Commit Graph

19408 Commits

Author SHA1 Message Date
Lars-Peter Clausen
00200107a2 ASoC: Move card field form platform/codec to component
Both the snd_soc_codec and snd_soc_platform struct do have a pointer to the
parent card and both handle this pointer in mostly the same way. This patch
moves the card field to the component level which will allow further code
consolidation between platforms and CODECS.

Since there are only a handful of users of the snd_soc_codec struct's card field
(and none of the snd_soc_platform's) these are update in this patch as well,
which allows it to be removed from the snd_soc_codec struct.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-22 23:15:57 +01:00
Lars-Peter Clausen
093c4e5c3c ASoC: tegra: Replace instances of rtd->codec->card with rtd->card
No need to go via the CODEC to get a pointer to the card. This will help to
eventually remove the card field from the snd_soc_codec struct.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-22 23:15:57 +01:00
Lars-Peter Clausen
9898e1ccf5 ASoC: Remove per card platform list
The platform_dev_list was added in commit f0fba2ad1b ("ASoC: multi-component -
ASoC Multi-Component Support") and while platforms are added and remove from
that list it is otherwise unused. This patch removes it again.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-22 23:15:57 +01:00
Takashi Iwai
c5d019c395 ALSA: hda - Fix loopback noise on Dell XPS 15
Dell XPS 15 (1028:05fe) gives the bad feedback noise from analog
loopback line even if the channels are muted.  The similar problem has
been seen on XPS 13, and it was fixed by simply disabling loopback.
The same fixup was confirmed to be applicable and fix the problem on
XPS 15, too.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=80821
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-07-22 17:33:32 +02:00
Takashi Sakamoto
eb12f72ee7 ALSA: bebob: Correction for return value of special_clk_ctl_put() in error
This commit is a supplement to my previous patch.
http://mailman.alsa-project.org/pipermail/alsa-devel/2014-July/079190.html

The special_clk_ctl_put() still returns 0 in error handling case. It should
return -EINVAL.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-07-22 17:31:28 +02:00
Takashi Sakamoto
f77ac91e8e ALSA: bebob: Correction for return value of .put callback
This commit is for correction of my misunderstanding about return value of
.put callback in ALSA Control interface.

According to 'Writing ALSA Driver' (*1), return value of the callback has
three patterns; 1: changed, 0: not changed, an negative value: fatal error.

But I misunderstood that it's boolean; zero or nonzero.

*1: Writing an ALSA Driver (2005, Takashi Iwai)
http://www.alsa-project.org/main/index.php/ALSA_Driver_Documentation

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-07-22 16:21:54 +02:00
Takashi Sakamoto
5a0438f4a6 ALSA: bebob: Use different labels for digital input/output
This commit uses different labels for control elements of digital input/output
interfaces to correct my misunderstanding about M-Audio Firewire 1814 and
ProjectMix I/O.

According to user manuals for these two models, they have two modes for
digital input; one is S/PDIF in both of optical and coaxial interfaces,
another is ADAT in optical interface only.

But in current implementation, a control element for it reduced labels which
a control element for digital output uses because of my misunderstanding
that optical interface is not available for digital input with S/PDIF mode.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-07-22 16:21:30 +02:00
Takashi Sakamoto
9014011664 ALSA: bebob: Fix a missing to unlock mutex in error handling case
In error handling case, special_clk_ctl_put() returns without unlock_mutex(),
therefore the mutex is still locked. This commit moves mutex_lock() after
the error handling case.

This commit is my solution for this post.

[PATCH -next] ALSA: bebob: Fix missing unlock on error in special_clk_ctl_put()
https://lkml.org/lkml/2014/7/20/12

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-07-22 16:21:17 +02:00
David Henningsson
214eef76eb ALSA: hda - Make vendor quirks lowest prio for ALC2xx
Add a new quirk table to make sure that pin quirks have a higher
priority than quirks that apply to an entire vendor.

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-07-22 14:26:53 +02:00
David Henningsson
f5662e1cbf ALSA: hda - Refactor quirk picking and change quirk priority
Previously, calling one quirk function first and another later
would make the latter one take priority, this is now changed
to make the former take priority.

By adding two special values for fixup_id we can also get rid of the
fixup_forced flag.

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-07-22 14:26:50 +02:00
Daniel Drake
9dc12862da ALSA: hda - Add mic fixup for Gigabyte BXBT-2807
The Gigabyte BRIX BXBT-2707 is a mini-PC with Realtek ALC283 HDA,
exposing a single headset jack.

However, the default pin config information only suggests
that one pin is connected: a HP out jack (pin 0x21, default
config 0x04211010).

The microphone input is behind pin 0x19, which has default
config 0x411111f0 (i.e. unused), so it does not show up in
userspace, and no microphone input is possible via the headset.

Override the pin config so that the headset mic can be used.

[rearranged the fixup entry position by tiwai]

Signed-off-by: Daniel Drake <drake@endlessm.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-07-22 12:48:04 +02:00
David Henningsson
423044744a ALSA: hda - Add mute LED pin quirk for HP 15 touchsmart
This makes the mute LED work on a HP 15 touchsmart machine.

BugLink: https://bugs.launchpad.net/bugs/1334950
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-07-22 12:34:43 +02:00
Takashi Iwai
c1ab70c10b Merge branch 'topic/monotonic' into for-next 2014-07-22 12:11:34 +02:00
Dan Murphy
e295a4a43e ASoC: tas2552: Fix PM sequencing
In the pm suspend/resume it is better
to disable the GPIO after the regmap_cache
setting calls so that if the call is interrupted
the new reg values will be cached and set on resume.

Also add pm_runtime_put in the remove call.

Signed-off-by: Dan Murphy <dmurphy@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-21 13:18:44 +01:00
Takashi Iwai
e58c295c04 ALSA: pcm: Add tstamp_type and proto to sw_params compat layer
I forgot to add the new fields in sw_params to 32bit compat layer.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-07-21 09:08:18 +02:00
Takashi Iwai
589008106b ALSA: pcm: Introduce protocol version field to sw_params
For controlling the new fields more strictly, add sw_params.proto
field indicating the protocol version of the user-space.  User-space
should fill the SNDRV_PCM_VERSION value it's built with, then kernel
can know whether the new fields should be evaluated or not.

And now tstamp_type field is evaluated only when the valid value is
set there.  This avoids the wrong override of tstamp_type to zero,
which is SNDRV_PCM_TSTAMP_TYPE_GETTIMEOFDAY.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-07-21 09:07:46 +02:00
Olof Johansson
f57c0e049a Samsung cleanup for v3.17
: Most of them are for exynos SoCs, remove useless
 codes and update for PMU consolidation.
 
 - remove unnecessary header file in mach-exynos/pmu.c
 - remove unused code in mach-exynos/common.h
 - remove mach-exynos/regs-pmu.h dependency from PD
 - remove file path from comment section in mach-exynos
 
 - move SYSREG definitions into mach-exynos/regs-sys.h
 
 - add mapping PMU base address via DT for PMU cleanup
 
 - use staic in mach-exynos/common.h
 - update Samsung UART config options for low-level debug
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Merge tag 'samsung-cleanup' of git://git.kernel.org/pub/scm/linux/kernel/git/kgene/linux-samsung into next/cleanup

Merge "Samsung cleanup for v3.17" from Kukjin Kim:

Most of them are for exynos SoCs, remove useless codes and update for
PMU consolidation.

- remove unnecessary header file in mach-exynos/pmu.c
- remove unused code in mach-exynos/common.h
- remove mach-exynos/regs-pmu.h dependency from PD
- remove file path from comment section in mach-exynos

- move SYSREG definitions into mach-exynos/regs-sys.h

- add mapping PMU base address via DT for PMU cleanup

- use staic in mach-exynos/common.h
- update Samsung UART config options for low-level debug

* tag 'samsung-cleanup' of git://git.kernel.org/pub/scm/linux/kernel/git/kgene/linux-samsung:
  ARM: EXYNOS: Add support for mapping PMU base address via DT
  ARM: EXYNOS: Remove "linux/bug.h" from pmu.c
  ARM: EXYNOS: Remove regs-pmu.h header dependency from pm_domain
  ARM: EXYNOS: Remove file path from comment section
  ARM: EXYNOS: Move SYSREG definition into sys-reg specific file
  ARM: EXYNOS: Make exynos machine_ops as static
  ARM: EXYNOS: Remove unused code in common.h
  ARM: debug: Update Samsung UART config options
  + Linux 3.16-rc5

Signed-off-by: Olof Johansson <olof@lixom.net>
2014-07-19 15:01:08 -07:00
Mark Brown
d0ab92d63c Merge remote-tracking branches 'asoc/fix/samsung', 'asoc/fix/sgtl5000', 'asoc/fix/simple' and 'asoc/fix/tlv320aic3x' into asoc-linus 2014-07-17 21:45:47 +01:00
Mark Brown
a3e048a5bb Merge remote-tracking branches 'asoc/fix/fsl-sai', 'asoc/fix/intel', 'asoc/fix/max98090' and 'asoc/fix/s6000' into asoc-linus 2014-07-17 21:45:44 +01:00
Mark Brown
1165d95644 Merge remote-tracking branches 'asoc/fix/adau1701', 'asoc/fix/adsp', 'asoc/fix/arizona', 'asoc/fix/blackfin', 'asoc/fix/cs42l56', 'asoc/fix/davinci' and 'asoc/fix/debugfs' into asoc-linus 2014-07-17 21:45:41 +01:00
Mark Brown
d1c46ef858 Merge remote-tracking branch 'asoc/fix/rcar' into asoc-linus 2014-07-17 21:45:40 +01:00
Mark Brown
1b6c03df8d Merge remote-tracking branch 'asoc/fix/pcm' into asoc-linus 2014-07-17 21:45:39 +01:00
Mark Brown
e122b3aec7 ASoC: Fixes for v3.16
Quite a few build coverage fixes in here among the usual small driver
 fixes includling the sigmadsp change from Lars - moving the driver to
 separate modules per bus (which is basically just code motion) avoids
 issues with some combinations of buses being enabled.
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Merge tag 'asoc-v3.16-rc1' into asoc-linus

ASoC: Fixes for v3.16

Quite a few build coverage fixes in here among the usual small driver
fixes includling the sigmadsp change from Lars - moving the driver to
separate modules per bus (which is basically just code motion) avoids
issues with some combinations of buses being enabled.

# gpg: Signature made Thu 19 Jun 2014 11:57:31 BST using RSA key ID 7EA229BD
# gpg: Good signature from "Mark Brown <broonie@sirena.org.uk>"
# gpg:                 aka "Mark Brown <broonie@debian.org>"
# gpg:                 aka "Mark Brown <broonie@kernel.org>"
# gpg:                 aka "Mark Brown <broonie@tardis.ed.ac.uk>"
# gpg:                 aka "Mark Brown <broonie@linaro.org>"
# gpg:                 aka "Mark Brown <Mark.Brown@linaro.org>"
2014-07-17 21:45:38 +01:00
Brian Austin
a046558719 ASoC: Fix SOC_DOUBLE_R_SX_TLV volume mixer arguments
Remove unnecessary bit shifts.
Correct min value to match datasheet.
Num steps = number of steps between min and max.

Reported-by: Ryan Harvey <ryan.harvey@cirrus.com>
Signed-off-by: Ryan Harvey <ryan.harvey@cirrus.com>
Signed-off-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-17 19:59:20 +01:00
Brian Austin
95b47f8de7 ASoC: cs42l56: Fix stereo channel register assignment for Headphone and LineOut volume mixers
Stereo Headphone and LineOut volume mixers are now attached to HPA+HPB, LOA+LOB.

Reported-by: Ryan Harvey <ryan.harvey@cirrus.com>
Signed-off-by: Ryan Harvey <ryan.harvey@cirrus.com>
Signed-off-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-17 19:57:50 +01:00
Nicolin Chen
4800f88b61 ASoC: fsl_sai: Fix incorrect register writing in fsl_sai_isr()
In the rx irq handling part, we should clear the flags in RCSR not TCSR.

Signed-off-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-17 18:10:41 +01:00
Nicolin Chen
eff952b733 ASoC: fsl_sai: Reset FIFOs after disabling TE/RE
SAI will not clear their FIFOs after disabling TE/RE. Therfore, the driver
should take care the task so as not to let useless data remain in the FIFO.

Signed-off-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-17 18:10:41 +01:00
Dan Murphy
5df7f71d5c ASoC: tas2552: Support TI TAS2552 Amplifier
Support the TI TAS2552 Class D amplifier.

The TAS2552 is a high efficiency Class-D audio
power amplifier with advanced battery current
management and an integrated Class-G boost
The device constantly measures the
current and voltage across the load and provides a
digital stream of this information.

Signed-off-by: Dan Murphy <dmurphy@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-17 17:57:05 +01:00
Scott Jiang
30443408fd ASoC: blackfin: use samples to set silence
The third parameter for snd_pcm_format_set_silence needs the number
of samples instead of sample bytes.

