I notice that the fixes were merged, minus one:
sound/soc/codecs/wm9705.c: At top level:
sound/soc/codecs/wm9705.c:445: warning: initialization from incompatible pointer type
so you might find this trivial patch useful.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The original implementation of the constraints were good against sane
applications.
If the opening sequence is:
stream1_open, stream1_hw_params, stream2_open, stream2_hw_params -> the
constraints are set correctly for stream2.
But if the sequence is:
stream1_open, stream2_open, stream2_hw_params, stream1_hw_params -> than stream2
would receive constraint rate = 0, sample_bits = 0, since the stream1 has not
yet called hw_params...
The command to trigger this event:
gst-launch-0.10 alsasrc device=hw:0 ! alsasink device=hw:0 sync=false
This patch does some 'black magic' in order to always set the correct
constraints and sets it only when it is needed for the other stream.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
My email address is going to expire soon so update it. Adding also
Peter Ujfalusi <peter.ujfalusi@nokia.com> as a second contact to OMAP core
drivers since I won't have anymore access to non-public OMAP documentation
in the future and Peter is working with these drivers as well.
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Cc: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Those macros are just screwed as soon as CONFIG_PXA25x is enabled.
This patch
- changes ssp_set_scr to take an ssp_dev pointer instead of ssp_device
- adds a corresponding ssp_get_scr function.
Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
DSP_A mode is similar to the DSP_B, but the MSB is delayed with
one bclk (appears after the FS pulse and not under it).
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use single-phase mode for the DSP mode and keep the dual phase
mode for the I2S mode.
The mono (1 channel) mode already used single phase mode,
now it is more cleaner. There is no need to configure the
second phase, when the single phase is used.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Using inverted FS polarity in OSK5912 must be an error since TLV320AIC23
do not have support for inverted polarities. This is mostly due the hassle
with the DSP formats in OMAP McBSP DAI and inversion on OMAP side probably
just made this configuration working at some point.
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Acked-by: Arun KS <arunks@mistralsolutions.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The DSP format wasn't still correct in OMAP McBSP DAI even after the commit
bd25867a6c.
Thanks to Peter Ujfalusi <peter.ujfalusi@nokia.com> for noticing and being
part of the fix. Now the FS length definition is more clear by defining
it with FWID(0).
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix accidental change of <mach/regs-gpio.h> to
<plat/regs-gpio.h> in s3c2412-i2s.c
Signed-off-by: Ben Dooks <ben-linux@fluff.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix the build error in s3c-i2s-v2.c caused by
a change to the snd_soc_dai ops field.
Signed-off-by: Ben Dooks <ben-linux@fluff.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The definition of s3c_i2sv2_iis_calc_rate was never
renamed from s3c2412_iis_calc_rate, so rename this
to allow the build to work.
Signed-off-by: Ben Dooks <ben-linux@fluff.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix build errors in sound/soc/s3c24xx/jive_wm8750.c
from changes to ASoC.
Signed-off-by: Ben Dooks <ben-linux@fluff.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
pxa_ssp_set_dai_fmt() currently has an early exit if the desired format
equals the current configuration. This is correct behaviour unless this
function is called with a zero value parameter for the first time.
Zero is a valid value for this function, but the early exit is bogus in
this case.
Hence, set priv->dai_fmt to -1 in the beginning so we can configure the
port.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Cc: pHilipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some limited volume controls (mostly simple attenuations) have only two
settings so the ASoC info functions misreport them as booleans. Since
we currently have no better information check for " Volume" in the
control name and always report any controls matching as being integer.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM8960 is a low power, high quality stereo codec designed for
portable digital audio applications.
Stereo class D speaker drivers provide 1W per channel into 8W loads.
Guaranteed low leakage, excellent PSRR and pop/click suppression
mechanisms enable direct battery connection for the speaker supply.
The device also integrates a complete microphone interface and a stereo
headphone driver. External component requirements are drastically
reduced as no separate microphone, speaker or headphone amplifiers are
required. Advanced on-chip digital signal processing performs automatic
level control for the microphone or line input.
Stereo 24-bit sigma-delta ADCs and DACs are used with low power
over-sampling digital interpolation and decimation filters and a
flexible digital audio interface.
The master clock can be input directly or generated internally by an
onboard PLL, supporting most commonly-used clocking schemes.
