Commit Graph

7909 Commits

Author SHA1 Message Date
Axel Lin
4eaac50552 ASoC: wm8974: fix a memory leak if another WM8974 is registered
wm8974 is allocated in wm8974_i2c_probe() but is not freed if wm8974_register()
return -EINVAL (if another WM8974 is registered).

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-08-03 07:46:07 +01:00
Axel Lin
6b5d071e8b ASoC: wm8961: fix resource reclaim in wm8961_register error path
This patch fixes the error path in wm8961_register to properly free resources.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-08-03 07:45:52 +01:00
Axel Lin
8089a49d99 ASoC: wm8955: fix resource reclaim in wm8955_register error path
This patch fixes the error path in wm8955_register to properly free resources.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-08-03 07:45:37 +01:00
Axel Lin
db1e18de98 ASoC: wm8940: fix a memory leak if wm8940_register return error
This patch adds checking for wm8940_register return value,
and does kfree(wm8940) if wm8940_register() fail.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-08-03 07:45:20 +01:00
Axel Lin
62f5ad6733 ASoC: wm8904: fix resource reclaim in wm8904_register error path
This patch includes below fixes:
1. wm8904 need to be kfreed in wm8904_register() error path before return.
2. fix the error path for snd_soc_register_codec() fail and
   snd_soc_register_dai() fail to properly free resources.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-08-03 07:45:00 +01:00
Axel Lin
2c2749de11 ASoC: wm8711: fix a memory leak if another WM8711 is registered
wm8711 is allocated in either wm8711_spi_probe() or wm8711_i2c_probe() but is
not freed if wm8711_register() return -EINVAL(if another ad1836 is registered).

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-08-03 07:44:29 +01:00
Axel Lin
ef99e9b5a1 ASoC: wm8523: fix resource reclaim in wm8523_register error path
This patch includes below fixes:
1. If another WM8523 is registered, need to kfree wm8523 before return -EINVAL.
2. If snd_soc_register_codec failed, goto error path to properly free resources.
3. Instead of using mixed in-line and goto style cleanup, use goto style error
   handling if snd_soc_register_dai failed.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-08-03 07:44:10 +01:00
Axel Lin
085efd28b6 ASoC: da7210: fix a memory leak if failed to initialise da7210 audio codec
da7210 should be kfreed if da7210_init() return error.
This patch also fixes the error handing in the case of snd_soc_register_dai()
fail by adding snd_soc_unregister_codec() in error path.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-08-03 07:43:52 +01:00
Axel Lin
7bcaad919b ASoC: ak4642: fix a memory leak if failed to initialise AK4642
ak4642 should be kfreed if ak4642_init() return error.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Reviewed-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-08-03 07:40:41 +01:00
Axel Lin
fd3c8ac9cb ASoC: ad1836: fix a memory leak if another ad1836 is registered
ad1836 is allocated in ad1836_spi_probe() but is not freed if ad1836_register()
return -EINVAL (if another ad1836 is registered).

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Barry Song <21cnbao@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-08-03 07:40:26 +01:00
Ian Lartey
992bee401c ASoC: Initial WM8741 CODEC driver
The WM8741 is a very high performance stereo DAC designed for audio
applications such as professional recording systems, A/V receivers and
high specification CD, DVD and home theatre systems. The device supports
PCM data input word lengths from 16 to 32-bits and sampling rates up to
192kHz.  The WM8741 also supports DSD bit-stream data format, in both
direct DSD and PCM-converted DSD modes.

TODO: Expand wm8741_set_dai_sysclk and rate_constraint members to
allow for all supported sample rate / Master Clock frequency combinations.
Fully enable control of supplies.

Signed-off-by: Ian Lartey <ian@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-08-03 07:38:15 +01:00
David Henningsson
7bfb9c031e ALSA: hda - Do not try to create speaker NIDs for ALC268 if there aren't any
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-08-02 14:51:01 +02:00
Takashi Iwai
988b0dc154 Merge branch 'for-2.6.36' of git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into topic/asoc 2010-08-02 12:10:52 +02:00
Peter Ujfalusi
998a8a69f3 ASoC: omap-mcbsp: Remove period size constraint in THRESHOLD mode
The use of sDMA packet mode in THRESHOLD mode removes the restriction on the
period size.
With the extended THRESHOLD mode user space can ask for any
period size it wishes, and the driver will configure the
sDMA and McBSP FIFO accordingly.

Replace the hw_rule for the period size with static constraint,
which will make sure that the period size will be always
even (to avoid prime period size, which could be possible in
mono stream)

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-08-02 10:38:16 +01:00
Peter Ujfalusi
cf80e15860 ASoC: omap-mcbsp: Support for sDMA packet mode
Utilize the sDMA controller's packet syncronization mode, when
the McBSP FIFO is in use (by extending the THRESHOLD mode).
When the sDMA is configured for packet mode, the sDMA frame size
does not need to match with the McBSP threshold configuration.
Uppon DMA request the sDMA will transfer packet size number of
words, and still trigger interrupt on frame boundary.

The patch extends the original THRESHOLD mode by doing the
following:

if (period_words <= max_threshold)
Current THRESHOLD mode configuration

Otherwise (period_words > max_threshold)
McBSP threshold = sDMA packet size
sDMA frame size = period size

With the extended THRESHOLD mode we can remove the constraint
for the maximum period size, since if the period size is
bigger than the maximum allowed threshold, than the driver
will switch to packet mode, and picks the best (biggest)
threshold value, which can divide evenly the period size.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-08-02 10:38:16 +01:00
Peter Ujfalusi
15d0143007 ASoC: omap-mcbsp: Code cleanup in omap_mcbsp_dai_hw_params
To make the code a bit more readable, change the indexed
references to the omap_mcbsp_dai_dma_params elements with
pointer.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-08-02 10:38:16 +01:00
Peter Ujfalusi
81ec027e64 ASoC: omap-mcbsp: Restructure the code within omap_mcbsp_dai_hw_params
In preparation for the extended threshold mode (sDMA packet mode
support), the code need to be restructured.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-08-02 10:37:43 +01:00
John S Gruber
dd2f8c2f81 ALSA: usb - Correct audio problem for Hauppage HVR-850 and others rel. to urb data align
Match usb ids in usb/quirks-table.h for some Hauppage HVR-950Q models
and for the HVR850 model to those ids at the end of au0828-cards.c

Thanks to nhJm449 for pointing out the problem.

