ASoC: fsi: merge fsi_data_push/pop to fsi_fifo_data_ctrl
Current FSI driver had data push/pop functions.
But the main operation of these 2 were very similar.
This mean it is possible to merge these to 1 function.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The sh/siu ASoC driver doesn't compile because of a function defined static in
the source and extern in a header. Remove the unneeded declaration in the
header.
Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The clkdev API doesn't use .name and .id members of struct clk for clock
lookup. Instead clocks should be added to a lookup list. Without this patch
audio om the Migo-R board fails silently.
Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Dzianis Kahanovich <mahatma@eu.by>
[Modified to move the location of the table] Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM8985 is a low power, high quality, feature-rich stereo
CODEC designed for portable multimedia applications that
require low power consumption and high quality audio.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Jarkko Nikula [Tue, 14 Sep 2010 12:24:38 +0000 (15:24 +0300)]
ASoC: Remove needless codec->bias_level assignment to SND_SOC_BIAS_OFF
This assignment is done by the snd_soc_register_codec so there is no need
to redo it in probe function of a codec driver.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch modify FIFO_DIPSTICK value of PCM TX FIFO to be a optimal one.
Privious value (0x20) did not support 'Almost_full' of PCM FIFO for the DMA
request.
Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> Acked-by: Jassi Brar <jassi.brar@samsung.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When PCM capture, sound recorded abnormally because of RX FIFO
threshold settings are missing. So, This patch modify PCM RX FIFO
setting codes same as TX.
And for DMA, if PCM RXFIFO_DIPSTICK is not '0', it doesn't effect
to DMA request, because DMA refer RX_FIFO_EMPTY flag as the DMA
request.
Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> Acked-by: Jassi Brar <jassi.brar@samsung.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The reg_cache_size is the number of elements in the register cache,
not the size of the cache itself. This is not a problem if the size
of each element of the cache is 1 byte but it matters in any other
case.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Axel Lin [Thu, 9 Sep 2010 09:50:56 +0000 (17:50 +0800)]
ASoC: ad1980 - set reg_cache_default to ad1980_reg
Signed-off-by: Axel Lin <axel.lin@gmail.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch solve below report from Guennadi.
But I didn't remove #include <sound/sh_fsi.h>.
Because it have FSI_PORT_B define which is used on this file.
Can it actually do mono? Maybe at probe time you could look at audio flags
from your previous patch and, e.g., for SPDIF set channels_min to 2?
> + .channels_max = 2,
That's the "smallest max," yes. With some other interfaces (I2S, DSD) it
can support up to 8 channels...
> + .rates = SNDRV_PCM_RATE_8000_48000,
Hm, in the datasheet I see supported frequencies 32kHz to 192kHz. And if
you promise support for multiple frequencies, don't you want to implement
.hw_params? Besides, not all of these frequencies will be available,
depending on your supplied clock and your willingness to implement
downsampling.
Reported-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Reviewed-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ASoC: fsi codecs: modify menu attribute on Kconfig
Current SND_FSI_xxx menu attributes were bool,
but it should be tristate.
This patch solve below report from Guennadi
"bool" means, if someone is linking the whole ASoC into the kernel, they
will not be able to build this as a module. Not a big deal, but you're
stealing some freedom from the user.
Reported-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Mark Brown [Tue, 7 Sep 2010 14:53:35 +0000 (15:53 +0100)]
ASoC: Remove export of CS4270 DAI
Not needed with multi-component.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Timur Tabi <timur@freescale.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Julia Lawall [Tue, 31 Aug 2010 15:48:56 +0000 (17:48 +0200)]
ASoC: pl022_ds.c: Add of_node_put to avoid memory leak
Add a call to of_node_put in the error handling code following a call to
of_parse_phandle.
This patch also moves the existing call to of_node_put tothe end of the
error handling code, to make it possible to jump to of_node_put without
doing the other cleanup operations. These appear to be disjoint
operations, so the ordering doesn't matter.