Signed-off-by: Scott Jiang <scott.jiang.linux@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org
2014-07-17 12:27:56 +01:00
Lars-Peter Clausen
3d9501aff3 ASoC: Remove unused 'r' variable from dapm_connect_dai_link_widgets()
It was accidentally added in commit 44ba2641 ("ASoC: dapm: Add support for DAI
multicodec").

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-17 10:12:42 +01:00
Liam Girdwood
afdb74fd70 ASoC: Intel: Add Broadwell Machine support
Add support for Broadwell based machines with SST DSP audio.

Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Jie Yang <yang.jie@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-16 23:16:51 +01:00
Mark Brown
0ca0c2d868 Merge remote-tracking branch 'asoc/topic/rt286' into HEAD 2014-07-16 23:16:47 +01:00
Benoit Cousson
93e6958a36 ASoC: pcm: Add soc_dai_hw_params helper
Add a function helper to factorize the hw_params code.

Suggested by Lars-Peter Clausen <lars@metafoo.de>

Signed-off-by: Benoit Cousson <bcousson@baylibre.com>
Tested-by: Lars-Peter Clausen <lars@metafoo.de>
Reviewed-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-16 23:06:27 +01:00
Benoit Cousson
8151d5e602 ASoC: compress: Prevent multicodec for compressed stream
Multiple codecs does not make sense in the case of compressed
stream.

Exit with error if it happens.

Signed-off-by: Benoit Cousson <bcousson@baylibre.com>
Tested-by: Lars-Peter Clausen <lars@metafoo.de>
Reviewed-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-16 23:06:27 +01:00
Benoit Cousson
44ba2641b7 ASoC: dapm: Add support for DAI multicodec
Add multicodec support in soc-dapm.c

Signed-off-by: Benoit Cousson <bcousson@baylibre.com>
Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
Signed-off-by: Fabien Parent <fparent@baylibre.com>
Tested-by: Lars-Peter Clausen <lars@metafoo.de>
Reviewed-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-16 23:06:26 +01:00
Benoit Cousson
2e5894d737 ASoC: pcm: Add support for DAI multicodec
Add multicodec support in soc-pcm.c

Signed-off-by: Benoit Cousson <bcousson@baylibre.com>
Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
Signed-off-by: Fabien Parent <fparent@baylibre.com>
Tested-by: Lars-Peter Clausen <lars@metafoo.de>
Reviewed-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-16 23:04:52 +01:00
Benoit Cousson
88bd870f02 ASoC: core: Add initial support for DAI multicodec
DAI link assumes a one to one mapping between CPU DAI and CODEC. In
some cases, the same CPU DAI can be connected to several codecs.
This is the case for example, if you connect two mono codecs to the
same I2S link in order to have a stereo card.
The current ASoC implementation does not allow such setup.

Add support for DAI links composed of a single CPU DAI and multiple
CODECs. Sound cards have to pass the CODECs array in the corresponding
DAI link through a new 'snd_soc_dai_link_component' struct. Each CODEC in
this array is described in the same manner single CODEC DAIs are
(either DT/OF node or codec_name).

Multi-codec links are not supported in the case of CODEC to CODEC links.
Just print a warning if it happens.

Based on an original code done by Misael.

Signed-off-by: Benoit Cousson <bcousson@baylibre.com>
Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
Signed-off-by: Fabien Parent <fparent@baylibre.com>
Tested-by: Lars-Peter Clausen <lars@metafoo.de>
Reviewed-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-16 22:58:49 +01:00
Mark Brown
f7acb3694a Merge remote-tracking branch 'asoc/topic/component' into asoc-multi 2014-07-16 22:58:45 +01:00
Omair Mohammed Abdullah
7523a27168 ASoC: core: add a helper for extended byte controls using TLV
ALSA supports arbitrary length TLVs for each kcontrol that can be used
to pass metadata about the control (e.g. volumes, enum information). The
same transport mechanism is now used for arbitrary length data by
defining a new helper.

Signed-off-by: Omair Mohammed Abdullah <omair.m.abdullah@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-16 22:43:59 +01:00
Jie Yang
249adddb1a ASoC: Intel: Update FW version readback
Update FW version readback. IPC_GLB_GET_FW_VERSION reads back
the ABI version whilst the release version is in the mailbox.
Update to use mailbox version for info logging.

Signed-off-by: Jie Yang <yang.jie@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-16 22:26:57 +01:00
Daniel Mack
8267525c99 ASoC: mcasp: don't override bclk divider if it was provided by the machine
If a machine driver provides an BCLK divider to the McASP driver, skip the
automatic calculation.

This fixes machines on which the physical sample transport always works
in 32 bits, even though not all of them are actually used.
snd_soc_params_to_bclk() will fail to address such cases properly.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-16 22:15:15 +01:00
Peter Ujfalusi
f3f9cfa8a1 ASoC: davinci-mcasp: Use dmaengine based platform driver for AM335x/447x
Use the edma-pcm with AM335x and AM437x SoCs.
Keep using the davinci-pcm for daVinci devices, they can be switched to use
the dmaengine based driver later when they are verified to work correctly.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Tested-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-16 22:09:13 +01:00
Peter Ujfalusi
e8ffacee0a ASoC: edma-pcm: Add empty functions for !CONFIG_SND_EDMA_SOC builds
So drivers mixing with other platform drivers, like davinci-mcasp do not
need to fiddle with CONFIG_SND_EDMA_SOC in their code.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Tested-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-16 22:09:08 +01:00
Peter Ujfalusi
24600b5472 ASoC: edma-pcm: Remove PCM_INFO_BATCH and add PCM_INFO_NO_PERIOD_WAKEUP flag
The SNDRV_PCM_INFO_BATCH will be added back based on the dmaengine driver's
capabilities.
Patches for edma dmaengine driver has been prepared to suppress the
interrupts. We can add this flag right away and have the benefit of not
having interrupts during audio activity when the edma and ASoC patches
are in the same tree.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Tested-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-16 22:08:55 +01:00
Mark Brown
91fff60b7d Merge remote-tracking branch 'asoc/fix/davinci' into asoc-davinci 2014-07-16 22:08:39 +01:00
Linus Torvalds
2da2944740 sound fixes for 3.16-rc6
Things seem to calm down so far, just a small few HD-audio fixes
 (regression fixes and a new codec ID addition) popping up.
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Merge tag 'sound-3.16-rc6' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "Things seem to calm down so far, just a small few HD-audio fixes
  (regression fixes and a new codec ID addition) popping up"

* tag 'sound-3.16-rc6' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
  ALSA: hda - Fix broken PM due to incomplete i915 initialization
  ALSA: hda - Revert stream assignment order for Intel controllers
  ALSA: hda - Add new GPU codec ID 0x10de0070 to snd-hda
  ALSA: hda: Fix build warning
2014-07-16 06:48:08 -10:00
Takashi Iwai
2d9772ef65 ALSA: hda - Add NULL check to all PM ops in hda_intel.c
Since devptr can be NULL due to asynchronous probe, all PM ops should
have NULL checks at the beginning.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-07-16 16:32:01 +02:00
Dan Carpenter
6217e5ede2 ALSA: compress: fix an integer overflow check
I previously added an integer overflow check here but looking at it now,
it's still buggy.

The bug happens in snd_compr_allocate_buffer().  We multiply
".fragments" and ".fragment_size" and that doesn't overflow but then we
save it in an unsigned int so it truncates the high bits away and we
allocate a smaller than expected size.

Fixes: b35cc82258 ('ALSA: compress_core: integer overflow in snd_compr_allocate_buffer()')
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-07-16 15:27:03 +02:00
Lv Zheng
1d34f3ef4b ASoC: intel: Cleanup useless ACPI inclusion.
The sst-haswell-dsp.c is an ACPI independent file, this patch removes
ACPI header files for it.

Signed-off-by: Lv Zheng <lv.zheng@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-16 13:06:41 +01:00
Olof Johansson
cb6a0fd4e4 This is a purge of all things <mach/gpio.h>, now I never
want to see it again.
 
 - Remove the need for <mach/gpio.h> from S5P
 - Kill CONFIG_NEED_MACH_GPIO_H
 - Kill remnants of ARM_GPIOLIB_COMPLEX
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Merge tag 'gpio-h-purge' of git://git.kernel.org/pub/scm/linux/kernel/git/linusw/linux-gpio into next/cleanup

This is a purge of all things <mach/gpio.h>, now I never
want to see it again.

- Remove the need for <mach/gpio.h> from S5P
- Kill CONFIG_NEED_MACH_GPIO_H
- Kill remnants of ARM_GPIOLIB_COMPLEX

* tag 'gpio-h-purge' of git://git.kernel.org/pub/scm/linux/kernel/git/linusw/linux-gpio:
  ARM: delete old reference to ARM_GPIOLIB_COMPLEX
  ARM: kill CONFIG_NEED_MACH_GPIO_H
  ARM: mach-s5p: get rid of all <mach/gpio.h> headers
  ARM: s5p: cut the custom ARCH_NR_GPIOS definition

Signed-off-by: Olof Johansson <olof@lixom.net>
2014-07-15 18:29:47 -07:00
Takashi Iwai
8695a003a5 ALSA: hda - Add the pin fixup for HP Envy TS bass speaker
NID 0x10 seems corresponding to the bass speaker.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-07-15 16:59:56 +02:00
Takashi Iwai
0cea76f339 ALSA: control: Define SNDRV_CTL_TLV_OP_* constants
Instead of hard-coded magic numbers, define constants for op_flag to
tlv callbacks.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-07-15 16:31:01 +02:00
Takashi Iwai
1618e84aa8 Merge branch 'for-linus' into for-next
Since init_failed flag was moved to struct hda_intel, its access in
the commit [4da63c6f: ALSA: hda - Fix broken PM due to incomplete
i915 initialization] is also replaced with hda->init_failed
appropriately.
2014-07-15 15:28:43 +02:00
Takashi Iwai
4da63c6fc4 ALSA: hda - Fix broken PM due to incomplete i915 initialization
When the initialization of Intel HDMI controller fails due to missing
i915 kernel symbols (e.g. HD-audio is built in while i915 is module),
the driver discontinues the probe.  However, since the probe was done
asynchronously, the driver object still remains, thus the relevant PM
ops are still called at suspend/resume. This results in the bad access
to the incomplete audio card object, eventually leads to Oops or stall
at PM.

This patch adds the missing checks of chip->init_failed flag at each
PM callback in order to fix the problem above.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=79561
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-07-15 15:19:43 +02:00
Arnd Bergmann
dd2380e31c ASoC: samsung: remove unused DMA data
The s3c_dma_client structures and the 'ch' and 'ops' members in
s3c_dma_params were only used by the legacy DMA driver and serve
no function any more. This removes any reference to them.

Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-14 19:53:43 +01:00
Arnd Bergmann
8f501c7b44 ASoC: samsung: s3c24xx dmaengine follow-up
Commit ae602456e8 ("ASoC: samsung: drop support for legacy
S3C24XX DMA API") removed the old code for the samsung specific
DMA interfaces, now that everybody can use dmaengine.

This picks up the few remaining pieces left over by that patch:

The most important one is the removal of the dma_data->ops->started()
calls in ac97. My understanding is that these are only required
for drivers that do not support cyclic transfers, which the new dma
engine driver now does, so we can simply remove them. This would also
fix at least one bug in the ac97 driver on newer machines, which
currently gives us a NULL pointer dereference from trying to call
dma_data->ops->started().

Further, we must no longer 'select' S3C2410_DMA, which conflicts
with the dmaengine driver. The SND_S3C_DMA symbol is now
useless, because it is always selected, so we can remove it
and build the dmaengine support unconditionally.

Finally, we should not 'select' S3C24XX_DMAC or S3C64XX_PL080,
which may have additional dependencies. This replaces it with
'depends on', to be more conservative.

Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-14 19:53:10 +01:00
Vinod Koul
e310fb9141 ASoC: Intel: mfld: add generic parameter interface
This interface will be used by subsequent patches to set/get parameters from DSP

Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-14 19:44:52 +01:00
Vinod Koul
8813e66db7 ASoC: Intel: mfld: add dsp error codes
DSP returns error codes for IPC return so add them in driver

Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-14 19:44:47 +01:00
Vinod Koul
1ad0e33060 ASoC: Intel: add sst shim register start-end variables
the shim registers start and end can be useful while parsing the shim addresses,
so add these

Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-14 19:44:47 +01:00
Bard Liao
23c4fd5c97 ASoC: rt286: make rt286_i2c_driver static
Signed-off-by: Fengguang Wu <fengguang.wu@intel.com>
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-14 19:14:14 +01:00
Jie Yang
85e63007bb ASoC: Intel: Start with all memory banks disabled
All required banks are enabled during boot procedure.

Signed-off-by: Jie Yang <yang.jie@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-14 19:12:09 +01:00
Jie Yang
548793232f ASoC: Intel: Use a table for ADSP SRAM shift
Use a table for ADSP IRAM/DRAM bit shift.

Signed-off-by: Jie Yang <yang.jie@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-14 19:12:09 +01:00
Jie Yang
15446c0b8d ASoC: Intel: Merge wild cat point ADSP DRAM regions
Merge D-SRAM0 D-SRAM1 D-SRAM2 to D-SRAM, for wild cat point ADSP mem regions.

