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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>
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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>
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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>
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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>
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Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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This also simplifies the code a bit.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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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>
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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>
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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>
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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>
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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>
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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>
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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>
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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>
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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>
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We always run in the first timeslot of one.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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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>
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The snd_soc_codec was moved into socdev->card, but this change wasn't
applied in some places. Fixed now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This removes the calls to pxa_gpio_mode from the pxa2xx-i2s driver.
Pin setup should be done during board init via pxa2xx_mfp_config
instead.
Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Acked-by: Eric Miao <eric.miao@marvell.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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This machine driver enables sound functions on Mitac mio
a701 smartphone. Build upon ASoC v1, it handles :
- rear speaker
- front speaker
- microphone
- GSM
A global "Mio Mode" switch is not yet provided to cope with
audio path setup. As balance on audio chip line is no more
assured, an incorrect setup can produce a lot of heat and
even fry the battery behind the wm9713 and the speaker
amplifier.
It doesn't cope with :
- headset jack
- mio master mode
- master volume control
This driver is backported from ASoc v2, and amputated from
scenario setups and master volume control.
[Minor mods for terminology in comments -- broonie]
Signed-off-by: Robert Jarzmik <robert.jarzmik@free.fr>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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PXA2xx/3xx SSP ports start from 1, not 0. Thus, the probe function
requested the wrong SSP port. Correcting this unveiled another bug
where ssp_init tries to request the already-requested SSP port again.
So this patch replaces the ssp_init/exit calls with their internals
from mach-pxa/ssp.c, leaving out the redundant ssp_request and the
unneeded IRQ request. Effectively, that leaves us with not much more
than enabling/disabling the SSP clock.
Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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pxa-regs.h and hardware.h are not intended for use directly in driver
code and references to them have been removed in other code - remove
them from the newly added e740 and e750 machine drivers.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Fix a merge issue caused by context overlap.
Reported-by: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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This patch provides suupport for the wm9705 AC97 codec on the Toshiba e740.
Note:
The e740 has a hard headphone switch that turns the speaker off and is not
software detectable or controlable. Also both headphone and speaker amps
share a common output enable.
Signed-off-by: Ian Molton <ian@mnementh.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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The Zylonite supports switching the MCLK for the WM9713 between the
AC97CLK and CLK_POUT outputs of the PXA processor via switch SW15 on
the board. This patch adds support for configuring the system to use
CLK_POUT.
Unfortunately it is not possible to read the state of SW15 from software
so this feature is controlled by a module option 'clk_pout' which should
be set to a non-zero value to enable the use of CLK_POUT.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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This patch adds support for the wm9712 ac97 codec as used in the Toshiba e800
PDA. It includes support for powering up / down the external headphone and
speaker amplifiers on this machine.
Signed-off-by: Ian Molton <ian@mnementh.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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This patch adds support for the wm9705 ac97 codec as used in the Toshiba e750
PDA. It includes support for powering up / down the external headphone and
speaker amplifiers on this machine.
Signed-off-by: Ian Molton <ian@mnementh.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Set the invalid dma channel to -1 (and check properly for it) in
pxa2xx_pcm_hw_free(). Was assuming 0 is an invalid channel number but 0
is a valid pxa dma channel num.
Signed-off-by: stephen <stephen.ware@eqware.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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* 'devel' of master.kernel.org:/home/rmk/linux-2.6-arm: (407 commits)
[ARM] pxafb: add support for overlay1 and overlay2 as framebuffer devices
[ARM] pxafb: cleanup of the timing checking code
[ARM] pxafb: cleanup of the color format manipulation code
[ARM] pxafb: add palette format support for LCCR4_PAL_FOR_3
[ARM] pxafb: add support for FBIOPAN_DISPLAY by dma braching
[ARM] pxafb: allow pxafb_set_par() to start from arbitrary yoffset
[ARM] pxafb: allow video memory size to be configurable
[ARM] pxa: add document on the MFP design and how to use it
[ARM] sa1100_wdt: don't assume CLOCK_TICK_RATE to be a constant
[ARM] rtc-sa1100: don't assume CLOCK_TICK_RATE to be a constant
[ARM] pxa/tavorevb: update board support (smartpanel LCD + keypad)
[ARM] pxa: Update eseries defconfig
[ARM] 5352/1: add w90p910-plat config file
[ARM] s3c: S3C options should depend on PLAT_S3C
[ARM] mv78xx0: implement GPIO and GPIO interrupt support
[ARM] Kirkwood: implement GPIO and GPIO interrupt support
[ARM] Orion: share GPIO IRQ handling code
[ARM] Orion: share GPIO handling code
[ARM] s3c: define __io using the typesafe version
[ARM] S3C64XX: Ensure CPU_V6 is selected
...
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The module init entries should be __init instead of __devinit.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This is done at modprobe time, mirroring current behaviour, except for
mpc5200_psc_i2s where we do registration at the same time as we register
with soc-of-simple. Since the core currently ignores registration this
has no practical impact.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Register all platform DAIs with the core. In line with current behaviour
this is done at module probe time rather than when the devices are probed
(since currently that only happens as the entire ASoC card is registered
except for those drivers that currently implement some kind of hotplug).
