ALSA mixers cannot classify this "Class-D Amplifier Gain" speaker output
gain control as a playback control. Fix this by changing the name as
"Class-D Playback Volume".
Signed-off-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
PIO handler is not good performance, but works on all platform.
So, switch to PIO handler if DMA handler was invalid case.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch used dmaengine helper functions instead of using hand setting.
And reduced local variables
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add platform-driver handling all DMA-activities.
Signed-off-by: Ola Lilja <ola.o.lilja@stericsson.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Adds a supply-widget variant for connection to the clock-framework.
This widget-type corresponds to the variant for regulators.
Signed-off-by: Ola Lilja <ola.o.lilja@stericsson.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In kernel 3.6, Seaboard will only be supported when booting using device
tree; the board files are being removed. Hence, remove the non-DT support
for Seaboard and derivatives Kaen and Aebl from the audio driver.
Harmony is the only remaining board supported by this driver when not
using DT. This support is currently scheduled for removal in 3.7.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
By the time any widget callbacks could be called, if the GPIO ID they
will manipulate is valid, it must have already been requested, or the
card would have failed to probe or initialize. So, testing for GPIO
validity is equivalent to testing whether the GPIO was successfully
requested at this point in the code. Making this change will allow later
patches to remove the gpio_requested variable.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The headphone jack GPIOs are added/initialized in the DAI link's init()
method, and hence in theory may not always have been added before remove()
is called in some unusual cases. In order to prevent calling
snd_soc_jack_free_gpios() if snd_soc_jack_add_gpios() had not been, the
code kept track of the initialization state to avoid the free call when
necessary.
However, it appears that snd_soc_jack_free_gpios() is robust in the face
of being called without snd_soc_jack_add_gpios() first succeeding, so
there is little point manually tracking this information. Hence, remove
the tracking code. All other machine drivers already operate this way.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Now that deferred probe exists, we can parse device tree and request
GPIOs from probe(), rather than deferring this to the DAI link's init().
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The headphone jack GPIOs are added/initialized in the DAI link's init()
method, and hence in theory may not always have been added before remove()
is called in some unusual cases. In order to prevent calling
snd_soc_jack_free_gpios() if snd_soc_jack_add_gpios() had not been, the
code kept track of the initialization state to avoid the free call when
necessary.
However, it appears that snd_soc_jack_free_gpios() is robust in the face
of being called without snd_soc_jack_add_gpios() first succeeding, so
there is little point manually tracking this information. Hence, remove
the tracking code. Almost all other machine drivers already operate this
way.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
By using this function, the driver no longer needs to explicitly free
the GPIOs. Hence, we can also remove the flags we use to track whether
we allocated these GPIOs.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Now that deferred probe exists, we can parse device tree and request
GPIOs from probe(), rather than deferring this to the DAI link's init().
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This allows the GPIOs to be available as soon as the I2C device has
probed, which in turn enables machine drivers to request the GPIOs in
their probe(), rather than deferring this to their ASoC machine init
function, i.e. after the whole sound card has been constructed, and
hence the WM8903 codec is available.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Small driver-specific updates.
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Merge tag 'asoc-3.5' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: A few bug fixes for the merge window
Small driver-specific updates.
On PCI RME MADI cards, the PLL register does not contain the proper
value, so the calculated system_sample_rate is wrong. In this case, we
simply return the cached rate from struct hdspm.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The array of sample rates is reallocated every time when opening
the PCM device, but was freed only once when unplugging the device.
Reported-by: "Alexander E. Patrakov" <patrakov@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
FSI DMAEngine has to be stopped certainly at the start/stop time.
Without this patch, it will include noise on playback.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
FSI driver is using dma_sync_single_xxx(),
but the dma area was not correct.
This patch fix it up.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
DMA stream handler didn't care about master clock.
This patch fixes it up.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The new clock subsystem was merged in linux-3.4 without any users, this
now moves the first three platforms over to it: imx, mxs and spear.
The series also contains the changes for the clock subsystem itself,
since Mike preferred to have it together with the platforms that require
these changes, in order to avoid interdependencies and conflicts.
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Merge tag 'clock' of git://git.kernel.org/pub/scm/linux/kernel/git/arm/arm-soc
Pull arm-soc clock driver changes from Olof Johansson:
"The new clock subsystem was merged in linux-3.4 without any users,
this now moves the first three platforms over to it: imx, mxs and
spear.
The series also contains the changes for the clock subsystem itself,
since Mike preferred to have it together with the platforms that
require these changes, in order to avoid interdependencies and
conflicts."
Fix up trivial conflicts in arch/arm/mach-kirkwood/common.c (code
removed in one branch, added OF support in another) and
drivers/dma/imx-sdma.c (independent changes next to each other).
* tag 'clock' of git://git.kernel.org/pub/scm/linux/kernel/git/arm/arm-soc: (97 commits)
clk: Fix CLK_SET_RATE_GATE flag validation in clk_set_rate().
clk: Provide dummy clk_unregister()
SPEAr: Update defconfigs
SPEAr: Add SMI NOR partition info in dts files
SPEAr: Switch to common clock framework
SPEAr: Call clk_prepare() before calling clk_enable
SPEAr: clk: Add General Purpose Timer Synthesizer clock
SPEAr: clk: Add Fractional Synthesizer clock
SPEAr: clk: Add Auxiliary Synthesizer clock
SPEAr: clk: Add VCO-PLL Synthesizer clock
SPEAr: Add DT bindings for SPEAr's timer
ARM i.MX: remove now unused clock files
ARM: i.MX6: implement clocks using common clock framework
ARM i.MX35: implement clocks using common clock framework
ARM i.MX5: implement clocks using common clock framework
ARM: Kirkwood: Replace clock gating
ARM: Orion: Audio: Add clk/clkdev support
ARM: Orion: PCIE: Add support for clk
ARM: Orion: XOR: Add support for clk
ARM: Orion: CESA: Add support for clk
...
This is the second updates for 3.5-rc1. It's mainly for OMAP4 HDMI
updates and the device tree updates for OMAP, in addition to a couple
of PCM accuray improvement and Realtek ALC269VD codec support.
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Merge tag 'sound-3.5' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound update from Takashi Iwai:
"This is the second updates for 3.5-rc1. It's mainly for OMAP4 HDMI
updates and the device tree updates for OMAP, in addition to a couple
of PCM accuray improvement and Realtek ALC269VD codec support."
* tag 'sound-3.5' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (21 commits)
ALSA: hda/realtek - Add new codec support for ALC269VD
ALSA: core: group read of pointer, tstamp and jiffies
ASoC: OMAP: HDMI: Rename sound card source file
ASoC: OMAP: HDMI: Make sound card naming more generic
ASoC: OMAP: HDMI: Make build config options more generic
ASoC: OMAP: HDMI: Expand capabilities of the HDMI DAI
ASoC: OMAP: HDMI: Improve how the display state is verified
ASoC: OMAP: HDMI: Expand configuration of hw_params
ASoC: OMAP: HDMI: Use the DSS audio interface
ASoC: OMAP: HDMI: Create a structure for private data of the CPU DAI
ASoC: OMAP: HDMI: Change error values in HDMI CPU DAI
ASoC: OMAP: HDMI: Update the platform device names
ASoC: omap-abe-twl6040: Introduce driver data for runtime parameters
ASoC: omap-abe-twl6040: Move Digital Mic widget into dapm table
ASoC: omap-abe-twl6040: Keep only one snd_soc_dai_link structure
ASoC: omap-dmic: Add device tree bindings
ASoC: omap-mcpdm: Add device tree bindings
ASoC: omap-mcbsp: buffer size constraint only applies to playback stream
ASoC: omap-mcbsp: Use the common interrupt line if supported by the SoC
ASoC: omap-mcbsp: Remove unused FRAME dma_op_mode
...
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Merge tag 'hda-switcheroo' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull VGA-switcheroo audio client support for HD-audio from Takashi Iwai.
This depended on the recent drm pull.
* tag 'hda-switcheroo' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda - unlock on error in azx_interrupt()
ALSA: hda - Support VGA-switcheroo
ALSA: hda - Export snd_hda_lock_devices()
ALSA: hda - Check the dead HDMI audio controller by vga-switcheroo
- Fix mismatch between DMA mapping direction (was wrong) and DMA synchronization
direction (was correct) of isochronous reception buffers of userspace drivers
if vma-mapped for R/W access. For example, libdc1394 was affected.
- more consistent retry stategy in device discovery/ rediscovery, and improved
failure diagnostics
- various small cleanups, e.g. use SCSI layer's DMA mapping API in firewire-sbp2
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Merge tag 'firewire-updates' of git://git.kernel.org/pub/scm/linux/kernel/git/ieee1394/linux1394
Pull IEEE 1394 (FireWire) subsystem updates from Stefan Richter:
- Fix mismatch between DMA mapping direction (was wrong) and DMA
synchronization direction (was correct) of isochronous reception
buffers of userspace drivers if vma-mapped for R/W access. For
example, libdc1394 was affected.
- more consistent retry stategy in device discovery/ rediscovery, and
improved failure diagnostics
- various small cleanups, e.g. use SCSI layer's DMA mapping API in
firewire-sbp2
* tag 'firewire-updates' of git://git.kernel.org/pub/scm/linux/kernel/git/ieee1394/linux1394:
firewire: sbp2: document the absence of alignment requirements
firewire: sbp2: remove superfluous blk_queue_max_segment_size() call
firewire: sbp2: use scsi_dma_(un)map
firewire: sbp2: give correct DMA device to scsi framework
firewire: core: fw_device_refresh(): clean up error handling
firewire: core: log config rom reading errors
firewire: core: log error in case of failed bus manager lock
firewire: move rcode_string() to core
firewire: core: improve reread_config_rom() interface
firewire: core: wait for inaccessible devices after bus reset
firewire: ohci: omit spinlock IRQ flags where possible
firewire: ohci: correct signedness of a local variable
firewire: core: fix DMA mapping direction
firewire: use module_pci_driver
Pull media updates from Mauro Carvalho Chehab:
- some V4L2 API updates needed by embedded devices
- DVB API extensions for ATSC-MH delivery system, used in US for mobile
TV
- new tuners for fc0011/0012/0013 and tua9001
- a new dvb driver for af9033/9035
- a new ATSC-MH frontend (lg2160)
- new remote controller keymaps
- Removal of a few legacy webcam driver that got replaced by gspca on
several kernel versions ago
- a new driver for Exynos 4/5 webcams(s5pp fimc-lite)
- a new webcam sensor driver (smiapp)
- a new video input driver for embedded (sta2x1xx)
- several improvements, fixes, cleanups, etc inside the drivers.
Manually fix up conflicts due to err() -> dev_err() conversion in
drivers/staging/media/easycap/easycap_main.c
* 'v4l_for_linus' of git://git.kernel.org/pub/scm/linux/kernel/git/mchehab/linux-media: (484 commits)
[media] saa7134-cards: Remove a PCI entry added by mistake
[media] radio-sf16fmi: add support for SF16-FMD
[media] rc-loopback: remove duplicate line
[media] patch for Asus My Cinema PS3-100 (1043:48cd)
[media] au0828: Move the Kconfig knob under V4L_USB_DRIVERS
[media] em28xx: simple comment fix
[media] [resend] radio-sf16fmr2: add PnP support for SF16-FMD2
[media] smiapp: Use v4l2_ctrl_new_int_menu() instead of v4l2_ctrl_new_custom()
[media] smiapp: Add support for 8-bit uncompressed formats
[media] smiapp: Allow generic quirk registers
[media] smiapp: Use non-binning limits if the binning limit is zero
[media] smiapp: Initialise rval in smiapp_read_nvm()
[media] smiapp: Round minimum pre_pll up rather than down in ip_clk_freq check
[media] smiapp: Use 8-bit reads only before identifying the sensor
[media] smiapp: Quirk for sensors that only do 8-bit reads
[media] smiapp: Pass struct sensor to register writing commands instead of i2c_client
[media] smiapp: Allow using external clock from the clock framework
[media] zl10353: change .read_snr() to report SNR as a 0.1 dB
[media] media: add support to gspca/pac7302.c for 093a:2627 (Genius FaceCam 300)
[media] m88rs2000 - only flip bit 2 on reg 0x70 on 16th try
...