Signed-off-by: Jie Yang <yang.jie@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-14 19:12:09 +01:00
Jie Yang
c761b58306 ASoC: Intel: Add dummy read for SRAM block enable
Add dummy read after each block enable, to workaround
SRAM write missing bytes issue.

Signed-off-by: Jie Yang <yang.jie@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-14 19:11:27 +01:00
Jie Yang
8e89761876 ASoC: Intel: Cleanup HSW pcm format support
This change removes unsupported formats from System,
Capture and Loopback FE DAIs.
Also it fixes S24_LE support on all DAIs.
While at this fix 24 bit flag for BYT as well.

Signed-off-by: Jie Yang <yang.jie@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org
2014-07-14 19:09:05 +01:00
Kukjin Kim
3a549fbffc ASoC: samsung: no more support for S5P6440 and S5P6450 SoCs
This patch removes s5p64x0 related WM8580 because of removing support
for s5p64x0 SoCs.

Signed-off-by: Kukjin Kim <kgene.kim@samsung.com>
[pebolle@tiscali.nl: reported missing dependency for smdk6450]
Reported-by: Paul Bolle <pebolle@tiscali.nl>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-14 19:04:57 +01:00
Arnd Bergmann
d21e1b1605 ASoC: samsung: add explicit i2c/spi dependencies
I got another build error from SND_SOC_SMARTQ selecting
SND_SOC_WM8750 without ensuring that I2C is enabled,
in this example, i2c is a loadable module:

sound/built-in.o: In function `wm8750_i2c_probe':
:(.text+0x3e6c0): undefined reference to `devm_regmap_init_i2c'
sound/built-in.o: In function `wm8750_exit':
:(.exit.text+0x3f4): undefined reference to `i2c_del_driver'
sound/built-in.o: In function `wm8750_modinit':
:(.init.text+0x13d0): undefined reference to `i2c_register_driver'

This changes the samsund ASoC Kconfig to have explicit
I2C dependencies for each driver that uses an i2c bus and
SPI dependencies for the ones using those.

Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-14 19:04:38 +01:00
Himangi Saraogi
9cb0fe9b0e ASoC: omap-dmic: Use devm_clk_get
This patch introduces the use of managed interfaces like devm_clk_get
and does away with the clk_puts in the probe and remove functions. A
label is also done away with.

Signed-off-by: Himangi Saraogi <himangi774@gmail.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-14 19:02:45 +01:00
xujianqun
4c5258acfe ASoC: rockchip: fix duplicated argument to && or ||
This resulted from an unfortunate copy/paste to similar DEFINE name.

Detected automagically by kbuild.

Signed-off-by: xujianqun <xjq@rock-chips.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-14 18:28:51 +01:00
Takashi Iwai
03fe805a09 Merge branch 'topic/monotonic' into for-next 2014-07-14 18:13:41 +02:00
Takashi Iwai
cd50065b3b ALSA: hda - Revert stream assignment order for Intel controllers
We got a regression report for 3.15.x kernels, and this turned out to
be triggered by the fix for stream assignment order.  On reporter's
machine with Intel controller (8086:1e20) + VIA VT1802 codec, the
first playback slot can't work with speaker outputs.

But the original commit was actually a fix for AMD controllers where
no proper GCAP value is returned, we shouldn't revert the whole
commit.  Instead, in this patch, a new flag is introduced to determine
the stream assignment order, and follow the old behavior for Intel
controllers.

Fixes: dcb32ecd9a ('ALSA: hda - Do not assign streams in reverse order')
Reported-and-tested-by: Steven Newbury <steve@snewbury.org.uk>
Cc: <stable@vger.kernel.org> [v3.15+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-07-14 10:45:31 +02:00
Linus Torvalds
28b1fae568 Merge branch 'fixes' of git://git.infradead.org/users/vkoul/slave-dma
Pull slave-dmaengine fixes from Vinod Koul:
 "We have two small fixes.  First one from Daniel to handle 0-length
  packets for usb cppi dma.  Second by Russell for imx-sdam cyclic
  residue reporting"

* 'fixes' of git://git.infradead.org/users/vkoul/slave-dma:
  Update imx-sdma cyclic handling to report residue
  dma: cppi41: handle 0-length packets
2014-07-12 22:24:50 -07:00
Sylwester Nawrocki
b97c60abf9 ASoC: samsung-i2s: Maintain CDCLK settings across i2s_{shutdown/startup}
Currently configuration of the CDCLK pad is being overwritten in
the i2s_shutdown() callback in order to gate the SoC output clock.
However if an ASoC machine driver doesn't restore that clock
settings each time after opening the sound device this results
in the CDCLK pin being permanently configured into input mode.
I.e. the output clock will always stay disabled.
Fix that by saving the CDCLKCON bit state in i2s_shutdown() and
and restoring it in the i2s_startup() callback.

Signed-off-by: Chen Zhen <zhen1.chen@samsung.com>
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-11 14:04:03 +01:00
xujianqun
1b21572f8f ASoC: rockchip: add missing module.h include
Missing module.h in linux directory will cause compile error with randconfig,
like as:
sound/soc/rockchip/rockchip_i2s.c:526:20: error: expected declaration
specifiers or ‘...’ before string constant
MODULE_DESCRIPTION("ROCKCHIP IIS ASoC Interface");

Signed-off-by: xujianqun <xjq@rock-chips.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-11 13:32:26 +01:00
Takashi Iwai
5646eda585 ALSA: pcm: Add timestamp type to sw_params
For allowing adjusting the timestamp type on the fly, add it to
sw_params.  The existing ioctl is still kept for compatibility.

Along with this, increment the PCM protocol version.

The extension was suggested by Clemens Ladisch.

Acked-by: Jaroslav Kysela <perex@perex.cz>
Reviewed-by: Mark Brown <broonie@linaro.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-07-10 16:58:15 +02:00
Takashi Iwai
2408c219df ALSA: pcm: simplify snd_pcm_tstamp()
No functional change.

Acked-by: Jaroslav Kysela <perex@perex.cz>
Reviewed-by: Mark Brown <broonie@linaro.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-07-10 16:58:01 +02:00
Charles Keepax
c8badda8ee ASoC: arizona: Update handling for input change on an active FLL
Currently, the driver places no restrictions on changes that can be
applied to an active FLL. However, it is only possible to change the
input for an active FLL, to change the output the FLL should be stopped
and then recofigured. This patch disallows changes in output frequency
and adds some additional handling to ensure the output remains
consistent across an input transition.

Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-10 12:09:16 +02:00
Charles Keepax
5e39a50bf8 ASoC: arizona: FLL freerun only required whilst disabling
The FLL freerun is only required whilst we disable the FLL not the
entire time the FLL is disabled. This patch moves the FLL freerun
disable from the enable sequence to the disable sequence.

Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-10 12:09:16 +02:00
Charles Keepax
c393aca94f ASoC: arizona: Correct return value of arizona_is_enabled_fll
arizona_is_enabled_fll currently returns a bool, but can throw an error.
The error will be basically ignored and we will treat the FLL as already
on. This patch changes the return to be an int and adds error code to
propagate the error up to the callback.

Reported-by: Anil Kumar <anil.kumar@wolfsonmicro.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-10 12:09:15 +02:00
Charles Keepax
613124ce64 ASoC: arizona: Coding standards, remove unneeded brackets
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-10 12:09:15 +02:00
Charles Keepax
35a730a079 ASoC: arizona: Correct relationship between VCO corner and Fref
When configuring the FLL we must ensure that the reference clock passed
to the FLL is under a certain limit. This limit was specified
incorrectly in the current code, this patch corrects this. Although the
error will only be encountered in some edge cases.

Reported-by: Ryo Tsutsui <ryo.tsutsui@wolfsonmicro.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-10 12:09:15 +02:00
Charles Keepax
29fee82940 ASoC: arizona: Correct checking of FLL ratio limitations
The check to ensure the Fref frequency is within the bounds for the
current ratio, was placed in the wrong loop. The initial configuration
will always be valid and the loop lowering the ratio will only reinforce
this validity. The check should be on the loop increasing the ratio.
This could on occasion cause an invalid ratio/Fref combination to be
selected.

Reported-by: Ryo Tsutsui <ryo.tsutsui@wolfsonmicro.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-10 12:09:15 +02:00
Charles Keepax
4714bc015d ASoC: arizona: Do not test ratio zero as it is not a valid setting
Zero is not a valid FRATIO for the FLL, as such we shouldn't test it
whilst refining the FRATIO. This patch does just that.

Reported-by: Ryo Tsutsui <ryo.tsutsui@wolfsonmicro.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-10 12:09:14 +02:00
Sylwester Nawrocki
a40712a3e6 ASoC: samsung: Add machine driver for Odroid X2/U3
This patch adds the sound subsystem driver for Odroid-X2 and
Odroid-U3 boards. The codec works in I2S master mode; there
are two separate audio routing paths defined, as there are
differences in the signal routing between the X2 and U3 boards,
i.e. U3 uses single jack for headphones and microphone.

Signed-off-by: Chen Zhen <zhen1.chen@samsung.com>
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-09 10:34:18 +02:00
Manish Badarkhe
159baadf59 ASoC: omap-dmic: Use devm_snd_soc_register_component
Replaced snd_soc_register_component with its devres equivalent,
devm_snd_soc_register_component.

Signed-off-by: Manish Badarkhe <badarkhe.manish@gmail.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-09 10:14:56 +02:00
Manish Badarkhe
36765c9c06 ASoC: omap-mcbsp: Use devm_snd_soc_register_component
Replaced snd_soc_register_component with its devres equivalent,
devm_snd_soc_register_component.

Signed-off-by: Manish Badarkhe <badarkhe.manish@gmail.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-09 10:14:52 +02:00
Fabio Estevam
e42be7e142 ASoC: sgtl5000: Fix driver unbound
Using the sgtl5000 codec driver as a module and trying to remove it causes the
followig kernel oops:

root@freescale /$ rmmod snd-soc-imx-sgtl5000
[  117.122920] ------------[ cut here ]------------
[  117.127609] WARNING: CPU: 0 PID: 631 at drivers/regulator/core.c:3604 regula)
[  117.137046] Modules linked in: snd_soc_imx_sgtl5000(-) snd_soc_sgtl5000 evbug
[  117.144315] CPU: 0 PID: 631 Comm: rmmod Not tainted 3.16.0-rc3-next-201407043
[  117.153366] Backtrace:
[  117.155865] [<80011e5c>] (dump_backtrace) from [<80011ff8>] (show_stack+0x18)
[  117.163484]  r6:802fcc48 r5:00000000 r4:00000000 r3:00000000
[  117.169228] [<80011fe0>] (show_stack) from [<80668cc0>] (dump_stack+0x88/0xa)
[  117.176508] [<80668c38>] (dump_stack) from [<80029a38>] (warn_slowpath_commo)
[  117.184696]  r5:00000009 r4:00000000
[  117.188322] [<800299c8>] (warn_slowpath_common) from [<80029a80>] (warn_slow)
[  117.197150]  r8:dd60d600 r7:ddfa6d00 r6:dd5d9064 r5:dd5e0f90 r4:dd5d9400
[  117.203983] [<80029a5c>] (warn_slowpath_null) from [<802fcc48>] (regulator_u)
[  117.212828] [<802fcb74>] (regulator_unregister) from [<7f0047c4>] (ldo_regul)
[  117.223475]  r4:dd59e300 r3:dd5e0f90
[  117.227100] [<7f00479c>] (ldo_regulator_remove [snd_soc_sgtl5000]) from [<7f)
[  117.238959]  r4:dd5d8000 r3:ddd51420
[  117.242623] [<7f0047dc>] (sgtl5000_remove [snd_soc_sgtl5000]) from [<804e5b1)
[  117.252489]  r5:00000000 r4:dd5d8000
[  117.256111] [<804e5af8>] (soc_remove_codec) from [<804e5ed4>] (soc_remove_da)
[  117.264933]  r4:ddfb640c r3:00000000
[  117.268555] [<804e5c10>] (soc_remove_dai_links) from [<804e5fbc>] (snd_soc_u)
[  117.277810]  r10:80359e48 r9:dd684000 r8:dd5ca800 r7:dd685e60 r6:00000001 r5c
[  117.285761]  r4:dd5d9064
[  117.288329] [<804e5f28>] (snd_soc_unregister_card) from [<804f29f0>] (devm_c)
[  117.297324]  r6:00000004 r5:ddd0fc10 r4:dd5a7980 r3:804f29dc
[  117.303095] [<804f29dc>] (devm_card_release) from [<8035a418>] (release_node)
[  117.311369] [<8035a2a8>] (release_nodes) from [<8035aab4>] (devres_release_a)
[  117.319583]  r10:00000000 r9:dd684000 r8:8000ed64 r7:00000081 r6:ddd0fc44 r54
[  117.327543]  r4:ddd0fc10
[  117.330108] [<8035aa7c>] (devres_release_all) from [<803571c8>] (__device_re)
[  117.339214]  r4:ddd0fc10 r3:dd5d9010
[  117.342989] [<80357148>] (__device_release_driver) from [<80357a48>] (driver)
[  117.351607]  r5:7f00c9e4 r4:ddd0fc10
[  117.355285] [<8035798c>] (driver_detach) from [<80357030>] (bus_remove_drive)
[  117.363454]  r6:00000880 r5:00000000 r4:7f00c9e4 r3:dd6c3a80
[  117.369195] [<80356fdc>] (bus_remove_driver) from [<803580b8>] (driver_unreg)
[  117.377683]  r4:7f00c9e4 r3:dd60d480
[  117.381305] [<80358088>] (driver_unregister) from [<80358bb4>] (platform_dri)
[  117.390642]  r4:7f00ca28 r3:7f00c35c
[  117.394309] [<80358ba0>] (platform_driver_unregister) from [<7f00c370>] (imx)
[  117.406188] [<7f00c35c>] (imx_sgtl5000_driver_exit [snd_soc_imx_sgtl5000]) f)
[  117.417452] [<8008c2b8>] (SyS_delete_module) from [<8000eba0>] (ret_fast_sys)
[  117.425753]  r6:5f636f73 r5:5f646e73 r4:00016f40
[  117.430428] ---[ end trace 8fd8a5cb39e46d0e ]---

This problem is well explained by Russell King:

"The sgtl5000 uses a two-stage initialisation process.  The first stage
is when the platform driver is probed, where some resources are found
and initialised.  The second stage is via the codec driver's probe
function, where regulators are found and initialised using the managed
resource support.