Since the core currently ignores DAI registration this has no practical
effect.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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None of the DAIs use it except s3c2412-i2s which only uses it for
dev_() printouts.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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As part of the deprecation of snd_soc_device push the registration of
the platform down into the card structure.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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The optimal change would be to move the AC97 register definitions into
the AC97 driver, unfortunately, the registers are shared between several
files. Move them into a dedicated regs-ac97.h first.
Signed-off-by: Eric Miao <eric.miao@marvell.com>
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Fix the wrong shutdown callback type. Also removed the unused variables
there:
sound/soc/pxa/corgi.c: In function 'corgi_shutdown':
sound/soc/pxa/corgi.c:114: warning: unused variable 'codec'
sound/soc/pxa/corgi.c: At top level:
sound/soc/pxa/corgi.c:175: warning: initialization from incompatible pointer type
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Signed-off-by: Dmitry Baryshkov <dbaryshkov@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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DAI type information is only ever used within ASoC in order to special
case AC97 and for diagnostic purposes. Since modern CPUs and codecs
support multi function DAIs which can be configured for several modes
it is more trouble than it's worth to maintain anything other than a
flag identifying AC97 DAIs so remove the type field and replace it with
an ac97_control flag.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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It's not exported.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Implement support for the Marvell Zylonite PXA3xx reference platform,
supporting standard AC97 stereo and AUX interfaces together with the
auxiliary I2S interface of the WM9713.
The board has two options for the MCLK of the WM9713: either the standard
AC97 system clock can be used or the 13MHz CLK_POUT output of the PXA3xx
can be used, selected via SW15 on the board. Currently only the AC97
system clock is supported by this driver.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Liam Girdwood's ASoC v2 work avoids having two different ops structures
for DAIs by merging the members of struct snd_soc_ops into struct
snd_soc_dai_ops, allowing per DAI configuration for everything.
Backport this change.
This paves the way for future work allowing any combination of DAIs to
be connected rather than having fixed purpose CODEC and CPU DAIs and
only allowing CODEC<->CPU interconnections.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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One of the issues with the ASoC v1 API which has been addressed in the
ASoC v2 work that Liam Girdwood has done is that the ALSA card provided
by ASoC is distributed around the ASoC structures. For example, machine
wide data such as the struct snd_card are maintained as part of the
CODEC data structure, preventing the use of multiple codecs. This has
been addressed by refactoring the data structures so that all the data
for the ALSA card is contained in a single structure snd_soc_card which
replaces the existing snd_soc_machine and snd_soc_device.
Begin the process of backporting this by renaming struct snd_soc_machine
to struct snd_soc_card, better reflecting its function and bringing it
closer to standard ALSA terminology.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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this patch adds asoc audio driver for pxa27x based Palm PDAs. I tested it for
palmtx, t5 and ld, it should work with palmz72 as well (slapin, please test).
I sent it here some time ago, but now I got to fixing bugs in it. It should
be somehow mostly ok and ready for applying.
[Converted to use snd_soc_dapm_nc_pin() and bool Kconfig -- broonie]
Signed-off-by: Marek Vasut <marek.vasut@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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The SSP ports PXA series processors can be used to implement a variety of
audio interface formats. This patch implements support for I2S, DSP A and
DSP B modes on these ports.
This patch is based on the previous out of tree pxa2xx-ssp driver (which
was originally written by Liam Girdwood with updates from Philipp Zabel
and Nicola Perrino) and pxa3xx-ssp driver (originally written by Seth
Forsee based on the pxa2xx-ssp driver). Testing coverage is not complete
currently.
Tested-by: Daniel Ribeiro <drwyrm@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (33 commits)
ALSA: ASoC codec: remove unused #include <version.h>
ALSA: ASoC: update email address for Liam Girdwood
ALSA: hda: corrected invalid mixer values
ALSA: hda: add mixers for analog mixer on 92hd75xx codecs
ALSA: ASoC: Add destination and source port for DMA on OMAP1
ALSA: ASoC: Drop device registration from GTA01 lm4857 driver
ALSA: ASoC: Fix build of GTA01 audio driver
ALSA: ASoC: Add widgets before setting endpoints on GTA01
ALSA: ASoC: Fix inverted input PGA mute bits in WM8903
ALSA: ASoC: OMAP: Set DMA stream name at runtime in McBSP DAI driver
ALSA: ASoC: OMAP: Add support for OMAP2430 and OMAP34xx in McBSP DAI driver
ALSA: ASoC: OMAP: Add multilink support to McBSP DAI driver
ALSA: ASoC: Make TLV320AIC26 user-visible
ALSA: ASoC - clean up Kconfig for TLV320AIC2
ALSA: ASoC: Make WM8510 microphone input a DAPM mixer
ALSA: ASoC: Implement WM8510 bias level control
ALSA: ASoC: Remove unused AUDIO_NAME define from codec drivers
ALSA: ASoC: tlv320aic3x: Use uniform tlv320aic naming
ALSA: ASoC: Add WM8510 SPI support
ALSA: ASoC: Add WM8753 SPI support
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