027b ALSA: hda - Fix S3/S4 problem on machines with VREF-pin mute-LED
The verb command in stac92xx_post_suspend caused the audio to stop
working after resuming from S3 mode on HP laptops with the VREF-pin
mute-LED control.  Removing relevant post_suspend registering.

Although removing D3 on AFG is no optimal solution, the impact should
be small in comparison with the broken S3/S4.

Signed-off-by: Charles Chin <Charles.Chin@idt.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-12-01 11:27:43 +01:00
Marc Vertes
4f8b6c7dc8 ALSA: hda_intel - revert a quirk that affect VIA chipsets
This quirk sould be reverted. It has the following probems:

1) The quirk was intended to "ASUS MV2-MX SE" motherboards only, but the
ID used matches a much broader range, potentially all boards containing a
VIA chipset model in the family of vendor VIA 0x1106 and audio device ID
0x3288, which encompasses VIA-VT82xx, VIA-VT1xx and VIA-VT20xx chipsets.

2) VIA chipsets rely on azx_via_get_position() to handle correctly dma
transfers during capture. Using POS_FIX_LPIB instead of POS_FIX_VIACOMBO
leads to partially corrupted input buffers during capture. The effects
of this bug are not immediately visible, it took strong DSP expertise,
some expensive signal generator and a spectrum analyzer to identify it
and verify correct behaviour using original default.

3) It's almost certain that the quirk did not fix the real problem,
if there was one. Refer to original submission:
http://mailman.alsa-project.org/pipermail/alsa-devel/2010-February/025109.html

Signed-of-by: Marc Vertes <mvertes@sigfox.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-11-29 13:04:03 +01:00
Takashi Iwai
542c9a0a2f ALSA: hda - Avoid touching mute-VREF pin for IDT codecs
Some HP laptops use a pin VREF for controlling the mute LED, and such a
pin shouldn't be powered off.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-11-29 13:01:30 +01:00
Lars-Peter Clausen
bda63586bc firmware: Sigma: Fix endianess issues
Currently the SigmaDSP firmware loader only works correctly on little-endian
systems. Fix this by using the proper endianess conversion functions.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Mike Frysinger <vapier@gentoo.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
2011-11-29 11:59:50 +00:00
Lars-Peter Clausen
c56935bdc0 firmware: Sigma: Skip header during CRC generation
The firmware header is not part of the CRC, so skip it. Otherwise the firmware
will be rejected due to non-matching CRCs.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Mike Frysinger <vapier@gentoo.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
2011-11-29 11:59:44 +00:00
Lars-Peter Clausen
4f718a29fe firmware: Sigma: Prevent out of bounds memory access
The SigmaDSP firmware loader currently does not perform enough boundary size
checks when processing the firmware. As a result it is possible that a
malformed firmware can cause an out of bounds memory access.

This patch adds checks which ensure that both the action header and the payload
are completely inside the firmware data boundaries before processing them.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Mike Frysinger <vapier@gentoo.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
2011-11-29 11:53:53 +00:00
John F Leach
ae7cc709f2 ALSA: usb-audio - Support for Roland GAIA SH-01 Synthesizer
Added table quirks entry for Roland GAIA SH-01 Synthesizer based upon
Roland SH-201 table entry as template.  USB MIDI and audio was tested
with Muse and Audacity.

Signed-off-by: John F Leach <jfleach@jfleach.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-11-29 08:23:15 +01:00
Mark Brown
fc8e6e8668 ASoC: Supply dcs_codes for newer WM1811 revisions
Based on initial data.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-11-28 23:18:38 +00:00
Mark Brown
fc07ecd851 ASoC: Error out if we can't generate a LRCLK at all for WM8994
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-11-28 22:10:05 +00:00
Axel Lin
a09452eeb7 ALSA: convert sound/* to use module_platform_driver()
This patch converts the drivers in sound/* to use the
module_platform_driver() macro which makes the code smaller and a bit
simpler.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-11-27 18:43:48 +01:00
Takashi Iwai
f339240dd8 Merge branch 'fix/hda' into for-linus 2011-11-27 17:59:07 +01:00
Takashi Iwai
187d333edc ALSA: hda - Fix jack-detection control of VT1708
VT1708 has no support for unsolicited events per jack-plug, the driver
implements the workq for polling the jack-detection.  The mixer element
"Jack Detect" was supposed to control this behavior on/off, but this
doesn't work properly as is now.  The workq is always started and the
HP automute is always enabled.