This driver was originally written by Liam Girdwood, with substantial
subsequent additions and updates for feature completeness and changes in
the ASoC framework from me.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
SCMODE(0): Data Driven (Falling), Data Sampled (Rising), Idle State (Low)
SCMODE(1): Data Driven (Rising), Data Sampled (Falling), Idle State (Low)
SCMODE(2): Data Driven (Rising), Data Sampled (Falling), Idle State (High)
SCMODE(3): Data Driven (Falling), Data Sampled (Rising), Idle State (High)
SCMODE(3) does not invert the clock polarity compared to the default SCMODE(0).
This patch also adds all possible NF/IF, NB/IB combinations to the DSP_A and
DSP_B modes.
Signed-off-by: Daniel Ribeiro <drwyrm@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This ensures that we sync with the DAPM powerdown sequencing properly
and don't need to bounce the power on the voice DAC so often.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is simple code motion, intended to support future refactoring of
the DAPM algorithms and (more immediately) the additon of events for
DACs and ADCs.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Due to the process and communications issues with the 2.6.30 S3C
platform merges none of the underlying arch/arm code for S3C64xx audio
support made it into mainline, rendering the drivers useless. Disable
them in Kconfig to avoid user confusion - users patching in the required
support can always reenable this too.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add DSP_A interface format support by setting the LRP bit in
DSP mode.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Eric Miao <eric.miao@marvell.com>
Cc: Philipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM8988 is a low power, high quality stereo CODEC designed for
portable digital audio applications.
The device integrates complete interfaces to 2 stereo headphone or line
out ports. External component requirements are drastically reduced as no
separate headphone amplifiers are required. Advanced on-chip digital
signal processing performs graphic equaliser, 3-D sound enhancement and
automatic level control for the microphone or line input.
The WM8988 can operate as a master or a slave, with various master clock
frequencies including 12 or 24MHz for USB devices, or standard 256fs
rates like 12.288MHz and 24.576MHz. Different audio sample rates such as
96kHz, 48kHz, 44.1kHz are generated directly from the master clock
without the need for an external PLL.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Many devices require symmetric configurations of capture and playback
data formats, often due to shared clocking but sometimes also due to
other shared playback and record configuration in the device. Start
providing core support for this by allowing the DAIs or the machine
to specify that the sample rates used should be kept symmetric.
A flag symmetric_rates is provided in the snd_soc_dai and
snd_soc_dai_link structures. If this is set in either of the DAIs or in
the machine then a constraint will be applied when a stream is already
open preventing any changes in sample rate.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (36 commits)
ALSA: hda - Add VREF powerdown sequence for another board
ALSA: oss - volume control for CSWITCH and CROUTE
ALSA: hda - add missing comma in ad1884_slave_vols
sound: usb-audio: allow period sizes less than 1 ms
sound: usb-audio: save data packet interval in audioformat structure
sound: usb-audio: remove check_hw_params_convention()
sound: usb-audio: show sample format width in proc file
ASoC: fsl_dma: Pass the proper device for dma mapping routines
ASoC: Fix null dereference in ak4535_remove()
ALSA: hda - enable SPDIF output for Intel DX58SO board
ALSA: snd-atmel-abdac: increase periods_min to 6 instead of 4
ALSA: snd-atmel-abdac: replace bus_id with dev_name()
ALSA: snd-atmel-ac97c: replace bus_id with dev_name()
ALSA: snd-atmel-ac97c: cleanup registers when removing driver
ALSA: snd-atmel-ac97c: do a proper reset of the external codec
ALSA: snd-atmel-ac97c: enable interrupts to catch events for error reporting
ALSA: snd-atmel-ac97c: set correct size for buffer hardware parameter
ALSA: snd-atmel-ac97c: do not overwrite OCA and ICA when assigning channels
ALSA: snd-atmel-ac97c: remove dead break statements after return in switch case
ALSA: snd-atmel-ac97c: cleanup register definitions
...
Replace all DMA_32BIT_MASK macro with DMA_BIT_MASK(32)
Signed-off-by: Yang Hongyang<yanghy@cn.fujitsu.com>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
Fix for compillation error introduced by the constrain patch.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The driver should pass a device that specifies internal DMA ops, but
substream->pcm is just a logical device, and thus doesn't have arch-
specific dma callbacks, therefore following bug appears:
Freescale Synchronous Serial Interface (SSI) ASoC Driver
------------[ cut here ]------------
kernel BUG at arch/powerpc/include/asm/dma-mapping.h:237!
Oops: Exception in kernel mode, sig: 5 [#1]
...