This is the first big chunk for 3.5 merges of sound stuff.
There are a few big changes in different areas. First off, the
streaming logic of USB-audio endpoints has been largely rewritten
for the better support of "implicit feedback". If anything about USB
got broken, this change has to be checked.
For HD-audio, the resume procedure was changed; instead of delaying
the resume of the hardware until the first use, now waking up immediately
at resume. This is for buggy BIOS.
For ASoC, dynamic PCM support and the improved support for digital links
between off-SoC devices are major framework changes.
Some highlights are below:
* HD-audio
- Avoid the accesses of invalid pin-control bits that may stall the codec
- V-ref setup cleanups
- Fix the races in power-saving code
- Fix the races in codec cache hashes and connection lists
- Split some common codes for BIOS auto-parser to hda_auto_parser.c
- Changed the PM resume code to wake up immediately for buggy BIOS
- Creative SoundCore3D support
- Add Conexant CX20751/2/3/4 codec support
* ASoC
- Dynamic PCM support, allowing support for SoCs with internal routing
through components with tight sequencing and formatting constraints
within their internal paths or where there are multiple components
connected with CPU managed DMA controllers inside the SoC.
- Greatly improved support for direct digital links between off-SoC
devices, providing a much simpler way of connecting things like digital
basebands to CODECs.
- Much more fine grained and robust locking, cleaning up some of the
confusion that crept in with multi-component.
- CPU support for nVidia Tegra 30 I2S and audio hub controllers and
ST-Ericsson MSP I2S controolers
- New CODEC drivers for Cirrus CS42L52, LAPIS Semiconductor ML26124, Texas
Instruments LM49453.
- Some regmap changes needed by the Tegra I2S driver.
- mc13783 audio support.
* Misc
- Rewrite with module_pci_driver()
- Xonar DGX support for snd-oxygen
- Improvement of packet handling in snd-firewire driver
- New USB-endpoint streaming logic
- Enhanced M-audio FTU quirks and relevant cleanups
- Increment the support of OSS devices to 256
- snd-aloop accuracy improvement
There are a few more pending changes for 3.5, but they will be
sent slightly later as partly depending on the changes of DRM.
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Merge tag 'sound-3.5' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"This is the first big chunk for 3.5 merges of sound stuff.
There are a few big changes in different areas. First off, the
streaming logic of USB-audio endpoints has been largely rewritten for
the better support of "implicit feedback". If anything about USB got
broken, this change has to be checked.
For HD-audio, the resume procedure was changed; instead of delaying
the resume of the hardware until the first use, now waking up
immediately at resume. This is for buggy BIOS.
For ASoC, dynamic PCM support and the improved support for digital
links between off-SoC devices are major framework changes.
Some highlights are below:
* HD-audio
- Avoid accesses of invalid pin-control bits that may stall the codec
- V-ref setup cleanups
- Fix the races in power-saving code
- Fix the races in codec cache hashes and connection lists
- Split some common codes for BIOS auto-parser to hda_auto_parser.c
- Changed the PM resume code to wake up immediately for buggy BIOS
- Creative SoundCore3D support
- Add Conexant CX20751/2/3/4 codec support
* ASoC
- Dynamic PCM support, allowing support for SoCs with internal
routing through components with tight sequencing and formatting
constraints within their internal paths or where there are multiple
components connected with CPU managed DMA controllers inside the
SoC.
- Greatly improved support for direct digital links between off-SoC
devices, providing a much simpler way of connecting things like
digital basebands to CODECs.
- Much more fine grained and robust locking, cleaning up some of the
confusion that crept in with multi-component.
- CPU support for nVidia Tegra 30 I2S and audio hub controllers and
ST-Ericsson MSP I2S controolers
- New CODEC drivers for Cirrus CS42L52, LAPIS Semiconductor ML26124,
Texas Instruments LM49453.
- Some regmap changes needed by the Tegra I2S driver.
- mc13783 audio support.
* Misc
- Rewrite with module_pci_driver()
- Xonar DGX support for snd-oxygen
- Improvement of packet handling in snd-firewire driver
- New USB-endpoint streaming logic
- Enhanced M-audio FTU quirks and relevant cleanups
- Increment the support of OSS devices to 256
- snd-aloop accuracy improvement
There are a few more pending changes for 3.5, but they will be sent
slightly later as partly depending on the changes of DRM."
Fix up conflicts in regmap (due to duplicate patches, with some further
updates then having already come in from the regmap tree). Also some
fairly trivial context conflicts in the imx and mcx soc drivers.
* tag 'sound-3.5' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (280 commits)
ALSA: snd-usb: fix stream info output in /proc
ALSA: pcm - Add proper state checks to snd_pcm_drain()
ALSA: sh: Fix up namespace collision in sh_dac_audio.
ALSA: hda/realtek - Fix unused variable compile warning
ASoC: sh: fsi: enable chip specific data transfer mode
ASoC: sh: fsi: call fsi_hw_startup/shutdown from fsi_dai_trigger()
ASoC: sh: fsi: use same format for IN/OUT
ASoC: sh: fsi: add fsi_version() and removed meaningless version check
ASoC: sh: fsi: use register field macro name on IN/OUT_DMAC
ASoC: tegra: Add machine driver for WM8753 codec
ALSA: hda - Fix possible races of accesses to connection list array
ASoC: OMAP: HDMI: Introduce codec
ARM: mx31_3ds: Add sound support
ASoC: imx-mc13783 cleanup
mx31moboard: Add sound support
ASoC: mc13783 codec cleanups
ASoC: add imx-mc13783 sound support
ASoC: Add mc13783 codec
mfd: mc13xxx: add codec platform data
ASoC: don't flip master of DT-instantiated DAI links
...
Pull trivial updates from Jiri Kosina:
"As usual, it's mostly typo fixes, redundant code elimination and some
documentation updates."
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/jikos/trivial: (57 commits)
edac, mips: don't change code that has been removed in edac/mips tree
xtensa: Change mail addresses of Hannes Weiner and Oskar Schirmer
lib: Change mail address of Oskar Schirmer
net: Change mail address of Oskar Schirmer
arm/m68k: Change mail address of Sebastian Hess
i2c: Change mail address of Oskar Schirmer
net: Fix tcp_build_and_update_options comment in struct tcp_sock
atomic64_32.h: fix parameter naming mismatch
Kconfig: replace "--- help ---" with "---help---"
c2port: fix bogus Kconfig "default no"
edac: Fix spelling errors.
qla1280: Remove redundant NULL check before release_firmware() call
remoteproc: remove redundant NULL check before release_firmware()
qla2xxx: Remove redundant NULL check before release_firmware() call.
aic94xx: Get rid of redundant NULL check before release_firmware() call
tehuti: delete redundant NULL check before release_firmware()
qlogic: get rid of a redundant test for NULL before call to release_firmware()
bna: remove redundant NULL test before release_firmware()
tg3: remove redundant NULL test before release_firmware() call
typhoon: get rid of redundant conditional before all to release_firmware()
...
Group read of hw_ptr, tstamp and jiffies in a sequence
for better correlation. Previous code took timestamp at the
end, which could introduce delays between audio time and
system time.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
With this, five platforms are moving to the relatively new pinctrl
subsystem for their pin management, replacing the older soc specific
in-kernel interfaces with common code.
There is quite a bit of net addition of code for each platform being
added to the pinctrl subsystem. but the payback comes later when adding
new boards can be done by only providing new device trees instead.
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Merge tag 'pinctrl' of git://git.kernel.org/pub/scm/linux/kernel/git/arm/arm-soc
Pull arm soc-specific pinctrl changes from Olof Johansson:
"With this, five platforms are moving to the relatively new pinctrl
subsystem for their pin management, replacing the older soc specific
in-kernel interfaces with common code.
There is quite a bit of net addition of code for each platform being
added to the pinctrl subsystem. But the payback comes later when
adding new boards can be done by only providing new device trees
instead."
Fix up trivial conflicts in arch/arm/mach-ux500/{Makefile,board-mop500.c}
* tag 'pinctrl' of git://git.kernel.org/pub/scm/linux/kernel/git/arm/arm-soc: (61 commits)
mtd: nand: gpmi: fix compile error caused by pinctrl call
ARM: PRIMA2: select PINCTRL and PINCTRL_SIRF in Kconfig
ARM: nomadik: enable PINCTRL_NOMADIK where needed
ARM: mxs: enable pinctrl support
video: mxsfb: adopt pinctrl support
ASoC: mxs-saif: adopt pinctrl support
i2c: mxs: adopt pinctrl support
mtd: nand: gpmi: adopt pinctrl support
mmc: mxs-mmc: adopt pinctrl support
serial: mxs-auart: adopt pinctrl support
serial: amba-pl011: adopt pinctrl support
spi/imx: adopt pinctrl support
i2c: imx: adopt pinctrl support
can: flexcan: adopt pinctrl support
net: fec: adopt pinctrl support
ARM: ux500: switch MSP to using pinctrl for pins
ARM: ux500: alter MSP registration to return a device pointer
ARM: ux500: switch to using pinctrl for uart0
ARM: ux500: delete custom pin control system
ARM: ux500: switch over to Nomadik pinctrl driver
...
Rename sound card driver source file to encompass not only OMAP4 but future
OMAP versions that feature HDMI.
Signed-off-by: Ricardo Neri <ricardo.neri@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
Rename all the relevant structures, variables and functions to not
make specific reference to OMAP4. This is to make the driver encompass
future OMAP versions that feature HDMI and not only OMAP4. These
changes are only in naming. There are not functional changes.
Signed-off-by: Ricardo Neri <ricardo.neri@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
Make Kconfig and Makefile more generic to encompass not only
OMAP4 but other OMAP processors featuring HDMI.
Also, relax the dependency list to depend only on any OMAP processor
supporting OMAP2_DSS and OMAP4_DSS_HDMI. As HDMI support for
future OMAP versions is added, the dependency list must change
accordingly.
Signed-off-by: Ricardo Neri <ricardo.neri@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
According to the HDMI specification, a source is permitted to
transmit L-PCM audio in the following sample rates: 32kHz, 44.1kHz,
48kHz, 88.2kHz, 96kHz, 176.4kHz or 192kHz.
It also supports up to 8 audio channels.
The sink may not necessarily support all these sample rates and
channels. However, as this CPU DAI describes the HDMI source, it
makes sense to include them. The limitation of capabilities as
supported by the sink should be done in the ASoC HDMI codec.
Signed-off-by: Ricardo Neri <ricardo.neri@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
Before starting to play audio, we need to make sure that the
display is active and the current video mode supports audio. instead
of using the overlay manager in the machine driver, we use the DSS audio
interface's audio_supported function. As we already have a pointer to
the correct dssdev, we do not have to look for it every time audio is
to be played. Also, the CPU DAI startup function is called earlier
than the card hw_param function. Hence and we can detect the state of
the display earlier.
While there, add a error message if the constraint cannot be applied.
Signed-off-by: Ricardo Neri <ricardo.neri@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
squash to improve err
Signed-off-by: Liam Girdwood <lrg@ti.com>
Expand the configuration of the hw_params to include the IEC-60958
channel status word and the CEA-861 audio infoframe. The configuration
of such structures depends on the snd_pcm_hw_params received. A
omap_dss_audio is used to pass the configuration parameters to DSS.
Signed-off-by: Ricardo Neri <ricardo.neri@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
Instead of accessing the HDMI IP directly, the CPU DAI driver takes
advantage of the audio interface provided by the DSS device driver.
The ASoC driver will link the DSS audio functionality with ALSA by
calling the appropriate DSS device driver functions at the relevant
moments. For this, three new DAI operations are added: trigger, prepare
and shutdown operations.