The semantic match that finds this problem is as follows:
(http://coccinelle.lip6.fr/)
*x =
(of_find_node_by_path
|of_find_node_by_name
|of_find_node_by_phandle
|of_get_parent
|of_get_next_parent
|of_get_next_child
|of_find_compatible_node
|of_match_node
|of_find_node_by_type
|of_find_node_with_property
|of_find_matching_node
|of_parse_phandle
)(...);
...
if (x == NULL) S
<... when != x = E
*if (...) {
... when != of_node_put(x)
when != if (...) { ... of_node_put(x); ... }
(
return <+...x...+>;
|
* return ...;
)
}
...>
(
E2 = x;
|
of_node_put(x);
)
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk> Acked-by: Timur Tabi <timur@freescale.com> Acked-by: Liam Girdwood <lrg@slimlogic.uo.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Ian Lartey [Tue, 31 Aug 2010 22:56:34 +0000 (23:56 +0100)]
ASoC: Added a missing 32-bit PCM format, to the wm8994 codec.
Signed-off-by: Ian Lartey <ian@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Mark Brown [Tue, 31 Aug 2010 18:39:48 +0000 (19:39 +0100)]
ASoC: Implement WM8994 DAC data source muxes
Allow selection of the channel used for input to the AIFnDAC signals.
This isn't integrated into DAPM since we treat the data as a single
mono channel until just beyond this selection so it ends up having
no visible effect on the routing.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
ASoC: uda1380: make driver more powersave-friendly
Disable some codec modules in standby mode, completely disable
codec in off mode to save some power.
Fix suspend/resume: mark mixer regs as dirty on resume to
restore mixer values, otherwise driver produces no sound
(master is muted by default).
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com> Acked-by: Marek Vasut <marek.vasut@gmail.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Jarkko Nikula [Tue, 31 Aug 2010 11:52:35 +0000 (14:52 +0300)]
ASoC: Swap bias level enumeration
Swapping the bias level enumeration is only meant to help debugging. It is
easier if number 0 means bias off and bigger number means bigger bias level.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Manuel Lauss [Thu, 26 Aug 2010 12:53:51 +0000 (14:53 +0200)]
ASoC: fix au1x platform
This patch fixes up the au1x audio platform after the multi-component
merge:
- compile fixes and updates to get DB1200 platform audio working again,
- removal of global variables in AC97/I2S/DMA(PCM) modules.
The AC97 part is limited to one instance only for now due to issues
with getting at driver data in the soc_ac97_ops.
Signed-off-by: Manuel Lauss <manuel.lauss@googlemail.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Jarkko Nikula [Fri, 27 Aug 2010 13:56:50 +0000 (16:56 +0300)]
ASoC: tlv320aic3x: Sanitize output controls
Currently output controls are not uniform. Some routes are adjusted by
mono controls that don't match to associated mixer switch, many routes are
not covered at all and stereo controls have following variants:
This patch attempts to fix these issues. First, for the convenience, only
direct L-to-L, R-to-R and [L | R]-to-Mono routes are controlled by the
stereo controls. This logic is also used with the output pin mute controls
so all of them except mono output are controlled by stereo switches.
Then rest of the swapped L-to-R and R-to-L routes are controlled by the
mono controls that map to mixer switches with a same name. Mixers can then
associate these switches and volumes together.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Jarkko Nikula [Fri, 27 Aug 2010 13:56:49 +0000 (16:56 +0300)]
ASoC: tlv320aic3x: Reimplement output mixers
It turned out that the output mixers and their routes were misdefined: They
are not mixing output pins to internal signals but opposite. This has worked
for direct left-to-left and right-to-right routes since for those there are
complete routes. For swapped left-to-right and right-to-left routes this is
not working since there are no routes defined between them.
Another consequence is that those misdefined mixers are incorrectly routed
to several output pins leading unnecessary pin powerings even if there is no
route active to them.
