Rev. E of the M-Audio Delta 66 is partially supported (commit
ef2cd2ccad), but the layout of the GPIO
pins was still unclear. This patch adds the GPIO definitions so that
communication to the CS8247 & 2x AK4524 works correctly.
ALSA bug#3327 has more details; users cap & jhunt report there that the
GPIO wiring is similar to the Digigram VX442 (chip select: pin 4 =
CS8427, pin 5 = AK4524 #0, pin 6 = AK4524 #1). There has been a lot of
conflicting information in the bug, but given these definitions, my
Delta 66E works; I tested analog in&out at 44.1kHz & 96kHz, analog gain
settings, S/PDIF clock sync, and S/PDIF in&out at 44.1kHz.
Signed-off-by: Brian Bloniarz <brian.bloniarz@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: http://launchpad.net/497546
Confirmed that the ideapad model works better than the current
quirk for Dell Vostro 320.
Cc: stable@kernel.org (2.6.35+)
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
create fixup function for the mario model and override amp capabilities
for NID 0x2
Signed-off-by: Todd Broch <tbroch@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Facilitate fixup for realtek codecs via modelname lookup of fixup
data. Fallback to quirk based lookup in absence of model definition.
Signed-off-by: Todd Broch <tbroch@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Uk Kim <w0806.kim@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM8994 supports mono signals - enable this in the driver. With DSP
mode an automatic data channel selector is available, activate this
when in mono mode.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
In case the codec driver did not provide a read/write function,
codec->driver->read|write will be NULL. Ensure that we use the one
specified in codec->read|write to avoid oopsing when we access
the debugfs entries. This is achieved by using snd_soc_read() and
snd_soc_write().
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some newer chips have more than one HDMI output, but usually not
all of them are exposed as physical jacks. Removing the unused
PCM devices (as indicated by BIOS in the pin config default) will
reduce user confusion as they currently have to choose between
several HDMI devices, some of them not working anyway.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a new HDMI/DP device is plugged in, hdmi_update_short_audio_desc()
is called for every SAD (Short Audio Descriptor) in the ELD data. For
LPCM coding type SAD defines the supported sample sizes. For several
other coding types (such as AC-3), a maximum bitrate is defined.
The maximum bitrate and sample size fields are not always cleared.
Therefore, if a device is unplugged and a different one is plugged in,
and the coding types of some SAD positions differ between the devices,
the old max_bitrate or sample_bits values will persist if the new SADs
do not define those values.
The leftover max_bitrate and sample_bits do not cause any issues other
than wrongly showing up in eld#X.Y procfs file and kernel log.
Fix that by always clearing sample_bits and max_bitrate when reading
SADs.
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit bbbe33900d added functionality to restrict PCM parameters
based on ELD info (derived from EDID data) of the audio sink.
However, according to CEA-861-D no SAD is needed for basic audio
(32/44.1/48kHz stereo 16-bit audio), which is instead indicated with a
basic audio flag in the CEA EDID Extension.
The flag is not present in ELD. However, as all audio capable sinks are
required to support basic audio, we can assume it to be always
available.
Fix allowed audio formats with sinks that have SADs (Short Audio
Descriptors) which do not completely overlap with the basic audio
formats (there are no reports of affected devices so far) by always
assuming that basic audio is supported.
Reported-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Cc: stable@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit bbbe33900d added functionality to restrict PCM parameters
based on ELD info (derived from EDID data) of the audio sink.
However, it wrongly assumes that the bits 0-2 of the first byte of
CEA Short Audio Descriptors mean a supported number of channels. In
reality, they mean the maximum number of channels (as per CEA-861-D
7.5.2). This means that the channel count can only be used to restrict
max_channels, not min_channels.
Restricting min_channels causes us to deny opening the device in stereo
mode if the sink only has SADs that declare larger numbers of channels
(like Primare SP32 AV Processor does).
Fix that by not restricting min_channels based on ELD information.
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Reported-by: Jean-Yves Avenard <jyavenard@gmail.com>
Tested-by: Jean-Yves Avenard <jyavenard@gmail.com>
Cc: stable@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In case of SNDRV_PCM_FORMAT_S32_LE, we need to set WM8580_AIF_LENGTH_32,
rather than WM8580_AIF_LENGTH_24.
Also, the BCLK has to be 64fs, for sample size of 20, 24 and 32 bits.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Properly free the resources in the case of snd_card_register failure
and soc_register_ac97_dai_link failure.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Properly free the resources in the case of soc_register_ac97_dai_link failure.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There is no need to mark this function as inline. Inline functions
usually are small and concise functions that benefit from not needing
to set up a stack frame and undergo a call/ret sequence upon each
invocation.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
By using strncpy() if the source string does not have a null byte in the
first n bytes, then the destination string is not null-terminated.
This can be fixed in a two-step process by manually null-terminating the
array after the use of strncpy() or by using strlcpy().
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Commit 2eea392 "ASoC: Add support for optional auxiliary dailess codecs"
added much of code that can be shared with DAI link codec probing/removal.
Merge now this common code into new soc_probe_codec, soc_remove_codec and
soc_post_component_init functions.
Error prints in these functions are converted to use dev_err and to print
the error code.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is a patch to the sound/ac97_bus.c file that fixes up a 80 character
line limit issue found by the checkpatch.pl tool.
Signed-off-by: Jeffrin Jose <ahiliation@yahoo.co.in>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added an optional name member to snd_soc_cache_ops to enable more
sensible diagnostic messages during cache init, exit and sync.
Remove redundant newline in source code.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch fixed intial return value to be a '0' as asuccess on
set_audio_clock_heirachy(). This avoids unintended error on initialize.
Signed-off-by: Seungwhan Youn <sw.youn@samsung.com>
Acked-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
As jack detection can trigger DAPM and the latency in debouncing can create
confusing windows in operation provide some trace events which will hopefully
help in diagnostics. The soc-jack core traces all reports that it gets and
the resulting notifications to upper layers. An event for jack IRQs is also
provided for instrumentation of debounce, and used in the GPIO jack code.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Reformat and update the comments that describe the hardware connections
on the various models.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of the hardcoded "CMI8788", show the actual chip name.
Note: This is neither what the chip is (it's always the same),
nor what the chip is actually called.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
To help with debugging, add the registers of the model-specific
codecs to the controller and AC97 register dump in the proc file.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The "Front Panel" switch on the Xonar D1/DX actually switches only the
output direction, so mark it appropriately.
The front panel microphone is controlled by the FMIC2MIC bit of the
CM9780. It was unconditionally enabled on the D1/DX and never set on
the ST(X); add a control for it. Selecting the front panel microphone
as source does not actually disable the microphone jack, but this is
bug-compatible with the Windows driver, and users rely on it.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The GPIO bit that enables analog output on the Xonar HDAV1.3 also
disables the HDMI audio output, so we better add a switch for it.
Hopefully, this is sufficient to make the HDMI output work.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Initialize the configuration of some unknown GPIO output bits (that
might not be used at all) to be the same as in the Windows driver, just
to be sure.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As the comments of snd_soc_instantiate_cards() said,
snd_soc_instantiate_cards() must be called with client_mutex.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
SPKOUTL_BOOST start from third bit, SPKOUTLR_BOOST start from 0 bit.
Signed-off-by: Uk Kim <w0806.kim@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
Currently the machine driver can only do bias level configuration before
the CODEC bias level is brought up. This means that the machine cannot do
any configuration which depends on the CODEC bias level being maintained.
