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Unsigned variables should use `%u' rather than `%d'.
Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Use default to optimize the switch/case in magicial_playback_hw_params(),
which also fixes the compile warnings below:
sound/soc/pxa/magician.c:89: warning: 'acds' may be used uninitialized in this function
sound/soc/pxa/magician.c:89: warning: 'acps' may be used uninitialized in this function
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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pxa2xx_i2s_resume is :
- unconditionnaly setting SACR0_ENB
- unsetting SACR0_ENB in saved SACR0 pxa_i2s.sacr0
fix these.
In pxa2xx_i2s_{resume,suspend}, save/restore registers even
when !dai->active.
Signed-off-by: Karl Beldan <karl.beldan@mobile-devices.fr>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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i2s_clk is 'put' for no reason in pxa2xx_i2s_shutdown.
Now we 'get' i2s_clk at probe and 'put' it at driver removal or when
probe fails.
Signed-off-by: Karl Beldan <karl.beldan@mobile-devices.fr>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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- hw_params enables both RPL and REC functions each time : Enable the
appropriate function in pxa2xx_i2s_trigger.
- pxa2xx_i2s_shutdown disables i2s anytime one of RPL or REC function is
off : Turn it off only when both functions are off.
Signed-off-by: Karl Beldan <karl.beldan@mobile-devices.fr>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Make sure we are in a know good state at end of probe :
Reset FIFO logic and registers, and make sure REC and RPL functions
along with FIFO service are disabled (SACR0_RST enables REC and RPL).
Resetting loses current settings so remove reset from stream startup.
Now reset occurs only at probe.
Signed-off-by: Karl Beldan <karl.beldan@mobile-devices.fr>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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There is a single I2S_SYNC pin on the chip.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Signed-off-by: Mike Rapoport <mike@compulab.co.il>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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This patch adds the ASoC side of the board support for the Crossbow
IMB400 daughter board.
Thanks to Crossbow for considerable assistance.
Signed-off-by: Jonathan Cameron <jic23@cam.ac.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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* master.kernel.org:/home/rmk/linux-2.6-arm:
[ARM] 5460/1: Orion: reduce namespace pollution
[ARM] 5458/1: pcmcia: pxa2xx-sharpsl: check if we do have Scoop config
[ARM] 5457/1: mach-imx gpio buildfix
[ARM] 5456/1: add sys_preadv and sys_pwritev
[ARM] pxa/pcm990: start external GPIOs immediately after built-in ones
[ARM] pxa/palm27x: General fix for Palm27x aSoC driver
[ARM] pxa/mioa701: use GPIO95 as AC97 reset line
[ARM] pxa: merge AC97 platform data structures
[ARM] pxa/magician: remove un-necessary #include of pxa-regs.h and hardware.h
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The original idea came from pHilipp, and this makes the code looks
more consistent.
Signed-off-by: Eric Miao <eric.miao@marvell.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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The SSP DMA parameters can actually be easily generated at run-time since
they are almost similar except for the FIFO width and direction. Another
benefit is the re-use of information from 'struct ssp_device', like SSDR
physical FIFO address and DRCMR register index for both directions.
Signed-off-by: Eric Miao <eric.miao@marvell.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Reviewed-by: pHilipp Zabel <philipp.zabel@gmail.com>
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* fix/asoc:
ASoC: Fix warning in wm9705
ASoC: OMAP: Update contact addresses
ASoC: pxa-ssp: Don't use SSCR0_SerClkDiv and SSCR0_SCR
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git://git.kernel.org/pub/scm/linux/kernel/git/ycmiao/pxa-linux-2.6
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Firstly, this patch makes the palm27x asoc driver a little more sane. Also,
since all affected devices use GPIO95 as AC97_nRESET, this patch sets that
properly. Affected are PalmT5, TX and LifeDrive.