Signed-off-by: John S Gruber <JohnSGruber@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-08-02 09:12:59 +02:00
Dominik Brodowski
ac8b422838 pcmcia: remove cs_types.h
Remove cs_types.h which is no longer needed: Most definitions aren't
used at all, a few can be made away with, and two remaining definitions
(typedefs, unfortunatley) may be moved to more specific places.

CC: linux-ide@vger.kernel.org
CC: linux-usb@vger.kernel.org
CC: laforge@gnumonks.org
CC: linux-mtd@lists.infradead.org
CC: alsa-devel@alsa-project.org
CC: linux-serial@vger.kernel.org
Acked-by: Marcel Holtmann <marcel@holtmann.org> (for drivers/bluetooth/)
Acked-by: David S. Miller <davem@davemloft.net>
Signed-off-by: Dominik Brodowski <linux@dominikbrodowski.net>
2010-07-30 21:07:39 +02:00
Takashi Iwai
c7a9434dd6 ALSA: hda - Add a warning for ignored pins with ALC259/268/269
The current ALC259/268/269 parser ignores some pins as unhandled,
but user won't notice what goes wrong.  So, added a warning message
for the ignored pins as a hint.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-07-30 14:10:43 +02:00
Takashi Iwai
b08b1637ce ALSA: hda - Handle pin NID 0x1a on ALC259/269
The pin NID 0x1a should be handled as well as NID 0x1b.
Also added comments.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-07-30 14:09:38 +02:00
Takashi Iwai
697c373e34 ALSA: hda - Shut up pins at power-saving mode with Conexnat codecs
Call snd_hda_shutup_pins() for power-saving and reboot-notifier in
patch_conexant.c as well as other codecs.  This will reduce the pop
noise in power-save mode.

Reference: bnc#624896
	https://bugzilla.novell.com/show_bug.cgi?id=624896

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-07-30 11:28:02 +02:00
Takashi Iwai
954a29c881 ALSA: hda - Prefer VREF50 if BIOS sets for Realtek codecs
If BIOS sets up the input pin as VREF 50, use the value as is instead of
overriding forcibly to VREF 80.  This fixes the quality of inputs on
some devices like Packard-Bell M5210.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-07-30 10:55:44 +02:00
Takashi Iwai
5d4abf93ea ALSA: hda - Handle missing NID 0x1b on ALC259 codec
Since ALC259/269 use the same parser of ALC268, the pin 0x1b was ignored
as an invalid widget.  Just add this NID to handle properly.
This will add the missing mixer controls for some devices.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-07-30 10:51:10 +02:00
Takashi Iwai
757899acee ALSA: hda - Share digital I/O parser in patch_realtek.c
Make a helper function to parse the digital I/Os of all Realtek codecs
to simplify the code and to ensure the setups.
Also, initialize digital I/O pins properly in init callbacks.  Some BIOS
seem to leave pins uninitialized.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-07-30 10:48:14 +02:00
Takashi Iwai
ce503f38bd ALSA: hda - Increase the connection list size for ALC662
Some ALC662-compatible codecs like ALC892 may have more than 4
connections for the input source.  Use HDA_MAX_CONNECTIONS instead of
the fixed magic number 4.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-07-30 10:37:29 +02:00
Takashi Iwai
5aacc2186c ALSA: hda - Make error messages more verbose
Add a prefix and more information for error messages regarding the
connection-list in hda_codec.c.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-07-30 10:36:29 +02:00
Linus Torvalds
e271e872a8 Merge branch 'fix/hda' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
* 'fix/hda' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
  ALSA: hda - Add a PC-beep workaround for ASUS P5-V
  ALSA: hda - Assume PC-beep as default for Realtek
  ALSA: hda - Don't register beep input device when no beep is available
  ALSA: hda - Fix pin-detection of Nvidia HDMI
2010-07-29 15:21:07 -07:00
Kuninori Morimoto
3bc280708e ASoC: fsi: Add new funtion for SPDIF
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-07-29 10:28:49 -07:00
Kuninori Morimoto
265c770d03 ASoC: fsi: remove device id check
Current FSI driver id is not only 0

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-07-29 10:28:37 -07:00
Kuninori Morimoto
bced8f5a36 ASoC: fsi: remove unnecessary clock processing
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-07-29 10:28:27 -07:00
David Henningsson
150b432f44 ALSA: hda - Rename iMic to Int Mic on Lenovo NB0763
The non-standard name "iMic" makes PulseAudio ignore the microphone.
BugLink: https://launchpad.net/bugs/605101

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-07-29 15:37:28 +02:00
Takashi Iwai
b0485610d6 Merge branch 'fix/hda' into topic/hda 2010-07-29 15:32:34 +02:00
Takashi Iwai
dc1eae256c ALSA: hda - Add a PC-beep workaround for ASUS P5-V
ASUS P5-V provides a SSID that unexpectedly matches with the value
compilant with Realtek's specification.  Thus the driver interprets
it badly, resulting in non-working PC beep.

This patch adds a white-list for such a case; a white-list of known
devices with working PC beep.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-07-29 15:30:02 +02:00
Kulikov Vasiliy
9c29490246 sound: oss: msnd: check request_region() return value
request_region() may fail, if so return -EBUSY.

Signed-off-by: Kulikov Vasiliy <segooon@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-07-29 13:48:57 +02:00
Kulikov Vasiliy
fa95a6471f ALSA: msnd: check request_region() return value
request_region() may fail, if so return -EBUSY.