NIP [c02259c4] snd_malloc_dev_pages+0x58/0xac
LR [c0225c74] snd_dma_alloc_pages+0xf8/0x108
Call Trace:
[df02bde0] [df02be2c] 0xdf02be2c (unreliable)
[df02bdf0] [c0225c74] snd_dma_alloc_pages+0xf8/0x108
[df02be10] [c023a100] fsl_dma_new+0x68/0x124
[df02be20] [c02342ac] soc_new_pcm+0x1bc/0x234
[df02bea0] [c02343dc] snd_soc_new_pcms+0xb8/0x148
[df02bed0] [c023824c] cs4270_probe+0x34/0x124
[df02bef0] [c0232fe8] snd_soc_instantiate_card+0x1a4/0x2f4
[df02bf20] [c0233164] snd_soc_instantiate_cards+0x2c/0x68
[df02bf30] [c0234704] snd_soc_register_platform+0x60/0x80
[df02bf50] [c03d5664] fsl_soc_platform_init+0x18/0x28
...
This patch fixes the issue by using card's device instead.
Signed-off-by: Anton Vorontsov <avorontsov@ru.mvista.com>
Acked-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
According to the data sheet data is clocked out on the falling edge
and latched on the rising edge of the bit clock. While the left sample
is transmitted the word clock line is low.
Signed-off-by: Daniel Glöckner <dg@emlix.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ak4535_remove() from sound/soc/codecs/ak4535.c calls
i2c_unregister_device() with a possibly null pointer.
This bug was found by smatch (http://repo.or.cz/w/smatch.git/).
Signed-off-by: Dan Carpenter <error27@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds machine specific code for the audio part of the Stretch
s6105 IP camera reference design.
The device uses the tlv320aic31(01) codec to generate the clock for
both I2S ports of the soc. While the master clock is generated by a
configurable PLL chip, the code assumes the factory default settings.
An additional kcontrol has been added to handle the special routing of
the board, connecting both HPLCOM and HPROUT to the same pin of the audio
jack. One of these should always be switched off.
Signed-off-by: Daniel Glöckner <dg@emlix.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds a driver for the I2S interface found on Stretch s6000
family processors.
Signed-off-by: Daniel Glöckner <dg@emlix.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Adds the needed code to be able to use 96KHz playback.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Without this the WM9705 driver fails badly when resuming.
Tested-by: Russell King <linux@arm.linux.org.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Ensure that any AC97 devices that bind to the CODEC are below the
ASoC device in the device tree so the suspend and resume code can
figure out what order to handle them in.
Reported-by: Russell King <linux@arm.linux.org.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
AC97 devices may have other drivers hanging off them directly so need to
have resumed when the resume function returns meaning that we can't defer
the resume - complete it immediately for them. Non-AC97 devices should
not have other drivers hanging directly off the ASoC devices.
We only really need the deferral for non-AC97 devices - it's there since
some I2C buses are very slow and non-AC97 codecs often have large numbers
of registers to restore and require delays to bring the codec up cleanly
leading to a substantial impact on overall resume time.
Reported-by: Russell King <linux@arm.linux.org.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
McBSP2 in OMAP3 has 1 ksample (1k x 32 bit) internal FIFO. During
initial playback startup, this FIFO is keeping the DMA request active
until the FIFO is full.
So now if ALSA buffer size is smaller, DMA is looping around it while
filling up the HW FIFO, generating burst of interrupts as well and SW
doesn't have any change to fill enough data.
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In case of duplex mode (capture and playback at the same time), the second
stream has to have the same parameters (rate, sample size) as the already
running stream.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
TWL4030 supports 96KHz sample playback, but only playback.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Optimize the display of SSI statistics in the Freescale MPC8610 sound driver
to display the status count only of the interrupts that were actually enabled.
Previously, it would display the counts of all SISR status bits, even those
that were not enabled.
Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Remove the delay from the trigger function in the Freescale MPC8610 sound
driver when capture is started. This delay was used to ensure that the DMA
controller was active when ALSA call the .pointer function to request a
DMA transfer status. A better approach is for the .pointer function to detect
that DMA has not started, and return zero instead. This change eliminates
the need for the delay.
Also add some related code to check for a DMA programming error, and report
XRUN if it occurs.
Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
HTC Magician has a Philips UDA1380 codec connected via
SSP1 (playback) and I2S (capture).
There is a flip-flop between the SSP frame clock output
and the codec's word select input pin. To make the codec
see proper I2S input, the SSP has to send two frames per
sample.
Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Now magician and similar boards can use network mode with only one
active slot to explicitly set 16 bit frame width, even for S16_LE
stereo sound.
Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Headset was declared previously as a Headphone widget connecting
HSMIC and HSOL/HSOR pins of TWL4030 codec in SDP430 machine driver.
The capture path becomes invalid as the Headphone widget is not a
valid input endpoint.
Instead of that, the Headset is declared as separate Microphone
and Headphone widgets. Current patch modifies audio map:
- Headset Mic: HSMIC with bias
- Headset Stereophone: HSOL, HSOR
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add functions "Headset" and "Mic" to the control "Jack Function" for
activating and de-activating codec input pin LINE1L which is connected to
the mic pin of 4-pole Nokia AV connecter.
Note there is no mic bias voltage management here since bias is coming from
Nokia ASIC and driver for it is not in mainline.
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There is an AVDD supply as well, normally one or more of the other
upplies would be tied to it.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The active discharge does not bring sufficient benefit to justify the
lengthy times involved so don't do that.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The commit 14fa43f53f ("ASoC: Only
register AC97 bus if it's not done already") added a condition for
calling of soc_ac97_dev_register() but not added for calling of
soc_ac97_dev_unregister(). This patch adds same condition for
soc_ac97_dev_unregister(). Without this fix, kernel crashes when
unloading an asoc driver.
Signed-off-by: Atsushi Nemoto <anemo@mba.ocn.ne.jp>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
CC sound/soc/codecs/twl4030.o
sound/soc/codecs/twl4030.c:1400: warning: braces around scalar initializer
sound/soc/codecs/twl4030.c:1400: warning: (near initialization for 'twl4030_dai.ops')
sound/soc/codecs/twl4030.c:1401: error: field name not in record or union initializer
sound/soc/codecs/twl4030.c:1401: error: (near initialization for 'twl4030_dai.ops')
sound/soc/codecs/twl4030.c:1401: warning: initialization from incompatible pointer type
sound/soc/codecs/twl4030.c:1402: error: field name not in record or union initializer
sound/soc/codecs/twl4030.c:1402: error: (near initialization for 'twl4030_dai.ops')
sound/soc/codecs/twl4030.c:1402: warning: excess elements in scalar initializer
sound/soc/codecs/twl4030.c:1402: warning: (near initialization for 'twl4030_dai.ops')
sound/soc/codecs/twl4030.c:1403: error: field name not in record or union initializer
sound/soc/codecs/twl4030.c:1403: error: (near initialization for 'twl4030_dai.ops')
sound/soc/codecs/twl4030.c:1403: warning: excess elements in scalar initializer
sound/soc/codecs/twl4030.c:1403: warning: (near initialization for 'twl4030_dai.ops')
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Two issues are fixed here:
- I2S transmits the left frame with the clock low but I don't seem to
get LRCLK out without SFRMDLY being set so invert SFRMP and set a
delay.
- I2S has a clock cycle prior to the first data byte in each channel
so we need to delay the data by one cycle.
Tested-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This switches the pxa ssp port usage from network mode to PSP mode.
Removed some comments and checks for configured TDM channels.
A special case is added to support configuration where BCLK = 64fs. We
need to do some black magic in this case which doesn't look nice but
there is unfortunately no other option than that.
Diagnosed-by: Tim Ruetz <tim@caiaq.de>
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Move headset jack registration to the codec/machine specific
initialization. Having the jack registration in machine init
causes that the jack device gets initialized but not registered
since the sound card is registered before the jack. Moving jack
registration to device initialization will register the jack
device along with all other devices associated to the card when
the card is registed. As a consequence of jack device registered
properly, the jack is detected as an input device.
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The drivers are basically duplicating the same code over and over.
As snd_soc_cnew is going to be made static some time after the next
merge window, we might as well convert them now.
Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Refactor the WM8580 device registration to probe via standard I2C device
registration, registering the DAIs once the device has probed via I2C.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Merge Eric Maio's patch to merge snd_soc_dai_ops out of line. Fixed
merge issues and updated drivers, plus an issue with the ops for the two
s3c2443 AC97 DAIs having been merged.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM8400 is a highly integrated audio CODEC and power management unit
intended for mobile multimedia application. This driver supports the
primary audio CODEC features, including:
- 1W speaker driver
- Fully differential headphone output
- Up to 4 differential microphone inputs
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Buildfix:
CC sound/soc/omap/osk5912.o
sound/soc/omap/osk5912.c: In function 'osk_soc_init':
sound/soc/omap/osk5912.c:189: error: implicit declaration of function 'clk_get_usecount'
make[3]: *** [sound/soc/omap/osk5912.o] Error 1
There's no such (standard) clock interface.
