After DMA burst mode has been introduced in sound/soc/omap/omap-pcm.c,
omap_pcm_prepare() unconditionally calls:
omap_set_dma_src_burst_mode(prtd->dma_ch, OMAP_DMA_DATA_BURST_16);
omap_set_dma_dest_burst_mode(prtd->dma_ch, OMAP_DMA_DATA_BURST_16);
Current implementation of those two functions found in
arch/arm/plat-ompa/dma.c doesn't support OMAP_DMA_DATA_BURST_16 on OMAP1 at
all, so they both end with BUG() on that machine. That results in
ASoC being completely unusable, at least on my OMAP5910 based Amstrad Delta.
The patch corrects the problem by not calling those two functions when run on
OMAP1 class based machines.
Created against linux-2.6.32-rc5.
Tested on Amstrad Delta.
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix the definition of DAC33_LTM field, the LTM bits in
FIFO_IRQ_MODE_B register are starting at bit 6.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Hi Mark,
Here is a patch that corrects small omissions I have found in my code.
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Move the remaining headers under plat-omap/include/mach
to plat-omap/include/plat. Also search and replace the
files using these headers to include using the right path.
This was done with:
#!/bin/bash
mach_dir_old="arch/arm/plat-omap/include/mach"
plat_dir_new="arch/arm/plat-omap/include/plat"
headers=$(cd $mach_dir_old && ls *.h)
omap_dirs="arch/arm/*omap*/ \
drivers/video/omap \
sound/soc/omap"
other_files="drivers/leds/leds-ams-delta.c \
drivers/mfd/menelaus.c \
drivers/mfd/twl4030-core.c \
drivers/mtd/nand/ams-delta.c"
for header in $headers; do
old="#include <mach\/$header"
new="#include <plat\/$header"
for dir in $omap_dirs; do
find $dir -type f -name \*.[chS] | \
xargs sed -i "s/$old/$new/"
done
find drivers/ -type f -name \*omap*.[chS] | \
xargs sed -i "s/$old/$new/"
for file in $other_files; do
sed -i "s/$old/$new/" $file
done
done
for header in $(ls $mach_dir_old/*.h); do
git mv $header $plat_dir_new/
done
Signed-off-by: Tony Lindgren <tony@atomide.com>
Some codec DAIs like stac9766, wm9712, wm9713, ad1980 don't register themselves
then it loses to the chance to be given a null_dai_ops in snd_soc_register_dai
if they have no ops. When functions like soc_pcm_open, soc_pcm_hw_params etc.
access the ops field in these DAIs, panic will happen.
Signed-off-by: Barry Song <21cnbao@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If the NULL test on jack is needed, then the derefernce should be after the
NULL test.
A simplified version of the semantic match that detects this problem is as
follows (http://coccinelle.lip6.fr/):
// <smpl>
@match exists@
expression x, E;
identifier fld;
@@
* x->fld
... when != \(x = E\|&x\)
* x == NULL
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Codec read/write functions: wait 21us between the pokings of hardware.
Add timeouts to unbounded loops waiting for bits to change.
Signed-off-by: Manuel Lauss <manuel.lauss@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Verify that the correct register has been received from the codec.
Signed-off-by: Manuel Lauss <manuel.lauss@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Do not rewrite the whole register, but only update the needed
bits in set_dai_sysclk functions.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Driver for Texas Instruments TLV320DAC33 (SLAS546) low power stereo
audio DAC.
TLV320DAC33 is a stereo audio codec with integrated 24KB FIFO for low
power audio playback.
The digital interface can use I2S, DSP (A or B), Right and Left
justified formats.
DAC33 has stereo analog input, which can be bypassed to the analog
outputs.
Regarding to the internal 24KB FIFO the driver implements 'FIFO bypass'
mode (default) and nSample mode (FIFO is in use).
a) In 'FIFO bypass' mode the internal FIFO is not in use, the codec is
working synchronously as a normal codec (it needs constant stream of
data on the digital interface).
b) The nSample mode implementation uses one interrupt line from DAC33 to
the host:
Alarm threshold is set to 10ms of audio data (limit by the driver
implementation).
DAC33 will signal an interrupt, when the FIFO level goes under the
Alarm threshold.
The host will write to nSample register a value (number of stereo
samples), to tell DAC33 how many samples it should read in a burst from
the host. When the DAC33 received the number of samples, it disables the
clocks on the I2S bus. When the FIFO use again goes under the Alarm
threshold, DAC33 signals the host with an interrupt, and the process is
repeated.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Igor Grinberg <grinberg@compulab.co.il>
Signed-off-by: Mike Rapoport <mike@compulab.co.il>
CC: Mark Brown <broonie@opensource.wolfsonmicro.com>
CC: alsa-devel@alsa-project.org
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The PM core will grow pm_link infrastructure in 2.6.33 which can be
used to implement the intended functionality of the ASoC-specific
device suspend and resume callbacks so drop them.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Remove the <plat/audio.h> include from arch/arm/plat-s3c/include/plat/audio.h
as it provides nothing to the current kernel and is not in any future plans
for the system.
Signed-off-by: Ben Dooks <ben@simtec.co.uk>
Signed-off-by: Simtec Linux Team <linux@simtec.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Access to damp_power_widgets() is assumed to be single-threaded.
Concurrent accesses to dapm_power_widgets() may result in
unpredictable behavior.
Calls from:
close_delayed_work()
soc_codec_close()
soc_pcm_prepare()
soc_suspend()
soc_resume_deferred()
to snd_soc_dapm_stream_event() do not have the codec->mutex
taken to cover the call to dapm_power_widgets(). Thus, take
the mutex in these paths also to assure single-threaded use
of dapm_power_widgets().
Signed-off-by: Eero Nurkkala <ext-eero.nurkkala@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The power for the amplifier should be handled internally
by the tpa6130a2 driver.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
- Staticise ttpa6130a2_client.
- Remove unneeded cast from void.
- Use explict NULL rather than 0.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Driver for Texas Instruments TPA6130A2 stereo headphone
amplifier.
The driver provides playback gain control and also pre-defined
DAPM_HP widgets and DAPM routings for power management.
The DAPM_HP widget names are:
"TPA6130A2 Headphone Left"
"TPA6130A2 Headphone Right"
From soc machine drivers to use with the tpa6130a2 amplifier,
the tpa6130a2_add_controls has to be called, which adds the alsa
controls and the DAPM routing needed for the tpa6130a2.
After that the machine driver can connect the codec's output
with 'TPA6130A2 Left' and 'TPA6130A2 Right':
{"TPA6130A2 Left", NULL, "CODEC LEFT OUT"},
{"TPA6130A2 Right", NULL, "CODEC RIGHT OUT"},
Internally the left and right channels are powered separately.
When none of the channels are needed the amplifier is powered
down:
hard power: valid GPIO number is passed within platform data
soft power: Using the software shutdown of the amplifier
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The modified revision of at91sam9g20 Evaluation Kit rev. C and onwards share
with previous ones its audio connexion to Wolfson wm8731. Modify the SoC file
to extend the machine ID checking.