Signed-off-by: Kulikov Vasiliy <segooon@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-07-29 13:48:39 +02:00
Kulikov Vasiliy
ec9d04b2a8 ALSA: asihpi: check return value of get_user()
get_user() may fail, if so return -EFAULT.

Signed-off-by: Kulikov Vasiliy <segooon@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-07-29 12:26:28 +02:00
Kulikov Vasiliy
b3390ceab9 sound: oss: midi_synth: check get_user() return value
get_user() may fail, if so return -EFAULT.

Signed-off-by: Kulikov Vasiliy <segooon@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-07-29 12:25:06 +02:00
Kulikov Vasiliy
5157cc8113 ALSA: sb: check get_user() return value
get_user() may fail, if so return -EFAULT.

[Fixed one missing place by tiwai]

Signed-off-by: Kulikov Vasiliy <segooon@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-07-29 12:24:22 +02:00
Peter Ujfalusi
a577b318fc ASoC: tlv320dac33: Add support for automatic FIFO configuration
Platform parameter to enable automatic FIFO configuration when
the codec is in Mode1 or Mode7 FIFO mode.
When this mode is selected, the controls for changing
nSample (in Mode1), and UTHR (in Mode7) are not added.
The driver configures the FIFO configuration based on
the stream's period size in a way, that every burst will
read period size of data from the host.
In Mode7 we need to use a formula, which gives close enough
aproximation for the burst length from the host point
of view.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-07-29 10:21:11 +01:00
Peter Ujfalusi
f430a27f05 ASoC: tlv320dac33: Revisit the FIFO Mode1 handling
Replace the hardwired latency definition with platform data
parameter, and simplify the nSample parameter calculation.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-07-29 10:21:04 +01:00
Takashi Iwai
b6cbe517b9 ALSA: hda - Assume PC-beep as default for Realtek
Enable PC-beep as default for hardwares that aren't compliant with the
SSID value Realtek requires.  In such a case, better to enable the beep
to avoid a regression.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-07-28 17:43:36 +02:00
Takashi Iwai
8af2591d63 ALSA: hda - Don't register beep input device when no beep is available
We check now the availability of PC beep and skip the build of beep
mixers, but the driver still registers the input device.  This should
be checked as well.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-07-28 17:37:16 +02:00
Takashi Iwai
a39afc8eb4 Merge branch 'fix/hda' into topic/hda 2010-07-28 14:26:47 +02:00
Takashi Iwai
38faddb1af ALSA: hda - Fix pin-detection of Nvidia HDMI
The behavior of Nvidia HDMI codec regarding the pin-detection unsol events
is based on the old HD-audio spec, i.e. PD bit indicates only the update
and doesn't show the current state.  Since the current code assumes the
new behavior, the pin-detection doesn't work relialby with these h/w.

This patch adds a flag for indicating the old spec, and fixes the issue
by checking the pin-detection explicitly for such hardware.

Tested-by: Wei Ni <wni@nvidia.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-07-28 14:26:14 +02:00
Axel Lin
63818c448a ALSA: hpimsgx: fix wrong sizeof
The correct size should be sizeof(gRESP_HPI_SUBSYS_FIND_ADAPTERS),
sizeof(&gRESP_HPI_SUBSYS_FIND_ADAPTERS) is incorrect.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-07-28 11:53:03 +02:00
Peter Ujfalusi
b93cc9f19b ASoC: TWL4030: Capture route DAPM event fix
There is no need to handle POST_PMU, POST_PMD event with
the Capture Route widget.
It is enough to handle POST_REG event, since that will come
when the user changes the routing, and we will switch the needed
bits in the registers.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-07-27 11:43:40 +01:00
Takashi Iwai
7899f81fe4 Merge branch 'devel' of git.alsa-project.org:alsa-kernel into topic/misc 2010-07-27 10:16:04 +02:00
Ralf Baechle
93871603a7 SOUND: Au1000: Fix section mismatch
WARNING: sound/soc/au1x/snd-soc-au1xpsc-i2s.o(.data+0xa8): Section mismatch in reference from the variable au1xpsc_i2s_driver to the function .init.text:au1xpsc_i2s_drvprobe()
The variable au1xpsc_i2s_driver references
the function __init au1xpsc_i2s_drvprobe()
If the reference is valid then annotate the
variable with __init* or __refdata (see linux/init.h) or name the variable:
*_template, *_timer, *_sht, *_ops, *_probe, *_probe_one, *_console,

Signed-off-by: Ralf Baechle <ralf@linux-mips.org>
2010-07-26 19:08:15 +01:00
Takashi Iwai
7ccc3eface ALSA: hda - Fix max amp cap calculation for IDT/STAC codecs
The commit afbd9b8448
    ALSA: hda - Limit the amp value to write
introduced a regression for codec setups with amp offsets like IDT/STAC
codecs.  The limit value should be a raw value without offset calculation.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-07-26 17:00:15 +02:00
Kulikov Vasiliy
e5de3dfc39 sound: oss: waveartist: simplify waveartist_sleep()
waveartist_sleep() uses loop with schedule_timeout() to unconditionally
wait for msec. Use schedule_timeout_uninteruptible() instead.

Signed-off-by: Kulikov Vasiliy <segooon@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-07-26 10:33:41 +02:00
Kulikov Vasiliy
2232e23829 sound: oss: au1550_ac97: simplify au1550_delay()
au1550_delay() uses loop with schedule_timeout() to unconditionally wait
for msec. Use schedule_timeout_uninteruptible() instead.

Signed-off-by: Kulikov Vasiliy <segooon@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-07-26 10:33:31 +02:00
David Henningsson
2385b789f1 ALSA: hda - Ensure codec patch files are checked for the correct codec ID
Signed-off-by: David Henningsson <diwic@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-07-26 10:28:01 +02:00
Grant Likely
1ab1d63a85 of/platform: remove all of_bus_type and of_platform_bus_type references
Both of_bus_type and of_platform_bus_type are just #define aliases
for the platform bus.  This patch removes all references to them and
switches to the of_register_platform_driver()/of_unregister_platform_driver()
API for registering.

