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BugLink: http://bugs.launchpad.net/bugs/705323
Thinkpad Edge 14 has one more SSID that suffers from disabled auto-mute.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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sound/pci/hda/patch_realtek.c: In function ‘alc_apply_fixup’:
sound/pci/hda/patch_realtek.c:1724:14: warning: unused variable ‘modelname’
snd_printdd() is evaluated only when CONFIG_SND_DEBUG_VERBOSE=y.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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It seems working well with model=hp-bpc.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This reverts commit 03b7a1ab557efe34e8f79b78660e514bd7374248.
This commit was mistakenly re-introduced. While the change is harmless
(as ALC887 uses patch_alc888() now), we should get rid of any wrong code.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: hda - Fix initialization for HP 2011 notebooks
ALSA: hda - Add support for VMware controller
ALSA: hda - consitify string arrays
ALSA: hda - Add add multi-streaming playback for AD1988
ASoC: EP93xx: fixed LRCLK rate and DMA oper. in I2S code
ASoC: WM8990: msleep() takes milliseconds not jiffies
ALSA : au88x0 - Limit number of channels to fix Oops via OSS emu
ALSA: constify functions in ac97
ASoC: WL1273 FM radio: Fix breakage with MFD API changes
ALSA: hda - More coverage for odd-number channels elimination for HDMI
ALSA: hda - Store PCM parameters properly in HDMI open callback
ALSA: hda - Rearrange fixup struct in patch_realtek.c
ALSA: oxygen: Xonar DG: fix CS4245 register writes
ALSA: hda - Suppress the odd number of channels for HDMI
ALSA: hda - Add fixup-call in init callback
ALSA: hda - Reorganize fixup structure for Realtek
ALSA: hda - Apply Sony VAIO hweq fixup only once
ALSA: hda - Apply mario fixup only once
ALSA: hda - Remove unused fixup entry for ALC262
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Changelog:
1. I2S module of EP93xx should be feed by 32bit DMA transfers. This is
hardware limitation and that's the way original Cirrus's driver worked.
This will fix distorted sound playback and make capture actually work in
present ep93xx drivers.
I've found, that author of code, on which modern ep93xx-i2s.c and
ep93xx-pcm.c are based, had faced this problem also in 2007:
http://blog.gmane.org/gmane.linux.ports.arm.cirrus/month=20070101/page=3
Now SoC code uses his developments, but not overcomes the hardware
issues. Some details from EP93xx users guide:
Both I2S transmitter and receiver have similar 16x32bit FIFO, where they
store 8 samples for both left and right channels. The FIFO is always
32bit wide and should be properly aligned if you use samples of other
width. Transmitter and receiver have configuration registers for
selection of I2S word length (16, 24, 32). They are I2STXWrdLen and
I2SRXWrdLen.
Yes, EP93xx DMA can do byte, word and quad-word transfers. The width for
transfers to and from peripherals is selected by particular module
configuration. Lucky AC97 module has such configuration: AC97RXCRx
registers, bit CM (Compact mode enable) switches between 16 and 32 bit
samples. AC97TXCRx registers have the same bits for transmitters.
ep93xx-ac97.c enables this compact mode and so has all the rights to use
S16_LE format.
No one has found such a configuration in I2S module until now in any
Cirrus manuals. I2S module always feeds it's 32bit wide FIFO with 32bit
samples consecutively for left and right channels. You cannot use 32-bit
DMA transfers to transfer two 16-bit samples.
So we can use two formats for AC97, but should remove all but S32_LE for
I2S. Always using 32 bit chunks is not a problem for I2S, the codec I
use uses less bits too (24), it's permitted by I2S standard.
In proposed patch formats list shortened to just S32_LE, this makes all
the DMA transactions right, while ALSA will do all sample format
translation for us.
2. Incorrect setting of LRCLK (2 times slower) in original ep93xx-i2s.c
masks the first problem.
DMA takes two 16 bit samples instead of one, overall sound speed seems
to be normal, but you get actually 4000 sampling rate instead of
requested 8000 and therefore some noise... This is also the reason why
the capture function not worked at all in this driver...
If we take a look into I2S specification, we will figure that LRCLK MUST
be equal to sample rate, if we are talking about stereo (in mono too,
but it's not our case at all).
In proposed patch SCLK and LRCLK rates are corrected, assuming we always
send 32 bits * 2 channels to codec.
Signed-off-by: Alexander Sverdlin <subaparts@yandex.ru>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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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>
Cc: stable@kernel.org
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These changes are needed to keep up with the changes in the
MFD core and V4L2 parts of the wl1273 FM radio driver.
Use function pointers instead of exported functions for I2C IO.
Also move all preprocessor constants from the wl1273.h to
include/linux/mfd/wl1273-core.h.