Signed-off-by: Nicolas Ferre <nicolas.ferre@atmel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Sometimes it is desirable to have a mux which does not reflect any
direct register configuration but which will instead only have an
effect implicitly (for example, as a result of changing which parts
of the device are powered up). Provide a virtual mux for this purpose.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We don't need to check for an event callback since we also check for
an appropriate event flag when applying mux status changes.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix for typo in commit 8d50e447d1
ASoC: Factor out I/O for Wolfson 8 bit data 16 bit register CODECs
Signed-off-by: Jonathan Cameron <jic23@cam.ac.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Change the way the debugfs entries are created:
If the codec->dev is valid, than use:
debugfs/asoc/{codec->name}.{dev_name(codec->dev)}/
if the codec->dev is NULL:
debugfs/asoc/{codec->name}/
as root for the debugfs entries.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since the SND_SOC_DAPM_LINE can be input or output, additional check is
needed in order to determine if the widget is connected as input or
output.
When checking for connected outputs, if the widget is line, than check
if the sources list is not empty (line is connected as output)
For input endpoint check, when the widget is line, also check if the
sinks list is not empty (line is connected as input).
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In order to support multiple codecs on the same system in the debugfs
the directory hierarchy need to be changed by adding directory per codec
under the asoc direcorty:
debugfs/asoc/{dev_name(socdev->dev)}-{codec->name}/codec_reg
/dapm_pop_time
/dapm/{widgets}
With the original implementation only the debugfs files are only
created for the first codec, other codecs loaded later would fail to
create the debugfs files (since they are already exist).
Furthermore in this situation any of the codecs has been removed, would
cause the debugfs entries to disappear, regardless if the codec, which
created them are still loaded (the one which loaded first).
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some of the Blackfin options don't directly follow the kconfig options
they depend on, so kconfig is unable to display the proper tree. So sort
the options such they expand/collapse properly.
Signed-off-by: Barry Song <barry.song@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix the missing argument of snd_soc_dai_set_pll() in neo1973_*.c,
which was forgotten in the commit 85488037bb.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix a couple of typos and a missing header file inclusion to build wm8711.c
properly with CONFIG_SPI_MASTER.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Don't use a static for WM8974 PLL factors - we don't support more than
one device so it won't happen but no sense in leaving the race condition
hanging around. Also, pre_div is a single bit and it's a bit simpler if
we move the handling of the factor of 4 in the output into the
coefficient setup.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The DMA params for McASP with FIFO has been updated so that it works for
various FIFO levels. A member- 'fifo_level' has been added to the DMA
params data structure. The fifo_level can be adjusted by the tx[rx]_numevt
platform data. This is relevant only for DA8xx/OMAP-L1xx platforms. This
implementation has been tested for numevt values 1, 2, 4, 8.
Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When running in TDM mode there can be more than 2 channels used. Datasheet has
figures for upto 8 channels so increase max_channels on all SSP interfaces to
this figure.
Signed-off-by: Graeme Gregory <dp@xora.org.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
McASP write FIFO registers should be modified for playback and read FIFO
registers for capture. Check the PCM mode before manipulating the
FIFO registers. Currently, irrespective of playback/capture both the
FIFOs are enabled or disbaled. This resulted in errors in audio loopback
mode.
Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch removes references to cpu_dai->dma_data.
It makes struct davinci_pcm_dma_params part of
struct davinci_mcbsp_dev or struct davinci_audio_dev.
It removes the unused name variable from davinci_pcm_dma_params.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When both playback and capture stream were open
davinci_i2s_hw_params was setting parameters for
the wrong stream. The fix for davinci_i2s_hw_params
is sufficient, but it looks like a race still happens
in davici_pcm_open. This patch also makes the race smaller
but the next patch provides a better fix.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* 'fix/asoc' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ASoC: wm8753: fix mapping when MONOMIX is set to Stereo
ASoC: some minor changes for AD1836 and AD1938 codec drivers
ASoC: DaVinci: Fixes to McASP configuration
ASoC: Blackfin I2S: fix resuming when device hasn't been used
ASoC: Blackfin I2S: add lost platform_device parameter to resume function
ASoC: fix typos in Blackfin headers
ASoC: bf5xx-sport: the irq save/restore funcs take an unsigned long
ASoC: Blackfin AC97: add a few missing multichannel define handling
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Barry Song <barry.song@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since the active field of the dai already tells us the stream activity,
the local counter variable is redundant and can be replaced.
Signed-off-by: Barry Song <barry.song@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When MONOMIX is set to Stereo, Left PGA was not powered on but should be.
Add a mapping from Capture Left Mux to Capture Left Mixer to fix the issue.
Signed-off-by: Phil Vandry <vandry@TZoNE.ORG>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
1. delete redundant assignment to bus field in spi_driver structure
2. fix lost assignment to set_bias_level entry in ad1938 codec dai
3. change spi driver name of ad1836 from "ad1836-spi" to "ad1836"
Signed-off-by: Barry Song <barry.song@analog.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
New machine driver for WM8580 I2S i/f on SMDK64XX.
By default SoC-Slave is set and WM8580 is configured to use it's
PLLA to generate clocks from a 12MHz crystal attached to WM8580.
[Added dependency on BROKEN since the IISv4 interface hasn't been merged
yet, fixed the PLL API usage and removed the disabling of the PLL in the
hw_free function since that'll break simultaneous playback and record
-- broonie.]
Signed-off-by: Jassi <jassi.brar@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* 'davinci-for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/khilman/linux-davinci: (62 commits)
DaVinci: DM646x - platform changes for vpif capture and display drivers
davinci: DM355 - platform changes for vpfe capture
davinci: DM644x platform changes for vpfe capture
davinci: audio: move tlv320aic33 i2c setup into board files
DaVinci: EDMA: Adding 2 new APIs for allocating/freeing PARAMs
DaVinci: DM365: Adding entries for DM365 IRQ's
DaVinci: DM355: Adding PINMUX entries for DM355 Display
davinci: Handle pinmux conflict between mmc/sd and nor flash
davinci: Add NOR flash support for da850/omap-l138
davinci: Add NAND flash support for DA850/OMAP-L138
davinci: Add MMC/SD support for da850/omap-l138
davinci: Add platform support for da850/omap-l138 GLCD
davinci: Macro to convert GPIO signal to GPIO pin number
davinci: Audio support for DA850/OMAP-L138 EVM
davinci: Audio support for DA830 EVM
davinci: Correct the number of GPIO pins for da850/omap-l138
davinci: Configure MDIO pins for EMAC
DaVinci: DM365: Add Support for new Revision of silicon
DaVinci: DM365: Fix Compilation issue due to PINMUX entry
DaVinci: EDMA: Updating default queue handling
...
Instead of always returnig pointer to the 'audio-bus' clock,
check which clock is used to generate internal clocks and
then return it's pointer.
Signed-off-by: Jassi <jassi.brar@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
McASP register settings are not correct for DSP mode of operation.
There is a channel swap initally. This patch provides fixes to
the register values for proper working.
Tested on DA830/OMAP-L137 EVM, DM6467 EVM.
Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If the sound system hasn't been utilized yet and we suspend, then we
attempt to save/restore using state that doesn't exist. So use a global
handle instead to reconfigure properly.
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* fix/asoc:
ASoC: remove unused #include <linux/version.h>
ASoC: S3C lrsync function made to work with IRQs disabled.
ASoC: Fix display of stream name in DAPM debugfs
ASoC: Clean up error handling in MPC5200 DMA setup
Commit dc7d7b830e trimmed the platform_device parameter from all of the
suspend functions, but it also accidentally removed it from the resume
function in the Blackfin I2S driver. So restore it.
Signed-off-by: Barry Song <barry.song@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Barry Song <barry.song@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Somewhere along the line, most of SND_BF5XX_MULTICHAN_SUPPORT handling was
merged, but two places were missed (the probe/resume functions). Restore
handling of this option so it gets initialized properly.
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
CDCLK can either be an output generated by the CPU, intended for use
as the CODEC master clock, or an input (probably from the CODEC)
providing a master clock for the IIS block.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch enables tlv320aic3101 support on DM365 EVM and
it was tested on DM365 EVM rev c.
Note: this patch was created based on temp/asoc branch.
Signed-off-by: Miguel Aguilar <miguel.aguilar@ridgerun.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
1) Explicitly set LRCLK polarity for I2S Vs LSM/MSB modes.
2) Convert from numerical to bit-field values for BCLK selection.
3) Use proper error checking for return value from clk_get
Signed-off-by: Jassi <jassi.brar@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
s3c2412_snd_lrsync() maybe reached with IRQs disabled and if LRCLK
is dead due to improper initialization of CPU or CODEC, the system
gets stuck in the loop because jiffies may never get updated.
Implemented counter based wait mechanism for atleast the same
timeout period.
Signed-off-by: Jassi <jassi.brar@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* 'devel' of master.kernel.org:/home/rmk/linux-2.6-arm: (257 commits)
[ARM] Update mach-types
ARM: 5636/1: Move vendor enum to AMBA include
ARM: Fix pfn_valid() for sparse memory
[ARM] orion5x: Add LaCie NAS 2Big Network support
[ARM] pxa/sharpsl_pm: zaurus c3000 aka spitz: fix resume
ARM: 5686/1: at91: Correct AC97 reset line in at91sam9263ek board
ARM: 5640/1: This patch modifies the support of AC97 on the at91sam9263 ek board
ARM: 5689/1: Update default config of HP Jornada 700-series machines
ARM: 5691/1: fix cache aliasing issues between kmap() and kmap_atomic() with highmem
ARM: 5688/1: ks8695_serial: disable_irq() lockup
ARM: 5687/1: fix an oops with highmem
ARM: 5684/1: Add nuc960 platform to w90x900
ARM: 5683/1: Add nuc950 platform to w90x900
ARM: 5682/1: Add cpu.c and dev.c and modify some files of w90p910 platform
ARM: 5626/1: add suspend/resume functions to amba-pl011 serial driver
ARM: 5625/1: fix hard coded 4K resource size in amba bus detection
MMC: MMCI: convert realview MMC to use gpiolib
ARM: 5685/1: Make MMCI driver compile without gpiolib
ARM: implement highpte
ARM: Show FIQ in /proc/interrupts on CONFIG_FIQ
...
Fix up trivial conflict in arch/arm/kernel/signal.c.
It was due to the TIF_NOTIFY_RESUME addition in commit d0420c83f ("KEYS:
Extend TIF_NOTIFY_RESUME to (almost) all architectures") and follow-ups.
The patch adds an interface to set the relationship between audio
channel number and slot number. The interface should be really useful
because audio channel n doesn't always use slot n in all platforms. And
for some devices, the relationship even can change with sound mode
switch in 2.1,3.1,4.1,5.1,6.1,7.1 etc.
Signed-off-by: Barry Song <21cnbao@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Error handling code following a kzalloc should free the allocated data.
Error handling code following an ioremap should iounmap the allocated data.
The semantic match that finds the first problem is as follows:
(http://www.emn.fr/x-info/coccinelle/)
// <smpl>
@r exists@
local idexpression x;
statement S;
expression E;
identifier f,f1,l;
position p1,p2;
expression *ptr != NULL;
@@
x@p1 = \(kmalloc\|kzalloc\|kcalloc\)(...);
...
if (x == NULL) S
<... when != x
when != if (...) { <+...x...+> }
(
x->f1 = E
|
(x->f1 == NULL || ...)
|
f(...,x->f1,...)
)
...>
(
return \(0\|<+...x...+>\|ptr\);
|
return@p2 ...;
)
@script:python@
p1 << r.p1;
p2 << r.p2;
@@
print "* file: %s kmalloc %s return %s" % (p1[0].file,p1[0].line,p2[0].line)
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* topic/asoc: (226 commits)
ASoC: au1x: PSC-AC97 bugfixes
ASoC: Fix WM835x Out4 capture enumeration
ASoC: Remove unuused hw_read_t
ASoC: fix pxa2xx-ac97.c breakage
ASoC: Fully specify DC servo bits to update in wm_hubs
ASoC: Debugged improper setting of PLL fields in WM8580 driver
ASoC: new board driver to connect bfin-5xx with ad1836 codec
ASoC: OMAP: Add functionality to set CLKR and FSR sources in McBSP DAI
ASoC: davinci: i2c device creation moved into board files
ASoC: Don't reconfigure WM8350 FLL if not needed
ASoC: Fix s3c-i2s-v2 build
ASoC: Make platform data optional for TLV320AIC3x
ASoC: Add S3C24xx dependencies for Simtec machines
ASoC: SDP3430: Fix TWL GPIO6 pin mux request
ASoC: S3C platform: Fix s3c2410_dma_started() called at improper time
ARM: OMAP: McBSP: Merge two functions into omap_mcbsp_start/_stop
ASoC: OMAP: Fix setup of XCCR and RCCR registers in McBSP DAI
OMAP: McBSP: Use textual values in DMA operating mode sysfs files
ARM: OMAP: DMA: Add support for DMA channel self linking on OMAP1510
ASoC: Select core DMA when building for S3C64xx
...
The AK4671 is a stereo CODEC with a built-in Microphone-Amplifier,
Receiver-Amplifier and Headphone-Amplifier.
The datasheet for the ak4671 can find at the following url:
http://www.asahi-kasei.co.jp/akm/en/product/ak4671/ak4671_f01e.pdf
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some chips with complex internal supply (particularly clocking)
arragements may have multiple options for some of the supply
connections. Since these don't affect user-visible audio routing
the expectation would be that they would be managed automatically
by one of the drivers.
Support these users by allowing routes to have a connected function
which is queried before the connectedness of the path is checked as
normal. Currently this is only done for supplies, other widgets
could be supported but are not currently since the expectation for
them is that audio routing will be under the control of userspace.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch fixes the following bugs:
- only reprogram bitdepth if it has changed since last call to hw_params.
- add locking inside ac97_read/write functions:
When reprogramming sample depth, the ac97 unit has to be disabled,
which should not be done in the middle of codec register accesses.
- retry timed-out codec register accesses.