Subsequent patches will convert each user of of_register_platform_driver()
into plain platform_drivers without the of_platform_driver shim.  At which
point the of_register_platform_driver()/of_unregister_platform_driver()
functions can be removed.

Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
Acked-by: David S. Miller <davem@davemloft.net>
2010-07-24 09:57:52 -06:00
Grant Likely
4e4f62bf73 Merge commit 'v2.6.35-rc6' into devicetree/next
Conflicts:
	arch/sparc/kernel/prom_64.c
2010-07-24 09:49:13 -06:00
Kuninori Morimoto
a7e7cd5bd7 ASoC: da7210: Add HeadPhone Playback Volume control
HeadPhone Playback Volume control register of DA7210 has
reserved area. This patch considered it as mute.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-07-23 10:17:47 +01:00
Linus Torvalds
84b37df419 Merge branch 'fix/asoc' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
* 'fix/asoc' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
  ASoC: Select wm_hubs automatically for WM8994
  ASoC: Remove duplicate AUX definition from WM8776
  ASoC:: remove a redundant snd_soc_unregister_codec call in wm8988_register
  ASoC: wm8727: add a missing return in wm8727_platform_probe
  ASoC: fsi: fixup wrong value setting order of TDM
  ASoC: fsi: fixup clock inversion operation
2010-07-21 09:29:39 -07:00
Christian Dietrich
ff388f270d sound/oss: Remove dead CONFIG_SOFTOSS*
CONFIG_SOFTOSS* doesn't exist in Kconfig or somewhere
else, therefore removing all references for it from the source code.

Signed-off-by: Christian Dietrich <qy03fugy@stud.informatik.uni-erlangen.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-07-21 15:02:46 +02:00
Takashi Iwai
49e7042799 Merge branch 'for-2.6.36' of git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into topic/asoc 2010-07-21 15:01:07 +02:00
Peter Ujfalusi
01ea6ba2bc ASoC: TWL4030: Add configurable delay after digimic enable
When digital microphones are connected to twl, delay is
needed after enabling the digimic interface of the codec.
Add new parameter for the setup data, which can be used
to pass the apropriate delay in ms after the digimic
interface has been enabled.

Without certain delay (in certain HW configuration) the
beggining of the recorded sample contains a glitch, which
is generated by the digital microphones.

Delaying the micbias1, 2 (which is the bias for the digimic0
or 1) does not help, since the glitch is coming after
switching the digimic interface.

Reversing the micbias and digimic enable order does not
work either (in that case the wait need to be added after
the micbias enabled).

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-07-21 11:57:58 +01:00
Jaroslav Kysela
cd7643bfb7 ALSA: hda-intel - fix function_id rework (add missing bitmask)
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2010-07-20 12:13:25 +02:00
Mark Brown
d1ce6b200c ASoC: Unconditionally enable WM8994 AIF1ADC TDM mode
AIF1ADC TDM mode has no effect other than causing the ADCDAT line to
be tristated rather than driven low on clock cycles where there is no
data to be transmitted. If the clock cycle is idle then there should
be no devices using the data so tristating should have no adverse
effects.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-07-20 10:27:05 +01:00
Sekhar Nori
48519f0ae0 ASoC: davinci: let platform data define edma queue numbers
Currently the EDMA queue to be used by for servicing ASP through
internal RAM is fixed to EDMAQ_0 and that to service internal RAM
from external RAM is fixed to EDMAQ_1.

This may not be the desirable configuration on all platforms. For
example, on DM365, queue 0 has large fifo size and is more suitable
for video transfers. Having audio and video transfers on the same
queue may lead to starvation on audio side.

platform data as defined currently passes a queue number to the driver
but that remains unused inside the driver.

Fix this by defining one queue each for ASP and RAM transfers in the
platform data and using it inside the driver.

Since EDMAQ_0 maps to 0, thats the queue that will be used if
the asp queue number is not initialized. None of the platforms
currently utilize ping-pong transfers through internal RAM so that
functionality remains unchanged too.

This patch has been tested on DM644x and OMAP-L138 EVMs.

Signed-off-by: Sekhar Nori <nsekhar@ti.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-07-20 09:57:20 +01:00
Chanwoo Choi
5c519767b6 ASoC:Support Samsung SoC(S5P) in I2Sv2
This patch modify I2Sv2 driver to support Samsung SoC(S5PV210).

Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com>
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-07-20 09:53:36 +01:00
Mark Brown
3b89b22358 Merge branch 'for-2.6.35' into for-2.6.36 2010-07-20 09:52:25 +01:00
Chanwoo Choi
41f9a314af ASoC: Select wm_hubs automatically for WM8994
Otherwise all machine drivers need to do so.

Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com>
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-07-20 09:51:12 +01:00
Mark Brown
a3257ba869 ASoC: Implement WM8994 AIF1ADC2 paths
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-07-20 09:48:25 +01:00
Mark Brown
395e4b7362 ASoC: Explicitly disable DC servo on WM hubs headphone powerdown
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-07-20 09:48:07 +01:00
Eric Bénard
8a0bbbeb58 ASoC: eukrea-tlv320: add support for cpuimx35sd
Signed-off-by: Eric Bénard <eric@eukrea.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-07-20 09:47:28 +01:00
Jerone Young
ab85457f0a ALSA: hda - Add conexant quirk for AMD based Lenovo G series machines
This is a follow on patch adds support for AMD based Lenovo G series
machines, such as the Lenovo G555.

Signed-off-by: Jerone Young <jerone.young@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-07-19 18:47:38 +02:00
Kulikov Vasiliy
68bf57001f ALSA: riptide: check kzalloc() result
If kzalloc() fails exit with -ENOMEM.

Signed-off-by: Kulikov Vasiliy <segooon@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-07-19 17:59:26 +02:00
Kulikov Vasiliy
0b6d092c8e ALSA: echoaudio: check kmalloc() result
If kmalloc() fails exit with -ENOMEM.