At the moment, it is assumed that only one HDMI display is available
in the system, as it is the case in OMAP4. However, in the future,
one DAI for each HDMI display should be provided.
Signed-off-by: Ricardo Neri <ricardo.neri@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
Create a struct hdmi_priv to store the relevant data of the CPU DAI
driver. As more data is added to the driver, having all the data
in the same location eases its handling. At the moment, only the DMA
configuration parameters are included in the structure.
Also, the required memory is allocated using devm_kzalloc rather than
using a static global variable.
Signed-off-by: Ricardo Neri <ricardo.neri@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
When getting the needed resources fails, return -ENODEV. This is more
in line with other drivers do and it gives a more descriptive error.
Signed-off-by: Ricardo Neri <ricardo.neri@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
In order to utilize the new OMAP HDMI codec and the updated name of
the device of the CPU DAI, update the names at the drivers accordingly.
While there, also update the name of the machine driver to be more
generic and encompass more OMAP processors featuring HDMI and not
only OMAP4.
Signed-off-by: Ricardo Neri <ricardo.neri@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
In preparation to Device Tree support.
With DT booted kernel we can not rely on pdata which used to
hold information needed for the driver at runtime.
Use the card's driver data to hold these informations from now on.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
The needed change in routing will be done runtime for the non
twl6040 connected widgets, like the Digital microphone.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
There is no need to have two snd_soc_dai_link structure for the two setup
the machine driver supports.
We can just tell core to register only the first link if the DMIC link is
not in use on the device.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
Device tree support for OMAP4+ dmic cpu dai driver.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Reviewed-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
Device tree support for OMAP4+ McPDM cpu dai driver.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Reviewed-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
In capture stream the buffer size does not need to be constrained to be
bigger then the McBSP FIFO.
In capture the FIFO content is taken out in period length burst, this
enusres that the FIFO is not going to overflow.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
With the common irq the driver only needs to use one interrupt line, and
it provides better debugging possibilites compared to the legacy TX/RX
interrupt lines.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
The frame dma_op_mode has never been used, and it is just creating
confusion for users/developers.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
Take the DMA packet mode into use when the McBSP is configured in element
dma_op_mode if the stream is not mono.
In this way we transfer one sample from/to McBSP FIFO upon DMA request.
This change only affects OMAP3+ versions, where the McBSP ports have FIFO.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
These are all new code, they've been in -next already so should be OK
for merge this time round. I'd been planning to send a pull request
today after they'd had a bit of exposure there to make sure breakage
didn't propagate into your tree.
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Merge tag 'asoc-3.5' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Last minute updates
These are all new code, they've been in -next already so should be OK
for merge this time round. I'd been planning to send a pull request
today after they'd had a bit of exposure there to make sure breakage
didn't propagate into your tree.
The major thing here is the addition of some helpers to factor code out
of drivers, making a fair proportion of regulators much more just data
rather than code which is nice.
- Helpers in the core for regulators using regmap, providing generic
implementations of the enable and voltage selection operations which
just need data to describe them in the drivers.
- Split out voltage mapping and voltage setting, allowing many more
drivers to take advantage of the infrastructure for selectors.
- Loads and loads of cleanups from Axel Lin once again, including many
changes to take advantage of the above new framework features
- New drivers for Ricoh RC5T583, TI TPS62362, TI TPS62363, TI TPS65913,
TI TWL6035 and TI TWL6037.
Some of the registration changes to support the core refactoring caused
so many conflicts that eventually topic branches were abandoned for this
release.
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Merge tag 'regulator-3.5' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/regulator
Pull regulator updates from Mark Brown:
"The major thing here is the addition of some helpers to factor code
out of drivers, making a fair proportion of regulators much more just
data rather than code which is nice.
- Helpers in the core for regulators using regmap, providing generic
implementations of the enable and voltage selection operations which
just need data to describe them in the drivers.
- Split out voltage mapping and voltage setting, allowing many more
drivers to take advantage of the infrastructure for selectors.
- Loads and loads of cleanups from Axel Lin once again, including many
changes to take advantage of the above new framework features
- New drivers for Ricoh RC5T583, TI TPS62362, TI TPS62363, TI
TPS65913, TI TWL6035 and TI TWL6037.
Some of the registration changes to support the core refactoring
caused so many conflicts that eventually topic branches were abandoned
for this release."
* tag 'regulator-3.5' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/regulator: (227 commits)
regulator: tps65910: use of_node of matched regulator being register
regulator: tps65910: dt: support when "regulators" node found
regulator: tps65910: add error message in case of failure
regulator: tps62360: dt: initialize of_node param for regulator register.
regulator: tps65910: use devm_* for memory allocation
regulator: tps65910: use small letter for regulator names
mfd: tpx6586x: Depend on regulator
regulator: regulator for Palmas Kconfig
regulator: regulator driver for Palmas series chips
regulator: Enable Device Tree for the db8500-prcmu regulator driver
regulator: db8500-prcmu: Separate regulator registration from probe
regulator: ab3100: Use regulator_map_voltage_iterate()
regulator: tps65217: Convert to set_voltage_sel and map_voltage
regulator: Enable the ab8500 for Device Tree
regulator: ab8500: Split up probe() into manageable pieces
regulator: max8925: Remove check_range function and max_uV from struct rc5t583_regulator_info
regulator: max8649: Remove unused check_range() function
regulator: rc5t583: Remove max_uV from struct rc5t583_regulator_info
regulator: da9052: Convert to set_voltage_sel and map_voltage
regulator: max8952: Use devm_kzalloc
...
Set some substream struct members to make the proc interface code work
again.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-by: Felix Homann <linuxaudio@showlabor.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The handling for some PCM states is missing for snd_pcm_drain().
At least, XRUN streams should be simply dropped to SETUP, and a few
initial invalid states should be rejected.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The module_platform_driver() conversion ended up tripping over the driver
name, leading to confusion in the macro with regards to 'driver' being
redefined. rename it to something slightly more suitable to avoid
namespace collisions.
sound/sh/sh_dac_audio.c:444:122: error: conflicting types for 'driver_init'
include/linux/device.h:773:6: note: previous declaration of 'driver_init' was here
make[3]: *** [sound/sh/sh_dac_audio.o] Error 1
Signed-off-by: Paul Mundt <lethal@linux-sh.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
SupherH FSI2 can use special data transfer,
but it depends on CPU-FSI2 connection style.
We can use 16bit data stream mode if it was valid connection,
and it is required for 16bit data DMA transfer / SPDIF sound output.
We can use 24bit data transfer if it was invalid connection.
We can select connection type if CPU is SH7372,
and it is always valid connection if latest SuperH.
This patch adds new bus_option and fsi_bus_setup()
for supporting these feature.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
fsi_hw_startup/shutdown() needs the setup of bus width,
but it is impossible to get parameter of snd_pcm_runtime at this timing.
So, these functions are changed so that be called from fsi_dai_trigger().
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds fsi_version() function for accessing version.
And there were some meaningless version check which never hit.
This patch removed it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Like the previous fixes for cache hash accesses, a protection over
accesses to the widget connection list array must be provided.
Together with this action, remove snd_hda_get_conn_list() which can be
always race, and replace it with either snd_hda_get_num_conns() or
snd_hda_get_connections() calls.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Introduce codec for HDMI. At the moment, this is a dummy codec. In the
future it will parse the EDID to modify the supported parameters, such
as the number of channels and the sample rates. At the moment, it blindly
supports all the sample rates and audio channels described in the HDMI
1.4a specification.
Signed-off-by: Ricardo Neri <ricardo.neri@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There is an spin_unlock() missing on this error path.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A few last-minute regression fixes for 3.4 final kernel.
All trivial, and Cc'ed to stable kernel.
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Merge tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"A few last-minute regression fixes for 3.4 final kernel. All trivial,
and Cc'ed to stable kernel."
* tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ASoC: wm8994: Fix AIF2ADC power down
ALSA: hda/idt - Fix power-map for speaker-pins with some HP laptops
ASoC: cs42l73: Sync digital mixer kcontrols to allow for 0dB
* 'clk-next' of git://git.linaro.org/people/mturquette/linux:
clk: Fix CLK_SET_RATE_GATE flag validation in clk_set_rate().
clk: Provide dummy clk_unregister()
ARM: Kirkwood: Replace clock gating
ARM: Orion: Audio: Add clk/clkdev support
ARM: Orion: PCIE: Add support for clk
ARM: Orion: XOR: Add support for clk
ARM: Orion: CESA: Add support for clk
ARM: Orion: SDIO: Add support for clk.
ARM: Orion: NAND: Add support for clk, if there is one.
ARM: Orion: EHCI: Add support for enabling clocks
ARM: Orion: SATA: Add per channel clk/clkdev support.
ARM: Orion: UART: Get the clock rate via clk_get_rate().
ARM: Orion: WDT: Add clk/clkdev support
ARM: Orion: Eth: Add clk/clkdev support.
ARM: Orion: SPI: Add clk/clkdev support.
ARM: Orion: Add clocks using the generic clk infrastructure.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Commit 4924082 "ASoC: core: Flip master for CODECs in the CPU slot of a
CODEC<->CODEC link" added code that was conditional on there being no
PCM/DMA driver for the link. However, it failed to cover the case where
the link was instantiated from device tree, and hence was specified by
DT node rather than name.
This prevents the following error on Toshiba AC100:
aplay: pcm_write:1603: write error: Input/output error
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some last minute fixes for ASoC. Small, focused changes to specific
drivers.
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Merge tag 'asoc-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Last minute fixes
Some last minute fixes for ASoC. Small, focused changes to specific
drivers.
* linus/master: (805 commits)
tty: Fix LED error return
openvswitch: checking wrong variable in queue_userspace_packet()
bonding: Fix LACPDU rx_dropped commit.
Linux 3.4-rc7
ARM: EXYNOS: fix ctrlbit for exynos5_clk_pdma1
ARM: EXYNOS: use s5p-timer for UniversalC210 board
ARM / mach-shmobile: Invalidate caches when booting secondary cores
ARM / mach-shmobile: sh73a0 SMP TWD boot regression fix
ARM / mach-shmobile: r8a7779 SMP TWD boot regression fix
ARM: mach-shmobile: convert ag5evm to use the generic MMC GPIO hotplug helper
ARM: mach-shmobile: convert mackerel to use the generic MMC GPIO hotplug helper
MAINTAINERS: Add myself as the cpufreq maintainer
dm mpath: check if scsi_dh module already loaded before trying to load
dm thin: correct module description
dm thin: fix unprotected use of prepared_discards list
dm thin: reinstate missing mempool_free in cell_release_singleton
gpio/exynos: Fix compiler warnings when non-exynos machines are selected
gpio: pch9: Use proper flow type handlers
powerpc/irq: Fix another case of lazy IRQ state getting out of sync
ks8851: Update link status during link change interrupt
...
Conflicts:
drivers/media/common/tuners/xc5000.c
drivers/media/common/tuners/xc5000.h
drivers/usb/gadget/uvc_queue.c
get_min_max_with_quirks() must be called after the control id name
string is determined, but the current code changes the id name string
after calling the function.
Reported-by: Christian Melki <christian.melki@ericsson.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch just sets the codec probe_mask=0x101 value for the WinFast VP200 H
PCoIP card based on Teradici hardware matching the PCI subsystem vendor/device
IDs 3a21:040d. The user reported no codec detection issues without this
explicit codec configuration.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Maintain both streams (playback, capture) synchronized. Previous code
didn't take in account the small byte count drifts caused by the irq
position rounding.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BIOS on some HP laptops don't set the speaker-pins as fixed but expose
as jacks, and this confuses the driver as if these pins are
jack-detectable. As a result, the machine doesn't get sounds from
speakers because the driver prepares the power-map update via jack
unsol events which never come up in reality. The bug was introduced
in some time in 3.2 for enabling the power-mapping feature.