Signed-off-by: Marek Vasut <marek.vasut@gmail.com>
Signed-off-by: Eric Miao <eric.miao@marvell.com>
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* fix/asoc:
ASoC: OMAP: Fix FS polarity in OSK5912 machine driver
ASoC: OMAP: Fix DSP_B format in OMAP McBSP DAI driver
ASoC: Fix include build error in s3c2412-i2s.c
ASoC: Fix s3c-i2s-v2.c snd_soc_dai changes
ASoC: s3c-i2s-v2.c fix for s3c_i2sv2_iis_calc_rate
ASoC: Fix jive_wm8750.c build problems
ASoC: pxa-ssp: allow setting of dai format 0
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Those macros are just screwed as soon as CONFIG_PXA25x is enabled.
This patch
- changes ssp_set_scr to take an ssp_dev pointer instead of ssp_device
- adds a corresponding ssp_get_scr function.
Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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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>
Cc: Mark Brown <broonie@opensource.wolfsonmicro.com>
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git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into topic/asoc
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SCMODE(0): Data Driven (Falling), Data Sampled (Rising), Idle State (Low)
SCMODE(1): Data Driven (Rising), Data Sampled (Falling), Idle State (Low)
SCMODE(2): Data Driven (Rising), Data Sampled (Falling), Idle State (High)
SCMODE(3): Data Driven (Falling), Data Sampled (Rising), Idle State (High)
SCMODE(3) does not invert the clock polarity compared to the default SCMODE(0).
This patch also adds all possible NF/IF, NB/IB combinations to the DSP_A and
DSP_B modes.
Signed-off-by: Daniel Ribeiro <drwyrm@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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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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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: (36 commits)
ALSA: hda - Add VREF powerdown sequence for another board
ALSA: oss - volume control for CSWITCH and CROUTE
ALSA: hda - add missing comma in ad1884_slave_vols
sound: usb-audio: allow period sizes less than 1 ms
sound: usb-audio: save data packet interval in audioformat structure
sound: usb-audio: remove check_hw_params_convention()
sound: usb-audio: show sample format width in proc file
ASoC: fsl_dma: Pass the proper device for dma mapping routines
ASoC: Fix null dereference in ak4535_remove()
ALSA: hda - enable SPDIF output for Intel DX58SO board
ALSA: snd-atmel-abdac: increase periods_min to 6 instead of 4
ALSA: snd-atmel-abdac: replace bus_id with dev_name()
ALSA: snd-atmel-ac97c: replace bus_id with dev_name()
ALSA: snd-atmel-ac97c: cleanup registers when removing driver
ALSA: snd-atmel-ac97c: do a proper reset of the external codec
ALSA: snd-atmel-ac97c: enable interrupts to catch events for error reporting
ALSA: snd-atmel-ac97c: set correct size for buffer hardware parameter
ALSA: snd-atmel-ac97c: do not overwrite OCA and ICA when assigning channels
ALSA: snd-atmel-ac97c: remove dead break statements after return in switch case
ALSA: snd-atmel-ac97c: cleanup register definitions
...
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Replace all DMA_32BIT_MASK macro with DMA_BIT_MASK(32)
Signed-off-by: Yang Hongyang<yanghy@cn.fujitsu.com>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
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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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Conflicts:
sound/soc/pxa/pxa2xx-i2s.c
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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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pxa-regs.h and hardware.h are not intended for use directly in driver
code, remove those unnecessary references.
Signed-off-by: Eric Miao <eric.miao@marvell.com>
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1. Driver code where pxa_request_dma() is called will most likely
reference DMA registers as well, and it is really unnecessary
to include pxa-regs.h just for this. Move the definitions into
<mach/dma.h> and make relevant drivers include it instead of
<mach/pxa-regs.h>.
2. Introduce DMAC_REGS_VIRT as the virtual address base for these
DMA registers. This allows later processors to re-use the same
IP while registers may start at different I/O address.
Signed-off-by: Eric Miao <eric.miao@marvell.com>
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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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