Signed-off-by: David Brownell <dbrownell@users.sourceforge.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In pxa_ssp_set_dai_fmt(), check whether there is anything to do at all.
If there would be but the SSP port is in use already, bail out.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This will break any boards that don't register the AC97 controller
device due to using ASoC.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
pxa-regs.h and hardware.h are not intended for use directly in driver
code, remove those unnecessary references.
Signed-off-by: Eric Miao <eric.miao@marvell.com>
1. Driver code where pxa_request_dma() is called will most likely
reference DMA registers as well, and it is really unnecessary
to include pxa-regs.h just for this. Move the definitions into
<mach/dma.h> and make relevant drivers include it instead of
<mach/pxa-regs.h>.
2. Introduce DMAC_REGS_VIRT as the virtual address base for these
DMA registers. This allows later processors to re-use the same
IP while registers may start at different I/O address.
Signed-off-by: Eric Miao <eric.miao@marvell.com>
This adds a driver for the SPI connected AK4104 S/PDIF transmitter
device. Its features are fairly simple, but as there is need to set up
certain bits in the IEC958 information, this better goes into a real
driver.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Cc: Mark Brown <broonie@sirena.org.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Removes numbers from the list of features/limitations and makes it
reflect recent changes to the code.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Acked-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add support for true pause and unpause. Without this, mplayer will drop some
audio (less than one second, but still noticeable) when pausing playback.
Remove support for PM suspend and resume from the trigger function, since the
driver doesn't support PM anyway.
Optimize the delay after starting capture. Instead of delaying 1ms, the driver
now polls the hardware. The new delay is shorter by over 90% yet still
effective.
Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Upgrade the severity of some failure messages from debug level so
they're displayed by default.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The recent set of S3C64xx patches re-added a lot of uses of DBG() that
had previously been removed - revert this so the standard pr_debug()
macro is used.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Reported-by: Rob Maris <maris.rob@vdi.de>
Signed-off-by: Mike Frysinger <vapier.adi@gmail.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Mike Frysinger <vapier.adi@gmail.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add headset jack detection for SDP3430 boards using SoC jack
reporting interface. Headset detection on SDP3430 board is
achieved through TWL4030 GPIO_2 pin.
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add a new device tree property for the SSI node: "fsl,ssi-asynchronous". If
defined, the SSI is programmed into asynchronous mode, otherwise it is
programmed into synchronous mode. In asynchronous mode, pin SRCK must be
connected to the same clock source as STFS, and pin SRFS must be connected to
the same signal as STFS. Asynchronous mode allows playback and capture to
use different sample sizes. It also technically allows different sample rates,
but the driver does not support that.
Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We now support the 64xx series as well as the 24xx series - make sure
people using Kconfig know this.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
snd_soc_dapm_switch ends up ends up in dapm_new_mixer() (since a switch
is a special case of a mixer with only one input) but this wasn't
correctly handled in the code.
Also fix the coding style for the switch below while we're here.
Reported-by: Joonyoung Shim <dofmind@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
A bit in PXA's SSCR0 register was erroneously named ADC but its name is
in fact ACS (audio clock select).
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Enum type for selecting the desired ramp delay for the headset output.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Select the relevant DMA implementation when the
sound driver is selected.
Signed-off-by: Ben Dooks <ben@simtec.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add the initial code to support the S3C64XX I2S hardware using the
s3c-i2s-v2 core code.
Signed-off-by: Ben Dooks <ben@simtec.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The S3C2412 I2S (IIS) interface is replicated on further Samsung SoC
parts in a broadly compatible way, so split the common code out into
a core called s3c-i2s-v2.[ch] so that the newer SoCs such as the
S3C6410 can make use of it.
As such, all the original s3c2412 functions are currently being left
with their original names, and will be renamed later in the series.
Signed-off-by: Ben Dooks <ben@simtec.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add support for the Jive's WM8750 codec attached via the S3C2412 IIS.
Signed-off-by: Ben Dooks <ben@simtec.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The <mach/audio.h> file needs to be common to both ARCH_S3C2410 and
ARCH_S3C64XX as they share common driver code, so move it to <plat/audio.h>.
Signed-off-by: Ben Dooks <ben@simtec.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Move the IIS headers to their correct place.
Signed-off-by: Ben Dooks <ben@simtec.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Considering the fact that most cpu_dai or codec_dai are using a same
'snd_soc_dai_ops' for several similar interfaces, 'ops' would be better
made a pointer instead, to make sharing easier and code a bit cleaner.