Provide a post-CODEC callback which allows the machine driver to do things
like enable the FLL on a CODEC which is brought down to BIAS_OFF when idle.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
BugLink: https://launchpad.net/bugs/685161
The reporter of the bug states that he must use position_fix=1 to enable
capture for the internal microphone, so set it for his machine's PCI
SSID. Verified using 2.6.35 and the 2010-12-04 alsa-driver build.
Reported-and-tested-by: Ralph Wabel <rwabel@gmx.net>
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Switch to the generic hdmi parser for codec id 1002:aa01 (ATI R6xx
HDMI), as the codec appears to work fine with it.
Note that the codec is still limited to stereo output only, despite it
reportedly being multichannel capable. Some as of yet unknown quirks
will be needed to get that working.
Testing was done on 2.6.36 by John Ettedgui.
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Tested-by: John Ettedgui <john.ettedgui@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added a quirk to cxt5066_cfg_tbl to enable jack sense for ThinkPad Edge 13.
Reference: http://launchpad.net/bugs/685015
Signed-off-by: Manoj Iyer <manoj.iyer@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ASoC: omap: N810: Don't select CONFIG_OMAP_MUX but make it as dependency
ALSA: hda: Use "alienware" model quirk for another SSID
ASoC: WM8731: Fix incorrect mask for bypass path disable
s6105-ipcam: fix compilation
s6000-pcm: fix compilation
s6000-i2s: fix compilation
ASoC: Fix missing spin_unlock_irqrestore
ALSA: Fix SNDCTL_DSP_RESET ioctl for OSS emulation
ASoC: Add missing dev_set_drvdata in p1022_ds_probe
ASoC: Add missing dev_set_drvdata in mpc8610_hpcd_probe
ASoC: Remove unneeded !! operations while checking return value of nuc900_checkready
ASoC: Fix compile error for nuc900-pcm.c
ASoC: Fix prototype for nuc900_ac97_probe and nuc900_ac97_remove
ASoC: Fix compile error for nuc900-ac97.c
ALSA: hda: Use BIOS auto-parsing instead of existing model quirk for MEDION MD2
When we disable the WM8994 FLL code path sharing means that we end up
writing out a configuration. Currently this is the currently active
input and output frequency (which causes snd_soc_update_bits() to
suppress actual writes both immediately and in the common case where
we reenable the same configuration later) but we allow machine drivers
to pass through a source of zero. Since the register values written
are one less than the source constants this causes corruption of other
bitfields in the register.
Fix this by using the most recently configured FLL source when none is
provided.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Cc: stable@kernel.org
We want the index of DAI's driver here.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Allow the CODEC driver structure to be marked const by making all
the APIs that use it do so.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
This patch allows machine drivers to override the compression type
provided by the codec driver.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Make sure to use codec->reg_def_copy instead of codec_drv->reg_cache_default
wherever necessary. This change is necessary because in the next patch we
move the cache initialization code outside snd_soc_register_codec() and by that
time any data marked as __devinitconst such as the original reg_cache_default
array might have already been freed by the kernel.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The snd_soc_codec_conf struct now holds codec specific configuration
information.
A new configuration option has been added to allow machine drivers to
override the compression type set by the codec driver.
In the absence of providing an snd_soc_codec_conf struct or when providing
one but not setting the compress_type member to anything, the one supplied
by the codec driver will be used instead. In all other cases the one
set in the snd_soc_codec_conf struct takes effect.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We need to keep a copy of the compress_type supplied by the codec driver
so that we can override it if necessary with whatever the machine driver
has provided us with. The reason for not modifying the codec->driver
struct directly is that ideally we'd like to keep it const.
Adjust the code in soc-cache and soc-core to make use of the compress_type
member in the snd_soc_codec struct.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Blind copy of codec finding algorithm from soc_bind_dai_link does not work
in soc_probe_aux_dev if matching codec name is not found. In that case the
code falls through and tries to start the probing procedure with invalid
codec pointer.
Fix this and add an error print showing the codec name that cannot be found.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In alc5623_i2c_probe(),
the default case for checking alc5623->id behaves the same as case 0x23.
However, In alc5623_probe() the default case for checking alc5623->id
becomes to be the same as case 0x21.
This makes the meaning of default case inconsistent.
Since we have checked codec id in alc5623_i2c_probe() by comparing
vid2 with id->driver_data, it is not possible to run into the default case now.
In case we may add more supported devices to alc5623_i2c_table in the future,
this patch changes the default case return -EINVAL to let people know that
they should not run into this case. They should also add a new case accordingly
for the new id.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Arnaud Patard <arnaud.patard@rtp-net.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Current FSI driver was checking register range on fsi_master_xxx function.
This runtime check was added to avoid an illegal access
from wrong/mistake implementation.
But it is useless check under the correct implementation.
This patch escape runtime check by using macro technique.
If there is wrong implementation, it will be compile error.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Current FSI driver was checking register range on fsi_reg_xxx function.
This runtime check was added to avoid an illegal access
from wrong/mistake implementation.
But it is useless check under the correct implementation.
This patch escape runtime check by using macro technique.
If there is wrong implementation, it will be compile error.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There was a strange part where fsi->xxx had been used
for fsi_master_xxx function instead of master->xxx in current FSI.
This patch modify it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We shouldn't be assigning to the driver structure (which really ought
to be const, further patch to follow) though there's unlikely to be any
actual problem except in the unlikely case that two devices with the
same driver but different bus types appear in the same system.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Silly little grammar nit but it bugs the hell out of me.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Instead of hard-coded magic numbers, properly define and use macros
for improve the readability. Also, dell_automute is handled samely
as thinkpad, since it also sets port_d_mode, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On the docking station for the Lenovo T410 and T410s, the line-out
doesn't work. The trouble seems to be that it generates a plug event,
but then doesn't report that the jack is connected. So automute mutes
the jack when you plug something into it. The following patch (next
message) fixes it.
Signed-off-by: John Baboval <john.baboval at virtualcomputer.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://launchpad.net/bugs/595482
The original reporter states that audible playback from the internal
speaker is inaudible despite the hardware being properly detected. To
work around this symptom, he uses the model=lg quirk to properly enable
both playback, capture, and jack sense. Another user corroborates this
workaround on separate hardware. Add this PCI SSID to the quirk table
to enable it for further LG P1 Expresses.
Reported-and-tested-by: Philip Peitsch <philip.peitsch@gmail.com>
Tested-by: nikhov
Cc: <stable@kernel.org> [2.6.32+]
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Update some of the default configuration for the device to improve
the performance.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
The deemphasis filter should be selected based on sample rate for
optimal performance.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
This is much more maintainable than the array.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
The use of DAPM widgets, and extra routing can cause ordering
problems in the system.
Machine drivers should use the exported direct interface with
SND_SOC_DAPM_HP's event callback to manage the state of the
amplifier.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Not all omap boards use kernel based pin multiplexing so
CONFIG_SND_OMAP_SOC_N810 should not select it by default as it can make
harm to other boards in multi-board kernels.
Therefore put CONFIG_OMAP_MUX as a dependency to N810 ASoC machine driver.
Thanks to Tony Lindgren <tony@atomide.com> for noticing.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Cc: Tony Lindgren <tony@atomide.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Tony Lindgren <tony@atomide.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
BugLink: https://launchpad.net/bugs/683695
The original reporter states that headphone jacks do not appear to
work. Upon inspecting his codec dump, and upon further testing, it is
confirmed that the "alienware" model quirk is correct.
Reported-and-tested-by: Cody Thierauf
Cc: <stable@kernel.org> [2.6.32+]
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The mute/unmute is controled by SMUTE (Soft Mute Control bit):
0: Normal Operation (Default)
1: DAC outputs soft-muted
I think this change improves readability.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The paramters of the WM8958 multiband compressor can be tuned by the
user for their system using a graphical configuration tool on the host.