Signed-off-by: Matti J. Aaltonen <matti.j.aaltonen@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Fixes for HP 2011 notebooks: enable dock ports and disable BTL
initialization in the driver.
Signed-off-by: Vitaliy Kulikov <Vitaliy.Kulikov@idt.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Add the new PCI ID 0x15ad and device ID 0x1977 for VMware HDAudio
Controller.
[changed to use AZX_DRIVER_GENERIC by tiwai]
Signed-off-by: Bankim Bhavsar <bbhavsar@vmware.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Attached a patch which add a new model to support multi-streaming
playback for ad1988.
playback another stereo stream through the front panel headphone on
device 2 while playback through the speakers connected to rear panel
on device 0 at the same time.
Tested with ad1988a rev2 codec on asus P5B-V motherboard.
Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The commit ad09fc9d2156f3d37537b34418a6b79309013d33 didn't cover the
case for Intel and Nvidia HDMIs, where hdmi_pcm_open() is called.
Put the hw_constraint there, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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In hdmi_pcm_open(), the evaluated PCM hw parameters are stored in
hinfo, but these aren't properly set back to the current runtime
record since these have been set beforehand in azx_pcm_open().
This patch fixes the behavior.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Signed-off-by: Takashi Iwai <tiwai@suse.de>
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It looks like that HDMI codecs don't support the odd number of channels
although HD-audio spec doesn't have the restriction. Add the
hw_constraint to limit to only the even number of channels.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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In some cases, the fix-up is required in the init callback to be called
both at the first initialization and at the resume. The new action type
ALC_FIXUP_ACT_INIT is used for this case.
So far, only ALC275_FIXUP_SONY_HWEQ uses this.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Instead of keeping various data types in a single record, put the
type field and keep a single value in each entry, but allows chaining
multiple fixup entries. This allows more flexible data management
(see ALC275_FIXUP_SONY_HWEQ for example).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This should be applied also only once as a part of the initialization.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The amp-override is necessary only once at initialization time.
Also fixed a coding style issue.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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... and a minor cleanup.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Fix playback/capture channels patch to change supported playback
channels of au8830 to 1,2,4 and capture channels to 1,2.
This prevent oops when oss emulation use SNDCTL_DSP_CHANNELS to
set 3 Channels
Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Signed-off-by: Hanno Boeck <hanno@hboeck.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Accidentally exchanging register addresses and register values leads to
many strange errors ...
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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git://git.kernel.org/pub/scm/linux/kernel/git/jbarnes/pci-2.6
* 'linux-next' of git://git.kernel.org/pub/scm/linux/kernel/git/jbarnes/pci-2.6:
PCI/PM: Report wakeup events before resuming devices
PCI/PM: Use pm_wakeup_event() directly for reporting wakeup events
PCI: sysfs: Update ROM to include default owner write access
x86/PCI: make Broadcom CNB20LE driver EMBEDDED and EXPERIMENTAL
x86/PCI: don't use native Broadcom CNB20LE driver when ACPI is available
PCI/ACPI: Request _OSC control once for each root bridge (v3)
PCI: enable pci=bfsort by default on future Dell systems
PCI/PCIe: Clear Root PME Status bits early during system resume
PCI: pci-stub: ignore zero-length id parameters
x86/PCI: irq and pci_ids patch for Intel Patsburg
PCI: Skip id checking if no id is passed
PCI: fix __pci_device_probe kernel-doc warning
PCI: make pci_restore_state return void
PCI: Disable ASPM if BIOS asks us to
PCI: Add mask bit definition for MSI-X table
PCI: MSI: Move MSI-X entry definition to pci_regs.h
Fix up trivial conflicts in drivers/net/{skge.c,sky2.c} that had in the
meantime been converted to not use legacy PCI power management, and thus
no longer use pci_restore_state() at all (and that caused trivial
conflicts with the "make pci_restore_state return void" patch)
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pci_restore_state only ever returns 0, thus there is no benefit in
having it return any value. Also, a large majority of the callers do
not check the return code of pci_restore_state. Make the
pci_restore_state a void return and avoid the overhead.
Acked-by: Mauro Carvalho Chehab <mchehab@redhat.com>
Signed-off-by: Jon Mason <jon.mason@exar.com>
Signed-off-by: Jesse Barnes <jbarnes@virtuousgeek.org>
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git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (348 commits)
ALSA: hda - Fix NULL-derefence with a single mic in STAC auto-mic detection
ALSA: hda - Add missing NID 0x19 fixup for Sony VAIO
ALSA: hda - Fix ALC275 enable hardware EQ for SONY VAIO
ALSA: oxygen: fix Xonar DG input
ALSA: hda - Fix EAPD on Lenovo NB ALC269 to low
ALSA: hda - Fix missing EAPD for Acer 4930G
ALSA: hda: Disable 4/6 channels on some NVIDIA GPUs.