- wait for status bits to set/clear when starting/stopping various
functional blocks; very important after reenabling AC97 unit else
sound may be distorted (e.g. high-pitch noise in 1kHz sine wave).
- clear fifos before/after starting/stopping RX/TX.
- longer timeouts waiting for PSC/AC97 ready after cold reset
with certain codecs this can take ridiculous amounts of time.
Run-tested on various Au1200 platforms with various codecs.
Signed-off-by: Manuel Lauss <manuel.lauss@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
It's the 8th enum of a zero indexed array. This is why I don't let
new drivers use these arrays of enums...
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
This patch is for the AK4671 codec driver using this format.
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
More and more devices feature PLLs and FLLs with the ability to select
between multiple input clocks. In order to better support these devices
a new argument, source, has been added to the set_pll() configuration
API. Using set_clkdiv() is often difficult due to the need to stop the
PLL/FLL before any reconfiguration can be done.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Avoids potential issues if we read back unexpected values during
a read/modify/write cycle.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Bug was caught while trying to use WM8580 as I2S master on SMDK.
Symptoms were lesser LRCLK read by CRO(41.02 instead of 44.1 KHz) Solved
by referring to WM8580A manual and setting mask value correctly and
making the code to not touch 'reserved' bits of PLL4 register.
Signed-off-by: Jassi <jassi.brar@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
As discussed, the patch uses the original TDM order without rewriting.
For the match between TDM slot number and audio channel number, a new
API need be added.
Signed-off-by: Barry Song <21cnbao@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The McBSP1 port in OMAP3 processors (I believe OMAP2 too but I don't have
specifications to check it) have additional CLKR and FSR pins for McBSP1
receiver. Reset default is that receiver is using bit clock and frame
sync signal from those pins but it is possible to configure to use
also CLKX and FSX pins as well. In fact, other McBSP ports are doing that
internally that transmitter and receiver share the CLKX and FSX.
Add functionaly that machine drivers can set the CLKR and FSR sources by
using the snd_soc_dai_set_sysclk.
Thanks to "Aggarwal, Anuj" <anuj.aggarwal@ti.com> for reporting the issue.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Also, the codec setup data structure has to remain for successful
probe.
Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Kevin Hilman <khilman@deeprootsystems.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If the requested FLL configuration is the one we're currently running
in it's at best pointless to reconfigure the FLL.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Now that we don't need the I2C address for the device the platform data
is redundant so allow it to be omitted.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Tested-by: Chaithrika U S <chaithrika@ti.com>
No point in building them for S3C64xx, certainly no sense in running
into build issues there.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
- restructure to support multiple channel controllers by using
additional struct resources for each CC
- interface changes visible to EDMA clients
Introduce macros to build IDs from controller and channel number,
and to extract them. Modify the edma_alloc_slot function to take an
extra argument for the controller.
Also update ASoC drivers to use API. ASoC changes
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
- Move queue related mappings to dm<soc>.c
EDMA in DM355 and DM644x has two transfer controllers while DM646x
has four transfer controllers. Moving the queue to tc mapping and
queue priority mapping to dm<soc>.c will be helpful to probe these
mappings from platform device so that the machine_is_* testing will
be avoided.
- add channel mapping logic
Channel mapping logic is introduced in dm646x EDMA. This implies
that there is no fixed association for a channel number to a
parameter entry number. In other words, using the DMA channel
mapping registers (DCHMAPn), a PaRAM entry can be mapped to any
channel. While in the case of dm644x and dm355 there is a fixed
mapping between the EDMA channel and Param entry number.
Signed-off-by: Naresh Medisetty <naresh@ti.com>
Signed-off-by: Sudhakar Rajashekhara <sudhakar.raj@ti.com>
Reviewed-by: David Brownell <dbrownell@users.sourceforge.net>
Signed-off-by: Kevin Hilman <khilman@deeprootsystems.com>
Fix the write to PMBR1 register through I2C. Also, the constant which
holds the value to write is now called TWL4030_GPIO6_PWM0_MUTE. This
name is based on TRM to avoid confusion.
Signed-off-by: Jorge Eduardo Candelaria <x0107209@ti.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
s3c24xx dma has the auto reload feature, when the the trnasfer is done,
CURR_TC(DSTAT[19:0], current value of transfer count) reaches 0, and DMA
ACK becomes 1, and then, TC(DCON[19:0]) will be loaded into CURR_TC. So
the transmission is repeated.
IRQ is issued while auto reload occurs. We change the DISRC and
DCON[19:0] in the ISR, but at this time, the auto reload has been
performed already. The first block is being re-transmitted by the DMA.
So we need rewrite the DISRC and DCON[19:0] for the next block
immediatly after the this block has been started to be transported.
The function s3c2410_dma_started() is for this perpose, which is called
in the form of "s3c2410_dma_ctrl(prtd->params->channel,
S3C2410_DMAOP_STARTED);" in s3c24xx_pcm_trigger().
But it is not correct. DMA transmission won't start until DMA REQ signal
arrived, it is the time s3c24xx_snd_txctrl(1) or s3c24xx_snd_rxctrl(1)
is called in s3c24xx_i2s_trigger().
In the current framework, s3c24xx_pcm_trigger() is always called before
s3c24xx_pcm_trigger(). So the s3c2410_dma_started() should be called in
s3c24xx_pcm_trigger() after s3c24xx_snd_txctrl(1) or
s3c24xx_snd_rxctrl(1) is called in this function.
However, s3c2410_dma_started() is dma related, to call this function we
should provide the channel number, which is given by
substream->runtime->private_data->params->channel. The private_data
points to a struct s3c24xx_runtime_data object, which is define in
s3c24xx_pcm.c, so s3c2410_dma_started() can't be called in s3c24xx_i2s.c
Fix this by moving the call to signal the DMA started to the DAI
drivers.
Signed-off-by: Shine Liu <liuxian@redflag-linux.com>
Signed-off-by: Shine Liu <shinel@foxmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Functionality of functions omap_mcbsp_xmit_enable and omap_mcbsp_recv_enable
can be merged into omap_mcbsp_start and omap_mcbsp_stop since API of
those omap_mcbsp_start and omap_mcbsp_stop was changed recently allowing
to start and stop individually the transmitter and receiver.
This cleans up the code in arch/arm/plat-omap/mcbsp.c and in
sound/soc/omap/omap-mcbsp.c which was the only user for those removed
functions.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Eero Nurkkala <ext-eero.nurkkala@nokia.com>
Cc: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Commit ca6e2ce086 is setting up few XCCR and
RCCR bits for I2S and DPS_A formats. Part of the bits are already set
for all formats and I believe that XDISABLE and RDISABLE bits are
format independent.
As XCCR and RCCR are found only from OMAP2430 and OMAP34xx, I move setup
of XDISABLE and RDISABLE to where those cpu's are tested and remove format
dependent part for simplicity.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Eero Nurkkala <ext-eero.nurkkala@nokia.com>
Cc: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Free socdev if snd_soc_init_card() fails.
Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently when built with DEBUG DAPM will dump information about
the power state decisions it is taking for each widget to dmesg.