This patch fixes the problem by replacing the check of the persistent
power-map bits with a proper is_jack_detectable() call.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=43240
Cc: <stable@vger.kernel.org> [v3.2+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Rename the function v4l2_dont_use_lock to v4l2_disable_ioctl_locking,
and rename v4l2_dont_use_cmd to v4l2_disable_ioctl.
Signed-off-by: Hans Verkuil <hans.verkuil@cisco.com>
Signed-off-by: Mauro Carvalho Chehab <mchehab@redhat.com>
aic3x_set_headset_detection() isn't made available outside the driver or
referenced within the driver which sparse notices and complains about.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add the support for VGA-switcheroo in the HD-audio controller side.
When the graphics controller is disabled, the HD-audio driver also delays
the initialization until it's activated by VGA-switcheroo.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=43155
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It's a preliminary work for the vga-switcher support.
Export the function to do pseudo-lock for the sound card to be used
in other places.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a discrete-GPU is disabled by the VGA switcheroo, the
corresponding HD-audio controller for HDMI output is also disabled.
Such a dead controller still appears in the PCI device list, but you
can't access properly any longer (even calling pci_read_config_*()
triggers Oops!) which leads the stall of the whole communication of
the driver.
This patch adds a check of graphics controller at the probe time to
see whether it's disabled by vga-switcheroo. If disabled, skip the
whole initialization of this controller.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=43155
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The tea575x-tuner framework can support the VIDIOC_S_HW_FREQ_SEEK for only
some of the tea575x-based boards. Mark this ioctl as invalid if the board
doesn't support it.
This fixes an issue with S_HW_FREQ_SEEK in combination with priority handling:
since the priority check is done first it could return -EBUSY, even though
calling the S_HW_FREQ_SEEK ioctl would return -ENOTTY. It should always return
ENOTTY in such a case.
Signed-off-by: Hans Verkuil <hans.verkuil@cisco.com>
Acked-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mauro Carvalho Chehab <mchehab@redhat.com>
Trying to flush completed packets is pointless when the pointer
callback was called from the packet completion callback; avoid it.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
By flushing all completed but not yet reported packets before reading
the PCM hardware position, the granularity of the pointer is improved
from the interrupt interval to the packet size.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The following patch might introduce this call chain:
PCM .pointer callback
+ fw_iso_context_flush_completions
+ packet callback
+ snd_pcm_period_elapsed
+ PCM .pointer callback
Recursive calls to the pointer callback are not possible due to the PCM
group locking, so avoid this by moving the period notification into
a separate tasklet.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In preparation for irq_domain support change the code to the not switch
based on the irq number. This actually makes things simpler, if slightly
repetitive.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use the devm_ versions of the regmap and memory allocation functions,
saving some error handling code.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since we only need to clock AIF2 when it's actively in use start up the
FLL for it using a supply widget which supplies AIF2CLK. This both makes
the sequencing more robust and ensures we minimise power consumption.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add device tree probe for mxs-sgtl5000 machine driver.
Signed-off-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add device tree probe for mxs-saif driver.
Signed-off-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Same as the commit 518de86 (ASoC: tegra: register 'platform' from DAIs,
get rid of pdev), it makes mxs-pcm not a platform_driver but helper to
register "platform", so that the platform_device for mxs-pcm can be
saved completely.
Signed-off-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
It turned out that the FLOAT format on CS4206 results in simple
noises, which implies that this is no right format as is.
Since CS4206 is the only codec supporting the float, let's disable it
until we find the correct format.
Reported-and-tested-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some of the Digital mixer kcontrol max values were off by 1 not allowing a max of 0dB.
Signed-off-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
Explained by Takashi in <s5hfwbtmz0q.wl%tiwai@suse.de>
> The reason is because get_min_max*() isn't called in the place you
> created these controls, and get_min_max() would be called only for
> integer volumes later even if uninitialized. A short cut for booleans.
Signed-off-by: Mark Hills <mark@pogo.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the PCM read/write loop, the driver calls snd_pcm_update_hw_ptr()
at each time at the beginning of the loop. Russell King reported that
this hogs CPU significantly.
The current code assumes that the pointer callback is very fast and
cheap, also not too much fine grained. It's not true in all cases.
When the pointer advances short samples while the read/write copy has
been performed, the driver updates the hw_ptr and gets avail > 0
again. Then it tries to read/write these small chunks. This repeats
until the avail really gets to zero.
For avoiding this situation, a simple workaround is to call
snd_pcm_update_hw_ptr() only once at starting the loop, assuming that
the read/write copy is performed fast enough. If the available count
becomes short, it goes to snd_pcm_wait_avail() anyway, and this
processes right.
Tested-by: Russell King <rmk+kernel@arm.linux.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When pdata->reset_pin is passed with a negative value (means gpio
is invalid), then chip->reset_pin will not be assigned to a vaule,
it will use default value 0. This will cause unexpected behavior.
So, add this patch to correct.
Signed-off-by: Bo Shen <voice.shen@atmel.com>
Acked-by: Nicolas Ferre <nicolas.ferre@atmel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Slightly more than expected as rc7, but all are reasonablly small fixes.
A few additions of HD-audio fixup entries, a couple of other regression
fixes including a revert, and a few other trivial oneliners.
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Merge tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"Slightly more than expected as rc7, but all are reasonablly small
fixes. A few additions of HD-audio fixup entries, a couple of other
regression fixes including a revert, and a few other trivial
oneliners."
* tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ASoC: sh: fix migor.c compilation
ALSA: HDA: Lessen CPU usage when waiting for chip to respond
Revert "ALSA: hda - Set codec to D3 forcibly even if not used"
ALSA: hda/realtek - Call alc_auto_parse_customize_define() always after fixup
ALSA: hdsp - Provide ioctl_compat
ALSA: hda/realtek - Add missing CD-input pin for MSI-7350 mobo
ALSA: hda/realtek - Add a fixup for Acer Aspire 5739G
ALSA: echoaudio: Remove incorrect part of assertion
The amp and caps hashes aren't protected properly for concurrent
accesses. Protect them via a new mutex now.
But it can't be so simple as originally thought: since the update of a
hash table entry itself might trigger the power-up sequence which
again accesses the hash table, we can't cover the whole function
simply via mutex. Thus the update part has to be split from the mutex
and revalidated.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
They pollute the global namespace and cause sparse to complain.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
It can only be used with a machine driver so the idiomatic thing for
ASoC is to select this driver from the machine driver.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The controller is compatible with HD-audio 1.0a with some specific
restrictions.
- The BDLE entries can't be over 4k boundary
- No position-buffer and no MSI
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Makes sparse happy and avoids polluting the global namespace.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Brian Austin <brian.austin@cirrus.com>
We need to read the real register values
Signed-off-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
An API update which wasn't sufficiently thorough in updating the tree...
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Merge tag 'asoc-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Build fix for SH in 3.4
An API update which wasn't sufficiently thorough in updating the tree...
We're trying to remove all usage of the ASoc level cache and I/O code and
for a device like this with a pretty sparse register map the rbtree cache
is a better idea anyway.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add driver for running I2S with the MSP-block.
Signed-off-by: Ola Lilja <ola.o.lilja@stericsson.com>
[Fixed trailing whitespace -- broonie]
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix a recent compilation breakage, caused by a change in SH clock API.
Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Check the power_transition up/down state instead of boolean bit, so
that the power-up sequence can cancel the pending power-down work
properly. Also, by moving cancel_delayed_work_sync() before the
actual power-up sequence, make sure that the delayed power-down is
completed.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since the recent commit, the resume procedure is always performed at
the resume time. This makes the pre_resume hack for VREF mute LED on
some HP laptops superfluous. As this is the only user of pre_resume
(and there is no user of post_suspend) ops, let's kill them again.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When an IRQ for some reason gets lost, we wait up to a second using
udelay, which is CPU intensive. This patch improves the situation by
waiting about 30 ms in the CPU intensive mode, then stepping down to
using msleep(2) instead. In essence, we trade some granularity in
exchange for less CPU consumption when the waiting time is a bit longer.
As a result, PulseAudio should no longer be killed by the kernel
for taking up to much RT-prio CPU time. At least not for *this* reason.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Tested-by: Arun Raghavan <arun.raghavan@collabora.co.uk>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move the fixup helper functions in patch_realtek.c to hda_auto_parser.c
so that they can be used in other codec drivers like patch_conexant.c.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some power-saving states have been left unchanged in
snd_hda_codec_reset(), and this is a potential danger because the
function may be called in various situations including the continuous
operation after that call.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is a fix for the problem in commit 785f857d1c, the pop noise
issue on some machines with ALC269. The problem was the uninitialized
state after the resume due to the delayed resume of the codec chips.
In that commit, we tried to fix by forcibly putting the codec to D3 at
suspend. But, this still also leaves the uninitialized state after
resume, and it _might_ be still problematic with some BIOS. Since the
commit turned out to regress another issues, we reverted it in the
end.
Now, in this fix, try to fix by turning on the codec immediately at
the resume path. We need to take care of the power-saving in this
case. When the device is woken up at the power-saved state, it should
go power-saving again after the resume.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The upper limit of the available minors isn't necessarily 128 + unit,
but it's rather up to 256. Fixing this allows more than 8 devices.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are left-over codes from the ancient days with the static device
number limitation of 8. Actaully OSS can support up to 16 cards.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This reverts commit 785f857d1c.
The commit causes a problem with the wrong D3 state after suspend
because the call of hda_set_power_state() involves with the power-up
sequence, which changes the power_count, and this confuses the resume
sequence that checks the power_count as well.
Originally, this go-to-D3 sequence should be a simple task without the
power-up sequence. But, it'd need some proper sanity checks in the
case of power-saved state, so it's not too easy to write now in the
3.4-rc cycle.
In short, the safest option now is to revert this affecting commit.
Of course, we need to clean up and robustify the power-saving code
better for 3.5 kernel.
Reported-by: Konstantin Khlebnikov <khlebnikov@openvz.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The call for alc_auto_parse_customize_define() must be done after the
fixup pre-probe initialization. Otherwise SKU_IGNORE fixup won't work
properly (e.g. HP RP5800 with ALC662 codec).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When we instantiate an aux_dev we use a fake rtd as part of the process
which doesn't have a dai_link associated with it. Fix the dpcm startup
code to cope with this.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
snd_hdsp uses its own ioctls to acquire config- and status information.
Expose the corresponding ioctl handler via ioctl_compat, so that 32bit applications can use it on 64bit kernels.
Signed-off-by: Andre Schramm <andre.schramm@iosono-sound.com>
Reviewed-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
None of the machines uses the gain ramp possibility for HS/HF.
This code path is mostly unused and it does not reduces the pop
noise on the output (it alters it to sound a bit different).
The preferred method to reduce pop noise is to use ABE.
Remove the gain ramp, and related features form the driver.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since this is a generic API which should support any userspace interface
for reporting jacks update the documentation a little to make that a bit
clearer.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Allocating the SSP DMA parameters in startup, freeing it in
shutdown instead of freeing and re-allocating it in hw_params.
After doing that, the logic is clear and more safe.
Signed-off-by: guoyh <guoyh@marvell.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acer Aspire 5739G requires the same fix-up for 4930G to support the
surround / bass speakers.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=43180
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This assertion seems to imply that chip->dsp_code_to_load is a pointer.
It's actually an integer handle on the actual firmware, and 0 has no
special meaning.
The assertion prevents initialisation of a Darla20 card, but would also
affect other models. It seems it was introduced in commit dd7b254d.
ALSA sound/pci/echoaudio/echoaudio.c:2061 Echoaudio driver starting...
ALSA sound/pci/echoaudio/echoaudio.c:1969 chip=ebe4e000
ALSA sound/pci/echoaudio/echoaudio.c:2007 pci=ed568000 irq=19 subdev=0010 Init hardware...
