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10340 commits

Author SHA1 Message Date
Julia Lawall ca9380fd68 ALSA: sound/core/pcm_compat.c: adjust array index
Convert array index from the loop bound to the loop index.

A simplified version of the semantic patch that fixes this problem is as
follows: (http://coccinelle.lip6.fr/)

// <smpl>
@@
expression e1,e2,ar;
@@

for(e1 = 0; e1 < e2; e1++) { <...
  ar[
- e2
+ e1
  ]
  ...> }
// </smpl>

Signed-off-by: Julia Lawall <julia@diku.dk>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-28 15:12:02 +02:00
Andy Whitcroft 8d34e6d3ec sound: oss: rename local change_bits to avoid powerpc bitsops.h definition
This collides with powerpc exported functions from bitops.h.  Rename the
local copy in the oss soundblaster mixer and ad1848 driver.

Signed-off-by: Andy Whitcroft <apw@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-27 19:16:05 +02:00
Takashi Iwai c48a8fb0d3 ALSA: hda - Fix duplicated DAC assignments for Realtek
Copying hp_pins and speaker_pins from line_out_pins may confuse the
parser, and it can lead to duplicated initializations for the same pin
with a wrong DAC assignment.  The problem appears in 3.0 kernel code.

Cc: <stable@kernel.org> (for 3.0)
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-27 16:44:40 +02:00
Dan Carpenter ae6ff61e43 ALSA: asihpi - off by one in asihpi_hpi_ioctl()
"adapter" is used as an array index in the adapters[] array so
the off by one would make us read past the end.

1c073b6797 "ALSA: asihpi - Remove spurious adapter index check"
reverted Dan Rosenberg's check that would have prevented the
overflow here.

Signed-off-by: Dan Carpenter <error27@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-27 16:06:16 +02:00
Takashi Iwai 60a6a8425a ALSA: hda - Fix Oops with Realtek quirks with NULL adc_nids
Somce quirk models don't set adc_nids but let the parser filling it.
But the recent code has unnecessary NULL-checks of spec->input_mux,
and it resulted in NULL dereferences.
This patch fixes that regression.

Reported-and-tested-by: Oliver Neukum <oneukum@suse.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-27 16:05:30 +02:00
Eliot Blennerhassett 767cd365b2 ALSA: asihpi - bug fix pa use before init.
Fixes bug introduced by 1c073b67.
Also declare pa local to block in which it is used.

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-27 10:08:26 +02:00
Vitaliy Kulikov 45eebda7b4 ALSA: hda - Add support for vref-out based mute LED control on IDT codecs
This patch also registers all necessary callbacks to support mute LED
only when such control is enabled. And it keeps codec AFG in D0 or D1
state all the time when aggressive power managemnt is enabled for vref-out
control (and mute LED) work correctly.

Signed-off-by: Vitaliy Kulikov <Vitaliy.Kulikov@idt.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-27 08:51:10 +02:00
Takashi Iwai 636f78581d Merge branch 'fix/asoc' into for-linus 2011-07-26 17:47:05 +02:00
Tim Howe 56487c279f ALSA: hda - Cirrus Logic CS421x support
This update includes the changes necessary for supporting the
CS421x family of codecs.  Previously this file only supported
the CS420x family of codecs.

This file also contains init verbs to correct several issues in
the CS421x hardware.

Behavior between the CS421x and CS420x codec families is similar,
so several functions have been reused with "if" statements to
determine which codec family (CS421x or CS420x) is present.

Also, this file will be updated sometime in the near future in
order to add support for a system using CS421x that requires
mono mix on the speaker output only.

[Fix const usages and adaption for new APIs by tiwai]

Signed-off-by: Tim Howe <tim.howe@cirrus.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-26 17:21:25 +02:00
Takashi Iwai 4d7fbdbcb1 ALSA: hda - Allow codec-specific set_power_state ops
The procedure for codec D-state change may have exceptional cases
depending on the codec chip, such as a longer delay or suppressing D3.