Allow the user to specify a set of such paramters in platform data and
select between them at runtime.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
We need a post notification as we need to shut down the MBC after the
data stops flowing rather than before.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Commit 2eea392 "ASoC: Add support for optional auxiliary dailess codecs"
causes a build failure in soc-core.c: soc_probe_aux_dev since code tries to
access non-existing struct snd_soc_dapm_context and struct snd_soc_card
members.
Root cause for this was a double accident. Author sent the RFC patch from
top of another patch set and the RFC got committed. Fix the build failure
by removing the code line that depends on that another patch set.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
According to the datasheet the bypass path enable/disable is
bit 3 therefore we need 0x8 and not 0x4.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The PXA-SPI driver relies on some files / defines which are arm specific
and are within the ARM tree. The CE4100 SoC which is x86 has also the
SPI core.
This patch moves the ssp and spi files from arm/mach-pxa and plat-pxa to
include/linux where the CE4100 can access them.
This move got verified by building the following defconfigs:
cm_x2xx_defconfig corgi_defconfig em_x270_defconfig ezx_defconfig
imote2_defconfig pxa3xx_defconfig spitz_defconfig zeus_defconfig
raumfeld_defconfig magician_defconfig
Signed-off-by: Sebastian Andrzej Siewior <bigeasy@linutronix.de>
Signed-off-by: Dirk Brandewie <dirk.brandewie@gmail.com>
Add support for the RME HDSP RPM IO box. Changes have been made in the identification of the IO box and the neccessary controls have been added.
Signed-off-by: Florian Faber <faberman@linuxproaudio.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
New DAPM widgets, and paths to enable both channels at the
same time (for stereo output).
With this path the switch time difference can be avoided
between left and right channels.
The original DAPM paths can be still used, if only one of
TPA's output has been connected to a speaker, but for most of
the cases, switching to the stereo path is better.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Reduce the amount of duplicated code by using single
event handler for PGA_E to enable the needed channel.
Use the w->shift to pass the channel information to
the handler function.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Do not enable the amplifier right after the power has been
restored to the amplifier.
The DAPM_SUPPLY widget turns on the amp early in the DAPM
power walk, and the unmuting of channel happens quite late.
Keeping the amp in SW reset state ensures better muting.
In this way the pop noise coming from other components (codec)
can be filtered out.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Use simpler way to avoid setting the same power state
for the amplifier.
Simplifies the check introduced by patch:
ASoC: tpa6130a2: Fix unbalanced regulator disables
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Cc: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
During playback start the codec has been already powered at
BIAS_ON event time, so there's no need to enable the codec again.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
No need to enable the codec at this time.
The codec will be enabled later by other events
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
This makes possible to register auxiliary dailess codecs in a machine
driver. Term dailess is used here for amplifiers and codecs without DAI or
DAI being unused.
Dailess auxiliary codecs are kept in struct snd_soc_aux_dev and those codecs
are probed after initializing the DAI links. There are no major differences
between DAI link codecs and dailess codecs in ASoC core point of view. DAPM
handles them equally and sysfs and debugfs directories for dailess codecs
are similar except the pmdown_time node is not created.
Only suspend and resume functions are modified to traverse all probed codecs
instead of DAI link codecs.
Example below shows a dailess codec registration.
struct snd_soc_aux_dev foo_aux_dev[] = {
{
.name = "Amp",
.codec_name = "codec.2",
.init = foo_init2,
},
};
static struct snd_soc_card card = {
...
.aux_dev = foo_aux_dev,
.num_aux_devs = ARRAY_SIZE(foo_aux_dev),
};
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The bitmap_zero() nbits argument was improperly set to reg_size
but the underlying buffer was bmp_size long. This caused the memset
to zero past the end of the allocated buffer and into the kernel heap
causing strange kernel crashes sometimes by overwriting critical
kernel structures.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Simplify pm860x_probe error handling and return actual error code we got.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Current FSI-Ak4642 device had niche settings which were
FSI2-A-AK4643 and FSI-A-AK4642.
This patch add platform_device_id which can control
FSI/FSI2, PortA/PortB, AK4642/AK4643 from platform data.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Paul Mundt <lethal@linux-sh.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When the s6105-ipcam ASoC driver had been converted to the
multi-component API, a single reference to a former structure
element remained, blocking successful compilation.
Signed-off-by: Daniel Glöckner <daniel-gl@gmx.net>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
s6000_soc_platform has lost its forward declaration and there no
longer is a name element in it, so use a string constant when
calling request_irq.
Signed-off-by: Daniel Glöckner <daniel-gl@gmx.net>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
A semicolon was missing.
Signed-off-by: Daniel Glöckner <daniel-gl@gmx.net>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In nuc900_dma_hw_params(), if snd_pcm_lib_malloc_pages failed
it returns without calling spin_unlock_irqrestore().
Since snd_pcm_lib_malloc_pages() does not touch struct nuc900_audio,
we don't need to hold the lock while calling snd_pcm_lib_malloc_pages().
Fix it by moving spin_lock_irqsave() down to after snd_pcm_lib_malloc_pages().
In nuc900_dma_prepare(), spin_unlock_irqrestore() is missing in the error path.
Fix it by removing the return in default case.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Wan ZongShun <mcuos.com@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In OSS emulation, SNDCTL_DSP_RESET ioctl needs the reset of the internal
buffer state in addition to drop of the running streams. Otherwise the
succeeding access becomes inconsistent.
Tested-by: Amit Nagal <helloin.amit@gmail.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Otherwise, calling dev_get_drvdata in p1022_ds_remove returns NULL.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Timur Tabi <timur@freescale.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Otherwise, calling dev_get_drvdata in mpc8610_hpcd_remove returns NULL.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Timur Tabi <timur@freescale.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
I think this unneededd !! operations just reduce the readability.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch fixes below error:
CC sound/soc/nuc900/nuc900-pcm.o
sound/soc/nuc900/nuc900-pcm.c: In function 'nuc900_dma_open':
sound/soc/nuc900/nuc900-pcm.c:267: error: 'nuc900_ac97_data' undeclared (first use in this function)
sound/soc/nuc900/nuc900-pcm.c:267: error: (Each undeclared identifier is reported only once
sound/soc/nuc900/nuc900-pcm.c:267: error: for each function it appears in.)
sound/soc/nuc900/nuc900-pcm.c: At top level:
sound/soc/nuc900/nuc900-pcm.c:337: error: expected ',' or ';' before 'static'
sound/soc/nuc900/nuc900-pcm.c:354: error: 'nuc900_soc_platform_probe' undeclared here (not in a function)
make[3]: *** [sound/soc/nuc900/nuc900-pcm.o] Error 1
make[2]: *** [sound/soc/nuc900] Error 2
make[1]: *** [sound/soc] Error 2
make: *** [sound] Error 2
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch fixes below compile warning:
CC sound/soc/nuc900/nuc900-ac97.o
sound/soc/nuc900/nuc900-ac97.c:300: warning: initialization from incompatible pointer type
sound/soc/nuc900/nuc900-ac97.c:301: warning: initialization from incompatible pointer type
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix below compile error by add a missing ';'.
CC sound/soc/nuc900/nuc900-ac97.o
sound/soc/nuc900/nuc900-ac97.c:300: warning: initialization from incompatible pointer type
sound/soc/nuc900/nuc900-ac97.c:301: warning: initialization from incompatible pointer type
sound/soc/nuc900/nuc900-ac97.c:318: error: expected ',' or ';' before 'static'
sound/soc/nuc900/nuc900-ac97.c:405: error: 'nuc900_ac97_drvprobe' undeclared here (not in a function)
make[3]: *** [sound/soc/nuc900/nuc900-ac97.o] Error 1
make[2]: *** [sound/soc/nuc900] Error 2
make[1]: *** [sound/soc] Error 2
make: *** [sound] Error 2
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Ensure that we report any errors encountered during reads/writes
in the cache syncing functions.