Fix these by reimplementing the output mixers and routes as they are in
hardware. For completeness add also a few missing links between internal
signals and outputs.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Jarkko Nikula [Fri, 27 Aug 2010 13:56:48 +0000 (16:56 +0300)]
ASoC: tlv320aic3x: Sort output pin control registers in header file
Each output pin has 7 consecutive control registers in tlv320aic3x register
map. First 6 of them control the signal mixing and one is for output level
and power control.
Sort these registers as they are sorted clearly in hardware, it makes also
definitions more readable and easier to pinpoint missing register
definitions.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Bit 3 in output pin_CTRL register mutes the whole output pin not just the
route from DAC so remove misleading DAC from control name. Currently only
"Line[L | R] Playback Switch" were correct.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Ian Lartey [Fri, 27 Aug 2010 14:26:27 +0000 (15:26 +0100)]
ASoC: Complete supported clock ratios and rate constraints for wm8741
Signed-off-by: Ian Lartey <ian@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
I think this is a typo, debugfs_pop_time should not be executable.
Signed-off-by: Axel Lin <axel.lin@gmail.com> Acked-by: Liam Girdwood <lrg@slimloogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Axel Lin [Wed, 25 Aug 2010 08:59:11 +0000 (16:59 +0800)]
ASoC: pxa-ssp: fix a memory leak in pxa_ssp_remove()
The "priv" allocated in pxa_ssp_probe() should be kfreed in pxa_ssp_remove().
Signed-off-by: Axel Lin <axel.lin@gmail.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Ian Lartey [Fri, 20 Aug 2010 16:18:45 +0000 (17:18 +0100)]
ASoC: Enable autoloading of pxa2xx CPU I2S driver with module alias
Signed-off-by: Ian Lartey <ian@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Ian Lartey [Fri, 20 Aug 2010 16:18:44 +0000 (17:18 +0100)]
ASoC: Make codec dai naming for WM8741 consistent
Signed-off-by: Ian Lartey <ian@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Ian Lartey [Fri, 20 Aug 2010 16:18:43 +0000 (17:18 +0100)]
ASoC: pxa2xx-i2s is the proper name of the I2S DAI, not pxa-i2s.
Signed-off-by: Ian Lartey <ian@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Jarkko Nikula [Mon, 23 Aug 2010 07:35:05 +0000 (10:35 +0300)]
omap: rx51: Remove redefinition of rx51_aic3x_data
Redefinition of rx51_aic3x_data got introduced when merging upstread changes
for 2.6.36 and ASoC Multi-Component Support.
This must go together with multi-component commit f0fba2a.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Cc: Tony Lindgren <tony@atomide.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Jarkko Nikula [Mon, 23 Aug 2010 07:36:41 +0000 (10:36 +0300)]
ASoC: omap: rx51: Use gpio_set_value_cansleep for speaker amp control
Speaker amplifier is controlled by TWL4030 GPIO which may sleep. Therefore
use gpio_set_value_cansleep to get rid of runtime warning that is introduced
after the commit 9c4ba94 and to get a stack trace if ever executing this
code in atomic context.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Jarkko Nikula [Mon, 23 Aug 2010 07:38:40 +0000 (10:38 +0300)]
ASoC: Fix tlv320aic3x GPIO initialization
aic3x_init does a soft reset first and thus TLV320AIC3x GPIO setup must be
done after doing the basic init. Before multi-component the init was done
at i2c probe time and GPIO setup at soc probe time.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Timur Tabi [Thu, 19 Aug 2010 20:26:58 +0000 (15:26 -0500)]
ASoC: mpc8610: replace of_device with platform_device
'struct of_device' no longer exists, and its functionality has been merged
into platform_device. Update the MPC8610 HPCD audio drivers (fsl_ssi, fsl_dma,
and mpc8610_hpcd) accordingly.
Also add a #include for slab.h, which is now needed for kmalloc and kfree.