This patch adds a new codec ops, set_power_state() to override the system
default function.  For ease of porting, snd_hda_codec_set_power_to_all()
helper function is extracted from the default set_power_state() function.

As an example, the Conexant codec-specific delay is removed from the
default routine but moved to patch_conexant.c.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-26 17:21:24 +02:00
Takashi Iwai e581f3dba5 ALSA: hda - Add post_suspend patch ops
Add a new ops, post_suspend(), which is called after suspend() ops is
performed.  This is called only in the case of the real PM suspend, and
the codec driver can use this for further changing of D-state or
clearing the LED, etc.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-26 17:21:23 +02:00
Takashi Iwai 2a43952a99 ALSA: hda - Make CONFIG_SND_HDA_POWER_SAVE depending on CONFIG_PM
It makes little sense to enable power-saving without PM.
This removes SND_HDA_NEEDS_RESUME define so that we can use CONFIG_PM
in all places.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-26 17:21:18 +02:00
Vitaliy Kulikov 7df1ce1a81 ALSA: hda - Make sure mute led reflects master mute state
This patch adds checking of mute state on all outputs besides just
speakers to calculate the master mute state for mute led support.
It also renames and splits the function that does it for better code
clarity.

Signed-off-by: Vitaliy Kulikov <Vitaliy.Kulikov@idt.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-26 09:39:31 +02:00
Vitaliy Kulikov d02667e620 ALSA: hda - Fix invalid mute led state on resume of IDT codecs
Codec state is not restored immediately on resume but on the first
access when power-save is enabled.  That leads to an invalid mute led
state after resume until either sound is played or some control is
changed.  This patch adds a possibility for a vendor specific patch to
restore codec state immediately after resume if required.  And it adds
code to restore IDT codecs state immediately on resume on HP systems
with mute led support.

Signed-off-by: Vitaliy Kulikov <Vitaliy.Kulikov@idt.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-26 09:38:36 +02:00
Mark Brown a0c27ab242 ASoC: Revert "ASoC: SAMSUNG: Add I2S0 internal dma driver"
This reverts commit d7c3e9525a as it does
not currently build due to missing dependencies in the Samsung tree.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Jassi Brar <jassisinghbrar@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-07-24 22:47:39 +01:00
Vitaliy Kulikov 0c27c18052 ALSA: hda - Add support of the 4 internal speakers on certain HP laptops
Signed-off-by: Vitaliy Kulikov <Vitaliy.Kulikov@idt.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-24 13:36:24 +02:00
Eliot Blennerhassett acb03d440b ALSA: Make snd_pcm_debug_name usable outside pcm_lib
Formatting a PCM name is useful for module debug too.
Add snd_prefix when making function public.

[minor coding-style fixes by tiwai]

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-24 13:34:32 +02:00
Takashi Iwai 8f398ae72f ALSA: hda - Fix DAC filling for multi-connection pins in Realtek parser
Fix a regression in the DAC filling code in patch_realtek.c.  The already
filled DACs in multiout.dac_nids[] were ignored because of num_dacs=0,
thus always pointed to the first DAC.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-23 18:57:11 +02:00
Takashi Iwai 76531d4166 Merge branch 'topic/hda' into for-linus 2011-07-22 08:43:27 +02:00
Takashi Iwai 7d339ae997 Merge branch 'topic/misc' into for-linus 2011-07-22 08:43:24 +02:00
Takashi Iwai 13b137ef03 Merge branch 'topic/asoc' into for-linus 2011-07-22 08:43:19 +02:00
Takashi Iwai 000477a0fe ALSA: asihpi - Replace with snd_ctl_boolean_mono_info()
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-22 07:57:44 +02:00
Eliot Blennerhassett 509a714744 ALSA: asihpi - HPI version 4.08
HPI Version is used to check for firmware compatibility.
This version  will accept 4.08.xx released firmware,
and will also accept 4.09.xx beta firmware

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-22 07:55:02 +02:00
Eliot Blennerhassett fe0aa88eec ALSA: asihpi - Add volume mute controls
Mute functionality was recently added to the DSP firmware

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-22 07:54:20 +02:00
Eliot Blennerhassett c830613574 ALSA: asihpi - Control name updates
Add names corresponding to new HPI node types.
Shorten some names so that constructed names don't overflow the
maximum name length.