ALSA sound/pci/echoaudio/darla20_dsp.c:36 init_hw() - Darla20
------------[ cut here ]------------
WARNING: at sound/pci/echoaudio/echoaudio_dsp.c:478 init_hw+0x1d1/0x86c [snd_darla20]()
Hardware name: Dell DM051
BUG? (!chip->dsp_code_to_load || !chip->comm_page)
Signed-off-by: Mark Hills <mark@pogo.org.uk>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As good as nothing exciting here; just a few trivial fixes for
various ASoC stuff.
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Merge tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound sound fixes from Takashi Iwai:
"As good as nothing exciting here; just a few trivial fixes for various
ASoC stuff."
* tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ASoC: omap-pcm: Free dma buffers in case of error.
ASoC: s3c2412-i2s: Fix dai registration
ASoC: wm8350: Don't use locally allocated codec struct
ASoC: tlv312aic23: unbreak resume
ASoC: bf5xx-ssm2602: Set DAI format
ASoC: core: check of_property_count_strings failure
ASoC: dt: sgtl5000.txt: Add description for 'reg' field
ASoC: wm_hubs: Make sure we don't disable differential line outputs
Add the PCI ID of the Asus Xonar DGX card; it's otherwise
identical with the DG.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Nothing terribly exciting here, a bunch of small and simple fixes
scattered around the place.
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Merge tag 'asoc-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Updates for 3.4
Nothing terribly exciting here, a bunch of small and simple fixes
scattered around the place.
Signed-off-by: Oleg Matcovschi <oleg.matcovschi@ti.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
This patch improves playback quality for few sample rates like 8000 and
11025 Hz.
This also fixes an issue observed during testing of pll slave mode. Due
to the issue, on some rare occasions there was no sound output for first
time playback after system boot, though all subsequent playbacks were
fine. It was mainly because of the sequence in which SRM bit was
enabled.
Signed-off-by: Ashish Chavan <ashish.chavan@kpitcummins.com>
Signed-off-by: David Dajun Chen <dchen@diasemi.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Rather than invalidating the cached DCS value every time the headphone
gain changes store multiple values, indexed by gain. This allows the
optimisation we get from the cache to take effect more often.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This binding doesn't include the nvidia,model or nvidia,audio-routing
properties the other Tegra audio DT bindings have, because this binding
is targetted at a single machine, rather than for any machine using the
tlv320aic23 codec.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
As s3c2412-i2s is using the s3c_i2sv2 it should call the more specialised
s3c_i2sv2_register_dai instead of simply calling snd_soc_register_dai.
Without this call the snd_soc_dai_ops structure isn't initialised correctly.
Signed-off-by: Heiko Stuebner <heiko@sntech.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is now very standard behaviour for CODECs so shouldn't be device
specific and we shouldn't really be trying to peer into the register
cache from atomic context anyway.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In the conversion to module_init_i2c() the original open coded module
exit function was left. Remove it.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds support for Cirrus Logic CS42L52 Low Power Stereo Codec
Signed-off-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Georgi Vlaev <joe@nucleusys.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The core allocates the live copies, we shouldn't try to duplicate it and
were buggy trying to do so as we were using uninitialised data for the
control data.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We should check dailess before dereferencing.
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* commit f9dfbf9 "ASoC: tlv320aic23: convert to soc-cache" leads to
a bug preventing resumeof the codec as regmap expects a 9 bits data
register but 0xFFFF is passed in tlv320aic23_set_bias_level and this
values gets cached preventing any write to the TLV320AIC23_PWR
register as the final value produced by regmap is (register << 9) | value
* this patch solves the problem by only working on the 9 bits the
register contains.
Signed-off-by: Eric Bénard <eric@eukrea.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
It tries to clk_get the clock. And if it failed, it assumes the clock
by default enabled.
Signed-off-by: Richard Zhao <richard.zhao@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Class W can be used for any path where only data from the DAC is routed
to the headphones. Currently we only enable it when the direct DAC to
headphone path is used but it can also be enabled for paths that go via
the output mixer providing the DAC is the only input to the output mixer.
Implement support for this, including updates to the class W status when
the output mixer configuration is changed. This also allows us to enable
the DC servo optimisations for DAC to headphone paths where the output
mixer is used.
In general the direct DAC path is still preferred as this will offer
better performance on most wm_hubs devices but these additional paths
can simplify use case management.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since the analogue portions of the checks for class W are the same over
all the devices factor out these checks into wm_hubs and while we're at
it also use wm_hubs_dac_hp_direct() to enable class W optimisations on
more paths.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The optimisations which we can do with caching the headphone DCS result in
wm_hubs have only been enabled in cases where class W is enabled. However,
there are more use cases which can benefit from the cache, especially with
WM8994 series devices with their more advanced digital routing.
Rather than keying off the class W information from the CODECs have a
check in wm_hubs for a suitable path and use that to determine if we can
deploy our headphone optimisations.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Remove writable debugFS permission, use simple_open() and
fix indentation.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch fixes a bug discovered during testing of non pll slave mode.
Due to the bug chip was not getting correctly configured and as a result
there was no sound output while playback. After applying this patch,
both pll and non pll modes work fine.
Signed-off-by: Ashish Chavan <ashish.chavan@kpitcummins.com>
Signed-off-by: David Dajun Chen <dchen@diasemi.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Reduce our stack consumption by moving the params off the stack, they
are reasonably large and might be an issue on platforms with small stacks.
Reported-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Ackeded-by: Liam Girdwood <lrg@ti.com>
A workaround for an ASUS laptop and a few ASoC changes;
most of the commits are tagged for stable, too.
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Merge tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"A workaround for an ASUS laptop and a few ASoC changes; most of the
commits are tagged for stable, too."
* tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ASoC: wm8994: Improve sequencing of AIF channel enables
ALSA: HDA: Add external mic quirk for Asus Zenbook UX31E
ASoC: fsi: update for dmaengine prep_slave_sg fallout.
ASoC: core: Fix card RTD count for deferred probe.
ASoC: cs42l73: don't use negative array index
ASoC: dapm: Ensure power gets managed for line widgets
If a driver using a custom mic detection callback has provided a table
of mic detection rates via platform data then use it.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use a slightly larger debounce when identifying accessory type and a
slightly smaller one when detecting buttons in response to user feedback
from large scale testing.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When we're not actively doing audio we don't need the microphone biases
to be regulated, noise is not important when we are not looking at the
audio signal. Save some power by putting the MICBIAS regulators into
bypass mode when not doing audio.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Provide an ioctl marshaller for ASoC platform drivers.
This will use the default ALSA handler if no platform
handler exists.
This is also required for DPCM BE PCMs as snd_pcm_info()
will call the ioctl as part of stream startup.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some on SoC DSP HW is very tightly coupled with DMA and DAI drivers. It's
necessary to allow some flexability wrt to PCM operations here so that we
can define a bespoke DPCM trigger() PCM operation for such HW.
A bespoke DPCM trigger() allows exact ordering and timing of component
triggering by allowing a component driver to manage the final enable
and disable configurations without adding extra complexity to other
component drivers. e.g. The McPDM DAI and ABE are tightly coupled on
OMAP4 so we have a bespoke trigger to manage the trigger to improve
performance and reduce complexity when triggering new McPDM BEs.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some component drivers will need to be able to look up their
DAI link substream and RTD data. Provide a mechanism for this.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch allows DPCM to dynamically alter the FE to BE PCM links
at runtime based on mixer setting updates. DAPM is looked up after
every mixer update and we perform a DPCM runtime update if the
mixer has a change of value.
This patchs adds/changes the following :-
o Adds DPCM runtime update core.
o Changes soc_dapm_mixer_update_power() and soc_dapm_mux_update_power()
to return if a change has occured rather than 0. No other users check
atm.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add debugFS files for DPCM link management information.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The Dynamic PCM core allows digital audio data to be dynamically
routed between different ALSA PCMs and DAI links on SoC CPUs with
on chip DSP devices. e.g. audio data could be played on pcm:0,0 and
routed to any (or all) SoC DAI links.
Dynamic PCM introduces the concept of Front End (FE) PCMs and Back
End (BE) PCMs. The FE PCMs are normal ALSA PCM devices except that
they can dynamically route digital audio data to any supported BE
PCM. A BE PCM has no ALSA device, but represents a DAI link and it's
substream and audio HW parameters.
e.g. pcm:0,0 routing digital data to 2 external codecs.
FE pcm:0,0 ----> BE (McBSP.0) ----> CODEC 0
+--> BE (McPDM.0) ----> CODEC 1
e.g. pcm:0,0 and pcm:0,1 routing digital data to 1 external codec.
FE pcm:0,0 ---
+--> BE (McBSP.0) ----> CODEC
FE pcm:0,1 ---
The digital audio routing is controlled by the usual ALSA method
of mixer kcontrols. Dynamic PCM uses a DAPM graph to work out the
routing based upon the mixer settings and configures the BE PCMs
based on routing and the FE HW params.
DPCM is designed so that most ASoC component drivers will need no
modification at all. It's intended that existing CODEC, DAI and
platform drivers can be used in DPCM based audio devices without
any changes. However, there will be some cases where minor changes
are required (e.g. for very tightly coupled HW) and there are
helpers to support this too.
Somethimes the HW params of a FE and BE do not match or are
incompatible, so in these cases the machine driver can reconfigure
any hw_params and make any DSP perform sample rate / format conversion.
This patch adds the core DPCM code and contains :-
o The FE and BE PCM operations.
o FE and BE DAI link support.
o FE and BE PCM creation.
o BE support API.
o BE and FE link management.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
commit 4183eed2 (ASoC: core: Add signed multi register control) introduced
the variable 'min',but it is not used.
Remove it to fix the following build warning:
sound/soc/soc-core.c: In function 'snd_soc_put_xr_sx':
sound/soc/soc-core.c:2990: warning: unused variable 'min'
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Mostly a one to one converion. On one occasion the patch replaces a
snd_soc_read-snd_soc_write sequence with regmap_update_bits though as it helps
to keep the conversion simple.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We have never really updated that version number and probably never will, so
just remove it.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Not all advertised rates are available for all sysclk frequencies. Add
additional sysclk based rate constraints.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The sysclock is fixed, so just set it up once in the init callback instead of
setting it repeatably in the hw_params callback.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Commit 980b0bc69 ("ASoC: blackfin: Use dai_fmt") converted the blackfin ASoC
machine drivers to use the dai_links dai_fmt field to setup their DAI format.
For the bf5xx-ssm2602 the commit removed the manual call to snd_soc_dai_set_fmt,
but missed to set the dai_links dai_fmt field.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If FLL bypass is left enabled when we disable the CODEC then the output
clock will be left running which consumes a small amount of additional
current. Only enable bypass when there is an output.
Signed-off-by: Kyung-Kwee Ryu <Kyung-Kwee.Ryu@wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
sound/usb/endpoint.c: In function 'queue_pending_output_urbs':
sound/usb/endpoint.c:298: warning: 'packet' may be used uninitialized in this function
Cc: Daniel Mack <zonque@gmail.com>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This adds controls for the effects section on the FTU devices.
Some of these controls need volume quirks. They are added to
mixer.c.
[fixed missing break by tiwai]
Signed-off-by: Felix Homann <linuxaudio@showlabor.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is in preparation for more FTU controls to come.
Should help keeping names a bit shorter.
Signed-off-by: Felix Homann <linuxaudio@showlabor.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This adds db gain information to M-Audio Fast Track Ultra (8R) devices.
Signed-off-by: Felix Homann <linuxaudio@showlabor.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Rename mixer_vol_tlv to snd_usb_mixer_vol_tlv and export it to make
it reuseable in mixer_quirks.c.
Signed-off-by: Felix Homann <linuxaudio@showlabor.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Merge snd_maudio_ftu_create_ctl() and snd_ebox44_create_ctl() into
snd_create_std_mono_ctl().
As opposed to the ftu and ebox-44 specific functions, a TLV callback
can be specified for controls created by snd_create_std_mono_ctl().
[fixed minor checkpatch.pl warnings by tiwai]
Signed-off-by: Felix Homann <linuxaudio@showlabor.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
While we need to clean up unused single ended line outputs we don't want
to do this if the outputs are in differential mode.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Added support for a control that strobes a bit in
a register to high then back to low (or the inverse).