Signed-off-by: Kulikov Vasiliy <segooon@gmail.com>
Ack-by: Giuliano Pochini <pochini@shiny.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-07-19 17:59:04 +02:00
Takashi Iwai
8d011cc7a9 Merge branch 'devel' of git.alsa-project.org:alsa-kernel into topic/misc 2010-07-19 17:42:09 +02:00
Jaroslav Kysela
9e216e8a40 ALSA: pcm core - add a safe check to the silence filling function
In situation when appl_ptr is far greater then hw_ptr, the hw_avail value
can be greater than buffer_size. Check for this.

Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2010-07-19 16:47:01 +02:00
Jaroslav Kysela
79c944ad13 ALSA: hda-intel - do not mix audio and modem function IDs
The function IDs are different for audio and modem. Do not mix them.
Also, show the unsolicited bit in the function_id register.

Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2010-07-19 16:46:56 +02:00
Uwe Kleine-König
25d1fbfdd9 fix comment typos concerning "challenge"
Signed-off-by: Uwe Kleine-König <u.kleine-koenig@pengutronix.de>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
2010-07-19 11:09:52 +02:00
James Bottomley
82f682514a pm_qos: Get rid of the allocation in pm_qos_add_request()
All current users of pm_qos_add_request() have the ability to supply
the memory required by the pm_qos routines, so make them do this and
eliminate the kmalloc() with pm_qos_add_request().  This has the
double benefit of making the call never fail and allowing it to be
called from atomic context.

Signed-off-by: James Bottomley <James.Bottomley@suse.de>
Signed-off-by: mark gross <markgross@thegnar.org>
Signed-off-by: Rafael J. Wysocki <rjw@sisk.pl>
2010-07-19 02:00:34 +02:00
Kulikov Vasiliy
50e8ce1469 ASoC: imx: check kzalloc() result and fix memory leak
If kzalloc() fails we must exit with -ENOMEM. Also we must free
allocated runtime->private_data on error as it would be lost on next
call to snd_imx_open().

Signed-off-by: Kulikov Vasiliy <segooon@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-07-17 19:45:56 +01:00
Kulikov Vasiliy
51b6dfb627 ASoC: imx: check kzalloc() result and fix memory leak
If kzalloc() fails we must exit with -ENOMEM. Also we must free
allocated runtime->private_data on error as it would be lost on next
call to snd_imx_open().

Signed-off-by: Kulikov Vasiliy <segooon@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-07-17 19:45:56 +01:00
Kulikov Vasiliy
55938b106f ASoC: davinci: check kzalloc() result (typo)
The code checks 'davinci_vc' after kzalloc() and do not checks
'davinci_vcif_dev' that kzalloc() result is assigned to. It seems that
it is a typo (autocompletion?).

Signed-off-by: Kulikov Vasiliy <segooon@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-07-17 19:45:56 +01:00
Kuninori Morimoto
3c2ef841c0 ASoC: fsi: Add specified ID for soc-audio
Specified ID is necessary, when some codecs are used with FSI.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-07-17 19:45:56 +01:00
Mark Brown
d947837410 Merge branch 'for-2.6.35' into for-2.6.36 2010-07-17 19:45:43 +01:00
Mark Brown
3c0709396d ASoC: Remove duplicate AUX definition from WM8776
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Cc: stable@kernel.org
2010-07-17 19:44:40 +01:00
Jorge Eduardo Candelaria
0fad4ed7b2 ASoC: TWL6040: Correct widget handling for drivers
In order to reduce pop-noise at powering up/down of the DACs and Drivers,
these components have to be handled in a specific sequence. Headset,
Handsfree, and Earphone drivers are now registered as PGA components to
ensure DACs are enabled first.

Also, add a delay to leave time for DACs to settle before
continuing power up/down sequence.

Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria@ti.com>
Signed-off-by: Margarita Olaya Cabrera <magi.olaya@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-07-17 19:27:18 +01:00
Eliot Blennerhassett
e2768c0c22 ALSA: asihpi - Avoid useless assignment of returned index values.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-07-16 11:34:23 +02:00
Eliot Blennerhassett
604a440a9d ALSA: asihpi - Avoid using c99 uintX types.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-07-16 11:33:47 +02:00
Eliot Blennerhassett
8d4bbee77e ALSA: asihpi - HPI version 4.04.01
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-07-16 11:31:37 +02:00
Kulikov Vasiliy
315e8f7501 ALSA: asihpi: fix sign bug
bytes_per_sec is unsigned, so if snd_pcm_format_width() return error we
would not see it.

Signed-off-by: Kulikov Vasiliy <segooon@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-07-16 08:30:08 +02:00
Michael Witten
1d8c1100fb ALSA: Kconfig: SND_AC97_POWER_SAVE description improvement
The description has been expanded to explain the time-out
value provided by the power_save module parameter.

Signed-off-by: Michael Witten <mfwitten@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-07-15 13:43:44 +02:00
Michael Witten
7a53cd16d4 Kconfig: fixo typo in "Xilinx'"
Signed-off-by: Michael Witten <mfwitten@gmail.com>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
2010-07-15 09:37:39 +02:00
Mark Brown
5164d74d74 ASoC: Handle read failures in codec_reg
When a device is powered down volatile registers can't be read so
attempts to display codec_reg will show error values, and obviously
it is also possible for there to be hardware errors too. Check for
errors from reads and display them more clearly when formatting
codec_reg.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-07-14 20:13:09 +01:00
Mark Brown
03b0dc02cf Merge branch 'for-2.6.35' into for-2.6.36 2010-07-14 20:12:57 +01:00
Axel Lin
cecb66fddf ASoC:: remove a redundant snd_soc_unregister_codec call in wm8988_register
snd_soc_unregister_codec is called twice if snd_soc_register_dai fail.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-07-14 20:12:31 +01:00
Axel Lin
c555b028f1 ASoC: wm8727: add a missing return in wm8727_platform_probe
otherwise the error path will always be executed.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-07-14 20:12:18 +01:00
Arnd Bergmann
992cbf7438 sound/oss-msnd-pinnacle: ioctl needs the inode
This broke in sound/oss: convert to unlocked_ioctl, when I missed one
of the ioctl functions still using the inode pointer.

Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-07-14 15:14:02 +02:00
Takashi Iwai
840b64c080 ALSA: hda - Add support of dual-ADCs for Realtek ALC275
Some VAIO models with ALC275 have dual ADCs for both internal and external
mics, and the driver needs to switch one of them appropriately.
This patch adds a basic support for this functionality, dynamic switching
between two ADCs per jack plug state.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-07-13 22:49:01 +02:00
Manuel Lauss
0c74a939d8 ASoC: au1x: fix section mismatch in psc-i2s.c
Annotate platform probe callback with __devinit instead of plain __init.

Signed-off-by: Manuel Lauss <manuel.lauss@googlemail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-07-13 12:39:14 +01:00
arnaud.patard@rtp-net.org
b424ec9533 ASoC: kirkwood-i2s: Handle mute/unmute playback/record
The controller has mute/unmute capability and some bootloader may mute
them at boot. If it's not handled, all things will seem to be working
but no sound will come out of the speaker/headphone.

Signed-off-by: Arnaud Patard <arnaud.patard@rtp-net.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-07-13 12:37:09 +01:00
arnaud.patard@rtp-net.org
dfe4c93627 ASoC: Fix kirkwood i2s mono playback
Kirkwood controller needs to be informed if the audio stream is mono
or not. Failing to do so will result in playing at the wrong speed.

Signed-off-by: Arnaud Patard <arnaud.patard@rtp-net.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-07-13 12:37:09 +01:00
Kuninori Morimoto
ccad7b44cc ASoC: fsi: Fixup for master mode
This patch add hw_params to snd_soc_dai_ops,
because board specific set_rate is needed
when FSI was used as master mode.

This patch remove fsi_clk_ctrl from fsi_dai_startup,
because clock should be disabled before set_rate.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-07-13 12:35:39 +01:00
Kuninori Morimoto
d78541473d ASoC: fsi: Add pr_err for noticing unsupported access
This patch didn't use dev_err,
because it is difficult to get struct device here.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-07-13 12:35:38 +01:00
Kuninori Morimoto
73b92c1fc0 ASoC: fsi: Change struct fsi_regs to fsi_core
Many registers which were grouped by category were added in FSI2.
To make easy to switch FSI/FSI2, fsi_core was added instead of fsi_regs.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-07-13 12:35:37 +01:00
Kuninori Morimoto
a7ffb52bb3 ASoC: fsi: remove noisy CR_FMT macro
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-07-13 12:35:36 +01:00
Kuninori Morimoto
a09370cb8c ASoC: fsi: remove un-used variable on fsi_dai_startup
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-07-13 12:35:35 +01:00
Joe Perches
4726a57b8c ASoC: Remove unnecessary casts of private_data
Signed-off-by: Joe Perches <joe@perches.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-07-13 12:34:06 +01:00
Joe Perches
8ff23610a6 ASoC: Remove unnecessary casts of private_data
Signed-off-by: Joe Perches <joe@perches.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-07-13 12:33:59 +01:00
Mark Brown
4d53952a39 Merge branch 'for-2.6.35' into for-2.6.36 2010-07-13 12:29:10 +01:00
Kuninori Morimoto
637727838a ASoC: fsi: fixup wrong value setting order of TDM
channel size should be set before setting register value

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-07-13 12:26:26 +01:00
Kuninori Morimoto
b427b44cc8 ASoC: fsi: fixup clock inversion operation
Clock inversion should be specified by each flags bit.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-07-13 12:26:26 +01:00
Peter Ujfalusi
27eeb1feed ASoC: TWL4030: DAC power optimization
Restructure the DAPM connections in order to enable
only the needed DAC (out of four in twl4030 series).
I need to keep the 'AIF Enable' supply connected to
the L2/R2 digital path, since the digital loopback
needs AIF and APLL running.
If no valid route available, than none of the DAC will
be powered, but the AIF and APLL is going to be enabled.
Furthermore, if only one audio path have valid route,
than only the corresponding DAC will be powered.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsomicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-07-13 11:30:12 +01:00
Peter Ujfalusi
8b0d31532e ASoC: TWL4030: Fix for digital loopback gain range
When the gain is configured using dB value it was
not possible to use -24dB since the loopback got
muted instead of -24dB.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsomicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-07-13 11:30:05 +01:00
Linus Torvalds
7e48c02829 Merge branch 'fix/hda' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
* 'fix/hda' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
  ALSA: hda - Restore cleared pin controls on resume
2010-07-12 14:44:43 -07:00
Arnd Bergmann
d209974cdc sound/oss: convert to unlocked_ioctl
These are the final conversions for the ioctl file operation so we can remove
it in the next merge window.

Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-07-12 22:36:47 +02:00
Uwe Kleine-König
a7ce2e0d04 fix comnment/printk typos concerning "empty"
Signed-off-by: Uwe Kleine-König <u.kleine-koenig@pengutronix.de>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
2010-07-12 18:03:50 +02:00
Arnd Bergmann
90dc763fef sound: push BKL into open functions
This moves the lock_kernel() call from soundcore_open
to the individual OSS device drivers, where we can deal
with it one driver at a time if needed, or just kill
off the drivers.

All core components in ALSA already provide
adequate locking in their open()-functions
and do not require the big kernel lock, so
there is no need to add the BKL there.

Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-07-12 17:41:05 +02:00
Clemens Ladisch
32e0191d79 ALSA: HDA: VT1708S: fix Smart5.1 mode
Correctly configure bidirectional pins when resuming; do not power down
widgets when they are needed for Smart5.1 output; and on 3-jack boards,
create the streams and controls needed for six channels.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Reported-and-tested-by: Viliam Kubis <viliam.kubis@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-07-12 17:25:45 +02:00
Clemens Ladisch
395c61d196 ALSA: via82xx: allow changing the initial DXS volume
As per-stream volume controls, the DXS controls are not intended to
adjust the overall sound level and so are initialized every time
a stream is opened.  However, there are special situations where one
wants to reduce the overall volume in the digital domain, i.e., before
the AC'97 codec's PCM volume control.  To allow this, add a module
parameter that sets the initial DXS volume.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Tested-by: Soeren D. Schulze <soeren.d.schulze@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-07-12 17:25:27 +02:00
Clemens Ladisch
d32d552e66 ALSA: usb-audio: silence a superfluous warning
It is not advisable to print a warning when a device does not support
setting the sample rate because this is perfectly valid for devices with
a single rate or where rates are implicitly changed by selecting another
alternate setting.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-07-09 15:08:12 +02:00
Takashi Iwai
f8fb27bd4a Merge branch 'fix/hda' into topic/hda 2010-07-09 10:09:00 +02:00
Takashi Iwai
afbd9b8448 ALSA: hda - Limit the amp value to write
Check the amp max value at put callbacks and set the upper limit
so that the driver won't write any invalid value over the defined
range.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-07-09 10:08:57 +02:00
Takashi Iwai
3507e2a8f1 ALSA: hda - Add beep mixer support to Conexant codecs
Added the beep mixer controls to Conexant codecs.
They simply control the digital beep generator widget.

For cx5047, I couldn't find any beep generator, so it's not implemented
there.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-07-09 10:08:56 +02:00
Takashi Iwai
ac0547dc62 ALSA: hda - Restore cleared pin controls on resume
Many codecs now clear the pin controls at suspend via snd_hda_shutup_pins()
for reducing the click noise at power-off.  But this leaves some pins
uninitialized, and they'll be never recovered after resume.

This patch adds the proper recovery of cleared pin controls on resume.
Also it adds a check of bus->shutdown so that pins won't be cleared at
module unloading.

Reference: Kernel bug 16339
	http://bugzilla.kernel.org/show_bug.cgi?id=16339

Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-07-09 08:42:29 +02:00
Mark Brown
66b47fdb85 ASoC: Implement WM8994 OPCLK support
The WM8994 can output a clock derived from its internal SYSCLK, called
OPCLK.  The rate can be selected as a sysclk, with a division from the
SYSCLK rate specified (multiplied by 10 since a division of 5.5 is
supported) and the clock can be disabled by specifying a divisor of
zero.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-07-09 08:50:12 +09:00
Mark Brown
e88ff1e6db ASoC: Include WM8994 GPIO and interrupt registers in codec_reg
Very handy for debug.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-07-09 01:37:06 +09:00
Takashi Iwai
7645054f18 Merge branch 'fix/misc' into for-linus 2010-07-08 16:55:26 +02:00
Takashi Iwai
b492c4e895 Merge branch 'fix/hda' into for-linus 2010-07-08 16:55:02 +02:00
Raffaele Recalcati
d9823ed9fa ASoC: DaVinci: More accurate continuous serial clock for McBSP (I2S)
i2s_accurate_sck switch can be used to have a better approximate
    sampling frequency.
    The clock is an externally visible bit clock and it is named
    i2s continuous serial clock (I2S_SCK).
    The trade off is between more accurate clock (fast clock)
    and less accurate clock (slow clock).
    The waveform will be not symmetric.
    Probably it is possible to get a better algorithm for calculating
    the divider, trying to keep a slower clock as possible.

    This patch has been developed against the
        http://git.kernel.org/pub/scm/linux/kernel/git/khilman/linux-davinci.git
    git tree and has been tested on bmx board (similar to dm365 evm, but using
    uda1345 as external audio codec).

Signed-off-by: Raffaele Recalcati <raffaele.recalcati@bticino.it>
Signed-off-by: Davide Bonfanti <davide.bonfanti@bticino.it>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Acked-by: Sudhakar Rajashekhara <sudhakar.raj@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-07-06 23:54:07 +09:00
Raffaele Recalcati
ec63755337 ASoC: DaVinci: Added selection of clk input pin for McBSP
When McBSP peripheral gets the clock from an external pin,
    there are three possible chooses, MCBSP_CLKX, MCBSP_CLKR
    and MCBSP_CLKS.
    evm-dm365 uses MCBSP_CLKR, instead in bmx board I have a different
    hardware connection and I use MCBSP_CLKS, so I have added
    this possibility.

    This patch has been developed against the:
        http://git.kernel.org/pub/scm/linux/kernel/git/khilman/linux-davinci.git
    git tree and has been tested on bmx board (similar to dm365 evm)

Signed-off-by: Raffaele Recalcati <raffaele.recalcati@bticino.it>
Signed-off-by: Davide Bonfanti <davide.bonfanti@bticino.it>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Acked-by: Sudhakar Rajashekhara <sudhakar.raj@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-07-06 23:54:06 +09:00
Raffaele Recalcati
a4c8ea2dda ASoC: DaVinci: Added two clocking possibilities to McBSP (I2S)
Added two clocking options for dm365 McBSP peripheral when used
    with I2S timings, that are SND_SOC_DAIFMT_CBS_CFS (the cpu generates
    clock and frame sync) and SND_SOC_DAIFMT_CBS_CFM (the cpu gets clock
    from external pin and generates frame sync).
    A slave clock management can be important when the external codec needs
    the system clock and the bit clock synchronized (tested with uda1345).
    This patch has been developed against the:
        http://git.kernel.org/pub/scm/linux/kernel/git/khilman/linux-davinci.git
    git tree and has been tested on bmx board (similar to dm365 evm, but using
    uda1345 as external audio codec).

Signed-off-by: Raffaele Recalcati <raffaele.recalcati@bticino.it>
Signed-off-by: Davide Bonfanti <davide.bonfanti@bticino.it>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Acked-by: Sudhakar Rajashekhara <sudhakar.raj@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-07-06 23:54:06 +09:00
Maurus Cuelenaere
088fbab406 ASoC: Invert speaker enabling behaviour in SmartQ sound driver
The speaker was enabled when the headphone was plugged in, which isn't the
wanted behaviour so correct this.