Remove redundant newline in the source code.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Properly free allocated resources in smdk_init() error path.
Add missing platform_device_unregister() in smdk_exit().
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Properly free allocated resources in neo1973_gta02_init() error path.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Properly free allocated resources in goni_init() error path.
Add missing snd_soc_unregister_dai() in goni_exit().
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
BugLink: https://launchpad.net/bugs/682199
A 2.6.35 (Ubuntu Maverick) user, burningphantom1, reported a regression
in audio: playback was inaudible through both speakers and headphones.
In commit 272a527c04 of sound-2.6.git, a new model was added with this
machine's PCI SSID. Fortunately, it is now sufficient to use the auto
model for BIOS auto-parsing instead of the existing quirk.
Playback, capture, and jack sense were verified working for both
2.6.35 and the alsa-driver snapshot from 2010-11-27 when model=auto is
used.
Reported-and-tested-by: burningphantom1
Cc: <stable@kernel.org> [2.6.35+]
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This allows us to communicate our power management state back to the
parent device, allowing it to do a full power down when the device is
idle.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
The WM8994 and WM8958 register map is relatively sparse so benefits
from compression. The rbtree compression gives better results than
LZO for both memory and CPU consumption on a map as sparse as this.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
In the process we convert the driver to read registers one at a time
when initialising the cache. This has the effect of working around
limitations in the sizes of I2C transactions which can be done by some
CPUs. Due to the sparseness of the register map the overhead from this
should be minimual unless I2C transactions are very expensive to start.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
The WM8958 contains an advanced accessory detection feature which allows
detection of up to seven different impedence levels on the microphone
bias output, including detection of video outputs. Since some of the
more involved accessory interfaces may involve noticable interactions
with external components a simple detection scheme is provided by
default with the option to provide custom handling of accessory detect.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
The WM8958 features a multi-band compressor which can be enabled on
any of the AIF inputs. The MBC allows different gains to be applied to
differnt audio bands, providing an improvement in perceived loudness
of the signal by avoiding overdriving the output transducers. This
patch enables support for the MBC.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Additional audio routing options are available on the WM8958 audio
interface 3. Add support for these.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
The WM8958 is a WM8994 derivative. This patch merely ensures that some
revision specific configuration for WM8994 is not enabled on WM8994,
additional patches will add support for the new features introduced on
the WM8958.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Devices which do not have a DCS correction applied can use the explicit
DC servo startup mode for optimal startup performance. This most
immediately affects the WM8958.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
When SKU assid gives no valid bits for 0x38, the driver didn't take
any action, so far. This resulted in the missing initialization for
external amps, etc, thus the silent output in the end.
Especially users hit this problem on ALC888 newly since 2.6.35,
where the driver doesn't force to use ALC_INIT_DEFAULT any more.
This patch sets the default initialization scheme to use
ALC_INIT_DEFAULT when no valid bits are set for SKU assid.
Reference:
https://bugzilla.redhat.com/show_bug.cgi?id=657388
Reported-and-tested-by: Kyle McMartin <kyle@redhat.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In current implementation, there are resources leak in the error path.
This patch properly reclaims the allocated resources in the error path.
Also adds a missing clk_put in osk_soc_exit.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If aic3x_read failed , val is used uninitialized.
Fix it by initializing val to 0.
This patch fixes below compile warning:
sound/soc/codecs/tlv320aic3x.c: In function 'aic3x_get_gpio':
sound/soc/codecs/tlv320aic3x.c:1183: warning: 'val' may be used uninitialized in this function
sound/soc/codecs/tlv320aic3x.c: In function 'aic3x_headset_detected':
sound/soc/codecs/tlv320aic3x.c:1211: warning: 'val' may be used uninitialized in this function
sound/soc/codecs/tlv320aic3x.c: In function 'aic3x_button_pressed':
sound/soc/codecs/tlv320aic3x.c:1219: warning: 'val' may be used uninitialized in this function
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Supply a module exit function so that the driver can be unloaded.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Fix the error path to properly free allocated resources.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix below compile error:
CC sound/soc/samsung/smartq_wm8987.o
sound/soc/samsung/smartq_wm8987.c: In function 'smartq_hifi_hw_params':
sound/soc/samsung/smartq_wm8987.c:42: error: 'struct snd_soc_pcm_runtime' has no member named 'dai'
sound/soc/samsung/smartq_wm8987.c:43: error: 'struct snd_soc_pcm_runtime' has no member named 'dai'
sound/soc/samsung/smartq_wm8987.c: In function 'smartq_wm8987_init':
sound/soc/samsung/smartq_wm8987.c:192: warning: passing argument 1 of 'snd_soc_jack_new' from incompatible pointer type
sound/soc/samsung/smartq_wm8987.c: At top level:
sound/soc/samsung/smartq_wm8987.c:216: warning: initialization from incompatible pointer type
make[3]: *** [sound/soc/samsung/smartq_wm8987.o] Error 1
make[2]: *** [sound/soc/samsung] Error 2
make[1]: *** [sound/soc] Error 2
make: *** [sound] Error 2
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix imx_phycore_init() error path and imx_phycore_exit() to properly free
allocated resources.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Sascha Hauer <s.hauer@pengutronix.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix imx_ssi_probe() error path and imx_ssi_remove() to properly free
allocated resources.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Sascha Hauer <s.hauer@pengutronix.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix the error path to properly free allocated resources.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Mika Westerberg <mika.westerberg@iki.fi>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix the error path to properly free allocated resources.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add missing platform_device_put() if platform_device_add() failed.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add missing platform_device_put() if platform_device_add() failed.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We call snd_soc_register_dais() in sh4_soc_dai_probe(),
thus we should call snd_soc_unregister_dais() in sh4_soc_dai_remove().
Otherwise, we got "too many arguments to function 'snd_soc_unregister_dai'"
error message.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The mixer nids passed to alc_auto_create_input_ctls are wrong: 0x15 is
a pin, and 0x09 is the ADC on both ALC660-VD/ALC861-VD. Thus with
current code, input playback volume/switches and input source mixer
controls are not created, and recording doesn't work. Select correct
mixers, 0x0b (input playback mixer) and 0x22 (capture source mixer).
Reference: https://qa.mandriva.com/show_bug.cgi?id=61159
Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The patch enables ALC887-VD to use the DAC at nid 0x26,
which makes it possible to use this DAC for e g Headphone
volume.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The WM8737 is a low power, flexible stereo ADC designed for portable
applications. This driver supports most of the functionality of the
WM8737, though some features such as the ALC are not yet implemented.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Acked-by: Wan ZongShun <mcuos.com@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Return PTR_ERR(omap3pandora_dac_reg) instead of 0 if regulator_get failed.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
MCLKDIV bit of Register 04h Clocking1:
0 : Divide by 1
1 : Divide by 2
Thus in the case of freq <= 16500000, we should clear MCLKDIV bit.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
DACSLOPE bit of Register 06h ADC and DAC Control 2:
0: Normal mode
1: Sloping stop-band mode
Thus in the case of normal mode, we should clear DACSLOPE bit.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
Current AP4 FSI didn't use set_rate for ak4642,
and used dummy rate when init.
And FSI driver was modified to always call set_rate.
The user which are using FSI set_rate is only AP4 now.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Paul Mundt <lethal@linux-sh.org>
Current AP4 FSI set_rate function used bogus clock process
which didn't care enable/disable and clk->usecound.