This is typically useful for hardware that requires
strobing a singe bit to trigger some functionality
and where exposing the bit in a normal single control
would require the user to first manually set then
again unset the bit again for the strobe to trigger.
Added convenience macro.
SOC_SINGLE_STROBE
Added accessor implementations.
snd_soc_get_strobe
snd_soc_put_strobe
Signed-off-by: Kristoffer KARLSSON <kristoffer.karlsson@stericsson.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Added control type that can span multiple consecutive codec registers
forming a single signed value in a MSB/LSB manner.
The control dynamically adjusts to the register word size configured
in driver.
Added convenience macro.
SOC_SINGLE_XR_SX
Added accessor implementations.
snd_soc_info_xr_sx
snd_soc_get_xr_sx
snd_soc_put_xr_sx
Signed-off-by: Kristoffer KARLSSON <kristoffer.karlsson@stericsson.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
While reading through sound/soc/codecs/wm8994.c I noticed a fair
amount of trailing whitespace. This patch gets rid of it.
Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Slightly larger than normal - the DAPM fix is a "this should always have
worked" type of thing which is very clear and should have no impact on
systems that don't need it. The WM8994 fix is driver specific but
pretty important for that driver.
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Merge tag 'asoc-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: updates for 3.4
Slightly larger than normal - the DAPM fix is a "this should always have
worked" type of thing which is very clear and should have no impact on
systems that don't need it. The WM8994 fix is driver specific but
pretty important for that driver.
This ensures a clean startup of the channels, without this change some
use cases could result in issues in a small proportion of cases.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
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Merge tag 'mfd-for-linus-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/sameo/mfd-2.6
Pull MFD fixes from Samuel Ortiz:
"We have 3 build fixes, a OMAP USB host PHY reset fix and the twl6040
conversion to an i2c driver. The latter may not sound like a fix but
the twl6040 MFD driver won't probe without it, triggering an OMAP4
audio regression."
* tag 'mfd-for-linus-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/sameo/mfd-2.6:
mfd: Fix modular builds of rc5t583 regulator support
mfd: Fix asic3_gpio_to_irq
ARM: OMAP3: USB: Fix the EHCI ULPI PHY reset issue
mfd: Convert twl6040 to i2c driver, and separate it from twl core
mfd : Fix dbx500 compilation error
Drop some struct members and definitions that became obsolete during
the refactorization of the driver.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixes for a few regressions of HD-audio, originated partly from 3.4
and partly 3.3.
The fixes for ThinkPad docking-station are for 3.3 kernels, thus they
are based on 3.3 then merged back to 3.4, so that they can be merged
to stable tree cleanly. The non-trivial merge conflicts are because
of this action.
In addition, a copule of trivial fixes for documentation and a long-
statnding issue in the listing of built-in sound driver at boot time.
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Merge tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"Fixes for a few regressions of HD-audio, originated partly from 3.4
and partly 3.3.
The fixes for ThinkPad docking-station are for 3.3 kernels, thus they
are based on 3.3 then merged back to 3.4, so that they can be merged
to stable tree cleanly. The non-trivial merge conflicts are because
of this action.
In addition, a couple of trivial fixes for documentation and a long-
standing issue in the listing of built-in sound driver at boot time."
* tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda/conexant - Set up the missing docking-station pins
ALSA: hda/conexant - Don't set HP pin-control bit unconditionally
ALSA: workaround: change the timing of alsa_sound_last_init()
ALSA: hda/sigmatel - Fix inverted mute LED
ALSA: hda/realtek - Fix regression on Quanta/Gericom KN1
ALSA: fix core/vmaster.c kernel-doc warning
Some old codecs like ALC880 seem to give a bogus pin capability value 0
occasionally. This breaks the new sanity check in snd_hda_set_pin_ctl().
Skip the sanity checks in such a case.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a new helper function to guess the default VREF pin control bits
for mic in. This can be used to set the pin control value safely
matching with the actual pin capabilities.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For setting the pin-control values more safely to match with the
actual pin capability bits, a copule of new helper functions,
snd_hda_set_pin_ctl() and snd_hda_set_pin_ctl_cache(), are
introduced. These are simple replacement of the codec verb write with
AC_VERB_SET_PIN_WIDGET but do more sanity checks and filter out
superfluous pin-control bits if they don't fit with the corresponding
pin capabilities.
Some codecs are screwed up or ignore the command when such a wrong bit
is set. These helpers will avoid such secret errors.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
According to the reporter, external mic starts to work if the
laptop-dmic model is used. According to BIOS pin config, all
pins are consistent with the alc269vb_laptop_dmic fixup, except
for the external mic, which is not present.
Cc: stable@kernel.org
BugLink: https://bugs.launchpad.net/bugs/950490
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Merge fixes for Thinkpad docking-station regressions for 3.3 kernels
back to 3.4. These were committed in that branch to make the stable
merging easier.
Conflicts:
sound/pci/hda/patch_conexant.c
ThinkPad 410,420,510,520 and X201 with cx50585 & co chips have the
docking-station ports, but BIOS doesn't initialize for these pins.
Thus, like the former X200, we need to set up the pins manually in the
driver.
The odd part is that the same PCI SSID is used for X200 and T400, thus
we need to prepare individual fixup tables for cx5051 and others.
Bugzilla entries:
https://bugzilla.redhat.com/show_bug.cgi?id=808559https://bugzilla.redhat.com/show_bug.cgi?id=806217https://bugzilla.redhat.com/show_bug.cgi?id=810697
Reported-by: Josh Boyer <jwboyer@redhat.com>
Reported-by: Jens Taprogge <jens.taprogge@taprogge.org>
Tested-by: Jens Taprogge <jens.taprogge@taprogge.org>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some output pins on Conexant chips have no HP control bit, but the
auto-parser initializes these pins unconditionally with PIN_HP.
Check the pin-capability and avoid the HP bit if not supported.
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Springbank module can support a range of sample rates, selected at
runtime via GPIO configuration. Allow these to be configured at runtime.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Current alsa_sound_last_init() was called as __initcall().
So, on current ALSA, only devices that had been properly
registered at this point were shown.
So, it will show "No soundcards found" if driver requests
probe deferment. it's often misleading.
This patch delays the timing of alsa_sound_last_init()
as workaround.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviwed-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
While refactoring the mute-LED handling for HP laptops, I messed up
the polarity check in a wrong way. The red (or the mute-LED if any)
should appear in the muted state, corresponding to GPIO on.
Reported-by: Mikko Vinni <mmvinni@yahoo.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Through the transition to the auto-parser, the support for
Quanta/Gericom KN1 got broken. There are two problems behind it:
- This machine doesn't like the default COEF setup for ALC260 we take
now as default
- BIOS doesn't set the pins correctly at all; especially the machine
uses only the pin 0x0f for both headphone and speaker
This patch adds the fixup as a workaround for these issues.
Reported-and-tested-by: Uros Vampl <mobile.leecher@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In preparation for ASoC DSP support.
Add a DAPM API call to determine whether a DAPM audio path is valid between
source and sink widgets. This also takes into account all kcontrol mux and mixer
settings in between the source and sink widgets to validate the audio path.
This will be used by the DSP core to determine the runtime DAI mappings
between FE and BE DAIs in order to run PCM operations.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Connect the WM1250-EV1 baseband simulator on Littlemill systems up to
the CODEC AIF2 using the new CODEC<->CODEC link support, allowing a wider
range of use cases to be represented.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Leading up to the ->device_prep_slave_sg change in
185ecb5f4f 'dmaengine: add context
parameter to prep_slave_sg and prep_dma_cyclic' a generic wrapper was
added in place to guard against the API change, though the fsi driver
wasn't updated in the process (presumably its dmaengine support hadn't
been merged yet at the time). This trivially switches over to the new
wrapper and gets it building again.
Signed-off-by: Paul Mundt <lethal@linux-sh.org>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix kernel-doc warning in sound/core/vmaster.c:
Warning(sound/core/vmaster.c:429): No description found for parameter 'private_data'
Signed-off-by: Randy Dunlap <rdunlap@xenotime.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There is nothing audio-specific about the rcode_string() helper, so move
it from snd-firewire-lib into firewire-core to allow other code to use it.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Stefan Richter <stefanr@s5r6.in-berlin.de> (fixed sound/firewire/cmp.c)
Since AIF3 shares clock signals with other audio interfaces in order to
ensure it doesn't drive undesirable clocks we need to tristate it. Rather
than forcing the machine driver to do so have the driver do this.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch converts multiple if conditions in to single if with "&&"s.
Signed-off-by: Ashish Chavan <ashish.chavan@kpitcummins.com>
Signed-off-by: David Dajun Chen <dchen@diasemi.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently we increment the number of RTD's per card during the DAI link
bind. This can cause an incorrect RTD count when we cannot find a component
and defer the probe (and hence perform the DAI link bind for the card again).
Fix the count so that it is cleared before every card registration
and bind attempt.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Current DA7210 driver does support PLL mode fully. It uses fixed
value of input master clock and PLL mode is enabled and disabled based
on the sampling frequency being used for playback or recording. It also
doesn't support Sample Rate Measurement feature of DA7210 hardware.
This patch adds full support for PLL and SRM. Basically following three
modes of operation are possible for DA7210 hardware,
(1) I2S SLAVE mode with PLL bypassed
(2) I2S SLAVE mode with PLL enabled
(3) I2S Master mode with PLL enabled
This patch adds support for all three modes. Also, in case of SLAVE mode
with PLL, it supports SRM (Sample Rate Measurement) feature of the chip.
Actually this patch was submitted earlier and received some review
comments, but after that the driver got update by other patches. Because
of that, I am considering this as new patch and not versioning it based
of previous patches. This version tries to take care of all review
comments received for earlier submissions.
Signed-off-by: Ashish Chavan <ashish.chavan@kpitcummins.com>
Signed-off-by: David Dajun Chen <dchen@diasemi.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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ASoC: Merge tag 'v3.4-rc3' into for-3.5
Linux 3.4-rc3 contains a bunch of Tegra changes which are conflicting
annoyingly with the new development that's going on for Tegra so merge
it up to resolve those conflicts.
Conflicts:
sound/soc/soc-core.c
sound/soc/tegra/tegra_i2s.c
sound/soc/tegra/tegra_spdif.c
Fix the following build warning:
sound/soc/soc-dapm.c: In function 'snd_soc_dai_link_event':
sound/soc/soc-dapm.c:2913: warning: format '%lx' expects type 'long unsigned int', but argument 3 has type 'u64'
'%llx' should be used with 'u64' type.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Rather than having the user half start a stream but avoid any DMA to
trigger data flow on links which don't pass through the CPU create a
DAPM route between the two DAI widgets using a hw_params configuration
provided by the machine driver with the new 'params' member of the
dai_link struct. If no configuration is provided in the dai_link then
use the old style even for CODEC<->CODEC links to avoid breaking
systems.
This greatly simplifies the userspace usage of such links, making them
as simple as analogue connections with the stream configuration being
completely transparent to them.
This is achieved by defining a new dai_link widget type which is created
when CODECs are linked and triggering the configuration of the link via
the normal PCM operations from there. It is expected that the bias
level callbacks will be used for clock configuration.
Currently only the DAI format, rate and channel count can be configured
and currently the only DAI operations which can be called are hw_params
and digital_mute(). This corresponds well to the majority of CODEC
drivers which only use other callbacks for constraint setting but there
is obviously much room for extension here. We can't simply call
hw_params() on startup as things like the system clocking configuration
may change at runtime and in future it will be desirable to offer some
configurability of the link parameters.
At present we are also restricted to a single DAPM link for the entire
DAI. Once we have better support for channel mapping it would also be
desirable to extend this feature so that we can propagate per-channel
power state over the link.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
We should always have a CODEC already there when registering a CODEC DAI
and for CODEC<->CODEC links a dai_link will have two CODECs so it's much
simpler to do things at registration time.