ALSA: hda - Add static_hdmi_pcm option to HDMI codec parser
ALSA: hda - Don't refer ELD when unplugged
ASoC: tpa6130a2: Fix compiler warning
ASoC: tlv320dac33: Add DAPM selection for LOM invert
ASoC: DMIC codec: Adding a generic DMIC codec
ALSA: snd-usb-us122l: Fix missing NULL checks
ALSA: snd-usb-us122l: Fix MIDI output
ASoC: soc-cache: Fix invalid memory access during snd_soc_lzo_cache_sync()
ASoC: Fix section mismatch in wm8995.c
ALSA: oxygen: add S/PDIF source selection for Claro cards
ALSA: oxygen: fix CD/MIDI for X-Meridian (2G)
ASoC: fix migor audio build
ALSA: include delay.h for msleep in Xonar DG support
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git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into topic/asoc
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The size of the lzo syncing bitmap was incorrectly set to the size
of the cache times the word size, however, the correct size is the
size of the cache.
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>
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Commit 6d803ba736abb5e122dede70a4720e4843dd6df4 "ARM: 6483/1: arm & sh:
factorised duplicated clkdev.c" broke compilation of migor audio. Use the
correct header to fix the problem.
Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Update the CS4270 driver to use ASoC's internal codec register cache feature.
This change allows ASoC to perform the low-level I2C operations necessary to
read the register cache. Support is also added for initializing the register
cache with an array of known power-on default values.
The CS4270 driver was handling the register cache itself, but somwhere along
the conversion to multi-compaonent, this feature broke.
Signed-off-by: Timur Tabi <timur@freescale.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Enable Mic Jack during glue driver init, otherwise capture will not work.
Signed-off-by: Vasily Khoruzhick <anarsoul@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>
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git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into topic/asoc
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sound/soc/codecs/tpa6130a2.c: In function 'tpa6130a2_add_controls':
sound/soc/codecs/tpa6130a2.c:342: warning: unused variable 'dapm'
Introduced by commit 39646871a47fd8808c08de0ce7d7ca8393af2805 ("ASoC:
tpa6130a2: Replace DAPM code with direct interface").
The DAPM code has been removed from the driver, but the
dapm struct remained.
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>
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The L/R LOM line can be invertined side of the
corresponding DAC, or inverted from the corresponding
LOP.
Add control for user space to select the source of the
LOM inversion.
When only the analog bypass is enabled, and the LOM
is inverted from DAC output, we need to power the
corresponding DAC.
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>
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This codec is to be used by the DMIC driver to
control the DMIC codec. This driver will be used on future
implementations of the DMIC driver to support codec specific
features.
At this time, the codec driver just registers the codec DAI.
Signed-off-by: David Lambert <dlambert@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
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Add support for 24 bit audio (with S32_LE msbits 24).
The reason to limit the msbits to 24, is that the FIFO
can be configured for 16 or 24 bit layout.
It is unknown how the codec would downsample from 32 to
24 bit, if the interface is configured to receive 32
bit data.
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>
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Change the structure of FIFO handling in order to
pave the way for adding 32/24 bit audio support.
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>
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The manual FIFO configuration was the first version to enable
the use of the FIFO in the codec.
It had served it's purpose as debugging aid, but the automatic
FIFO configuration is much safer to use.
The removal of the manual controls, and configuration makes
it easier to add new features for the codec later, since
the manual mode neded different ways to calculate, and
protect against misconfiguration.
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>
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Make the phoenix HS and HF drivers use the new DAPM driver
widget in order to guarantee power ON/OFF order sequence.
Signed-off-by: Margarita Olaya Cabrera <magi.olaya@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
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This patch adds McBSP support for the OMAP4 CPU
Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria@ti.com>
Signed-off-by: Margarita Olaya Cabrera <magi.olaya@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
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__devinitconst can't be used for data referred in driver struct.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The soc-core takes the platform and codec driver reference during probe. Few of
these references are not released during remove. This cause the platform and
codec driver module unload to fail.
This patch fixes by the taking only one reference to platform and codec module
during probe and releases them correctly during remove. This allows load/unload
properly
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Harsha Priya <priya.harsha@intel.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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After multi-component conversion these machine drivers don't actually need
anything from sound/soc/codecs/tlv320aic3x.h so don't include 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>
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The rest of ASoC is using SND_SOC_ as the prefix for all the Kconfig
symbols so do so for the new Samsung drivers too, rather than using
ASOC_ as they currently are.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Seungwhan Youn <sw.youn@samsung.com>
Acked-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
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Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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