This isn't an ideal way of getting the information - it requires
a kernel build to turn it on and off and for large hub CODECs the
volume of information is so large as to be illegible. When the
output goes to the console it can also cause a noticable impact
on performance simply to print it out.
Improve the situation by adding a dapm directory to our debugfs
tree containing a file per widget with the same information in
it. This still requires a decision to build with debugfs support
but is easier to navigate and much less intrusive.
In addition to the previously displayed information active streams
are also shown in these files.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch is tested by ms7724se
Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is very simple driver for ALSA
It supprt headphone output and stereo input only
This patch is tested by ms7724se
Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add support for the tlv320aic3x CODEC on the Simtec Hermes board.
Signed-off-by: Ben Dooks <ben@simtec.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fixup the device changes by modifying the files that we just removed the
explicit device creation from with i2c_register_board_info() until this
can be moved into the relevant board files.
Signed-off-by: Ben Dooks <ben@simtec.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The tlv320aic3x driver managed its own i2c device, instead of an extant
one created by the board support code. Change the code to make it so that
the driver binds to an extant (in this case i2c) device.
Add explict tlv320aic33 as well as tlv320aic3x to the supported device
table and remove the old driver bindings from the users of this code.
Signed-off-by: Ben Dooks <ben@simtec.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add core support for the range of S3C24XX Simtec boards with TLV320AIC23
CODECs on them. Since there are also boards with similar IIS routing the
AMP and the configuration code is placed in a core file for re-use with
other CODEC bindings.
Signed-off-by: Ben Dooks <ben@simtec.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Configures DMA sync mode depending on McBSP operating mode value.
The value is configurable by McBSP instance. So, depending
on McBSP operating mode, the DMA sync mode is passed from
omap-mcbsp to omap-pcm. Besides that, it also configures
McBSP threshold value depending on which McBSP mode is activated.
Signed-off-by: Eduardo Valentin <eduardo.valentin@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch changes the way DMA is done in omap-pcm.c
in order to reduce power consumption. There is no need
to have so much SW control in order to have DMA in idle
state during audio streaming. Configuring McBSP threshold value
and DMA to FRAME_SYNC are sufficient.
Signed-off-by: Eduardo Valentin <eduardo.valentin@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
All these steps are required for ASoC to behave correctly.
rccr and xccr are format dependent, for example TDM audio
has different values than I2S or DSP_A. Also the
omap_mcbsp_xmit_enable and/or omap_mcbsp_recv_enable must
be called right after the DMA has started.
This provides no longer L and R channels switching at random.
Signed-off-by: Eero Nurkkala <ext-eero.nurkkala@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is, no RFIG or XFIG (Not defined in 34xx), correct
initiliazation of rccr and xccr.
Signed-off-by: Eero Nurkkala <ext-eero.nurkkala@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Align DMA address to DMA burst transaction sizes.
Signed-off-by: Eduardo Valentin <eduardo.valentin@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Improve DMA transfers by enabling Burst transaction.
Signed-off-by: Eduardo Valentin <eduardo.valentin@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This driver is very simple.
It support playback only now.
This patch is tested by ms7724se board.
Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
> Then it's a driver bug. If unaligned period size is allowed, it means
> that the irq is really generated in that period, not at the buffer
> boundary. Otherwise, it must have a proper hw-constraint to align the
> period size to the buffer size.
This patch will fix the bug metioned in the above mail. Force the peroid
size to be aligned with the buffer size.
Based and tested on linux-2.6.31-rc6.
Signed-off-by: Shine Liu <shinel@foxmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If the CODEC does not provide a set_bias_level() then update the
bias_level variable for it since other parts of the system expect
that to be maintained.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM8711 or WM8711L (WM8711/L) is a low power stereo DAC with an
integrated headphone driver. The WM8711/L is designed specifically for
portable MP3 audio and speech players. The WM8711/L is also ideal for
MD, CD machines and DAT players.
Signed-off-by: Mike Arthur <Mike.Arthur@wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM8993 provides digital sidetone paths and also allows each
channel on the audio interface to be routed separtately to the
DACs and ADCs.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently DAPM interfaces with the audio streams to and from the
processor at the DAC and ADC widgets. As the digital capabilities
of parts increases this is becoming a less and less able to meet
the needs of parts.
To meet the needs of these devices create new widgets interfacing
with the TDM bus but not integrated into any other functionality.
Audio can then be routed to and from these widgets using existing
routing widgets.
A slot number is provided in the definition but this is currently
not used yet. This is intended to support devices which can use
more than one TDM slot on a single interface.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Joonyoung Shim reports that S3C64xx I2S is working on the NCP boards so
allow it to be selected in Kconfig.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmciro.com>
The data format configuration for S3C64xx IISv2 was hardcoded for IISMOD
register. This patch changes to the defined values it.
And instead of bits 9 and 10 of IISMOD we should clear bits 13 and 14.
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Note that the number of slots used internally is specified in terms
of stereo slots while the external API works with mono slots.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When used without the PLL we were accidentally clearing the MCLK/2
divider, resulting in a double rate SYSCLK when the divider should
have been used.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Speaker and headphone outputs do not need to be handled separately
since they can't be part of the same path.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If the system doesn't have any DAPM widgets then we can't use their
state to check if the bias level for the codec should be up.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There is a mistake in current uda134x_mute function: mute_reg has been
changed in line 162 or line 164, so uda134x_write should write
"mute_reg" but not "mute_reg & ~(1<<2)" to
UDA134X_DATA010.
Signed-off-by: Shine Liu <shinel@foxmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Enhance period_index accuracy, particularly just before buffer rewind, by
making use of DMA interrupt status flags in addition to simply counting up
interrupts.
Created against linux-2.6.31-rc5.
Tested on Amstrad Delta.
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use newly implemented DMA channel self linking on OMAP1510 like on other OMAP
models. Remove unnecessary DMA transfer restart from interrupt handler
routine.
The interrupt routine used to maintain a period index, originally needed for
counting up periods up to a full buffer in order to restart the DMA transfer.
For some time, this counter is also used as a replacement for hardware DMA
progress counter that has been found unusable on OMAP1510 in case of playback.
Thus, the period index calculation cannot be omitted completely. However, the
accuracy of this counter can still suffer from missing DMA interrupts.
In order to work correctly, it requires patch 1 from this series also applied:
[RFC][PATCH 1/3] ARM: OMAP: DMA: Add support for DMA channel self linking on OMAP1510
Created against linux-2.6.31-rc5.
Tested on Amstrad Delta.
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Change the strings related to capture in order to be
interpreted correctly by alsamixer and possible other
UI based mixer applications.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Move the definition of the "generic" IRQ in the process.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Ben Dooks <ben-linux@fluff.org>
This patch adds support for passing platform data to ac97 bus devices
from PXA2xx-AC97 driver..
Signed-off-by: Marek Vasut <marek.vasut@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There is one instance of McASP on DA850/OMAP-L138 SoC. This is
connected to TLV320AIC3106 codec for audio playback and capture.
This patch adds audio support on this platform. Some of the
structure prefix names which are common for DA830/OMAP-L137 EVM and
DA850/OMAP-L138 EVM have been renamed to da8xx from da830.
Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The patch adds a DAI format: Codec bit clock master and frame sync slave,
to the driver.
Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
On DA830/OMAP-L137 and DA850/OMAP-L138 SoCs, the McASP peripheral has FIFO
support. This FIFO provides additional data buffering. It also provides
tolerance to variation in host/DMA controller response times.
The read and write FIFO sizes are 256 bytes each. If FIFO is enabled,
the DMA events from McASP are sent to the FIFO which in turn sends DMA requests
to the host CPU according to the thresholds programmed.
More details of the FIFO operation can be found at
http://focus.ti.com/general/docs/lit/getliterature.tsp?literatureNumber=
sprufm1&fileType=pdf
This patch adds support for FIFO configuration. The platform data has a
version field which differentiates the McASP on different SoCs.
Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM8993 analogue control is shared with other devices in the same
product line. Since this is a very substantial proportion of the
driver move the definitions of these controls into a new wm_hubs module
which allows them to be shared between the two.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There has been an ad1836 driver in sound/blackfin based on traditional alsa.
The new driver is based on asoc. The architecture of ad1836 codec driver is
very much like ad1938.
Signed-off-by: Barry Song <21cnbao@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Dynamically control and control only the needed output amplifier
muting/un-muting.
The original code was muting and un-muting the following output
amplifiers: Earpiece PreDrivL/R, CarkitL/R at the same time
regardless which pin is actually in use at the given moment.
Move these as separate PGA so only the needed amplifier will be touched.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
According to the function dapm_dac_check_power() in
sound/soc/soc-dapm.c, dac power can't be on/off stand-alone without any
output widget as sink. And according to dapm_adc_check_power(), adc
power can't be on/off stand-alone without any input widget as source. So
we can't only define some stand-alone SND_SOC_DAPM_DAC/SND_SOC_DAPM_ADC
to hope their power can be managed dynamically.
Signed-off-by: Barry Song <21cnbao@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Store the TDM slot width then if it's set use that rather than the
sample size to calculate BCLK. Leave imposing constraints to the
core (which should do this but doesn't yet) or machine driver.
Also allow 0 TDM slots to be configure (for use when disabling TDM).
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix soc build errors when I2C is built as a loadable module:
(.text+0x5d26b): undefined reference to `i2c_master_send'
soc-cache.c:(.text+0x5d32d): undefined reference to `i2c_transfer'
Signed-off-by: Randy Dunlap <randy.dunlap@oracle.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add support for audio on DA830 EVM- here McASP1 is interfaced to
TLV320AIC3106 codec.
Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This fixes a build failure for 2.6.31-rc4-rt1 (ARCH=arm, s3c2410_defconfig):
CC [M] sound/soc/s3c24xx/s3c2443-ac97.o
sound/soc/s3c24xx/s3c2443-ac97.c:50: warning: type defaults to 'int' in declaration of 'DECLARE_MUTEX'
sound/soc/s3c24xx/s3c2443-ac97.c:50: warning: parameter names (without types) in function declaration
sound/soc/s3c24xx/s3c2443-ac97.c: In function 's3c2443_ac97_read':
sound/soc/s3c24xx/s3c2443-ac97.c:59: error: 'ac97_mutex' undeclared (first use in this function)
sound/soc/s3c24xx/s3c2443-ac97.c:59: error: (Each undeclared identifier is reported only once
sound/soc/s3c24xx/s3c2443-ac97.c:59: error: for each function it appears in.)
sound/soc/s3c24xx/s3c2443-ac97.c: In function 's3c2443_ac97_write':
sound/soc/s3c24xx/s3c2443-ac97.c:93: error: 'ac97_mutex' undeclared (first use in this function)
Signed-off-by: Uwe Kleine-König <u.kleine-koenig@pengutronix.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The patch changes the line discipline name registered in include/linux/tty.h
and updates the ams-delta machine driver to use it.
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The dma setup code assumes that the buffer size is a multiple
of the period size.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
dai is a parameter to the functions, so use it instead of
looking it up.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Simultaneous audio playback and capture on OMAP1510 can cause that second
stream is stalled if there is enough delay between startup of the audio
streams.
Current implementation of the omap_mcbsp_start is starting both transmitter
and receiver at the same time and it is called only for firstly started
audio stream from the OMAP McBSP based ASoC DAI driver.
Since DMA request lines on OMAP1510 are edge sensitive, the DMA request is
missed if there is no DMA transfer set up at that time when the first word
after McBSP startup is transmitted. The problem hasn't noted before since
later OMAPs are using level sensitive DMA request lines.
Fix the problem by changing API of omap_mcbsp_start and omap_mcbsp_stop by
allowing to start and stop individually McBSP transmitter and receiver
logics. Then call those functions individually for both audio playback
and capture streams. This ensures that DMA transfer is setup before
transmitter or receiver is started.
Thanks to Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> for detailed problem
analysis and Peter Ujfalusi <peter.ujfalusi@nokia.com> for info about DMA
request line behavior differences between the OMAP generations.
Reported-and-tested-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Tony Lindgren <tony@atomide.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Extend set_tdm_slot to allow the user to arbitrarily set the frame width
and active TX/RX slots.
Updates magician.c and wm9081.c for the new set_tdm_slot(). wm9081.c
still doesn't handle the slot_width override.
While being there, correct an incorrect use of SlotsPerFrm(7) use in
bitmask on pxa-ssp.c (SSCR0_SlotsPerFrm(x) is (((x) - 1) << 24)) ).
(this series is meant for Mark's for-2.6.32 branch)
Signed-off-by: Daniel Ribeiro <drwyrm@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch is a workaround for the problem of several subsequent control
statements not being applied correctly to the codec controller (modem).
In order to follow the hook switch state change from handset to handsfree
while
in full duplex mode, two consecutive +VLS control commands were sent to the
modem. The first one was M1 (microphone only), the seconds one was M1S1 (both
microphone and speaker). As there was no real modem handshaking procedure
implemented, neither in the codec nor in the machine driver part of the line
discipline, the modem was having the second command missed.
Since a possibility to switch to microphone only mode (and speaker only mode
as well) seams of no value, I have modified the code to issue single M1S1
command only for any of those cases.
Tested on my Amstrad Delta.
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds debugging statement that can help in tracing
how the driver is trying to control the codec device.
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM8776 is a high performance, stereo audio CODEC with five channel
input selector. The WM8776 is ideal for surround sound processing
applications for home hi-fi, DVD-RW and other audio visual equipment.
This driver implements support for most WM8776 features - currently the
ADC automatic level control/limiter functionality is omitted.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This adds support for i.mx27_visstrim_sm10 board machine driver which
uses an i.mx27 processor plus a wm8974 codec.
It has been tested on a visstrim_sm10 board.
Signed-off-by: Javier Martin <javier.martin@vista-silicon.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This adds support for DAI platform for the SSI present in MXC platforms.
It currently does not support i.MX3, the only thing necessary to do
this is to export DMA data for i.MX3 interface which I haven't done
because I don't have a i.MX3 based board available.