To solve this issue, this patch also modify FSI driver to call
set_rate with enough options.
This patch modify it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Paul Mundt <lethal@linux-sh.org>
If CONFIG_SND_DYNAMIC_MINORS is used, assign /dev/snd/seq and
/dev/snd/timer the usual static minors, and export specific
module aliases to generate udev module on-demand loading
instructions:
$ cat /lib/modules/2.6.33.4-smp/modules.devname
# Device nodes to trigger on-demand module loading.
microcode cpu/microcode c10:184
fuse fuse c10:229
ppp_generic ppp c108:0
tun net/tun c10:200
uinput uinput c10:223
dm_mod mapper/control c10:236
snd_timer snd/timer c116:33
snd_seq snd/seq c116:1
The last two lines instruct udev to create device nodes, even
when the modules are not loaded at that time.
As soon as userspace accesses any of these nodes, the in-kernel
module-loader will load the module, and the device can be used.
The header file minor calculation needed to be simplified to
make __stringify() (supports only two indirections) in
the MODULE_ALIAS macro work.
This is part of systemd's effort to get rid of unconditional
module load instructions and needless init scripts.
Cc: Lennart Poettering <lennart@poettering.net>
Signed-off-by: Kay Sievers <kay.sievers@vrfy.org>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (41 commits)
ALSA: hda - Identify more variants for ALC269
ALSA: hda - Fix wrong ALC269 variant check
ALSA: hda - Enable jack sense for Thinkpad Edge 11
ALSA: Revert "ALSA: hda - Fix switching between dmic and mic using the same mux on IDT/STAC"
ALSA: hda - Fixed ALC887-VD initial error
ALSA: atmel - Fix the return value in error path
ALSA: hda: Use hp-laptop quirk to enable headphones automute for Asus A52J
ALSA: snd-atmel-abdac: test wrong variable
ALSA: azt3328: period bug fix (for PA), add missing ACK on stop timer
ALSA: hda: Add Samsung R720 SSID for subwoofer pin fixup
ALSA: sound/pci/asihpi/hpioctl.c: Remove unnecessary casts of pci_get_drvdata
ALSA: sound/core/pcm_lib.c: Remove unnecessary semicolons
ALSA: sound/ppc: Use printf extension %pR for struct resource
ALSA: ac97: Apply quirk for Dell Latitude D610 binding Master and Headphone controls
ASoC: uda134x - set reg_cache_default to uda134x_reg
ASoC: Add support for MAX98089 CODEC
ASoC: davinci: fixes for multi-component
ASoC: Fix register cache setup WM8994 for multi-component
ASoC: Fix dapm_seq_compare() for multi-component
ASoC: RX1950: Fix hw_params function
...
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Finally, move the 's3c24xx' directory to 'samsung'
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Enable the ASoC Machine driver to run on SMDKC100 as well.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Kukjin Kim <kgene.kim@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We plan to use the same ASoC Machine driver for most of
latest SMDK platforms. So rename the 64XX specific driver
to generic named.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Now that we have better I2S CPU drivers and no need for the old
ones, discard them.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Modify the smdk64xx_wm8580.c to use new i2s controller driver.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Modify the goni_wm8994.c to use new i2s controller driver.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Modify the smartq_wm8987.c to use new i2s controller driver.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The I2S controllers since S3C64XX are incremental revisions, with
a new feature added to the last one. The programming i/f doesn't
conflict between these revisions, so it is possible to have one
common driver that could manage various versions of I2S (v3, 4 & 5)
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since the I2S of S3C64XX and newer SoCs are incremental
versions of each other with changes managable in a single
driver, rename the 's3c64xx-iis' -> 'samsung-i2s'
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Call the AC97 controller devices found in S3C, S5P and newer
SoCs as 'samsung-ac97' rather than 's3c-ac97'.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Kukjin Kim <kgene.kim@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some Samsung SoCs have a PCM(DSP) controller. So the name
s3c24xx-pcm-audio for DMA driver is not very appropraite.
This patch moves :-
s3c24xx-pcm-audio -> samsung-audio
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Kukjin Kim <kgene.kim@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
AQUILA and GONI are essentially the same h/w w.r.t ASoC.
They only differ by the fact that GONI has stereo speaker-out
whereas AQUILA has mono.
The difference can easily be handled in the same MACHINE driver
by making machine-specific runtime changes.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The refactoring commit d433a67831
ALSA: hda - Optimize the check of ALC269 codec variants
introduced a wrong check for ALC269-vb type. This patch corrects it.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a quirk entry for Thinkpad Edge 11 as well as other TP Edge models.
Signed-off-by: Manoj Iyer <manoj.iyer@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
They went AWOL during the multi-component merge.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
After clk_get() mclk is checked second time instead of pllb check.
In patch v1 Jarkko Nikula noticed that PTR_ERR() is also has wrong argument.
Signed-off-by: Vasiliy Kulikov <segoon@openwall.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There is no need to include soc-dai.h since soc.h includes it. Convert
drivers to include only soc.h.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There is no need to include soc-dapm.h since soc.h includes it.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Commit ce6120c require that soc-dapm.h cannot be included before soc.h but
these two drivers were not checked. Fix them by including only soc.h as it
includes soc-dapm.h.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Looks like this is missing during multi-component conversion.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
After clk_get() mclk is checked three times instead of mout_epll
and sclk_spdif checks.
Signed-off-by: Vasiliy Kulikov <segoon@openwall.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add a generic callback function for fixup elements. This can be used
to do some unusual things like overriding the AMP cache, etc.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALC887-VD is like ALC888-VD. It can not be initialized as ALC882.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the commit c0763e687d
ALSA: snd-atmel-abdac: test wrong variable
the return value via PTR_ERR() had to be fixed as well.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://launchpad.net/bugs/677652
The original reporter states that, in 2.6.35, headphones do not appear
to work, nor does inserting them mute the A52J's onboard speakers. Upon
inspecting the codec dump, it appears that the newly committed hp-laptop
quirk will suffice to enable this basic functionality. Testing was done
with an alsa-driver build from 2010-11-21.
Reported-and-tested-by: Joan Creus
Cc: <stable@kernel.org> [2.6.35+]
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
After clk_get() pclk is checked second time instead of sample_clk check.
Signed-off-by: Vasiliy Kulikov <segoon@openwall.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
. Fix PulseAudio "ALSA driver bug" issue
(if we have two alternated areas within a 64k DMA buffer, then max
period size should obviously be 32k only).
Back references:
http://pulseaudio.org/wiki/AlsaIssueshttp://fedoraproject.org/wiki/Features/GlitchFreeAudio
. In stop timer function, need to supply ACK in the timer control byte.
. Minor log output correction
When I did my first PA testing recently, the period size bug resulted
in quite precisely observeable half-period-based playback distortion.
PA-based operation is quite a bit more underrun-prone (despite its
zero-copy optimizations etc.) than raw ALSA with this rather spartan
sound hardware implementation on my puny Athlon.
Note that even with this patch, azt3328 still doesn't work for both
cases yet, PA tsched=0 and tsched
(on tsched=0 it will playback tiny fragments of periods, leading to tiny
stuttering sounds with some pauses in between, whereas with
timer-scheduled operation playback works fine - minus some quite increased
underrun trouble on PA vs. ALSA, that is).
Signed-off-by: Andreas Mohr <andi@lisas.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://launchpad.net/bugs/677830
The original reporter states that the subwoofer does not mute when
inserting headphones. We need an entry for his machine's SSID in the
subwoofer pin fixup list, so add it there (verified using hda_analyzer).