Signed-off-by: Timur Tabi <timur@freescale.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is an ALSA codec for the Texas Instruments WL1273 FM Radio.
Signed-off-by: Matti J. Aaltonen <matti.j.aaltonen@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Timur Tabi [Thu, 19 Aug 2010 22:11:40 +0000 (17:11 -0500)]
ASoC: add support for the Freescale P1022 DS reference board
The Freescale P1022 is a dual-core e500-based SOC with multimedia capabilities,
specifically the same SSI audio controller on the MPC8610. The P1022 DS
reference board includes a P1022 and a Wolfson Microelectronics WM8776
codec.
Signed-off-by: Timur Tabi <timur@freescale.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Axel Lin [Fri, 20 Aug 2010 05:23:36 +0000 (13:23 +0800)]
ASoC: remove include of pxa2xx-pcm.h in pxa2xx-ac97.c
Fix reference to moved header file, which was unused anyway.
This change fixes below build error:
CC sound/soc/pxa/pxa2xx-ac97.o
sound/soc/pxa/pxa2xx-ac97.c:27:24: error: pxa2xx-pcm.h: No such file or directory
make[3]: *** [sound/soc/pxa/pxa2xx-ac97.o] Error 1
make[2]: *** [sound/soc/pxa] Error 2
make[1]: *** [sound/soc] Error 2
make: *** [sound] Error 2
Signed-off-by: Axel Lin <axel.lin@gmail.com> Signed-off-by: Haojian Zhuang <haojian.zhuang@gmail.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Randolph Chung [Fri, 20 Aug 2010 04:47:53 +0000 (12:47 +0800)]
ASoC: Add support for tlv320aic3007 to tlv320aic3x codec.
This patch adds support for the tlv320aic3007 codec to the tlv320aic3x
driver.
The tlv320aic3007 is similar to the aic31, but has an additional class-D
speaker amp. The speaker amp control register overlaps with the mono
output register of other codecs in this family, so we add logic to
identify the actual codec being registered to set things up accordingly.
Signed-off-by: Randolph Chung <tausq@parisc-linux.org> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Timur Tabi [Thu, 19 Aug 2010 21:43:42 +0000 (16:43 -0500)]
ASoC: add support for separate codec DAIs to the fsl_dma driver
Some codecs have separate DAIs for playback and capture, so the DMA driver
should allocate a DMA buffer only for the streams that are valid when the
driver is opened.
Signed-off-by: Timur Tabi <timur@freescale.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Axel Lin [Thu, 19 Aug 2010 06:40:25 +0000 (14:40 +0800)]
ASoC: e740_wm9705 - free gpio in e740_exit()
In e740_init(), we call gpio_request() for
GPIO_E740_MIC_ON, GPIO_E740_AMP_ON and GPIO_E740_WM9705_nAVDD2.
We should free the these gpio accordingly in e740_exit().
Signed-off-by: Axel Lin <axel.lin@gmail.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ASoC: Use a more adequate name for the CX20442 codec DAI
In the process of unification of codec DAI names while implementing
multi-component, the CX20442 codec DAI has been renamed to "cx20442-hifi".
This new name seems not adequate for a 8kHz voice codec.
Use a better name, "cx20442-voice", as suggested by Liam Girdwood.
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Timur Tabi [Fri, 6 Aug 2010 17:16:12 +0000 (12:16 -0500)]
asoc/multi-component: fsl: add support for variable SSI FIFO depth
Add code that programs the DMA and SSI controllers differently based on the
FIFO depth of the SSI.
The SSI devices on the MPC8610 and the P1022 are identical in every way except
one: the transmit and receive FIFO depth. On the MPC8610, the depth is eight.
On the P1022, it's fifteen. The device tree nodes for the SSI include a
"fsl,fifo-depth" property that specifies the FIFO depth.
Signed-off-by: Timur Tabi <timur@freescale.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>