This patch fixes the jack-detect control behavior by triggering / stopping
the work appropriately at the state change.  Also the work checks the
internal state to continue scheduling or not.

Cc: <stable@kernel.org> [v3.1]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-11-27 17:56:17 +01:00
Dan Carpenter
92bb43e6aa ALSA: hda - cut and paste typo in cs420x_models[]
The CS420X_IMAC27 was copied from the line before but CS420X_APPLE
was clearly intented.

Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-11-27 17:56:07 +01:00
Mark Brown
5b895eec11 ASoC: Correct name of Speyside Main Speaker widget
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-11-27 16:03:51 +00:00
Axel Lin
51451b8d60 ALSA: Convert mips directory to module_platform_driver
Factor out some boilerplate code.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-11-24 13:03:02 +01:00
Takashi Iwai
77088cc973 Merge branch 'fix/asoc' into for-linus 2011-11-23 17:07:16 +01:00
Eric Miao
5ff1ddf22b ASoC: skip resume of soc-audio devices without codecs
There are cases where there is no working codec on the soc-audio devices,
and snd_soc_suspend() will skip such device when suspending. Yet its
counterpart snd_soc_resume() does not check this, causing complaints
about spinlock lockup:

[  176.726087] BUG: spinlock lockup on CPU#0, kworker/0:2/1067, d8ab82a8
[  176.732539] [<80014a14>] (unwind_backtrace+0x0/0xec) from [<805b3fc8>] (dump_stack+0x20/0x24)
[  176.741082] [<805b3fc8>] (dump_stack+0x20/0x24) from [<80322208>] (do_raw_spin_lock+0x118/0x158)
[  176.749882] [<80322208>] (do_raw_spin_lock+0x118/0x158) from [<805b7874>] (_raw_spin_lock_irqsave+0x5c/0x68)
[  176.759723] [<805b7874>] (_raw_spin_lock_irqsave+0x5c/0x68) from [<8002a020>] (__wake_up+0x2c/0x5c)
[  176.768781] [<8002a020>] (__wake_up+0x2c/0x5c) from [<804a6de8>] (soc_resume_deferred+0x3c/0x2b0)
[  176.777666] [<804a6de8>] (soc_resume_deferred+0x3c/0x2b0) from [<8004ee20>] (process_one_work+0x2e8/0x50c)
[  176.787334] [<8004ee20>] (process_one_work+0x2e8/0x50c) from [<8004fd08>] (worker_thread+0x1c8/0x2e0)
[  176.796566] [<8004fd08>] (worker_thread+0x1c8/0x2e0) from [<80053ec8>] (kthread+0xa4/0xb0)
[  176.804843] [<80053ec8>] (kthread+0xa4/0xb0) from [<8000ea70>] (kernel_thread_exit+0x0/0x8)

Signed-off-by: Eric Miao <eric.miao@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-11-23 14:56:36 +00:00
Axel Lin
b284362b6b ASoC: cs42l51: Fix off-by-one for reg_cache_size
Just checking the code in cs42l51_fill_cache():
The cache pointer points to codec->reg_cache + 1.
I think it is because CS42L51_FIRSTREG is 0x01,
so codec->reg_cache[0] is not used here.

Then we read CS42L51_NUMREGS bytes to cache.
So we need reg_cache_size to be CS42L51_NUMREGS + 1.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-11-23 11:34:21 +00:00
Paul Bolle
4ca8af579c ASoC: drop support for PlayPaq with WM8510
SoC Audio support for PlayPaq with WM8510 got added in commit 9aaca9683b
("[ALSA] Revised AT32 ASoC Patch"). That support depends on
BOARD_PLAYPAQ. That Kconfig symbol didn't exist when that support got
added in v2.6.27. It still doesn't. It has never been possible to even
build this driver. Drop it.

Signed-off-by: Paul Bolle <pebolle@tiscali.nl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-11-23 10:28:38 +00:00
Takashi Iwai
61071594f6 ALSA: hda/realtek - Minor cleanup
Use an inline function for the common pattern for assigning a capsrc.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-11-23 07:52:15 +01:00
Takashi Iwai
6759dc3238 ALSA: hda/realtek - Fix missing inits of item indices for auto-mic
When the imux entries are rebuilt in alc_rebuild_imux_for_auto_mic(),
the initialization of index field is missing.  It may work without it
casually when the original imux was created by the auto-parser, but
it's definitely broken in the case of static configs where no imux was
parsed beforehand.  Because of this, the auto-mic switching doesn't
work properly on some model options.