Reported-and-tested-by: i-NoD
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Increase the default timer limit so that snd-hrtimer.ko can be
automatically loaded when needed, e.g., when used as the default
sequencer timer. This replaces the check for the obsolete
CONFIG_SND_HPET.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a lightweight condition on top of the xrun checking so that we can
avoid the division when the application is calling the update function
often enough.
Suggested-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When period wakeups are disabled, successive calls to the pointer update
function do not have a maximum allowed distance, so xruns cannot be
detected with the pointer value only.
To detect xruns, compare the actually elapsed time with the time that
should have theoretically elapsed since the last update. When the
hardware pointer has wrapped around due to an xrun, the actually elapsed
time will be too big by about hw_ptr_buffer_jiffies.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Allow disabling period wakeup interrupts for all PCM streams.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch allows to disable period interrupts which are
not needed when the application relies on a system timer
to wake-up and refill the ring buffer. The behavior of
the driver is left unchanged, and interrupts are only
disabled if the application requests this configuration.
The behavior in case of underruns is slightly different,
instead of being detected during the period interrupts the
underruns are detected when the application calls
snd_pcm_update_avail, which in turns forces a refresh of the
hw pointer and shows the buffer is empty.
More specifically this patch makes a lot of sense when
PulseAudio relies on timer-based scheduling to access audio
devices such as HDAudio or Intel SST. Disabling interrupts
removes two unwanted wake-ups due to period elapsed events
in low-power playback modes. It also simplifies PulseAudio
voice modules used for speech calls.
To quote Lennart "This patch looks very interesting and
desirable. This is something have long been waiting for."
Support for this in hardware drivers is optional.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@intel.com>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://launchpad.net/bugs/677652
The original reporter states that, in 2.6.35, headphones do not appear
to work, nor does inserting them mute the A52J's onboard speakers. Upon
inspecting the codec dump, it appears that the newly committed hp-laptop
quirk will suffice to enable this basic functionality. Testing was done
with an alsa-driver build from 2010-11-21.
Reported-and-tested-by: Joan Creus
Cc: <stable@kernel.org> [2.6.35+]
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
After clk_get() pclk is checked second time instead of sample_clk check.
Signed-off-by: Vasiliy Kulikov <segoon@openwall.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
. Fix PulseAudio "ALSA driver bug" issue
(if we have two alternated areas within a 64k DMA buffer, then max
period size should obviously be 32k only).
Back references:
http://pulseaudio.org/wiki/AlsaIssueshttp://fedoraproject.org/wiki/Features/GlitchFreeAudio
. In stop timer function, need to supply ACK in the timer control byte.
. Minor log output correction
When I did my first PA testing recently, the period size bug resulted
in quite precisely observeable half-period-based playback distortion.
PA-based operation is quite a bit more underrun-prone (despite its
zero-copy optimizations etc.) than raw ALSA with this rather spartan
sound hardware implementation on my puny Athlon.
Note that even with this patch, azt3328 still doesn't work for both
cases yet, PA tsched=0 and tsched
(on tsched=0 it will playback tiny fragments of periods, leading to tiny
stuttering sounds with some pauses in between, whereas with
timer-scheduled operation playback works fine - minus some quite increased
underrun trouble on PA vs. ALSA, that is).
Signed-off-by: Andreas Mohr <andi@lisas.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://launchpad.net/bugs/677830
The original reporter states that the subwoofer does not mute when
inserting headphones. We need an entry for his machine's SSID in the
subwoofer pin fixup list, so add it there (verified using hda_analyzer).
Reported-and-tested-by: i-NoD
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: http://launchpad.net/bugs/669092
ALC887 does not have any volume control ability on the mixer NIDs,
so put the volume controls on the dac NIDs instead. Without this
patch, ALC887 users cannot use alsamixer at all.
Cc: stable@kernel.org
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://launchpad.net/bugs/669279
The original reporter states: "The Master mixer does not change the
volume from the headphone output (which is affected by the headphone
mixer). Instead it only seems to control the on-board speaker volume.
This confuses PulseAudio greatly as the Master channel is merged into
the volume mix."
Fix this symptom by applying the hp_only quirk for the reporter's SSID.
The fix is applicable to all stable kernels.
Reported-and-tested-by: Ben Gamari <bgamari@gmail.com>
Cc: <stable@kernel.org> [2.6.32+]
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As we allocate memory for twl4030 in twl4030_codec_probe(),
twl4030_codec_remove() is a better place to free the memory.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Makes the WM8994 driver file itself substantially smaller.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
After checking the code in 2.6.36,
I found this is missing during multi-component conversion.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Allow the application to choose if the ADC data presented on the left
and right channels is sourced from the internal left or right channel.
This allows a mono recording to be presented as stereo on the external
bus.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
This patch adds initial support for the MAX98089 CODEC.
Signed-off-by: Jesse Marroquin <jesse.marroquin@maxim-ic.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Multi-component commit f0fba2ad broke a few things which this patch should
fix. Tested on the DM355 EVM. I've been as careful as I can, but it would be
good if those with access to other Davinci boards could test.
--
The multi-component commit put the initialisation of
snd_soc_dai.[capture|playback]_dma_data into snd_soc_dai_ops.hw_params of the
McBSP, McASP & VCIF drivers (davinci-i2s.c, davinci-mcasp.c & davinci-vcif.c).
The initialisation had to be moved from the probe function in these drivers
because davinci_*_dai changed from snd_soc_dai to snd_soc_dai_driver.
Unfortunately, the DMA params pointer is needed by davinci_pcm_open (in
davinci-pcm.c) before hw_params is called. I have moved the initialisation to
a new snd_soc_dai_ops.startup function in each of these drivers. This fix
indicates that all platforms that use davinci-pcm must have been broken and
need to test with this fix.
--
The multi-component commit also changed the McBSP driver name from
"davinci-asp" to "davinci-i2s" in davinci-i2s.c without updating the board
level references to the driver name. This change is understandable, as there
is a similarly named "davinci-mcasp" driver in davinci-mcasp.c.
There is probably no 'correct' name for this driver. The DM6446 datasheet
calls it the "ASP" and describes it as a "specialised McBSP". The DM355
datasheet calls it the "ASP" and describes it as a "specialised ASP". The
DM365 datasheet calls it the "McBSP". Rather than fix this problem by
reverting to "davinci-asp", I've elected to avoid future confusion with the
"davinci-mcasp" driver by changing it to "davinci-mcbsp", which is also
consistent with the names of the functions in the driver. There are other
fixes required, so it was never going to be as simple as a revert anyway.
--
The DM365 only has one McBSP port (of the McBSP platforms, only the DM355 has
2 ports), so I've changed the the id of the platform_device from 0 to -1.
--
In davinci-evm.c, the DM6446 EVM can no longer share a snd_soc_dai_link
structure with the DM355 EVM as they use different cpu DAI names (the DM355
has 2 ports and the EVM uses the second port, but the DM6446 only has 1 port).
This also means that the 2 boards need different snd_soc_card structures.
--
The codec_name entries in davinci-evm.c didn't match the i2c ids in the board
files. I have only checked and fixed the details of the names used for the
McBSP based platforms. Someone with a McASP based platform (eg DA8xx) should
check the others.
Signed-off-by: Chris Paulson-Ellis <chris@edesix.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
During the multi-component conversion the WM8994 register cache init
got lost.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
The big kernel lock has been removed from all these files at some point,
leaving only the #include.
Remove this too as a cleanup.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
DAPM widgets may be associated with non-CODEC devices so compare based
on the DAPM context rather than the CODEC pointer.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
We allocated memory for wm8962 in wm8962_i2c_probe,
and will free the memory in either wm8962_i2c_probe error path
or wm8962_i2c_remove.