Signed-off-by: Maurus Cuelenaere <mcuelenaere@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-07-06 23:54:06 +09:00
Eliot Blennerhassett
f978d36da4 ALSA: asihpi - Remove unneeded ;
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-07-06 08:20:43 +02:00
Eliot Blennerhassett
36ed8bdd86 ALSA: asihpi - Minor HPI error handling fixes
Handle errors in tuner level caching,
Ccorrect error code for aesebu rx status.

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-07-06 08:20:21 +02:00
Eliot Blennerhassett
108ccb3f0f ALSA: asihpi - Change compander API and tidy
Compander API changed to one function per parameter.
Factor out some common code for stereo log value reading.
Make some more entity functions static.

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-07-06 08:19:56 +02:00
Eliot Blennerhassett
3843914635 ALSA: asihpi - Add ASI5200 family
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-07-06 08:19:35 +02:00
Eliot Blennerhassett
1dd6aaaafc ALSA: asihpi - Use version string instead of printf formatting
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-07-06 08:19:06 +02:00
Eliot Blennerhassett
168f1b07cc ALSA: asihpi - HPI API updates
Remove some deprecated items.
Change compander api to one function per parameter.
Add a version string define.

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-07-06 08:18:27 +02:00
Mark Brown
db059c0f6e ASoC: Automatically manage ALC coefficients for WM8960
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-07-06 08:46:10 +09:00
John Kacur
171d9f7d78 soundcore_open: Reduce the area BKL coverage
Most of this function is protected by the sound_loader_lock.
We can push down the BKL to this call out err = file->f_op->open(inode,file);

In order to build the sound core without the BKL, we
will need to push the lock_kernel() call into the ~20
device drivers that register their file operations.

Signed-off-by: John Kacur <jkacur@redhat.com>
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Acked-by: Alan Cox <alan@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-07-05 18:07:30 +02:00
Takashi Iwai
f189efcd1c ALSA: hda - Enable beep on Realtek codecs with PCI SSID override
When the PCI SSID gives an overriding SKU assno, PC-beep bit isn't
detected (since it's located over 16bit), resulting in no PC beep.
Also, many devices seem ignoring the requirement by Realtek's spec
for SSID numbers, and it also confuses the PC beep detection.

This patch assumes the PC beep is available on every machine with
PCI SSID override.  It's a regression fix from 2.6.34.

Reference: Kernel bug 16251
	http://bugzilla.kernel.org/show_bug.cgi?id=16251

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-07-05 17:28:17 +02:00
Mark Brown
afd6d36a0d ASoC: Automatically manage DAC deemphasis rate for WM8960
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-07-05 23:41:18 +09:00
Mark Brown
4faaa8d968 ASoC: Remove current WM8960 deemphasis control
It will be replaced with automatic deemphasis rate configuration but since
we have an enumeration table in this driver this is done in a separate
commit to make the renumbering of the enumeration items clear.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-07-05 23:37:17 +09:00
Mark Brown
9af8381023 ASoC: Fix sorting of Makefile and Kconfig
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-07-05 23:35:29 +09:00
Takashi Iwai
65ee2ba310 Merge branch 'devel' of git://git.alsa-project.org/alsa-kernel into topic/misc 2010-07-05 15:37:27 +02:00
Maurus Cuelenaere
ce93a37028 ASoC: Add SmartQ sound driver
This adds sound support for the SmartQ board.

The hardware consists of a S3C6410 coupled with a WM8987 over I²S. The WM8750
driver is used for driving the WM8987, as they are register compatible.

Signed-off-by: Maurus Cuelenaere <mcuelenaere@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-07-04 18:04:12 +09:00
Maurus Cuelenaere
0d9c15e45b ASoC: codec: Add WM8987 device id to WM8750 driver
The WM8987 codec is register compatible with the WM8750, so just add it to the
SPI and I²C device table.

Signed-off-by: Maurus Cuelenaere <mcuelenaere@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-07-04 18:02:07 +09:00
Kuninori Morimoto
a300de3cff ASoC: ak4642: Add Digital Playback Volume control
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-07-01 17:08:47 +01:00
Vladimir Zapolskiy
338de9d9da ASoC: uda134x: correct bias level setup for codecs family
For UDA1341 codec power control is managed in STATUS1 register, and
for all other codecs in DATA011 register.

Signed-off-by: Vladimir Zapolskiy <vzapolskiy@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-06-30 14:46:56 +01:00
Vladimir Zapolskiy
ed632ad3b8 ASoC: uda134x: add DATA011 register found in codecs family
In UDA1340, UDA1344 and UDA1345 codecs there is one more functional
register in part of DATA0 tranfser. For UDA1341 this register
coincides with EA register.

Signed-off-by: Vladimir Zapolskiy <vzapolskiy@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-06-30 14:46:56 +01:00
Mark Brown
af51b5c0f0 Merge remote branch 'takashi/topic/asoc' into for-2.6.36 2010-06-30 14:46:53 +01:00
Grant Likely
1636f8ac2b sparc/of: Move of_device fields into struct pdev_archdata
This patch moves SPARC architecture specific data members out of
struct of_device and into the pdev_archdata structure.  The reason
for this change is to unify the struct of_device definition amongst
all the architectures.  It also remvoes the .sysdata, .slot, .portid
and .clock_freq properties because they aren't actually used by
anything.

A subsequent patch will replace struct of_device entirely with struct
platform_device and the of_platform support code will share common
routines with the platform bus (but the bus instances themselves can
remain separate).

This patch also adds 'struct resources *resource' and num_resources
to match the fields defined in struct platform_device.  After this
change, 'struct platform_device' can be used as a drop-in replacement
for 'struct of_platform'.

This change is in preparation for merging the of_platform_bus_type
with the platform_bus_type.

Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
Acked-by: David S. Miller <davem@davemloft.net>
Cc: Stephen Rothwell <sfr@canb.auug.org.au>
2010-06-28 12:41:33 -07:00