The patch below is rather preliminary since the asoc tree is being
actively developed, and this touches almost every piece of code,
(and possibly many others in development need to be changed as
well). Building of all codecs are OK, yet to every SoC, I didn't test
that.
Signed-off-by: Eric Miao <eric.miao@marvell.com>
Acked-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When setting WM8510_MCLKDIV the pll was turned off.
When setting pll frequency you got twice the expected freq, because
the code calculated with postscaler of 8, but the hardware divide by 4.
Signed-off-by: Jonas Andersson <jonas@microbit.se>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add GPIO support to jack reporting framework in ASoC using gpiolib calls.
The gpio support exports two new functions: snd_soc_jack_add_gpios and
snd_soc_jack_free_gpios.
Client drivers using gpio feature must pass an array of jack_gpio pins
belonging to a specific jack to the snd_soc_jack_add_gpios function. The
framework will request the gpios, set the data direction and request irq.
The framework will update power status of related jack_pins when an event on
the gpio pins comes according to the reporting bits defined for each gpio.
All gpio resources allocated when adding jack_gpio pins can be released
using snd_soc_jack_free_gpios function.
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Only allow SND_SOC_DAIFMT_CBS_CFS for the playback DAI.
Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
For consistency with 24-bit and 32-bit modes, don't send 16-bit stereo
in one 32-bit transfer. Use 2 slots instead on Zylonite. It should result
in exactly the same behaviour.
Now it is possible to use 16-bit single slot transfers in pxa-ssp, which
are needed for Magician to get two frame clock pulses per sample
(one for each channel).
Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Tested-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If the UDA1380's interpolator or decimator are set to be clocked from
the WSPLL (which syncs to the WSI signal), the DAI link must be running
to change the interpolator/decimator registers (which include volume
controls and digital mute setting).
* Queue work in the alsa PCM_START .trigger to flush registers
as soon as the link is running. This replaces the .prepare
and .digital_mute callbacks.
* Use the SILENCE override instead of MTM for muting and remove
its alsa control to avoid confusion.
Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This removes a misspelled comment and got rid of superfluous switch
case.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Acked-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
On some systems it is desirable for control for DAPM pins to be provided
to user space. This is the case with things like GSM modems which are
controlled primarily from user space, for example. Provide a helper which
exposes the state of a DAPM pin to user space for use in cases like this.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Added support for scenarios where the Cirrus CS4270 audio codec is slave
to the bitclk and lrclk. Mixed setups are unsupported.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix the copyright statements in two of the S3C24XX ASoC files
that have (c) when we require the full word.
Signed-off-by: Ben Dooks <ben@simtec.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
sound/soc/codecs/wm8753.c: In function 'wm8753_probe':
sound/soc/codecs/wm8753.c:1577: error: implicit declaration of function 'wm8753_add_controls'
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This will reduce the number of writes done on resume, allowing that to
complete faster (especially on systems with very slow I2C like the
current Samsung driver).
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The base support for the only in-tree user, the GTA01, is out of tree
and will be updated separately.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch should be pure code motion, separating that out from the
functional changes to move to new style device registration.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This avoids temporarily enabling the ouput stages during startup which
can cause audible effets in the output stages.
Reported-by: Fredrik Redgård <rik@svep.se>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds the digital loopback/bypass support for twl4030 codec.
The digital loopback will let the digimic0 (routed in the TX1 capture path
inside of TWL4030) data to be routed back to the RX2 playback path
(I2S stereo). It can also route the analog capture date routed through the
TX1 back to RX2.
Effectively the digital loopback is routing the audio from the TX1 capture path
to the RX2 playback path.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Move the WM8731 driver to use a more standard device registration
scheme where the device can be registered independantly of the ASoC
probe.
As a transition measure push the current manual code for registering
the WM8731 into the individual machine driver probes. This allows
separate patches to update the relevant architecture files with less
risk of merge issues.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is a pure code motion patch intended to improve reviewability of a
following patch moving WM8731 to use more standard device registration.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is a bit more idiomatic and makes identifying a configuration
based on the board type work better.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We have software control of the MCLK for the WM8731 so save a bit of
power by actively managing it within the machine driver, enabling it
only while the codec is active.
Once ASoC supports multiple boards and doesn't require the soc-audio
device the initial clock setup should be pushed down into the arch/arm
code but for now this reduces merge issues.
Tested-by: Sedji Gaouaou <sedji.gaouaou@atmel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Takashi Iwai