Thus we should not call kfree(wm8962) in wm8962_probe, otherwise
we have double free of wm8962.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We allocated memory for wm8731 in wm8731_spi_probe / wm8731_i2c_probe,
and will free the memory in either wm8731_spi_probe / wm8731_i2c_probe
error path or wm8731_spi_remove / wm8731_i2c_remove.
Thus we should not call kfree(wm8731) in wm8731_probe, otherwise
we have double free of wm8731.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We allocated memory for aic3x in aic3x_i2c_probe,
and will free the memory in either aic3x_i2c_probe error path or
aic3x_i2c_remove.
Thus we should not call kfree(aic3x) in aic3x_probe, otherwise
we have double free of aic3x.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We allocated memory for ad193x in ad193x_spi_probe,
and will free the memory in either ad193x_spi_probe error path or
ad193x_spi_remove.
Thus we should not call kfree(ad193x) in ad193x_probe, otherwise
we have double free of ad193x.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We allocated memory for ad1836 in ad1836_spi_probe,
and will free the memory in either ad1836_spi_probe error path or
ad1836_spi_remove.
Thus we should not call kfree(ad1836) in ad1836_probe, otherwise
we have double free of ad1836.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Joe Perches <joe@perches.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There is a need to prefix codec kcontrol, widget and internal route names in
an ASoC machine that has multiple codecs with conflicting names. The name
collision would occur when codec drivers try to registering kcontrols with
the same name or when building audio paths.
This patch introduces optional prefix_map into struct snd_soc_card. With it
machine drivers can specify a unique name prefix to each codec that have
conflicting names with anothers. Prefix to codec is matched with codec
name.
Following example illustrates a machine that has two same codec instances.
Name collision from kcontrol registration is avoided by specifying a name
prefix "foo" for the second codec. As the codec widget names are prefixed
then second audio map for that codec shows a prefixed widget name.
static const struct snd_soc_dapm_route map0[] = {
{"Spk", NULL, "MONO"},
};
static const struct snd_soc_dapm_route map1[] = {
{"Vibra", NULL, "foo MONO"},
};
static struct snd_soc_prefix_map codec_prefix[] = {
{
.dev_name = "codec.2",
.name_prefix = "foo",
},
};
static struct snd_soc_card card = {
...
.prefix_map = codec_prefix,
.num_prefixes = ARRAY_SIZE(codec_prefix),
};
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Ensure that we keep all widget powerups in DAPM sequence by making
the CODEC the last thing we compare on rather than the first thing.
Also fix the fact that we're currently comparing the widget pointers
rather than the CODEC pointers when we do the substraction so we
won't get stable results.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
OpenRD Ultimate & Client are similar machines so enable OpenRD client sound
support on Ultimate too
Tested-by: Robas Teodor <teodor.robas@gmail.com>
Signed-off-by: Arnaud Patard <arnaud.patard@rtp-net.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Unfortunatelly, I misunderstood datasheet, and on s3c244x-iis
when MPLLin source for master clock is selected, prescaler has
no effect. Remove dividor calculation for 44100 rate; remove 88200
rate at all, rx1950 can't do it.
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If the device associated with a GPIO jack is wakeup capable then disable
suspend while we're debouncing the jack so that we skip suspends that race
with the jack.
Note that currently the GPIO based jack has a CODEC associated with it
which we're using right now. These jacks should be reparented against the
card itself and this code adjusted.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
The value makes no odds and it makes life easier with caches.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
This patch adds support for rbtree compression when storing the
register cache. It does this by not adding any uninitialized registers
(those whose value is 0). If any of those registers is written
with a nonzero value they get added into the rbtree.
Consider a sample device with a large sparse register map. The
register indices are between [0, 0x31ff]. An array of 12800 registers
is thus created each of which is 2 bytes. This results in a 25kB
region. This array normally lives outside soc-core, normally in the
driver itself. The original soc-core code would kmemdup this region
resulting in 50kB total memory. When using the rbtree compression
technique and __devinitconst on the original array the figures are
as follows. For this typical device, you might have 100 initialized
registers, that is registers that are nonzero by default. We build
an rbtree with 100 nodes, each of which is 24 bytes. This results
in ~2kB of memory. Assuming that the target arch can freeup the
memory used by the initial __devinitconst array, we end up using
about ~2kB bytes of actual memory. The memory footprint will increase
as uninitialized registers get written and thus new nodes created in
the rbtree. In practice, most of those registers are never changed.
If the target arch can't freeup the __devinitconst array, we end up
using a total of ~27kB. The difference between the rbtree and the LZO
caching techniques, is that if using the LZO technique the size of
the cache will increase slower as more uninitialized registers get
changed.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds support for LZO compression when storing the register
cache. The initial register defaults cache is marked as __devinitconst
and the only change required for a driver to use LZO compression is
to set the compress_type member in codec->driver to SND_SOC_LZO_COMPRESSION.
For a typical device whose register map would normally occupy 25kB or 50kB
by using the LZO compression technique, one can get down to ~5-7kB. There
might be a performance penalty associated with each individual read/write
due to decompressing/compressing the underlying cache, however that should not
be noticeable. These memory benefits depend on whether the target architecture
can get rid of the memory occupied by the original register defaults cache
which is marked as __devinitconst. Nevertheless there will be some memory
gain even if the target architecture can't get rid of the original register
map, this should be around ~30-32kB instead of 50kB.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch introduces the new caching API and migrates the
old caching interface into the new one. The flat register caching
technique does not use compression at all and it is equivalent to
the old caching technique. One can still access codec->reg_cache
directly but this is not advised as that will not be portable
across different caching strategies.
None of the existing drivers need to be changed to adapt to this
caching technique. There should be no noticeable overhead associated
with using the new caching API.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Trace events for DAPM allow us to monitor the performance and behaviour
of DAPM with logging which can be built into the kernel permanantly, is
more suited to automated analysis and display and less likely to suffer
interference from other logging activity.
Currently trace events are generated for:
- Start and stop of DAPM processing
- Start and stop of bias level changes
- Power decisions for widgets
- Widget event execution start and stop
giving some view as to what is happening and where latencies occur.
Actual changes in widget power can be seen via the register write trace in
soc-core.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
The trace subsystem provides a convenient way of instrumenting the kernel
which can be left on all the time with extremely low impact on the system
unlike prints to the kernel log which can be very spammy. Begin adding
support for instrumenting ASoC via this interface by adding trace for the
register access primitives.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Make the DAPM sequence execution look a bit nicer by factoring out the
code to invoke an event into a single function since it's all the same
pretty much.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Add Kconfig dependency on AT91_PROGRAMMABLE_CLOCKS for the Atmel SoC
audio SAM9G20-EK and PlayPaq boards. Fixes link errors on missing
clk_set_parent and clk_set_rate when building without
AT91_PROGRAMMABLE_CLOCKS.
Signed-off-by: Ryan Mallon <ryan@bluewatersys.com>
Acked-by: Geoffrey Wossum <gwossum@acm.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The Atmel SSC can divide by even numbers, not only powers of two.
Signed-off-by: Peter Rosin <peda@axentia.se>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In each function, the value apcm is stored in the private_data field of
runtime. At the same time the function ct_atc_pcm_free_substream is stored
in the private_free field of the same structure. ct_atc_pcm_free_substream
dereferences and ultimately frees the value in the private_data field. But
each function can exit in an error case with apcm having been freed, in
which case a subsequent call to the private_free function would perform a
dereference after free. On the other hand, if the private_free field is
not initialized, it is NULL, and not invoked (see snd_pcm_detach_substream
in sound/core/pcm.c). To avoid the introduction of a dangling pointer, the
initializations of the private_data and private_free fields are moved to
the end of the function, past any possible free of apcm. This is safe
because the previous calls to snd_pcm_hw_constraint_integer and
snd_pcm_hw_constraint_minmax, which take runtime as an argument, do not
refer to either of these fields.