The problem here is that this works fine until the codec driver is
unbound.  When this occurs, sgtl5000_remove() is called which is a no-op
as far as the managed resource code is concerned.  The regulators remain
allocated, and their pointers in sgtl5000_priv remain valid.

If the codec is now re-probed, it will again try and find the regulators,
which will now be busy.  This will fail.

That's not the only problem - if using the LDO regulator, the regulator
is unregistered from the regulator core code, but it still has a user.
When the user is cleaned up (eg, by removing the module) it hits the
free'd regulator, and this can oops the kernel.

This bug was originally introduced by 63e54cd9ca ("ASoC: sgtl5000:
Use devm_regulator_bulk_get()")."

This reverts commit 63e54cd9ca.

Tested on a imx53-qsb board.

Reported-by: Russell King <rmk+kernel@arm.linux.org.uk>
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-09 10:12:37 +02:00
Tushar Behera
7e07e7c085 ASoC: core: Fix possible NULL pointer dereference
snd_soc_of_parse_card_name() may be called before card->dev has been
set, which results in a kernel panic.

Unable to handle kernel NULL pointer dereference at virtual address 00000210
PC is at snd_soc_of_parse_card_name+0x18/0x54
LR is at snow_probe+0x5c/0xd4

Add an error check in snd_soc_of_parse_card_name() to take care of this
case and print out a message in case of error.

Signed-off-by: Tushar Behera <tushar.b@samsung.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-09 10:06:46 +02:00
Jianqun Xu
4495c89fcf ASoC: add driver for Rockchip RK3xxx I2S controller
Add driver for i2s controller found on rk3066, rk3168 and rk3288
processors from rockchip.

Tested on the RK3288 SDK board.

Signed-off-by: Jianqun Xu <xjq@rock-chips.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-09 09:39:29 +02:00
Brian Austin
c4324bfa54 ASoC: cs42l56: Move ADC/PCM Swap to DAPM
The Swap controls for ADC/PCM paths should be in the DAPM domain.

Signed-off-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-08 18:20:53 +02:00
Hui Wang
50dd9050e4 ALSA: hda - add capture mute led support in led_power_filter
Now the led_power_filter() needs to handle 3 situations:
    - only mute_led_nid is set
    - only cap_mute_led_nid is set
    - both mute_led_ind and cap_mute_led_nid are set

BugLink: https://bugs.launchpad.net/bugs/1329580
Cc: David Henningsson <david.henningsson@canonical.com>
Cc: Kailang Yang <kailang@realtek.com>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-07-08 12:41:21 +02:00
Hui Wang
fc1fad93c9 ALSA: hda - fix a typo by changing mute_led_nid to cap_mute_led_nid
BugLink: https://bugs.launchpad.net/bugs/1329580
Cc: David Henningsson <david.henningsson@canonical.com>
Cc: Kailang Yang <kailang@realtek.com>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-07-08 12:41:08 +02:00
Aaron Plattner
91947d8cc5 ALSA: hda - Add new GPU codec ID 0x10de0070 to snd-hda
Vendor ID 0x10de0070 is used by a yet-to-be-named GPU chip.

Signed-off-by: Aaron Plattner <aplattner@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-07-08 11:13:57 +02:00
Thierry Reding
121eb44413 ASoC: max98090: Fix build warning
The max98090_{suspend,resume}() functions are used for system sleep and
therefore need to be guarded by CONFIG_PM_SLEEP rather than CONFIG_PM.

Signed-off-by: Thierry Reding <treding@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-07 17:06:35 +02:00
Bard Liao
bc6c4e455a ASoC: RT286: Fix silent at the beginning of stream
This patch fix the issue that the output is almost silent at the
beginning of starting a playback.

Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-07 16:38:44 +02:00
Thierry Reding
525549d734 ALSA: hda: Fix build warning
The hda_tegra_disable_clocks() function is only used by the suspend and
resume code, so it needs to be included in the #ifdef CONFIG_PM_SLEEP
block to prevent the following warning:

  CC      sound/pci/hda/hda_tegra.o
sound/pci/hda/hda_tegra.c:238:13: warning: 'hda_tegra_disable_clocks' defined but not used [-Wunused-function]
 static void hda_tegra_disable_clocks(struct hda_tegra *data)
             ^

Signed-off-by: Thierry Reding <treding@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-07-07 15:15:10 +02:00
Bard Liao
61a414c412 ASoC: RT286: move initial settings to _i2c_probe
Move codec initial settings from asoc probe to i2c probe.

Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-07 12:26:40 +02:00
Bard Liao
4b21768a95 ASoC: RT286: check ID in i2c level
Move ID check from asoc level to i2c level.

Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-07 12:26:40 +02:00
Bard Liao
305b8d8782 ASoC: RT286: remove test code
Remove test code.

Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-07 12:26:40 +02:00
Hui Wang
549d878235 ALSA: hda - Add several entries for enabling HP mute led
BugLink: https://bugs.launchpad.net/bugs/1329580
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-07-07 11:35:34 +02:00
Oder Chiou
f18803a375 ASoC: rt5677: Modify the voltage level in the BIAS OFF stage
The patch modifies the voltage level in the BIAS OFF stage

Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-07 10:09:34 +02:00
Oder Chiou
cc6b0ae962 ASoC: rt5677: Remove the redundant definition in head file
The patch removes the redundant definition in head file

Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-07 10:09:33 +02:00
Joe Perches
d82353e534 ALSA: hda - Fix and neaten print_nid_path/debug_badness
print_nid_path has a possible buffer overflow if
struct nid_path.path values are > 256.

Avoid this and neaten the output to remove the leading ':'

Neaten debug_badness to always verify arguments.

Signed-off-by: Joe Perches <joe@perches.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-07-07 09:34:16 +02:00
Mark Brown
1bb60b3a5a ASoC: s6105-ipcam: Fix build for I2C dependencies
The s6105-ipcam driver calls i2c_register_board_info() which means that
not only does it need I2C to be enabled to be built but since that
symbol is not exported it means that the driver also can't be linked as
a module.  Use a bool instead of a tristate and add a dependency on I2C
to fix these issues.

Commit b5dd60fd26 (ASoC: s6000: Allow to build when COMPILE_TEST is
enabled) made this problem more visible since it enabled build of the
driver on more common architectures but this has always been an issue.

Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-05 18:00:34 +01:00
Tushar Behera
00ad93e263 ASoC: samsung: Make card name for Snow configurable
Snow sound-card driver supports multiple boards with different
audio codecs. Updating the sound card name per board basis would provide
some more information to the end-user.

Signed-off-by: Tushar Behera <tushar.b@samsung.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-04 20:31:45 +01:00
Tushar Behera
eba843201a ASoC: max98090: Remove redundant max98090_handle_pdata()
max98090_handle_pdata() is not doing anything other than printing
a message if pdata is not valid. This can be removed.

Signed-off-by: Tushar Behera <tushar.b@samsung.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-04 20:30:27 +01:00
Tushar Behera
978b641f95 ASoC: max98090: Add check for CODEC type
CODEC type (MAX98090/MAX98091) can be specified from device-tree file,
it can also be obtained from the CODEC during runtime.

Add an explicit check to figure out if both are matching, else print
a message warning about the same.

Signed-off-by: Tushar Behera <tushar.b@samsung.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-04 20:30:27 +01:00
Sylwester Nawrocki
d3d4e5247b ASoC: samsung: Correct I2S DAI suspend/resume ops
We should save/restore relevant I2S registers regardless of
the dai->active flag, otherwise some settings are being lost
after system suspend/resume cycle. E.g. I2S slave mode set only
during dai initialization is not preserved and the device ends
up in master mode after system resume.

Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org
2014-07-04 20:29:27 +01:00
Arnd Bergmann
90eb1ab9e4 ASoC: sirf: provide pm-runtime functions when needed
The newly introduced sirf-usp driver defines sirf_usp_pcm_{suspend,resume}
functions only when PM_RUNTIME is enabled, but also uses them when that
is disabled and only PM_SLEEP is turned on, resulting in this error:

../sound/soc/sirf/sirf-usp.c: In function 'sirf_usp_pcm_suspend':
../sound/soc/sirf/sirf-usp.c:308:3: error: implicit declaration of function 'sirf_usp_pcm_runtime_suspend' [-Werror=implicit-function-declaration]
   sirf_usp_pcm_runtime_suspend(dev);
   ^
../sound/soc/sirf/sirf-usp.c: In function 'sirf_usp_pcm_resume':
../sound/soc/sirf/sirf-usp.c:319:3: error: implicit declaration of function 'sirf_usp_pcm_runtime_resume' [-Werror=implicit-function-declaration]
   ret = sirf_usp_pcm_runtime_resume(dev);
   ^
cc1: some warnings being treated as errors

To fix that, this patch changes the #ifdef to CONFIG_PM, which
is enabled when at least one of PM_SLEEP or PM_RUNTIME are enabled.

Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-04 20:28:44 +01:00
Lars-Peter Clausen
13447cdc76 ASoC: s6105-ipcam: Automatically disconnect non-connected pins
All CODEC input and output widgets are either in the DAPM routing table or
manually marked as non-connected. This means the card is fully routed and we can
let the core take care of disconnecting non-connected pins.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-04 18:57:58 +01:00
Lars-Peter Clausen
b5dd60fd26 ASoC: s6000: Allow to build when COMPILE_TEST is enabled
Most of the ASoC s6000 code is architecture independent. This patch makes it
possible to select the platform when COMPILE_TEST is enabled.

The only architecture dependent code is the PCM driver which will still only be
selected if XTENSA_VARIANT_S6000 is enabled.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-04 18:57:58 +01:00
Lars-Peter Clausen
001eaa25ca ASoC: s6000-i2s: Fix s6000_i2s_remove() return type
The platform_driver remove callback return type is int not void.

Fixes the following warning:
	sound/soc/s6000/s6000-i2s.c:604:19: warning: incorrect type in initializer (different base types)
	sound/soc/s6000/s6000-i2s.c:604:19:    expected int ( *remove )( ... )
	sound/soc/s6000/s6000-i2s.c:604:19:    got void ( static [toplevel] *<noident>)( ... )

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-04 18:57:57 +01:00
Peter Ujfalusi
75f850fc22 ALSA: pcm_dmaengine: Correct support for 24bits physical sample widths
In case of _3LE/_3BE formats the samples are stored in 3 consecutive bytes
without padding it to 4 bytes. This means that the DMA needs to be able to
support 3 bytes word length in order to read/write the samples from memory
correctly. Originally the code treated 24 bits physical length samples as
they were 32 bits which leads to corruption when playing or recording audio.

The hw.formats field has already been prepared to exclude formats not
supported by the DMA engine in use, which means that only on platforms where
3 bytes is supported by the DMA will be able to use this format.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Acked-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-04 18:55:25 +01:00
Peter Ujfalusi
2d38df1283 ASoC: generic-dmaengine-pcm: Prepare formats mask for valid physical sample sizes
Based on the dma_slave_caps's addr_widths queried from the dma driver
prepare the hw.formats mask to include only formats which is supported by
the DMA engine.
In case the dma driver does not implement the slave_caps the default
assumption is that it supports 1, 2 and 4 bytes widths.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Acked-by: Takashi Iwai <tiwai@suse.de>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-04 18:55:21 +01:00
Peter Ujfalusi
732814c8ff ALSA: pcm_dmaengine: Use the available wrapper to get physical width
params_physical_width() is available via pcm_params.h

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Acked-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-04 18:55:17 +01:00
Bard Liao
07cf7cbadb ASoC: add RT286 CODEC driver
This patch adds Realtek ALC286 codec driver.