This results in a slight change in the error handling for failed CODEC
DAI registrations but practically speaking these are never supposed to
fail so there shouldn't be much issue. The change is that we don't fail
the overall CODEC registration if the DAI registration fails; this seems
more robust anyway as we may not need to use a given DAI in a particular
system.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When two CODEC DAIs are linked directly to each other then if we give the
same master mode settings to both devices things won't work as either
neither will drive or they'll drive against each other. Flip the settings
for the DAI in the CPU slot of the DAI link.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In order to allow CODEC<->CODEC links to function we will need to allow
DAPM paths to be created that pass through DAIs rather than only ones
that are source or sunk at the DAI.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
This helps us ignore errors in callers if the operation failed due to not
being available as opposed to an error.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Complete the separation of the twl6040 from the twl core since
it is a separate chip, not part of the twl6030 PMIC.
Make the needed Kconfig changes for the depending drivers at the
same time to avoid breaking the kernel build (vibra, ASoC components).
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Reviewed-by: Mark Brown <broonie@opensource.wolfsonicro.com>
Acked-by: Tony Lindgren <tony@atomide.com>
Acked-by: Dmitry Torokhov <dtor@mail.ru>
Signed-off-by: Samuel Ortiz <sameo@linux.intel.com>
A few regression fixes for Realtek HD-audio codecs, mainly specific to
some laptop models.
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Merge tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull another round of sound fixes from Takashi Iwai:
"A few regression fixes for Realtek HD-audio codecs, mainly specific to
some laptop models."
* tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda/realtek - Fix mem leak (and rid us of trailing whitespace).
ALSA: hda/realtek - Add quirk for Mac Pro 5,1 machines
ALSA: hda/realtek - Add a fixup entry for Acer Aspire 8940G
ALSA: hda/realtek - Fix GPIO1 setup for Acer Aspire 4930 & co
ALSA: hda/realtek - Add a few ALC882 model strings back
The mixer units from the firmware are corrupt, and even where they
are valid they presents mono controls as L and R channels of
stereo.
Signed-off-by: Mark Hills <mark@pogo.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some interfaces reference endpoints which do not exists. To
accomodate these, do not fail completely, but skip over them.
This allows the Electrix Ebox-44 with earlier firmware to be
detected and used for audio.
Signed-off-by: Mark Hills <mark@pogo.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We added locking here but there were a couple error paths where we
forgot to drop the lock before returning.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
All Tegra ASoC drivers will be reworked to use MMIO regmaps. Select
this in Kconfig.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Many fields have been moved to struct snd_usb_endpoint.
Also fix the proc output to correspond to the new structure.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ep->fill_max is a 1 bit flag, thus it has to be boolean.
sound/usb/endpoint.c: In function 'snd_usb_endpoint_set_params':
sound/usb/endpoint.c:785: warning: overflow in implicit constant conversion
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Current SuperH FSI require simple-card driver as sound card.
This patch select it on Kconfig when FSI was selected.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch uses simple-card driver instead of fsi-da7210 on each board.
To select DA7210 driver, each boards select it on Kconfig.
This patch removes fsi-da7210 driver which is no longer needed
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch uses simple-card driver instead of fsi-hdmi on each board.
This patch removes fsi-hdmi driver which is no longer needed
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch uses simple-card driver instead of fsi-ak4642 on each board.
To select AK4642 driver, each boards select it on Kconfig.
This patch removes fsi-ak4642 driver which is no longer needed
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Current ASoC requires card.c file to each platforms in order to
specifies its CPU and Codecs pair.
But the differences between these were only value/strings of setting.
In order to reduce duplicate driver, this patch adds generic/simple-card.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This adds Kconfig options for the Tegra30 AHUB and I2S controller, and
updates the Tegra+WM8903 machine driver Kconfig to select those.
Includes a squashed bugfix from Sumit Bhattacharya <sumitb@nvidia.com>
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This provides an ASoC DAI interface for Tegra 30's I2S controller.
Includes a squashed bugfix from Sumit Bhattacharya <sumitb@nvidia.com>
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The AHUB (Audio Hub) is a mux/crossbar which links all audio-related
devices except the HDA controller on Tegra30. The devices include the
DMA FIFOs, DAM (Digital Audio Mixers), I2S controllers, and SPDIF
controller. Audio data may be routed between these devices in various
combinations as required by board design/application.
Includes a squashed bugfix from Nikesh Oswal <noswal@nvidia.com>
Includes squashed bugfixes from Sumit Bhattacharya <sumitb@nvidia.com>
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If cs42l73_get_mclkx_coeff() returns < 0 (which it can) in
sound/soc/codecs/cs42l73.c::cs42l73_set_mclk(), then we'll be using
the (negative) return value as array index on the very next line of
code - that's bad.
Catch the negative return value and propagate it to the caller (which
checks for it) and things are a bit more sane :-)
Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Document the new streaming code and some of the functions so that
contributers can catch up easier.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Implicit feedback is a streaming mode that does not rely on dedicated
sync endpoints but uses the information provided by record streams to
clock output streams. Now that the streaming logic is decoupled from the
PCM streams, this is easy to implement.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
With the previous commit that added the new streaming model, all
endpoint and streaming related code is now in endpoint.c, and pcm.c
only acts as a wrapper for handling the packet's payload.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds a new generic streaming logic for audio over USB.
It defines a model (snd_usb_endpoint) that handles everything that
is related to an USB endpoint and its streaming. There are functions to
activate and deactivate an endpoint (which call usb_set_interface()),
and to start and stop its URBs. It also has function pointers to be
called when data was received or is about to be sent, and pointer to
a sync slave (another snd_usb_endpoint) that is informed when data has
been received.
A snd_usb_endpoint knows about its state and implements a refcounting,
so only the first user will actually start the URBs and only the last
one to stop it will tear them down again.
With this sort of abstraction, the actual streaming is decoupled from
the pcm handling, which makes the "implicit feedback" mechanisms easy to
implement.
In order to split changes properly, this patch only adds the new
implementation but leaves the old one around, so the the driver doesn't
change its behaviour. The switch to actually use the new code is
submitted separately.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In sound/pci/hda/patch_realtek.c::alc_auto_fill_dac_nids(), in the
'for (;;)' loop, if the 'badness' value returned from
fill_and_eval_dacs() is negative, then we'll return from the function
without freeing the memory we allocated for 'best_cfg', thus leaking.
Fix the leak by kfree()'ing the memory when badness is negative.
While I was there I also noticed some trailing whitespace in the
function that I removed (along with all other trailing whitespace in
the file) - it didn't seem worth-while to do that as two patches, so I
hope it's OK that I just did it all as one patch.
Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Line widgets had not been included in either the power up or power down
sequences so if a widget had an event associated with it that event would
never be run. Fix this minimally by adding them to the sequences, we
should probably be doing away with the specific widget types as they all
have the same priority anyway.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
A user reported that setting model=imac24 used to allow sound to work on their
Mac Pro 5,1 machine. Commit 5671087ffa "Move ALC885 macpro and imac24 models
to auto-parser" removed this model option. All Mac machines are now explicitly
handled with a quirk and the auto-parser. This adds a quirk for the device
found on the Mac Pro 5,1 machines.
This (partially) fixes https://bugzilla.redhat.com/show_bug.cgi?id=808559
[sorted the new entry in the ID number order by tiwai]
Reported-by: Gabriel Somlo <somlo@cmu.edu>
Signed-off-by: Josh Boyer <jwboyer@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It's compatible with 8930G.
Using the same fixup gives the proper 5.1 sound back.
Reported-and-tested-by: Dany Martineau <dany.luc.martineau@gmail.com>
Cc: <stable@kernel.org> [v3.3+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add GPIO1 setup explicitly for Acer Aspire 493x & co.
This could be set by alc_auto_init_amp(), but it's safer to set it
more explicitly in the fixup table.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- A series of fixes for Conexant 20549 HD-audio codec chip
- A workaround for HDMI hotplug debug prints that annoyed people
- A fix for the new support of platform DAPM contexts
- Many driver-specific minor fixes
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Merge tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
- A series of fixes for Conexant 20549 HD-audio codec chip
- A workaround for HDMI hotplug debug prints that annoyed people
- A fix for the new support of platform DAPM contexts
- Many driver-specific minor fixes
* tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda - hide HDMI/ELD printks unless snd.debug=2
ALSA: sound/isa/sscape.c: add missing resource-release code
sound: sound/oss/msnd_pinnacle.c: add vfrees
ALSA: hda - clean up CX20549 test mixer setup
ALSA: hda - CX20549 doesn't need pin_amp_workaround.
ALSA: hda - Remove CD control from model=benq for CX20549
ALSA: hda - fix record volume controls of CX20459 ("Venice")
ALSA: hda - Rename capture sources of CX20549 to match common conventions
ALSA: hda - Fix proc output for ADC amp values of CX20549
ASoC: tegra: fix i2s compilation when !CONFIG_DEBUG_FS
ASoC: set idle_bias_off=1 for all platform DAPM contexts
ASoC: imx-audmux: Check for NULL pointer
ASoC: imx-audmux: Fix ssi port numbers in sysfs
ASoC: ak4642: fixup: mute needs +1 step
MAINTAINERS: Don't list everyone working on Wolfson drivers
MAINTAINERS: Add missing ASoC OMAP co-maintainer
ASoC: pxa: pxa2xx-i2s: add io.h for IOMEM macro
ASoC: tegra: ensure clocks are enabled when touching registers
ASoC: sgtl5000: Enable VAG when DAC/ADC up
ALSA: asihpi - fix return value of hpios_locked_mem_alloc()
Pull media fixes from Mauro Carvalho Chehab:
- dvb core: there is a regression found when used with xine. For
whatever unknown reason, xine (and xine-lib clients) wants that the
frontend to tell what frequency he is using even before the PLL lock
(or at least, it expects a non-zero frequency).
On DVB, the frequency is only actually known after a frequency
zig-zag seek, done by the DVB core. Anyway, the fix was trivial.
That solves Fedora BZ#808871.
- ivtv: fix a regression when selecting the language channel
- uvc: fix a race-related crash
- it913x: fixes firmware loading
- two trivial patches (a dependency issue at a radio driver at sound
Kconfig, and a warning fix on dvb).
* 'v4l_for_linus' of git://git.kernel.org/pub/scm/linux/kernel/git/mchehab/linux-media:
[media] uvcvideo: Fix race-related crash in uvc_video_clock_update()
[media] Drivers/media/radio: Fix build error
[media] dvb_frontend: fix compiler warning
[media] it913x: fix firmware loading errors
[media] ivtv: Fix AUDIO_(BILINGUAL_)CHANNEL_SELECT regression
[media] dvb_frontend: regression fix: userspace ABI broken for xine
Since there are still many Acer models that might not be covered by
the current fixup table, let's add back a few typical model names so
that user can test the fixup without recompiling.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
sound/soc/codecs/wm9705.c: In function 'ac97_prepare':
sound/soc/codecs/wm9705.c:251: error: 'runtime' undeclared (first use in this function)
This was caused by commit e6968a (ASoC: codecs: Remove rtd->codec usage from CODEC drivers),
which removed the 'struct snd_pcm_runtime *runtime = substream->runtime' definition.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix the following build error:
sound/soc/codecs/ac97.c: In function 'ac97_prepare':
sound/soc/codecs/ac97.c:33: error: 'runtime' undeclared (first use in this function)
This was caused by commit e6968a (ASoC: codecs: Remove rtd->codec usage from CODEC drivers),
which removed the 'struct snd_pcm_runtime *runtime = substream->runtime' definition.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix the following build error:
sound/soc/codecs/wm9712.c:482:32: error: 'runtime' undeclared (first use in this function)
sound/soc/codecs/wm9712.c:499:33: error: 'runtime' undeclared (first use in this function)
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Recent list discussions concluded that drivers should not be calling
of_have_populated_dt(), and hence of_have_populated_dt() should not be
exported. Use a different mechanism to detect DT vs. non-DT boot.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Make sure we don't dereference the "quirk" pointer when it is null.