This patch adds the missing initialization of index field.

Reported-by: Dmitry Nezhevenko <dion@inhex.net>
Cc: <stable@kernel.org> [v3.1]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-11-23 07:45:21 +01:00
Takashi Iwai
6dfeb703e3 ALSA: hda - Fix invalid pin and GPIO for Apple laptops with CS codecs
The PCI SSID 8086:7270 is commonly used for multiple Apple machines,
thus we can't use it as identifier for a unique model.  Because of this
conflict, some machines show weird behavior.  For example, MacBook Air
shows Front and Surround speakers although only Surround works due to
the wrongly overridden pin-configuration for imac27.

This patch fixes two things:
- Stop the wrong pin-config override of imac27 by removing PCI SSID
  entry for avoiding the wrong mappings,
- Add the generic GPIO setup for Apple machines by checking the codec
  SSID vendor bits

Tested-by: Linus Torvalds <torvalds@linux-foundation.org>
Tested-by: Dirk Hohndel <hohndel@infradead.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-11-23 07:31:49 +01:00
Takashi Iwai
e2301a4de2 ALSA: hda - Check subdevice mask in snd_hda_check_board_codec_sid_config()
In snd_hda_check_board_codec_sid_config(), not only comparing with the
exact value but allow the bit-mask comparison for vendor-only, etc.

Tested-by: Linus Torvalds <torvalds@linux-foundation.org>
Tested-by: Dirk Hohndel <hohndel@infradead.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-11-23 07:31:36 +01:00
Timur Tabi
380c883038 ASoC: mpc8610: tell the CS4270 codec that it's the master
Commit ac601555 ("ASoC: Return early with -EINVAL if invalid dai format is
detected") requires the machine driver to tell the CS4270 codec driver
whether the CS4270 should be configured for master or slave operation.

Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-11-22 23:06:25 +00:00
Daniel Mack
d66b8537b3 ASoC: cs4720: use snd_soc_cache_sync()
Replace the manual register restore mechanism in cs4270.c and call the
generic snd_soc_cache_sync() handler instead.

This factors code out in favour of core facilities and also fixes a
bus confusion that is most probably caused by intermixing i2c-regmap
functions and i2c_smbus_* accessors.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-and-tested-by: Sven Neumann <s.neumann@raumfeld.com>
Acked-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-11-22 23:06:07 +00:00
Boojin Kim
3d94a2a53a ASoC: SAMSUNG: Fix build error
This patch adds <linux/modules.h> to fix following build errors.

sound/soc/codecs/wm8994.c: In function 'wm8994_readable':
sound/soc/codecs/wm8994.c:58: warning: unused variable 'wm8994'
sound/soc/samsung/smdk_wm8994.c:176: error: expected declaration specifiers or '...' before string constant
sound/soc/samsung/smdk_wm8994.c:176: warning: data definition has no type or storage class
sound/soc/samsung/smdk_wm8994.c:176: warning: type defaults to 'int' in declaration of 'MODULE_DESCRIPTION'
sound/soc/samsung/smdk_wm8994.c:176: warning: function declaration isn't a prototype
sound/soc/samsung/smdk_wm8994.c:177: error: expected declaration specifiers or '...' before string constant

Signed-off-by: Boojin Kim <boojin.kim@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-11-22 19:40:29 +00:00
Axel Lin
5c4b2aa3fd ASoC: max9877: Update register if either val or val2 is changed
In the case of ((max9877_regs[reg] >> shift) & mask) != val
but ((max9877_regs[reg2] >> shift) & mask) == val2,
current code does not update the registers.

Fix the logic to update registers if either val or val2 is changed.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-11-22 13:06:21 +00:00
Axel Lin
72531c9434 ASoC: Fix wrong define for AD1836_ADC_WORD_OFFSET
According to the datasheet:
The BIT[5:4] of ADC Control Register 2 is to control the word width.
        00 = 25 Bits
        01 = 20 Bits
        10 = 16 Bits
        11 = Invalid

Thus, the AD1836_ADC_WORD_OFFSET should be defined as 4.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
2011-11-22 13:06:14 +00:00
Wu Fengguang
a370fc62b9 ALSA: hda - fail ELD reading early
With the ELD repoll mechanism, we can (and should) fail the ELD reading
immediately when find something obviously wrong and let the caller retry
after some delay.

Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-11-22 12:09:45 +01:00