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-22 07:53:45 +02:00
Eliot Blennerhassett 3d0591eee4 ALSA: asihpi - Use size_t for sizeof result
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-22 07:53:07 +02:00
Eliot Blennerhassett 5ddc5bef5c ALSA: asihpi - Explicitly include mutex.h
Because mutex is used in adapter struct defined here.

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-22 07:52:31 +02:00
Eliot Blennerhassett b7f12482ca ALSA: asihpi - Add new node and message defines
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-22 07:52:15 +02:00
Eliot Blennerhassett 33162d2dfa ALSA: asihpi - Make local function static
Fixes a sparse warning.

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-22 07:52:02 +02:00
Eliot Blennerhassett 938c565a82 ALSA: asihpi - Fix minor typos and spelling
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-22 07:51:41 +02:00
Eliot Blennerhassett 4bf8cff05a ALSA: asihpi - Remove unused structures, macros and functions
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-22 07:50:57 +02:00
Eliot Blennerhassett 1c073b6797 ALSA: asihpi - Remove spurious adapter index check
Subsystem requests don't have or need a valid adapter index.
The adapter index is already checked further on, before it is used to index
the adapters array. (Reverts 4a122c10f)

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-22 07:50:44 +02:00
Eliot Blennerhassett 0a17e99307 ALSA: asihpi - Revise snd_pcm_debug_name, get rid of DEBUG_NAME macro
Work towards moving the function into alsa common header.

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-22 07:50:03 +02:00
Eliot Blennerhassett 95a4c6e785 ALSA: asihpi - DSP code loader API now independent of OS
The loader API has been revised so that OS specific data is kept
local to hpidspcd.c, and the public API is unchanged across OSes.

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-22 07:49:23 +02:00
Eliot Blennerhassett 58fbf77ff5 ALSA: asihpi - Remove controlex structs and associated special data transfer code
Some cobranet control data would not fit in an original HPI message.
Now that HPI is able to transfer larger messages, this special handling
is no longer required.

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-22 07:46:14 +02:00
Eliot Blennerhassett c6c2c9aba1 ALSA: asihpi - Increase request and response buffer sizes
Allow for up to 256 bytes of extra data on top of standard hpi
request and response sizes.

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-22 07:45:26 +02:00
Eliot Blennerhassett 82b5774fe0 ALSA: asihpi - Give more meaningful name to hpi request message type
Having a 'request message' makes more sense than a 'message message'

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-22 07:45:06 +02:00
David G Turner 6a6d822e12 ALSA: usb-audio - Add quirk for Roland / BOSS BR-800
Add support for Roland/BOSS BR-800 (0582:011e) to snd-usb-audio driver.

This allows playback and recording, which has been tested and found to
work. The third interface should be MIDI (MTC/SMPTE?) for DAW interface
and is set as per ME-25, but this has not been tested. SDHC card access
is already supported by usb-storage for Backup/Rhythm Editor/Wave
Convertor mode which should not conflict with this.

Signed-off-by: David G Turner <dgturner@iee.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-21 19:00:57 +02:00
Takashi Iwai a353fbb179 ALSA: hda - Remove a superfluous argument of via_auto_init_output()
"force" argument is always true, so let's strip it off.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-21 14:24:25 +02:00
Takashi Iwai 020066d1ec ALSA: hda - Fix indep-HP path (de-)activation for VT1708* codecs
This patch fixes non-working indep-HP control on VT1708* codecs.
The problems are that via_independent_hp_put() wasn't fixed to follow
the recent change of three HP paths, and hp_indep_path didn't contain
the amp nids of mixer elements.

Together with the fixes, a few code clean-ups are done.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-21 13:55:10 +02:00
Liam Girdwood 4805608ac1 ASoC: dapm - Add methods to retrieve snd_card and soc_card from dapm context.
In preparation for ASoC Dynamic PCM (AKA DSP) support.