It has been tested on i.MX27 board.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This adds support for DMA platform valid for i.MX1 and i.MX2 platforms.
This is not valid for i.MX3 since it doesn't share the same DMA
interface than i.MX1 and i.MX2.
It has been tested on i.MX27 board.
Signed-off-by: Javier Martin <javier.martin@vista-silicon.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Power management for the cs4270 codec is currently implemented as part
of the i2c_driver struct. The disadvantage of doing it this way is that
the callbacks registered in the snd_soc_card struct are called _before_
the codec's callbacks.
That doesn't work, because the snd_soc_card callbacks will most likely
switch down the codec's power domains or pull the reset GPIOs, and
hence make the i2c communication bail out.
Fix this by binding the suspend and resume code to the
snd_soc_codec_device driver model and let the I2C functions only call
the SoC core function for resume and suspend, which do nothing currently
but will do later.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Cc: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The code in psc_dma_bcom_enqueue_tx() didn't account for the fact that
s->runtime->control->appl_ptr can wrap around to the beginning of the
buffer. This change fixes this problem.
Signed-off-by: John Bonesio <bones@secretlab.ca>
Acked-by: Grant Likely <grant.likely@secretlab.ca>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This converts all the Wolfson drivers using this format (the only devices
that do) except WM8753 to use it.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
As part of this refactoring the type of the CODEC hw_read operation
is changed to match the regular read operation.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
While writes tend to be able to use a fairly bus independant format to
do the writes reads are all bus specific. To allow us to factor out
this code include the bus type as a parameter when setting up the
cache.
Initially just use this to factor out hw_write_t for I2C.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds machine support for Amstrad E3 (Delta) videophone to ASoC.
Created and tested against linux-2.6.31-rc3.
Applies and works with linux-omap-2.6 commit
7c5cb7862d32cb344be7831d466535d5255e35ac as well.
Depends on:
1) latest version of the CX20442 codec driver that exposes v253_ops
structure[1],
2) patch 2/3 form this series: TTY: Add definition of a new line
discipline required by Amstrad E3 (Delta) ASoC driver[2].
CPU DAI parameters best matching the codec DAI has been selected out
empirically for best user experience.
Board specific audio function control (with related DAPM widgets) has been
modeled after empirically discovered codec capabilities.
Unlike other ASoC machine drivers, this one makes use of a codec provided line
discipline that is required for talking to a modem chip that can control the
codec behavoiur. As the line discipline operations must call board specific
bits as well, the machine driver registers its own line discipline ops, not
the codec provided, and then calls those codec provided from inside its own
callbacks.
If some kind of a glue, like a bus over a tty, exsited that could help in
runtime detection of a modem (bus adapter) over a more generic line discipline
(bus driver)[3], the line discipline code could be probably designed in a
more generic way.
In order to work at all, this driver requires a working McBSP1. On OMAP1510
based machines (not sure if other OMAP1 variants as well), where McBSP1 is a
DSP public peripheral, that means the kernel must provide basic DSP support,
ie. omap_dsp_init(), in order to power up the DSP. This used to be included in
linux-omap-2.6 tree up to commit 2512fd29db4eb09e82d182596304c7aaf76d2c5c.
Without that, the driver would not work, ie. not shift in/out any bits over
the CPU DAI[4]. This limitation is not board, but CPU specific, and may apply
to other code that makes use of McBSP1/McBSP3 on affected machines. I provide
an extra patch (4/3) as a temporary solution.
To work correctly in playback mode, this driver requires my prevoiusly
submitted patch that corrects pcm pointer calculation for OMAP1510 based
machines[5] (already included in linux-2.6.31-rc3).
To support codec controls, this driver requires my previously submitted patch
that adds support for modem found on Amstrad Delta[6].
[1] http://mailman.alsa-project.org/pipermail/alsa-devel/2009-July/019780.html
[2] http://www.spinics.net/lists/linux-serial/msg01862.html
[3] http://www.spinics.net/lists/linux-serial/msg01856.html
[4] http://www.spinics.net/lists/linux-omap/msg15114.html
[5] http://mailman.alsa-project.org/pipermail/alsa-devel/2009-June/018950.html
[6] http://www.spinics.net/lists/linux-omap/msg15432.html
Credits to:
Mark Underwood - for his initial, omap-alsa based sound driver for
this machine,
Mark Brown - for his help, patience and excellent subsytem maintainer support.
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This corrected patch adds machine independent line discipline code, prevoiusly
exsiting inside my Amstrad Delta ASoC machine dirver, to the Conexant CX20442
codec driver. The code can be used as a standalone line discipline, or as a
set of codec specific functions called from machine's line discipline
callbacks. Anyway, the line discipline itself must be registered by a machine
driver.
Applies on top of the followup to my initial driver version:
http://mailman.alsa-project.org/pipermail/alsa-devel/2009-July/019757.html
Suggested by ASoC manintainer Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The irq can fire as soon as it has been requested, thus all fields accessed
from within the irq handler must be initialized prior to requesting the irq.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This helps CODECs with sparse register maps work better with the
register cache display interface.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
1. fix "line over 80 characters" checkpatch warnings
2. ‘DMA_nnBIT_MASK’ is deprecated, use DMA_BIT_MASK instead
3. fix typos
Signed-off-by: Barry Song <21cnbao@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
With the s3c platform has implementing gpiolib support the s3c_gpio api has been
deprecated.
This patch gets rid of all s3c_gpio calls and replaces them by using gpiolib.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The I2S DAI driver for blackfin SPORT, but works in TDM mode.
I2S is not a special case of TDM with only left and right two slots for
SPORT interface. I2S coordinates with TDM in SPORT, but not a part of
TDM. TDM require different hardware configuration with I2S, not only
different slot number. One is "Stereo Serial Operation" mode of SPORT,
the other one is "Multichannel Operation" mode. They are incompatible
at the same time.
Hardware and DMA description and data transfer flow are much different
for I2S and TDM. Merging them as a whole will be very ugly and difficult
to maintain.
So we don't define a new DAI type, but give two DAI instances for standard
I2S and TDM, both in I2S-family DAI type. The TDM instance still uses the
I2S-family DAI type.
Signed-off-by: Barry Song <21cnbao@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The patch fixes some checkpatch identified issues and adds a comment about
line discipline interaction to my driver code, as requested by Mark on my
inital submission (thank you Mark for applying my imperfect patch anyway).
It also fixes MODULE_ALIAS mismatch as used in my machine driver.
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch removes the old method of jack detection from palm27x-asoc
driver and adds jack detection api. It also removes some other (now)
useless stuff from the driver and corrects pin configuration for the
codec.
Signed-off-by: Marek Vasut <marek.vasut@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The patch adds a few small enhancements to the ASoC jack handling, as
suggested by Mark in his comments to my Amstrad Delta driver, and a few fixes
for related bugs found while learning Mark's code and testing results.
Enhancements:
1. Update status of an ASoC jack while associating it with new gpios.
2. Really update DAPM pins while associating them with an ASoC jack.
3. Export ASoC jack gpios over gpiolib sysfs for diagnostic purposes.
Fixes:
1. Apply mask on jack status report before using it, just for case.
2. While updating jack associated DAPM pins, use full resulting jack status,
not the status report passed as an argument.