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Also remove two warnings when CONFIG_SND_DEBUG is not set:
sound/pci/hda/patch_hdmi.c: In function ‘hdmi_intrinsic_event’:
sound/pci/hda/patch_hdmi.c:761:6: warning: unused variable ‘eldv’ [-Wunused-variable]
sound/pci/hda/patch_hdmi.c:760:6: warning: unused variable ‘pd’ [-Wunused-variable]
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Later WM8994 class devices can bypass the FLL from BCLK. Do this
automatically when the FLL input and output frequencies match up.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
release_firmware() does its own NULL ptr testing, it's redundant to
also test before calling it.
Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
release_firmware() deals gracefully with NULL pointers, no need to check first.
Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
At the point of this error-handling code, both regions and the dma have
been allocated, so free it as done in previous and subsequent
error-handling code.
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
At the point of this error-handling code, HAVE_DSPCODEH may be undefined,
so free INITCODE and PERMCODE as done elsewhere. A jump and label are
introduced to avoid code duplication.
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On Sunday, April 01, 2012 21:09:34 Tracey Dent wrote:
> radio-maxiradio depends on SND_FM801_TEA575X_BOOL to build or will
> result in an build error such as:
>
> Kernel: arch/x86/boot/bzImage is ready (#1)
> ERROR: "snd_tea575x_init" [drivers/media/radio/radio-maxiradio.ko] undefined!
> ERROR: "snd_tea575x_exit" [drivers/media/radio/radio-maxiradio.ko] undefined!
> WARNING: modpost: Found 6 section mismatch(es).
> To see full details build your kernel with:
> 'make CONFIG_DEBUG_SECTION_MISMATCH=y'
> make[1]: *** [__modpost] Error 1
> make: *** [modules] Error 2
>
> Select CONFIG_SND_TEA575X to fixes problem and enable
> the driver to be built as desired.
>
> v2:
> instead of selecting CONFIG_SND_FM801_TEA575X_BOOL, select
> CONFIG_SND_TEA575X, which in turns selects CONFIG_SND_FM801_TEA575X_BOOL
> and any other dependencies for it to build.
No, this is the correct patch:
RADIO_MAXIRADIO should be treated just like RADIO_SF16FMR2, I just didn't
realize at the time that it had to be added as a SND_TEA575X dependency.
Signed-off-by: Hans Verkuil <hans.verkuil@cisco.com>
Tested-by: Shea Levy <shea@shealevy.com>
Acked-by: Mauro Carvalho Chehab <mchehab@infradead.org>
Signed-off-by: Mauro Carvalho Chehab <mchehab@redhat.com>
Rather than adding new arguments to regulator_register() every time we
want to add a new bit of dynamic information at runtime change the function
to take these via a struct. By doing this we avoid needing to do further
changes like the recent addition of device tree support which required each
regulator driver to be updated to take an additional parameter.
The regulator_desc which should (mostly) be static data is still passed
separately as most drivers are able to configure this statically at build
time.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Merge tag 'v3.4-rc2' into for-3.5
Linux 3.4-rc2 contains some bug fixes we need, including the addition of
an export for regcache_sync_region().
Merge with latest Linus' tree, as I have incoming patches
that fix code that is newer than current HEAD of for-next.
Conflicts:
drivers/net/ethernet/realtek/r8169.c
A bunch of driver-specific fixes and one generic fix for the new support
for platform DAPM contexts - we were picking the wrong default for the
idle_bias_off setting which was meaning we weren't actually achieving
any useful runtime PM on platform devices.
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Merge tag 'asoc-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: fixes for 3.4
A bunch of driver-specific fixes and one generic fix for the new support
for platform DAPM contexts - we were picking the wrong default for the
idle_bias_off setting which was meaning we weren't actually achieving
any useful runtime PM on platform devices.
name pins consistently (MIC1/LINE1/HP-OUT/CD) on all controls
affecting those pins.
remove duplicate SET_AMP_GAIN_MUTE to 0x17/index 0 and 0x17/index 1
really select MIC1, not Mixer out for recording
"Mixer out" for recording is not a "pin", adjust comment
Signed-off-by: Michael Karcher <kernel@mkarcher.dialup.fu-berlin.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
CX20549 (ctx5045) doesn't accept data on index 1 for output pins,
as shown in the following hda-var transaction:
$ hda-verb /dev/snd/hwC0D0 0x10 set_amp_gain 0xb126
nid = 0x10, verb = 0x300, param = 0xb126
value = 0x0
$ hda-verb /dev/snd/hwC0D0 0x10 get_amp_gain 0x8001
nid = 0x10, verb = 0xb00, param = 0x8001
value = 0x0
Signed-off-by: Michael Karcher <kernel@mkarcher.dialup.fu-berlin.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The ID used for detection of the BenQ R55E actually identifies the
Quanta TW3 ODM design, which is also used for the Gigabyte W551 laptop
series. Schematics on the internet clearly indicate that the "Port C"
(analog input connected to record source #4 and mixer input #4) is
unconnected.
Playing an audio CD through analog playback (using cdplay from cdtools)
produces no sound, even with the mixer input labelled "CD" enabled, and
the volume control in the CD drive set to maximum. This indicates the
connection is really not present.
Signed-off-by: Michael Karcher <kernel@mkarcher.dialup.fu-berlin.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The "input converter" widget of the CX20459 has only one input amplifier,
expose that one as "Capture Volume/Capture Switch". The actual record
source selection is already exposed through the separately installed
input mux.
Signed-off-by: Michael Karcher <kernel@mkarcher.dialup.fu-berlin.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This includes renaming "Line In" to line, also in the mixer settings.
Signed-off-by: Michael Karcher <kernel@mkarcher.dialup.fu-berlin.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The CX20549 has only one single input amp on it's input converter
widget. Fix printing of values in the codec file in /proc/asound.
Signed-off-by: Michael Karcher <kernel@mkarcher.dialup.fu-berlin.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Tegra30 has some additional clocks that need to be manipulated, names
some clocks differently, runs PLLs at different base rates, etc. The
utility code needs to handle this.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Initialize the audio clock tree appropriately for some reasonable rate.
This makes sure the PLLs etc. are actually programmed to something
reasonable when the audio driver is loaded.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The devm_ APIs remove the need to manually clean up allocations,
thus removing some code.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
They were harmless but also unnecessary, probably a leftover from earlier code.
Signed-off-by: Maarten ter Huurne <maarten@treewalker.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Rename Tegra20-specific Kconfig variables, module filenames, all internal
symbol names, clocks, and platform devices, to reflect the fact the DAS
and I2S drivers are for a specific HW version.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Rename these files so they include a specific hardware version in their
filenames. The contents is only touched minimally so that git's rename
tracking operates correctly; renaming all symbols in the files results
in a diff so large that the rename detection fails.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Commit d4a2eca "ASoC: Tegra I2S: Remove dependency on pdev->id" changed
the prototype of tegra_i2s_debug_add, but didn't update the dummy inline
used when !CONFIG_DEBUG_FS. Fix that.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: <stable@vger.kernel.org> # 3.3
Merge batch of fixes from Andrew Morton:
"The simple_open() cleanup was held back while I wanted for laggards to
merge things.
I still need to send a few checkpoint/restore patches. I've been
wobbly about merging them because I'm wobbly about the overall
prospects for success of the project. But after speaking with Pavel
at the LSF conference, it sounds like they're further toward
completion than I feared - apparently davem is at the "has stopped
complaining" stage regarding the net changes. So I need to go back
and re-review those patchs and their (lengthy) discussion."
* emailed from Andrew Morton <akpm@linux-foundation.org>: (16 patches)
memcg swap: use mem_cgroup_uncharge_swap fix
backlight: add driver for DA9052/53 PMIC v1
C6X: use set_current_blocked() and block_sigmask()
MAINTAINERS: add entry for sparse checker
MAINTAINERS: fix REMOTEPROC F: typo
alpha: use set_current_blocked() and block_sigmask()
simple_open: automatically convert to simple_open()
scripts/coccinelle/api/simple_open.cocci: semantic patch for simple_open()
libfs: add simple_open()
hugetlbfs: remove unregister_filesystem() when initializing module
drivers/rtc/rtc-88pm860x.c: fix rtc irq enable callback
fs/xattr.c:setxattr(): improve handling of allocation failures
fs/xattr.c:listxattr(): fall back to vmalloc() if kmalloc() failed
fs/xattr.c: suppress page allocation failure warnings from sys_listxattr()
sysrq: use SEND_SIG_FORCED instead of force_sig()
proc: fix mount -t proc -o AAA
Many users of debugfs copy the implementation of default_open() when
they want to support a custom read/write function op. This leads to a
proliferation of the default_open() implementation across the entire
tree.
Now that the common implementation has been consolidated into libfs we
can replace all the users of this function with simple_open().
This replacement was done with the following semantic patch:
<smpl>
@ open @
identifier open_f != simple_open;
identifier i, f;
@@
-int open_f(struct inode *i, struct file *f)
-{
(
-if (i->i_private)
-f->private_data = i->i_private;
|
-f->private_data = i->i_private;
)
-return 0;
-}
@ has_open depends on open @
identifier fops;
identifier open.open_f;
@@
struct file_operations fops = {
...
-.open = open_f,
+.open = simple_open,
...
};
</smpl>
[akpm@linux-foundation.org: checkpatch fixes]
Signed-off-by: Stephen Boyd <sboyd@codeaurora.org>
Cc: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Cc: Al Viro <viro@zeniv.linux.org.uk>
Cc: Julia Lawall <Julia.Lawall@lip6.fr>
Acked-by: Ingo Molnar <mingo@elte.hu>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
sound/soc/codecs/lm49453.c: In function 'lm49453_set_dai_fmt':
sound/soc/codecs/lm49453.c:1189:4: warning: overflow in implicit
constant conversion [-Woverflow]
sound/soc/codecs/lm49453.c:1193:4: warning: overflow in implicit
constant conversion [-Woverflow]
sound/soc/codecs/lm49453.c:1197:4: warning: overflow in implicit
constant conversion [-Woverflow]
Reported-by: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: M R Swami Reddy <mr.swami.reddy@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Without this, the Tegra20 drivers can be built into a kernel that's
built only for Tegra30.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The DAS, I2S, and SPDIF Kconfig options are intended to be selected by
the Kconfig options for ASoC machine drivers. As such, they don't need
to be user-visible themselves. Drop the description from the DAS variable
to achieve this. I2S and SPDIF already don't have a description.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The ASoC core currently defaults to using STANDBY rather than OFF for
idle ASoC platform devices, which causes a permanent pm_runtime_get() on
them. This keeps the device active unnecessarily. This can be especially
problematic when the ASoC platform device and DAI device are the same
device.
The distinction between OFF and STANDBY is likely not relevant for ASoC
platform drivers, since they aren't analog devices. So, solve this issue
by hard-coding idle_bias_off = 1 for all ASoC platform devices. If this
turns out to be a problem, this value could be sourced from the
snd_soc_platform_driver, similarly to soc_probe_codec().
Note: Prior to this change, this caused a large (10) runtime_active count
for the Tegra I2S controller even when not in use, and a leak in that
value as streams were started and stopped. This change probably hides a
bug.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The internal mic input is phase inverted on one channel.
To avoid people in userspace summing the channels together
and get zero result, use a separate mixer control for the
inverted channel.
BugLink: https://bugs.launchpad.net/bugs/903853
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Check for NULL pointer before accessing it.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ak4642 out_tlv is +12.0dB to -115.0 dB, and it supports mute.
But current settings didn't care +1 step for mute.
This patch adds it
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
In order to support CODEC<->CODEC links remove the assumption that there
is only a single CODEC on a DAI link by removing the use of the CODEC
pointer in the rtd from the CODEC drivers. They are already being passed
their DAI whenever they are passed an rtd and can get the CODEC from
there.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Jarkko Nikula