Provide convenience methods to retrieve the soc_card or snd_card from a
DAPM context.

Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-20 21:15:51 +01:00
Sangbeom Kim d7c3e9525a ASoC: SAMSUNG: Add I2S0 internal dma driver
I2S in Exynos4 and S5PC110(S5PV210) has a internal dma.
It can be used low power audio mode and 2nd channel transfer.
This patch can support idma.

Signed-off-by: Sangbeom Kim <sbkim73@samsung.com>
Acked-by: Jassi Brar <jassisinghbrar@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Acked-by: Jassi Brar <jassisinghbrar@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-20 20:58:12 +01:00
Sangbeom Kim 61100f405d ASoC: SAMSUNG: Modify I2S driver to support idma
Previously, I2S driver only can support system dma.
In this patch, i2s driver can support internal dma too.
IDMA h/w configuration is initialized on idma.c

Signed-off-by: Sangbeom Kim <sbkim73@samsung.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Acked-by: Jassi Brar <jassisinghbrar@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-20 20:58:07 +01:00
Rajashekhara, Sudhakar 82d1d52103 ASoC: davinci: add missing break statement
In davinci_vcif_trigger() function, a break() statement was missing
causing the davinci_vcif_stop() function to be called as a fallback
after calling davinci_vcif_start().

Signed-off-by: Rajashekhara, Sudhakar <sudhakar.raj@ti.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
2011-07-20 20:51:35 +01:00
Rajashekhara, Sudhakar 3012f43eaf ASoC: davinci: fix codec start and stop functions
According to DM365 voice codec data sheet at [1], before starting
recording or playback, ADC/DAC modules should follow a reset and
enable cycle. Writing a 1 to the ADC/DAC bit in the register resets
the module and clearing the bit to 0 will enable the module. But the
driver seems to be doing the reverse of it.

[1] http://focus.ti.com/lit/ug/sprufi9b/sprufi9b.pdf

Signed-off-by: Rajashekhara, Sudhakar <sudhakar.raj@ti.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
2011-07-20 20:51:23 +01:00
Mark Brown 3198b9eb51 ASoC: Acknowledge WM8962 interrupts before acting on them
This closes the small race between a status being read in response to an
interrupt and clearing the interrupt, meaning that if the status changes
between those periods we might not get a reassertion of the interrupt.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-07-20 20:50:08 +01:00
Wolfram Sang 09bddc8eb2 ASoC: sgtl5000: guide user when regulator support is needed
Print a hint when the user has a setup where CONFIG_REGULATOR is really
needed to make the driver work.

Signed-off-by: Wolfram Sang <w.sang@pengutronix.de>
Tested-by: Dong Aisheng <b29396@freescale.com>
Tested-by: Shawn Guo <shawn.guo@freescale.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-19 16:19:30 +01:00
Wolfram Sang e94a4062c8 ASoC: sgtl5000: refactor registering internal ldo
The code for registering the internal ldo was present twice. Turn it
into a function instead. Also, inform the user if LDO is used now.

Signed-off-by: Wolfram Sang <w.sang@pengutronix.de>
Tested-by: Dong Aisheng <b29396@freescale.com>
Tested-by: Shawn Guo <shawn.guo@freescale.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-19 16:19:22 +01:00
Wolfram Sang 1c8371d61e ASoC: core: make comments fit the code
In one comment, cpu_dai was mentioned although codec_dai was used in the
code. Also, fix the name for the card dai list which has no seperation
into card_dai and codec_dai.

Signed-off-by: Wolfram Sang <w.sang@pengutronix.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-19 16:19:15 +01:00
Mark Brown 7be4ba24a3 ASoC: Mark cache as dirty when suspending
Since quite a few drivers are not managing to flag the cache as needing
to be resynced after suspend and it's a reasonable thing to do flag the
cache as needing sync automatically when suspending.

The expectation is that systems will mainly only keep the CODEC powered
when doing audio through the CODEC so we won't actually suspend the
device anyway; drivers which want to can override this behaviour when
they resume.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Cc: stable@kernel.org
2011-07-19 16:16:00 +01:00