Created and tested on linux-2.6.31-rc3
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch allows passing platform_data to devices attached to AC97 bus
(like touchscreens, battery measurement chips ...).
Signed-off-by: Marek Vasut <marek.vasut@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The MAX9877 needs an address of start register when we write values to
registers through i2c_master_send(), but the code for this was missed in
max9877_write_regs().
If the value of control is 0 in the max9877_set_out_mode(), the value is
not increased to 1, but actually the value to write to the register
should be 1.
And the register bits for out_mode and osc_mode should be cleared before
writing.
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds support for Conexant CX20442-11 voice modem codec, suitable
for use by the ASoC board driver for Amstrad E3 (Delta) videophone. Related
sound card driver will follow.
This codec is an optional part of the Conexant SmartV three chip modem design.
As such, documentation for its proprietary digital audio interface is not
available. However, on Amstrad Delta board, thanks to Mark Underwood who
created an initial, omap-alsa based sound driver a few years ago[1], the codec
has been discovered to be accessible not only from the modem side, but also
over the OMAP McBSP based CPU DAI. Thus, the driver can be used by any sound
card that can access the codec DAI directly. The DAI configuration parameters
(sample rate and format, number of channels) has been selected out empirically
for best user experience.
The codec analogue interface consists of two pairs of analogue I/O pins:
speakerphone interface or telephone handset/headset interface. Furthermore, it
seams to provide two operation modes for speakerphone I/O: standard and
advanced, with automatic gain control and echo cancelation. Even if the codec
control interface is unknown and not available, all those interfaces and modes
can be selected over the modem chip using V.253 commands. The driver is able
to issue necessary commands over a suitable hw_write function if provided by a
sound card driver. Otherwise, the codec can be controlled over the modem from
userspace while inactive.
Even if nothig is known about the codec internal power management
capabilities, DAPM widgets has been used to model the codec audio map.
Automatically performed powering up/down of those virtual widgets results in
corresponding V.253 commands being issued.
Some driver features/oddities may be board specific, but I have no way to
verify that with any board other than Amstrad Delta.
[1] http://www.earth.li/pipermail/e3-hacking/2006-April/000481.html
Created and tested against linux-2.6.31-rc3.
Applies and works with linux-omap-2.6 commit
7c5cb7862d32cb344be7831d466535d5255e35ac as well.
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
PLL was not being enabled when it was not bypassed. This patch
enables the PLL when it is used. Additionally, it disables the PLL
when it is bypassed.
Without this patch, the audio on TI DM646x EVM and DM355 EVM
does not work properly. The bit clocks and the frame sync signals
from the codec are not correct and hence the playback/record are faster
than usual for most sample rates. The reason for this was that the PLL
was not enabled when it was not bypassed.
Tested on DM6467 EVM, playback tested on DM355 EVM.
Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The callback function to control register was used by whole controls in
MAX9877 driver, but this causes using many if statement for double
register control or invert.
So, the callback function for double register control is separate
differently, and the code for invert is added in the callback function.
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This corrects a bug with ADC Inversion Switch in wm8974 codec.
Signed-off-by: Javier Martin <javier.martin@vista-silicon.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Reset was failing with the original udelay(50) between the code in
psc_ac97_cold_reset() and the call to psc_ac97_warm_reset(). Through testing
it was found that a delay of 1ms was necessary for the cold_reset code to
consistently complete successfully.
Signed-off-by: John Bonesio <bones@secretlab.ca>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
GCC 4.4.0 doesn't appear to be able to spot that we don't apply any FLL
configuration if the output frequency is zero.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Board sdp3430 has hardware support for EXTMUTE using TWL4030 GPIO6
line, controlled by register INTBR_PMBR1. Machine driver takes care
of enabling gpio line through i2c and codec driver manipulates the
line during headset ramp up/down sequence.
Signed-off-by: Jorge Eduardo Candelaria <x0107209@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
clock name strings are no longer passed on platform_data. Instead,
we rely entirely on struct device and clkdev to find the right clock.
Signed-off-by: Kevin Hilman <khilman@deeprootsystems.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The MAX9877 combines a high-efficiency Class D audio power amplifier
with a stereo Class AB capacitor-less DirectDrive headphone amplifier.
The max9877_add_controls() is called to register the MAX9877 specific
controls on machine specific init() of the machine driver.
The datasheet for the MAX9877 can find at the following url:
http://datasheets.maxim-ic.com/en/ds/MAX9877.pdf
[Slight edit to sort the ALL_CODECS entries -- broonie.]
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Due to the flexibility of the WM9081 FLL this should never happen
in a real system.
Reported-by: Jaswinder Singh Rajput <jaswinder@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We need to use the best value we picked, not the last value we
looked at.
Reported-by: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There is no need to manually start playback/capture ourselves as the PCM
driver will handle things for us.
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There is no need to manually start playback/capture ourselves as the PCM
driver will handle things for us.
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Without MODULE_LICENCE("GPL"), when built as a module it will fail
to load because it uses other GPL symbols from kernel.
Signed-off-by: Kevin Hilman <khilman@deeprootsystems.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In SOC DAPM layer of SOUND subsystem, when add signal route (in the
function snd_soc_dapm_add_route() ), the original code has wrong logic
when dapm layer check each widget whether an external one.
Signed-off-by: Rongrong Cao <rrcao@ambarella.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is the last in-kernel direct usage of driver_data, replace it with
the proper dev_get/set_drvdata() calls.
Cc: Takashi Iwai <tiwai@suse.de>
Cc: Jaroslav Kysela <perex@perex.cz>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
The WM8731 driver has been updated to allow registration via normal
device model methods rather than from within the ASoC driver probe
so update the AT91SAM9G20-EK to make use of this.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Andrew Victor <linux@maxim.org.za>
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Provide an interval after the end of DAPM sequencing so that we
can distinguish between a pop in the final step of the sequence
and a pop generated from some other source outside DAPM.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The system clock is currently fixed by the driver and this avoids
the need for us to handle errors with enabling and disabling MCLK
(which was incorrect previously so this fixes bugs in error
handling).
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
As shipped the board does not have inputs but it is relatively
straightforward to modify the board to hook them up so support
is provided in the driver. When these modifications have not
been made enabling the microphone stage can cause problems.
Add an ifdef to disable this by default. Don't put it into
Kconfig since users will have to get their soldering irons
out to change things.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This simplifies the driver by removing the need to manually
configure dividers within the CPU and improve audio performance
by ensuring that the optimal phase relationships between the
clocks in the system are maintained.
Note that currently this means that for playback to work the
Output Mixer HiFi switch must be enabled since otherwise CODEC
will not generate the DAC clock.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
While the hardware is capable of some limited asynmmetric modes the
driver does not currently support those modes so tell applications
that only symmetric rates are available.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM8993 is a highly integrated ultra-low power hi-fi CODEC designed
for portable devices such as multimedia phones.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Update the dma link with correct data as soon as
the master channel has copied it. Otherwise, the
1st period will play twice.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