ALC286 is a dual mode codec, which can run as HD-A or I2S mode.
It is controlled by HD-A verb commands via I2C protocol.
The following is the I/O difference between ALC286 and general I2S codecs.
1. A HD-A verb command contains three parts, NID, VID, and PID.
   And an I2S command contains only two parts: address and data.
2. Not only the register address is written, but the read command also
   includes the entire write command.
3. rt286 uses different registers for read and write the same bits.

We map verb command to regmap structure. However, we read most registers from
cache to prevent the asymmetry read/write issue in rt286.

Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Gustaw Lewandowski <gustaw.lewandowski@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-04 18:50:51 +01:00
Praveen Diwakar
0a37c6efec ASoC: wm_adsp: Add missing MODULE_LICENSE
Since MODULE_LICENSE is missing the module load fails,
so add this for module.

Signed-off-by: Praveen Diwakar <praveen.diwakar@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Reviewed-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org
2014-07-04 17:53:31 +01:00
Konstantinos Tsimpoukas
2f6ba2b31b ALSA: ice1712: Correcting/completing #defines for REGS
This small patch completes #defines for Control/Status Register,
adds comments for the missing ones there and on the Interrupt Mask
Register and additionally corrects "#define ICE1712_SERR_LEVEL 0x04 -> 0x08",
according to documentation.

Signed-off-by: Konstantinos Tsimpoukas <kostaslinuxxx@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-07-04 09:18:36 +02:00
Takashi Iwai
1a0e3f9639 Merge branch 'for-linus' into for-next
Conflicts:
	sound/pci/hda/hda_intel.c
2014-07-04 07:48:57 +02:00
Mengdong Lin
e4d9e513de ALSA: hda - restore BCLK M/N value as per CDCLK for HSW/BDW display HDA controller
For HSW/BDW display HD-A controller, hda_set_bclk() is defined to set BCLK
by programming the M/N values as per the core display clock (CDCLK) queried from
i915 display driver.

And the audio driver will also set BCLK in azx_first_init() since the display
driver can turn off the shared power in boot phase if only eDP is connected
and M/N values will be lost and must be reprogrammed.

Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-07-04 07:47:22 +02:00
Sachin Kamat
63d36f8814 ASoC: wm8996: Remove unused variable
'ret' is not used in the function. Remove it.

Signed-off-by: Sachin Kamat <sachin.kamat@samsung.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-03 19:46:23 +01:00
Sachin Kamat
d1498b13ae ASoC: wm8350: Remove unused variable
'irq' is not used in the function. Remove it.

Signed-off-by: Sachin Kamat <sachin.kamat@samsung.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-03 19:46:23 +01:00
Sachin Kamat
bb17bc7888 ASoC: twl4030: Remove unused variable
'status' is not used in the function. Remove it.

Signed-off-by: Sachin Kamat <sachin.kamat@samsung.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-03 19:46:23 +01:00
Daniel Mack
3ad80b828b ASoC: adau1701: fix adau1701_reg_read()
Fix a long standing bug in the read register routing of adau1701.
The bytes arrive in the buffer in big-endian, so the result has to be
shifted before and-ing the bytes in the loop.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org
2014-07-03 19:42:40 +01:00
Rongjun Ying
2bd8d1d5cf ASoC: sirf: Add audio usp interface driver
This patch adds ASoC support for SiRF SoCs USP interface.
Features include:
1. Only support slave mode.
2. Support I2S and DSP_A mode.
3. Support S16_LE, S24_LE and S24_3LE formats.
4. Support stereo and mono mode.
5. The biggest Support is 192Khz sample rate.

Signed-off-by: Rongjun Ying <rongjun.ying@csr.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-03 17:11:40 +01:00
Daniel Mack
c46af3124b ASoC: tas5086: add regulator consumer support
The TAS5086 has two power domains, DVDD and AVDD. Enable them both as
long as the codec is in use.

Also, switch on the power to identify the chip at device probe level,
and switch it off again afterwards. The codec level will take care for
power handling later.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-03 16:09:42 +01:00
Lars-Peter Clausen
5f3484ac87 ASoC: core: Move non-shared code paths out of snd_soc_post_component_init()
There are two call sites for snd_soc_post_component_init(), one passes 0 and the
other 1 for the 'dailess' parameter of snd_soc_post_component_init(). Depending
on whether 'dailess' is 0 or 1 snd_soc_post_component_init() runs different code
at the beginning and the end of the function. The patch moves this conditional
code out of snd_soc_post_component_init() and into the call sites. This removes
the need for snd_soc_post_component_init() to know whether it is called for a
DAI link or a aux dev.

Also do the initialization of rtd->card when the rtd struct is allocated.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-02 21:24:13 +01:00
Lars-Peter Clausen
44c69bb139 ASoC: core: Bind aux devs early
Currently in snd_soc_instantiate_card() we only check if the aux dev exists, but
do not yet assign it to its rtd. This means that we need to lookup the aux dev
again in soc_probe_aux_dev(). This patch changes the behavior to assign the aux
dev to the rtd in soc_check_aux_dev() (and renames it to soc_bind_aux_dev()).
This simplifies the implementation a bit and also removes the need for
soc_post_component_init() to know about the specific CODEC that was assigned to
the rtd. The later is necessary for componentization as the code should work for
all types of components not just CODECs. This new behavior is also more in sync
with how soc_bind_dai_link()/soc_probe_link_dais() works.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-02 21:24:10 +01:00
Lars-Peter Clausen
48f466d112 ASoC: core: Replace soc_find_matching_codec() with soc_find_codec()
soc_find_matching_codec() works in the same way as soc_find_codec() except that
it only works for auxdevs. It can easily be replaced by the generic
soc_find_codec().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-02 21:24:07 +01:00
Lars-Peter Clausen
b8257be5ff ASoC: core: Remove duplicated rtd->codec initialization
rtd->codec is already initialized in soc_bind_dai_link(), so there is no need to
do it again in soc_dai_link_init(). Removing the rtd->codec initialization from
soc_dai_link_init() also removes the need for soc_dai_link_init() to know about
the CODEC at all.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-02 21:24:04 +01:00
Bard Liao
5e8351de74 ASoC: add RT5670 CODEC driver
This patch adds a minimum support of Realtek ALC5670 codec.

Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-02 20:49:25 +01:00
Kuninori Morimoto
34cb6123f9 ASoC: rsnd: add DT support to DVC
Now, DVC can use DT

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-02 12:35:56 +01:00
Kuninori Morimoto
340371005d ASoC: rsnd: DMA start address is properly used for each DMAC
R-Car sound uses Audio DMAC and Audio DMAC peri peri.
Audio DMAC peri peri transfers data inside circuit.

DMA transfer needs source / destination address,
and destination address can be set via dmaengine_slave_config().
The source address can be set when starting DMAEngine.
Because Audio DMAC peri peri always ignores its value,
current driver always used same source address for
Audio DMAC / Audio DMAC peri peri
(Audio DMAC peri peri source / destination address
is always fixed value)
But, This is not good match for DT booting.
This patch properly uses DMA start address
for Audio DMAC / Audio DMAC peri peri.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-02 12:35:56 +01:00
Kuninori Morimoto
3752303485 ASoC: rsnd: DMA cleanup for flexible SSI/SRC selection
Current R-Car sound SSI/SRC/DVC selection has feature limit.
(It is assuming that SSI/SRC are using same index number)

So that enabling SSI/SRC flexible selection,
this patch modifies DMA settings.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-02 12:18:02 +01:00
Benoit Cousson
c8dd1fec47 ASoC: pcm: Refactor soc_pcm_apply_msb for multicodecs
Refactor the function to facilitate the migration to
multiple codecs.

Fix a trailing space in the header as well.

No functional change.

Signed-off-by: Benoit Cousson <bcousson@baylibre.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-01 18:20:07 +01:00
Benoit Cousson
3f901a028f ASoC: core: Change soc_link_dai_widgets signature for multiple codecs
Since multiple codecs DAI will be usable in the future, remove
explicit unique codec_dai and cpu_dai parameters.
Replace them with snd_soc_pcm_runtime pointer that will contain
every instances.

No functionale change.

Signed-off-by: Benoit Cousson <bcousson@baylibre.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-01 18:17:22 +01:00
Sachin Kamat
d5471e6722 ALSA: hda: Remove unused variable
'status' is not used in the function. Remove it.

Signed-off-by: Sachin Kamat <sachin.kamat@samsung.com>
Tested-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-07-01 17:55:25 +02:00
Sachin Kamat
330fb10df8 ALSA: mixart: Remove unused variable
'err' is not used in the function. Remove it.

Signed-off-by: Sachin Kamat <sachin.kamat@samsung.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-07-01 14:06:31 +02:00
Sachin Kamat
427f42e4cf ALSA: echoaudio: Remove unused variable
'chip' is not used in the function. Remove it.

Signed-off-by: Sachin Kamat <sachin.kamat@samsung.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-07-01 14:06:24 +02:00
Takashi Iwai
178942b69f ALSA: hda - Fix build error in hda_tegra.c
The "list" field has been omitted from struct azx, but its
initialization remained mistakenly in hda_tegra.c, which leads to a
compile error:
   sound/pci/hda/hda_tegra.c: In function 'hda_tegra_create':
   sound/pci/hda/hda_tegra.c:481:22: error: 'struct azx' has no member
named 'list'

Reported-by: kbuild test robot <fengguang.wu@intel.com>
Fixes: 9a34af4a33 ('ALSA: hda - Move more PCI-controller-specific stuff from generic code')
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-07-01 14:02:02 +02:00
Sachin Kamat
e2ff8406ad ALSA: trident: Remove unused variable in trident_memory.c
'prev' is not used in the function. Remove it.

Signed-off-by: Sachin Kamat <sachin.kamat@samsung.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-07-01 11:24:39 +02:00
Sachin Kamat
8d9048643f ALSA: trident: Remove unused variable in trident_main.c
'private_data' is not used in the function. Remove it.

Signed-off-by: Sachin Kamat <sachin.kamat@samsung.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-07-01 11:24:31 +02:00
Takashi Iwai
84526820c4 Merge branch 'topic/hda-cleanup' into for-next 2014-07-01 11:24:06 +02:00
Russell King - ARM Linux
d1a792f3b4 Update imx-sdma cyclic handling to report residue
I received a report this morning from one of the Novena developers that
the behaviour of the iMX6 ASoC codec driver (using imx-pcm-dma.c) was
sub-optimal under high system load.

While there are issues relating to system load remaining, upon reviewing
the ASoC imx-pcm-dma.c driver, it was noticed that it not using the
residue support, because SDMA doesn't support it.  This has the effect
that SDMA has to make multiple calls into the ASoC and ALSA code, one
for each period.

Since ALSA's snd_pcm_elapsed() does not need to be called multiple times
and it is entirely sufficient to call it once to update ALSA with the
current buffer position via the pointer method, we can do better here.
We can also avoid stopping the DMA entirely, just like real cyclic DMA
implementations behave.  While this means that we replay some old samples,
this is a nicer behaviour than having audio stop and restart.

The changes to achieve this are relatively minor - imx-sdma.c can track
where the DMA is to the nearest descriptor boundary - it does this
already when deciding how many callbacks to issue.  In doing this,
buf_tail always points at the descriptor which will complete next.

The residue is defined by the bytes remaining to the end of the buffer,
when the buffer is viewed as a single block of memory [start...end].
So, when we start out, there's a full buffer worth of residue, and this
counts down as we approach the end of the buffer, eventually becoming
zero at the end, before returning to the full buffer worth when we
wrap back to the start.

Moving the walking of the descriptors into the interrupt handler means
that we can update the BD_DONE flag at interrupt time, thus avoiding
a delayed tasklet stopping the cyclic DMA.

This means that the residue can be calculated from (total descriptors -
buf_tail) * descriptor size.  This is what the change below does.  We
update imx-pcm-dma.c to remove the NO_RESIDUE flag since we now provide
the residue.

Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
2014-07-01 12:23:42 +05:30
Linus Walleij
2fd48f9442 ARM: mach-s5p: get rid of all <mach/gpio.h> headers
This renames all the local <mach/gpio.h> headers in the S5P platforms
to <mach/gpio-samsung.h> indicating a scope local to this platform,
and cuts the implicit inclusion of <mach/gpio.h> from <linux/gpio.h>
by removing the use of NEED_MACH_GPIO_H from all S5P variants.

Acked-by: Alexandre Courbot <acourbot@nvidia.com>
Signed-off-by: Linus Walleij <linus.walleij@linaro.org>
2014-06-30 21:37:21 +02:00
Tushar Behera
46aed59752 ASoC: samsung: Extend snow driver to support MAX98091
Peach-pi board has MAX98091 CODEC. Extend snow machine driver to support
this board.