In each function, there is one error case where apcm needs to be freed, and
a call to kfree is added.
The sematic match that finds this problem is as follows:
(http://coccinelle.lip6.fr/)
// <smpl>
@@
expression e,e1,e2,e3;
identifier f,free1,free2;
expression a;
@@
*e->f = a
... when != e->f = e1
when any
if (...) {
... when != free1(...,e,...)
when != e->f = e2
* kfree(a)
... when != free2(...,e,...)
when != e->f = e3
}
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If the platform already provides a definition for these accessors
do not redefine them. The warning was caught on MIPS.
Signed-off-by: Florian Fainelli <florian@openwrt.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: http://launchpad.net/bugs/673075
According to the datasheet of 92HD87B, there is a digital mic
at nid 0x11, so enable it in order to be able to use the mic.
Cc: stable@kernel.org
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The [vk][cmz]alloc(_node) family of functions return void pointers which
it's completely unnecessary/pointless to cast to other pointer types since
that happens implicitly.
This patch removes such casts from sound/oss/
Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Ensure that whatever ran before us leaves the WM835x with a sane default
audio interface configuration as we do not override the companding,
loopback or tristate settings and do not reset the chip at startup (as it
is a PMIC).
Reported-by: Keiji Mitsuhisa <Keiji.Mitsuhisa@wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
The WM8350 driver was using some custom constants to interpret the direction
of the MCLK signal which had the opposite values to those used as standard
by the ASoC core, causing confusion in machine drivers such as the 1133-EV1
board.
Reported-by: Tommy Zhu <Tommy.Zhu@wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Prints from pop_dbg are enabled when dapm_pop_time != 0. Convert it to
use dev_info so that parent device of DAPM context is printed.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Switch printk and pr_ prints to dev_ variants. It is helpful to see
parent device of DAPM context especially when there are multiple DAPM
contexts (codecs currently).
This is mostly simple conversion. Exceptions are in snd_soc_dapm_set_pin
that prints also pin state, uniform "dapm: unknown pin" error prints from
snd_soc_dapm_set_pin, snd_soc_dapm_force_enable_pin and
snd_soc_dapm_ignore_suspend, and pop_dbg which is converted by an another
patch.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There are no known problems with current power-up sequence which first sets
the /shutdown pin high and then enables the supply. However, swap the order
so that the device is kept in shutdown/reset mode during the supply voltage
transition since slowly rising voltages can usually cause problems if the
device is not kept in reset.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Cc: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Add soc_init_card_debugfs and soc_cleanup_card_debugfs functions to fix below error.
CC sound/soc/soc-core.o
sound/soc/soc-core.c: In function 'soc_probe':
sound/soc/soc-core.c:1689: error: implicit declaration of function 'soc_init_card_debugfs'
sound/soc/soc-core.c: In function 'soc_remove':
sound/soc/soc-core.c:1718: error: implicit declaration of function 'soc_cleanup_card_debugfs'
make[2]: *** [sound/soc/soc-core.o] Error 1
make[1]: *** [sound/soc] Error 2
make: *** [sound] Error 2
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There is no need to use '&' in this case. Either way, if a is an array
of some type, then a == &a == &a[0].
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Instead of dereferencing a NULL function pointer and falling apart
use BUG_ON() for any unimplemented hw_read() calls.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When make mini2440_defconfig compilation end with undefined
references to DMA functions. There was missing selection
for S3C2410_DMA when compile ASoC audio for S3C24xx CPU.
Tested on mini2440 board.
Signed-off-by: Marek Belisko <marek.belisko@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM8770 is a high performance, multi-channel audio
codec. The WM8770 is ideal for surround sound processing
applications for home hi-fi, automotive and other audio
visual equipment.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Based on discussion the dapm_pop_time in debugsfs should be per card rather
than per device. Single pop time value for entire card is cleaner when the
DAPM sequencing is extended to cross-device paths.
debugfs/asoc/{card->name}/{codec dir}/dapm_pop_time
->
debugfs/asoc/{card->name}/dapm_pop_time
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Make use of sound card debugfs directory and move codec directories under
the parent card debugfs directory.
debugfs/asoc/{codec dir} -> debugfs/asoc/{card->name}/{codec dir}.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There will be need to have sound card specific debugfs entries. This patch
introduces a new debugfs/asoc/{card->name}/ directory but does not add yet
any entries there.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Decoupling Dynamic Audio Power Management (DAPM) from codec devices is
required when developing ASoC further. Such as for other ASoC components to
have DAPM widgets or when extending DAPM to handle cross-device paths.
This patch decouples DAPM related variables from struct snd_soc_codec and
moves them to new struct snd_soc_dapm_context that is used to encapsulate
DAPM context of a device. ASoC core and API of DAPM functions are modified
to use DAPM context instead of codec.
This patch does not change current functionality and a large part of changes
come because of structure and internal API changes.
Core implementation is from Liam Girdwood <lrg@slimlogic.co.uk> with some
minor core changes, codecs and machine driver conversions from
Jarkko Nikula <jhnikula@gmail.com>.
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Cc: Nicolas Ferre <nicolas.ferre@atmel.com>
Cc: Manuel Lauss <manuel.lauss@googlemail.com>
Cc: Mike Frysinger <vapier.adi@gmail.com>
Cc: Cliff Cai <cliff.cai@analog.com>
Cc: Kevin Hilman <khilman@deeprootsystems.com>
Cc: Ryan Mallon <ryan@bluewatersys.com>
Cc: Timur Tabi <timur@freescale.com>
Cc: Sascha Hauer <s.hauer@pengutronix.de>
Cc: Lars-Peter Clausen <lars@metafoo.de>
Cc: Arnaud Patard (Rtp) <arnaud.patard@rtp-net.org>
Cc: Wan ZongShun <mcuos.com@gmail.com>
Cc: Eric Miao <eric.y.miao@gmail.com>
Cc: Jassi Brar <jassi.brar@samsung.com>
Cc: Daniel Gloeckner <dg@emlix.com>
Cc: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The member reg_cache is not used at all and therefore it should be
removed. This member was usually needed for older versions of ASoC
that did not handle caching automatically and had to be done in the
driver itself.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When doing anything with the system, especially DAPM, we need to hold the
CODEC mutex.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Whether we can do mono or not depends on the codec. No need
to limit this in the ssi driver.
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We have different codecs on the pcm038 (ac97 wm9712 and mc13783).
To make alsactl restore work correctly these should have different
names.
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We have two different transfer methods on i.MX: FIQ and DMA. Since
the merge of the ASoC multicomponent support the DMA device is lost.
Add it again. Also, imx_ssi_dai_probe has to be called for !AC97
aswell.
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Ensure that the system does not suspend while we process a WM8962 jack
event by using pm_wakeup_event() to block the suspend while we're waiting
for the jack to settle. Use a slightly longer timeout than the jack waits
to allow for other stuff to take over and delays in scheduling.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
We now have trace in the ASoC core so we don't need to our own trace in
the driver.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
No need to print the register-value pair again, as we've already hooked
snd_soc_write() for that matter.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Facilitating adding trace type stuff. For a first pass add some dev_dbg()
statements into them.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Allow the standard soc-jack GPIO based jack handling to handle the use of
GPIOs which may sleep (such as those on GPIO expanders) by converting the
code to use request_any_context_irq().
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
strict_strtoul() has been made __must_check so do so.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