Signed-off-by: Tushar Behera <tushar.b@samsung.com>
Reviewed-by: Doug Anderson <dianders@chromium.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-30 19:52:40 +01:00
Wonjoon Lee
053e69d57c ASoC: max98090: Add max98091 compatible string
The MAX98091 CODEC is the same as MAX98090 CODEC, but with an extra
microphone. Existing driver for MAX98090 CODEC already has support
for MAX98091 CODEC. Adding proper compatible string so that MAX98091
CODEC can be specified from device tree.

Signed-off-by: Wonjoon Lee <woojoo.lee@samsung.com>
Signed-off-by: Doug Anderson <dianders@chromium.org>
Signed-off-by: Tushar Behera <tushar.b@samsung.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-30 19:51:23 +01:00
Axel Lin
099d334e3d ASoC: rt5677: Convert to use rl6231_pll_calc
The implementation of rt5677_pll_calc() has the same logic of rl6231_pll_calc().
The only difference is the lower boundary checking for freq_in.

This patch calls rl6231_pll_calc() instead of open-coded.
The k_bp of struct rt5677_pll_code is always false, thus also remove the
code to check pll_code.k_bp.

Signed-off-by: Axel Lin <axel.lin@ingics.com>
Tested-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-30 19:42:47 +01:00
Peter Ujfalusi
182bef863c ASoC: davinci-mcasp: Fix S24_LE and U24_LE support
In case of S24_LE/U24_LE modes we expect 24bits on the bus while the samples
are stored and transferred in memory on 32bits (lower 3 bytes of the 4
bytes).

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Tested-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-30 15:53:25 +01:00
Peter Ujfalusi
2a11a10abe ASoC: tlv320aic3x: Add support for S24_LE format
The codec need to be configured to 24bit mode in case of S24_LE format.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-30 15:52:32 +01:00
Peter Ujfalusi
25ccb22ed5 ASoC: tlv320aic3x: Correct S24_3LE support
Correct the hw_params callback to configure the codec correctly in case of
S24_3LE format since in case of S24_3LE the codec has been configured to
16bit format mode.
S24_LE is not defined as supported format for the codec.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-30 15:51:40 +01:00
Takashi Iwai
e8750940ce ALSA: hda - Fix invalid function call in snd_hda_add_vmaster()
The recent commit [6194b99d: ALSA: hda - Kill the rest of snd_print*()
usages] changed the callback map_slaves(), but one call was forgotten
to be replaced due to the cast, which leads to kernel Oops due to
invalid function.  This patch replaces it with a proper function.

Fixes: 6194b99de9 ('ALSA: hda - Kill the rest of snd_print*() usages')
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-06-30 14:05:17 +02:00
Paul Handrigan
59f5cbecf9 ASoC: cs4265: Change return values to boolean.
The cs4265_volatile_register reutrns a bool. The function now returns
true or false vs 1 and 0.

Signed-off-by: Paul Handrigan <Paul.Handrigan@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-30 11:11:58 +01:00
Mark Brown
211bcc6c3a Merge remote-tracking branch 'asoc/fix/debugfs' into asoc-component
Conflicts:
	sound/soc/soc-core.c
2014-06-28 14:47:12 +01:00
Kuninori Morimoto
65f459923b ASoC: rsnd: enable DVC when capture
Current DVC can be enabled only when playback,
but, this came from misunderstanding.
It is not correct.

DVC <-> DMA relationship is...

Playback: MEM -> DMAC  -> SRC -> DVC -> DMACp -> SSI
Capture:  SSI -> DMACp -> SRC -> DVC -> DMAC  -> MEM

DVC can be used for both Playback/Capture

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-28 14:41:19 +01:00
Kuninori Morimoto
ccd01559ea ASoC: rsnd: use dmaengine_prep_dma_cyclic() instead of original method
Current R-Car sound driver is using DMAEngine directly,
but, ASoC is requesting to use common DMA transfer method,
like snd_dmaengine_pcm_trigger() or dmaengine_pcm_ops.
It is difficult to switch at this point, since Renesas
driver is also supporting PIO transfer.
This patch uses dmaengine_prep_dma_cyclic() instead
of dmaengine_prep_slave_single().
It is used in requested method,
and is good first step to switch over.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-28 14:41:19 +01:00
Kuninori Morimoto
d9288d0ba1 ASoC: rsnd: SSI + DMA can select BUSIF
Sound data needs to be sent to R-Car sound SSI when playback.
But, there are 2 interfaces for it.
1st is SSITDR/SSIRDR which are mapped on SSI.
2nd is SSIn_BUSIF which are mapped on SSIU.

2nd SSIn_BUSIF is used when DMA transfer,
and it is always used if sound data came from via SRC.
But, we can use it when SSI+DMA case too.
(Current driver is assuming 1st SSITDR/SSIRDR for it)

2nd SSIn_BUSIF can be used as FIFO.
This is very helpful/useful for SSI+DMA.

But DMA address / DMA ID are not same between 1st/2nd cases.
This patch care about these settings.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-28 14:41:19 +01:00
Kuninori Morimoto
8457e0e9e2 ASoC: fsi: use dmaengine_prep_dma_cyclic() for DMA transfer
Current FSI driver is using DMAEngine directly,
but, ASoC is requesting to use common DMA transfer method,
like snd_dmaengine_pcm_trigger() or dmaengine_pcm_ops.
It is difficult to switch at this point, since Renesas
driver is also supporting PIO transfer.
This patch uses dmaengine_prep_dma_cyclic() instead
of dmaengine_prep_slave_single().
It is used in requested method,
and is good first step to switch over.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-28 14:41:18 +01:00
Kuninori Morimoto
d403e24908 ASoC: fsi: add fsi_pointer_update() for common pointer method
fsi PIO/DMA handler are using each own pointer update method,
but these can be share.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-28 14:41:18 +01:00
Kuninori Morimoto
ffb83e8cb1 ASoC: fsi: use SNDRV_DMA_TYPE_DEV for sound buffer
Current fsi driver is using SNDRV_DMA_TYPE_CONTINUOUS
for snd_pcm_lib_preallocate_pages_for_all().
But, it came from original dma-sh7760.c,
and no longer needed.
This patch exchange its parameter, and removed
original dma mapping and un-needed
dma_sync_single_xxx() from driver.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-28 14:41:18 +01:00
Mark Brown
6f2a06cd42 Merge remote-tracking branch 'asoc/fix/rcar' into asoc-rcar 2014-06-28 14:41:15 +01:00
Russell King
e73f3de5c5 ASoC: fix debugfs directory creation bug
Avoid creating duplicate directories by prefixing codecs and platforms
with their separate identifiers.  This avoids snd-soc-dummy (which can
appear both as a dummy platform and a dummy codec on the same card)
from clashing.

Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Tested-by: Andrew Lunn <andrew@lunn.ch>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-28 13:45:39 +01:00
Russell King
920ec4e595 ASoC: kirkwood: implement NO_PERIOD_WAKEUP support
Permit ALSA to run without hardware interrupts from the audio interface.
Instead, ALSA will use a kernel timer to decide when to check the buffer
state, resulting in a lighter workload for the CPU.

Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Tested-by: Andrew Lunn <andrew@lunn.ch>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-28 13:18:46 +01:00
Russell King
a622251c01 ASoC: kirkwood: allow smaller audio periods and smaller number of periods
There is no hardware restriction requiring a minimum of 8 periods, or
a minimum of 2048 bytes in a period.  Let's drop these values so that
userspace has more flexibility in choosing these parameters.

Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Tested-by: Andrew Lunn <andrew@lunn.ch>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-28 13:18:41 +01:00
Russell King
4d2097e517 ASoC: kirkwood-i2s: fix pause handling some more
We still see the occasional timeout waiting for busy to clear.  As the
spec is contradictory, and we know that the current implementation
doesn't work, try an alternative interpretation from the spec.  This
one appears to work - I have yet to find any issue with it during my
testing over several months.

Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Tested-by: Andrew Lunn <andrew@lunn.ch>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-28 13:18:33 +01:00
Russell King
2fbc38219c ASoC: kirkwood-i2s: fix mute handling
The spec requires that the mute bits must be set while the channel
is disabled.  Ensure that this is the case by providing a helper
which ensures that the appropriate mute bit is set while the enable
bit is clear.

Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Tested-by: Andrew Lunn <andrew@lunn.ch>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-28 13:18:28 +01:00
Russell King
6772190632 ASoC: kirkwood-i2s: fix RECCTL masking
Since we wish to disable capture inputs for some formats, we need to
ensure that we clear the enable bits in our cached record control
register.  This seems to have been missed, resulting in the register
only accumulating enable bits.

Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Tested-by: Andrew Lunn <andrew@lunn.ch>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-28 13:18:22 +01:00
Russell King
52b896cfef ASoC: kirkwood-i2s: provide helper KIRKWOOD_RECCTL_ENABLE_MASK definition
Add a KIRKWOOD_RECCTL_ENABLE_MASK definition to complement the existing
PLAYCTL definition, and make use of it where we wish to clear both
enable bits.

Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Tested-by: Andrew Lunn <andrew@lunn.ch>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-28 13:18:17 +01:00
Fabian Frederick
5bca396919 ASoC: wm0010.c: add static to local variable
Also add const to array

 text	   data	    bss	    dec	    hex	filename
 10946	   2904	   3528	  17378	   43e2	sound/soc/codecs/wm0010.o-before
 10891	   2840	   3512	  17243	   435b	sound/soc/codecs/wm0010.o-after

Signed-off-by: Fabian Frederick <fabf@skynet.be>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-27 12:53:52 +01:00
Sachin Kamat
ba54668708 ASoC: wm_hubs: Remove redundant OOM message
Let memory subsystem handle the error logging.

Signed-off-by: Sachin Kamat <sachin.kamat@samsung.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-27 12:48:23 +01:00
Sachin Kamat
0463585ce5 ASoC: wm9090: Remove redundant OOM message
Let memory subsystem handle the error logging.

Signed-off-by: Sachin Kamat <sachin.kamat@samsung.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-27 12:48:23 +01:00
Sachin Kamat
549f66e028 ASoC: wm8994: Remove redundant OOM message
Let memory subsystem handle the error logging.

Signed-off-by: Sachin Kamat <sachin.kamat@samsung.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-27 12:48:23 +01:00
Sachin Kamat
2cec4ff7f0 ASoC: wm8958: Remove redundant OOM message
Let memory subsystem handle the error logging.

Signed-off-by: Sachin Kamat <sachin.kamat@samsung.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-27 12:48:23 +01:00
Sachin Kamat
d931099beb ASoC: wm8904: Remove redundant OOM message
Let memory subsystem handle the error logging.

Signed-off-by: Sachin Kamat <sachin.kamat@samsung.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-27 12:48:23 +01:00
Sachin Kamat
a0f62118b7 ASoC: wm2000: Remove redundant OOM message
Let memory subsystem handle the error logging.

Signed-off-by: Sachin Kamat <sachin.kamat@samsung.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-27 12:48:23 +01:00
Sachin Kamat
611d7a7ba8 ASoC: wm1250-ev1: Remove redundant OOM message
Let memory subsystem handle the error logging.

Signed-off-by: Sachin Kamat <sachin.kamat@samsung.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-27 12:48:23 +01:00
Sachin Kamat
84cbc75f9a ASoC: wm0010: Remove redundant OOM message
Let memory subsystem handle the error logging.

Signed-off-by: Sachin Kamat <sachin.kamat@samsung.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-27 12:48:22 +01:00
Sachin Kamat
ac872d3d72 ASoC: wl1273: Remove redundant OOM message
Let memory subsystem handle the error logging.

Signed-off-by: Sachin Kamat <sachin.kamat@samsung.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-27 12:48:22 +01:00
Sachin Kamat
04cc41a809 ASoC: twl4030: Remove redundant OOM message
Let memory subsystem handle the error logging.

Signed-off-by: Sachin Kamat <sachin.kamat@samsung.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-27 12:48:22 +01:00
Sachin Kamat
656e343575 ASoC: tpa6130a2: Remove redundant OOM message
Let memory subsystem handle the error logging.

Signed-off-by: Sachin Kamat <sachin.kamat@samsung.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-27 12:48:22 +01:00
Sachin Kamat
b1117f5294 ASoC: tlv320aic3x: Remove redundant OOM message
Let memory subsystem handle the error logging.

Signed-off-by: Sachin Kamat <sachin.kamat@samsung.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-27 12:48:21 +01:00
Sachin Kamat
5c1573a342 ASoC: sta529: Remove redundant OOM message
Let memory subsystem handle the error logging.

Signed-off-by: Sachin Kamat <sachin.kamat@samsung.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-27 12:48:21 +01:00
Sachin Kamat
be81333415 ASoC: sgtl5000: Remove redundant OOM message
Let memory subsystem handle the error logging.

Signed-off-by: Sachin Kamat <sachin.kamat@samsung.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-27 12:48:21 +01:00
Sachin Kamat
10d95ad48b ASoC: cs42l73: Remove redundant OOM message
Let memory subsystem handle the error logging.

Signed-off-by: Sachin Kamat <sachin.kamat@samsung.com>
Acked-by: Paul Handrigan <Paul.Handrigan@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-27 12:48:21 +01:00
Sachin Kamat
0e0327f2ab ASoC: cs4270: Remove redundant OOM message
Let memory subsystem handle the error logging.

Signed-off-by: Sachin Kamat <sachin.kamat@samsung.com>
Acked-by: Paul Handrigan <Paul.Handrigan@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-27 12:48:21 +01:00
Sachin Kamat
41adf9056a ASoC: samsung: Remove unused variable from idma.c
‘iiscon’ is not used in the function. Remove it.

Signed-off-by: Sachin Kamat <sachin.kamat@samsung.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-27 12:02:20 +01:00
Takashi Iwai
a12137e779 ALSA: hda - Add a fixup for Thinkpad T540p
The similar fixup as T440 is needed for supporting the dock on T540.

Reported-by: Jim Minter <jminter@redhat.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-06-27 12:14:35 +02:00
David Henningsson
e03fdbde8a ALSA: hda - Add another headset pin quirk for some Dell machines
Another quirk to make the headset mic work on some new Dell machines.

Cc: Hui Wang <hui.wang@canonical.com>
BugLink: https://bugs.launchpad.net/bugs/1297581
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-06-27 12:09:57 +02:00
Grant Likely
ccdb8ed3b3 of: Migrate of_find_node_by_name() users to for_each_node_by_name()
There are a bunch of users open coding the for_each_node_by_name() by
calling of_find_node_by_name() directly instead of using the macro. This
is getting in the way of some cleanups, and the possibility of removing
of_find_node_by_name() entirely. Clean it up so that all the users are
consistent.

Signed-off-by: Grant Likely <grant.likely@linaro.org>
Cc: Rob Herring <robh+dt@kernel.org>
Cc: Benjamin Herrenschmidt <benh@kernel.crashing.org>
Cc: "Rafael J. Wysocki" <rjw@rjwysocki.net>
Cc: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Cc: Takashi Iwai <tiwai@suse.de>
2014-06-26 17:12:24 +01:00
Takashi Iwai
fb1d8ac299 ALSA: hda - Replace ICH6_ prefix
ICH6_ prefix doesn't mean that it's specific to ICH6 chipset but
rather its generic for all HD-audio (or "Azalia") devices.
Use AZX_ prefix instead to align with other constants.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-06-26 18:00:02 +02:00
Takashi Iwai
c6bf1d8e8c ALSA: hda - Remove obsoleted SFX definitions
It's no longer referred by anyone after standardizing with dev_*()
macros.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-06-26 18:00:02 +02:00
Takashi Iwai
33124929a2 ALSA: hda - Move SD nums definitions to hda_intel.c
The defined numbers of SDs are specific to hda-intel, so move them to
there.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-06-26 18:00:01 +02:00
Takashi Iwai
703c759f38 ALSA: hda - Use common reboot notifier
The very same notifier code is used in both hda_intel.c and
hda_tegra.c.  Move it to the generic code.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-06-26 18:00:01 +02:00
Takashi Iwai
9a34af4a33 ALSA: hda - Move more PCI-controller-specific stuff from generic code
Just move struct fields between struct azx and struct hda_intel, and
move some definitions from hda_priv.h to hda_intel.c.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-06-26 18:00:01 +02:00
Takashi Iwai
b6050ef664 ALSA: hda - Make position_fix as generic callback
... and move most parts into hda_intel.c from the generic controller
code.  This is a clean up, and there should be no functional change by
this patch.

Now, struct azx obtains the generic callbacks for getting the position
and the delay.  As default NULL, posbuf is read.  These replace the
old position_fix[], and each is implemented as a callback.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-06-26 18:00:01 +02:00
Takashi Iwai
085ec0d945 ALSA: hda - Remove superfluous MAX_AZX_DEV
MAX_AZX_DEV is no longer referred anywhere, let's kill it.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-06-26 18:00:01 +02:00
Takashi Iwai
7e9c2eb626 Merge branch 'for-linus' into for-next 2014-06-26 15:49:20 +02:00
Mengdong Lin
a07187c992 ALSA: hda - restore BCLK M/N values when resuming HSW/BDW display controller
For Intel Haswell/Broadwell display HD-A controller, the 24MHz HD-A link BCLK
is converted from Core Display Clock (CDCLK): BCLK = CDCLK * M / N
And there are two registers EM4 and EM5 to program M, N value respectively.
The EM4/EM5 values will be lost and when the display power well is disabled.

BIOS programs CDCLK selected by OEM and EM4/EM5, but BIOS has no idea about
display power well on/off at runtime. So the M/N can be wrong if non-default
CDCLK is used when the audio controller resumes, which results in an invalid
BCLK and abnormal audio playback rate. So this patch saves and restores valid
M/N values on controller suspend/resume.

And 'struct hda_intel' is defined to contain standard HD-A 'struct azx' and
Intel specific fields, as Takashi suggested.

Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-06-26 15:47:42 +02:00
Takashi Iwai
92a586bdc0 ALSA: usb-audio: Fix races at disconnection and PCM closing
When a USB-audio device is disconnected while PCM is still running, we
still see some race: the disconnect callback calls
snd_usb_endpoint_free() that calls release_urbs() and then kfree()
while a PCM stream would be closed at the same time and calls
stop_endpoints() that leads to wait_clear_urbs().  That is, the EP
object might be deallocated while a PCM stream is syncing with
wait_clear_urbs() with the same EP.

Basically calling multiple wait_clear_urbs() would work fine, also
calling wait_clear_urbs() and release_urbs() would work, too, as
wait_clear_urbs() just reads some fields in ep.  The problem is the
succeeding kfree() in snd_pcm_endpoint_free().

This patch moves out the EP deallocation into the later point, the
destructor callback.  At this stage, all PCMs must have been already
closed, so it's safe to free the objects.

Reported-by: Alan Stern <stern@rowland.harvard.edu>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-06-26 10:33:35 +02:00
Takashi Iwai
8b3dfdaf0c ALSA: hda - Adjust speaker HPF and add LED support for HP Spectre 13
HP Spectre 13 has the IDT 92HD95 codec, and BIOS seems to set the
default high-pass filter in some "safer" range, which results in the
very soft tone from the built-in speakers in contrast to Windows.
Also, the mute LED control is missing, since 92HD95 codec still has no
HP-specific fixups for GPIO setups.

This patch adds these missing features: the HPF is adjusted by the
vendor-specific verb, and the LED is set up from a DMI string (but
with the default polarity = 0 assumption due to the incomplete BIOS on
the given machine).

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=74841
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-06-25 17:50:24 +02:00
Takashi Iwai
db8e8a9dc9 ALSA: hda - Remove the obsoleted static quirk codes from patch_cmedia.c
The static quirk code has been disabled for a while and it seems
working fine, so it's time to actually get rid of it.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-06-25 14:51:24 +02:00
Takashi Iwai
d0ea6d270b ALSA: hda - Remove the obsoleted static quirk codes from patch_conexant.c
The static quirk code has been disabled for a while and it seems
working fine, so it's time to actually get rid of it.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-06-25 14:51:23 +02:00
Takashi Iwai
6194b99de9 ALSA: hda - Kill the rest of snd_print*() usages
Pass the codec object so that we can replace all the rest of
snd_print*() usages with the proper device-specific print helpers.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-06-25 14:51:23 +02:00
Takashi Iwai
79514d473b ALSA: hda - Kill snd_printd*() in HDMI debug / info prints
Pass codec instance to each function that still prints info and debug
outputs via snd_printd*().

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-06-25 14:51:23 +02:00
Rickard Strandqvist
a53613a67e sound: oss: mpu401.c: Cleaning up variable is set more than once
A struct member variable is set to the same value more than once

This was found using a static code analysis program called cppcheck.

Signed-off-by: Rickard Strandqvist <rickard_strandqvist@spectrumdigital.se>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-06-25 14:32:03 +02:00
Jarkko Nikula
b29d7c5f71 ASoC: Intel: byt-max98090: Do not change speaker and DMIC with jack state
Kernel should not enable/disable speakers and digital microphone whenever
jack is inserted/removed. This is more use-case than kernel specific
decision. For instance one may want to play VoIP ring tones using both
speakers and headphone but play music only from one of them.

Because of above reason remove "Ext Spk" and "Int Mic" update when jack
state is changed. Also this update was illogical anyway: "Ext Spk" was
enabled when jack was inserted and disabled when jack was removed.

Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-24 16:21:36 +01:00
Jarkko Nikula
ab6f7d0d93 ASoC: Intel: byt-max98090: Do not enable MAX98090 microphone detection
It turned out there is no need to enable microphone detection in MAX98090
codec. Headset microphone is anyway detected by a GPIO signal from another
chip and headset button presses cannot be detected either because a signal
needed for it is not connected.

Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-24 16:20:43 +01:00
Jarkko Nikula
2498899293 ASoC: Intel: byt-max98090: Fix jack type in order to report correctly
Pass actual jack type bitmask to snd_soc_jack_new() in order to report
also microphone detections and not only headphone. While at it change also
jack name and pass also SND_JACK_LINEOUT type.

Reported-by: Jin Yao <yao.jin@intel.com>
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-24 16:20:43 +01:00
Jarkko Nikula
6a0cdccad8 ASoC: Intel: byt-max98090: Do not report SND_JACK_LINEIN
Headset jack has only mono microphone input.

Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-24 16:20:43 +01:00
Jarkko Nikula
725a6dfd01 ASoC: Intel: byt-max98090: Fix mic detect GPIO polarity
Mic detect GPIO is active low when headset microphone is detected. Found
both by debugging and checking the schematics.

Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-24 16:20:43 +01:00
Jarkko Nikula
a5b37bf36f ASoC: Intel: byt-max98090: Move MICBIAS widget to supply of Headset Mic
Move "MICBIAS" as a supply widget to "Headset Mic" instead of keeping it
between input pin "IN34" and "Headset Mic".

Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-24 16:20:42 +01:00
Paul Handrigan
fb6f806967 ASoC: Add support for the CS4265 CODEC
This patch adds support for the Cirrus Logic CS4265 Stereo I2C CODEC.

Signed-off-by: Paul Handrigan <paul.handrigan@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-24 16:16:13 +01:00
David Henningsson
76c2132ec9 ALSA: hda - Make the pin quirk tables use the SND_HDA_PIN_QUIRK macro
This is cosmetical - it makes the pin quirk table look better.

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-06-24 14:48:34 +02:00
David Henningsson
a2d2fa02b2 ALSA: hda - Make a SND_HDA_PIN_QUIRK macro
This is cosmetical - it makes the new pin quirk table look better.

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-06-24 14:48:31 +02:00
Rickard Strandqvist
21d7216ca2 sound: oss: mpu401.c: Cleaning up missing break in a case
Added a missed break in a case statement

This was found using a static code analysis program called cppcheck.

Signed-off-by: Rickard Strandqvist <rickard_strandqvist@spectrumdigital.se>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-06-24 14:16:58 +02:00
David Henningsson
7a52cd79fa ALSA: hda - Add pin quirk for Dell XPS 15
Two bug reporters with Dell XPS 15 report that they need to use the
dell-headset-multi model to get the headset mic working.

The two bug reporters have different PCI SSID (1028:05fd and 1028:05fe)
but this pin quirk matches both.

BugLink: https://bugs.launchpad.net/bugs/1331915
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-06-24 14:16:15 +02:00
Vasily Khoruzhick
c1ae59c7bd ASoC: samsung: s3c24xx-i2s: Move to clk_prepare_enable/clk_disable_unprepare
Use clk_prepare_enable/clk_disable_unprepare to make the driver
work properly with common clock framework.

Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-24 11:57:43 +01:00
Vasily Khoruzhick
77ea6bf777 ASoC: samsung: s3c2412-i2s: Move to clk_prepare_enable/clk_disable_unprepare
Use clk_prepare_enable/clk_disable_unprepare to make the driver
work properly with common clock framework.

Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-24 11:57:43 +01:00
Vasily Khoruzhick
ae602456e8 ASoC: samsung: drop support for legacy S3C24XX DMA API
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-24 11:57:43 +01:00
Vasily Khoruzhick
87b132bc03 ASoC: samsung: s3c24{xx,12}-i2s: port to use generic dmaengine API
Use dmaengine instead of legacy s3c24xx DMA API for s3c24xx and s3c2412

Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-24 11:57:43 +01:00
Rasmus Villemoes
b245a822a4 ALSA: seq: seq_memory.c: Fix closing brace followed by if
Add a newline and, while at it, remove a space and redundant braces.

Signed-off-by: Rasmus Villemoes <linux@rasmusvillemoes.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-06-23 17:58:33 +02:00
Vinod Koul
0ec66fed40 ASoC: Intel: use common stream allocation method for compressed stream
As added in previosu patch along with stream to piep conversion si required for
compressed audio too

Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-23 12:24:28 +01:00
Vinod Koul
61b165caa6 ASoC: Intel: add mrfld pipelines
Merrifield DSP used various pipelines to identify the streams and processing
modules. Add these defination in the pcm driver and also add a table for device
entries to firmware pipeline id conversion

Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-23 12:24:27 +01:00
Pierre Ossman
a283368382 ALSA: hda - hdmi: call overridden init on resume
We need to call the proper init function in case it has been
overridden, as it might restore things that the generic routing
doesn't know anything about. E.g. AMD cards have special verbs
that need resetting.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=77901
Fixes: 5a61358433 ('ALSA: hda - hdmi: Add ATI/AMD multi-channel audio support')
Signed-off-by: Pierre Ossman <pierre@ossman.eu>
Cc: <stable@vger.kernel.org> [v3.13+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-06-23 12:38:28 +02:00
David Henningsson
8fffe7d1f0 ALSA: hda - Fix usage of "model" module parameter
A recent refactoring broke the possibility to manually specify
model name as a module parameter. This patch restores the desired
functionality.

Fixes: c21c8cf77f ('ALSA: hda - Add fixup_forced flag')
Reported-by: Kent Baxley <kent.baxley@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-06-23 12:21:12 +02:00
Kailang Yang
9c5dc3bf12 ALSA: hda/realtek - Support HP mute led for output and input
HP mute led support output mute led and input mute led.
ALC280:
GPIO3 to control output mute led.
Mic1 vref to control input mute led.
ALC282:
Line1 vref to control output mute led.
Mic1 vref to control input mute led.

Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-06-23 12:11:52 +02:00
Takashi Iwai
78fcce4d2c Merge branch 'for-linus' into for-next 2014-06-23 12:11:46 +02:00
Shahina Shaik
423ca88eb5 ASoC: tlv320aic31xx: Fixed Coding Style Issues
Fixed coding style issues of "Missing Blank line after declaration"

Signed-off-by: Shahina Shaik <sharab.shaik@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-22 12:03:26 +01:00
Shahina Shaik
43bf38ba56 ASoC: tlv320aic32x4: Fixed Coding Style Issues
Fixed Coding style issues of lines over 80 characters.

Signed-off-by: Shahina Shaik <sharab.shaik@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-22 12:03:17 +01:00
Shahina Shaik
eb72cbdf51 ASoC: tlv320aic32x4: Fixed Coding style Issues
Fixed a brace coding style issue in the tlv320aic32x4.c

Signed-off-by: Shahina Shaik <sharab.shaik@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-22 12:03:05 +01:00
Vinod Koul
a870cdce9e ASoC: Intel: mfld-pcm: modularize stream allocation code
Tis will be used to add table based support for pcm front ends in subsequent
patches

Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-22 12:01:49 +01:00
Vinod Koul
aa9b045f70 ASoC: Intel: add the mrfld fw IPC definations
This will be used to update current driver as well as in support for the mrfld
patches

Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-22 12:01:49 +01:00
Arnd Bergmann
ff40260f79 ASoC: fsl: refine DMA/FIQ dependencies
Commit 31ee2bfd72 ("ASoC: fsl: select SND_SOC_IMX_PCM_DMA
where needed") started selecting SND_SOC_IMX_PCM_DMA and
SND_SOC_IMX_PCM_FIQ for two drivers when building for i.MX.
This has turned out too aggressive, as FIQ is only available
for i.mx2 through i.mx5, but not i.mx6 or vybrid.

Further, two more drivers have become user-selectable in the
meantime, and they both depend on DMA for the imx platform
as well.

This changes the selection of FIQ to depend on the TZIC or
AVIC interrupt controllers that actually export the imx
specific FIQ interfaces, and adds the missing select statements
for SAI and ESAI.

Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-22 11:57:10 +01:00
Arnd Bergmann
5264d0e6ef ASoC: samsung: Add I2C dependency for snow
Both codecs used by snow, max98090 and max98095 require the use
of I2C, so we can only select this driver if I2C is there, otherwise
we get a build error like:

codecs/max98090.c:2494:1: warning: data definition has no type or storage class [enabled by default]
 module_i2c_driver(max98090_i2c_driver);
 ^
codecs/max98095.c:2443:1: warning: data definition has no type or storage class [enabled by default]
 module_i2c_driver(max98095_i2c_driver);
 ^

This adds one more I2C dependency to the hundreds we already
have.

Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Acked-by: Tushar Behera <tushar.behera@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-22 11:52:58 +01:00
Lars-Peter Clausen
88a8fe3df6 ASoC: dapm: Remove platform field from widget and dapm context struct
The platform field in the snd_soc_dapm_widget and snd_soc_dapm_context structs
is now unused can be removed. New code that wants to get the platform for a
widget or dapm context should use snd_soc_dapm_to_platform(w->dapm) or
snd_soc_dapm_to_platform(dapm).

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-21 21:36:09 +01:00
Lars-Peter Clausen
9420d97b3f ASoC: dapm: Remove DAI DAPM context
The DAI DAPM context was added in commit be09ad90 ("ASoC: core: Add platform DAI
widget mapping") and the only user was removed again in commit ae10e7e8f ("ASoC:
core: Only add platform DAI widgets once."). Now that we have a per component
DAPM context it is unlikely that we'll need the DAI DAPM context again.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-21 21:36:08 +01:00
Lars-Peter Clausen
14e8bdebfb ASoC: Add component level stream_event() and seq_notifier() support
This patch adds stream_event() and seq_notifier() callbacks similar to those
found in the snd_soc_codec_driver and snd_soc_platform driver struct to the
snd_soc_component_driver struct. This is meant to unify the handling of these
callbacks across different types of components and will eventually allow their
removal from the CODEC and platfrom driver structs.

The new callbacks are slightly different from the old ones in that they take a
snd_soc_component as a parameter rather than a snd_soc_dapm_context. This was
done since otherwise casting from the DAPM context to the component would
typically be the first thing to do in the callback. And the interface becomes
slightly cleaner by passing a snd_soc_component to all callbacks in the
snd_soc_component_driver struct.

The patch also already removes the stream_event() callback from the
snd_soc_codec_driver and snd_soc_platform_driver structs as it is currently
unused.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-21 21:34:15 +01:00
Lars-Peter Clausen
bc9af9fa9b ASoC: Use component DAPM context for platforms
The snd_soc_platform dapm field is not accessed outside of the ASoC core. Switch
it over to using the snd_soc_component DAPM context.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-21 21:34:15 +01:00
Lars-Peter Clausen
ce0fc93ae5 ASoC: Add DAPM support at the component level
This patch adds full DAPM support at the component level. Previously there was
only full DAPM support for CODECs and partial DAPM support (e.g. no Mixers nor
MUXs) for platforms. Having DAPM support at the component level will allow all
types of components to use DAPM and also help in consolidating the DAPM support
between CODECs and platforms.

Since the DAPM context is directly embedded into the snd_soc_codec and
snd_soc_platform struct and the 'dapm' field is directly referenced in a lot of
drivers moving the field just right now is not possible without causing code
churn. The approach this patch takes is to add two new fields to the component
struct. One field which is the pointer to the actual DAPM context used by the
component and one DAPM context that will be used as the default if no other
context was specified. For CODECs and platforms the pointer is initialized to
point to the CODEC or platform DAPM context. All generic code when referencing
a component's DAPM struct will go via the pointer. This will make it possible to
eventually seamlessly move the DAPM context from snd_soc_codec and
snd_soc_platform struct over once all direct references have been eliminated.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-21 21:34:15 +01:00
Lars-Peter Clausen
68f831c272 ASoC: Add a set_bias_level() callback to the DAPM context struct
Currently the DAPM code directly looks at the CODEC driver struct to get a
handle to the set_bias_level() callback. This patch adds a new set_bias_level()
callback to the DAPM context struct. The DAPM code will use this new callback
instead of the CODEC callback. For CODECs the new callback is set up to call the
CODEC specific set_bias_level callback(). Not looking directly at the CODEC
driver struct will allow non CODEC DAPM contexts to implement a set_bias_level()
callback.

This is also similar to how the seq_notifier() and stream_event() callbacks are
currently handled.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-21 21:34:15 +01:00
Mark Brown
647d62d9ff Merge remote-tracking branch 'asoc/fix/core' into asoc-component 2014-06-21 21:33:18 +01:00
Lars-Peter Clausen
7df3788410 ASoC: Auto disconnect pins from all DAPM contexts
Currently only pins in CODEC DAPM contexts are automatically marked as
non-connected if the card has the fully_routed flag set. This makes sense since
widgets which qualify for auto-disconnection are only found in CODEC DAPM
contexts. But with componentisation this is going to change, so consider all
widgets for auto-disconnection.

Also it is probably faster to walk the widgets list only once rather than once
for each CODEC.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-21 21:06:56 +01:00
Lars-Peter Clausen
bb13109d85 ASoC: Split component registration into two steps
Split snd_soc_component_register() into snd_soc_component_initialize() and
snd_soc_component_add(). Using a 2-stage registration approach has the advantage
that it is possible to modify the component after it has been initialized, but
before it is made visible to the system. This e.g. allows CODECs or platforms to
overwrite some of the default settings made in snd_soc_component_initialize().

Similar snd_soc_component_unregister() is split into two steps as well,
snd_soc_component_delete(), which removes the component from the system, and
snd_soc_component_cleanup(), which frees all the resources allocated by the
component.

Furthermore this patch makes sure that if a component is visible on two list
(e.g. the component list and the CODEC list) it is added or removed to both
lists atomically.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-21 21:05:13 +01:00
Lars-Peter Clausen
f4333203ec ASoC: Move name and id from CODEC/platform to component
The component struct already has a name and id field which are initialized to
the same values as the same fields in the CODEC and platform structs. So remove
them from the CODEC and platform structs and used the ones from the component
struct instead.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-21 21:04:24 +01:00
Lars-Peter Clausen
94f99c875c ASoC: Move name_prefix from CODEC to component
Move the name_prefix from the CODEC struct to the component struct. This will
eventually allow to specify prefixes for all types of components. It is also
necessary to make the DAPM code component type independent (i.e. a DAPM context
does not need to know whether it belongs to a CODEC or a platform or something
else).

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-21 21:03:22 +01:00
Lars-Peter Clausen
9f98cd69c1 ASoC: sh/fsi: Make one-bit bitfields unsigned
One-bit signed bitfields have two possible values: 0 and -1. This sometimes
leads to unexpected results (e.g. foo.bar = 1; foo.bar == 1 => false) which is
why it is recommended to make one-bit bitfields unsigned.

This fixes the following sparse warnings:
	sound/soc/sh/fsi.c:267:25: error: dubious one-bit signed bitfield
	sound/soc/sh/fsi.c:268:22: error: dubious one-bit signed bitfield
	sound/soc/sh/fsi.c:269:20: error: dubious one-bit signed bitfield
	sound/soc/sh/fsi.c:270:28: error: dubious one-bit signed bitfield
	sound/soc/sh/fsi.c:271:26: error: dubious one-bit signed bitfield
	sound/soc/sh/fsi.c:272:25: error: dubious one-bit signed bitfield

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-21 21:02:08 +01:00
Lars-Peter Clausen
cd7bcc6000 ASoC: rcar: Fix dma direction type
dmaengine_prep_slave_single() expects a enum dma_transfer_direction and not a
enum dma_data_direction. Since the integer representations of both DMA_TO_DEVICE
and DMA_MEM_TO_DEV aswell as DMA_FROM_DEVICE and DMA_DEV_TO_MEM have the same
value the code worked fine even though it was using the wrong type.

Fixes the following warning from sparse:
	sound/soc/sh/rcar/core.c:227:49: warning: mixing different enum types
	sound/soc/sh/rcar/core.c:227:49:     int enum dma_data_direction  versus
	sound/soc/sh/rcar/core.c:227:49:     int enum dma_transfer_direction

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-21 21:01:01 +01:00
Lars-Peter Clausen
76a77f4712 ASoC: omap-pcm: Include omap-pcm.h
omap_pcm_platform_register() is declared in omap-pcm.h and defined in
omap-pcm.c. To make sure that the function signature matches for both omap-pcm.c
should include omap-pcm.h

Fixes the following warning from sparse:
	sound/soc/omap/omap-pcm.c:235:5: warning: symbol
	'omap_pcm_platform_register' was not declared. Should it be static?

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-21 20:59:20 +01:00
Lars-Peter Clausen
afb7bb45bb ASoC: cs42xx8: Make of match table static
The cs42xx8_of_match table is not used outside of the driver, hence it can and
should be made static.

Fixes the following warning from sparse:
	sound/soc/codecs/cs42xx8.c:425:27: warning: symbol 'cs42xx8_of_match' was
	not declared. Should it be static?

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-21 20:58:19 +01:00
Lars-Peter Clausen
914bc160ef ASoC: tlv320aic31xx: Remove duplicate const
SOC_ENUM_SINGLE_DECL() already adds the const qualifier, so there is no need to
manually specify it.

Fixes the following warnings from sparse:
	sound/soc/codecs/tlv320aic31xx.c:253:1: warning: duplicate const
	sound/soc/codecs/tlv320aic31xx.c:255:1: warning: duplicate const
	sound/soc/codecs/tlv320aic31xx.c:257:1: warning: duplicate const
	sound/soc/codecs/tlv320aic31xx.c:260:1: warning: duplicate const
	sound/soc/codecs/tlv320aic31xx.c:262:1: warning: duplicate const

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-21 20:57:18 +01:00