| Age | Commit message (Collapse) | Author |
|---|
|
ADSP devices report information on the algorithms loaded on them. Parse
this data and use it to allow coefficients to be configured for specific
algorithms.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
The free running mode can cause problems when attempting to bring up the
FLL running from a defined clock source. This patch disables
free-running mode.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
saif->id and saif->master_id are unsigned, so they can not be negative.
Fix the following warning when building with W=1 option:
sound/soc/mxs/mxs-saif.c: In function 'mxs_saif_probe':
sound/soc/mxs/mxs-saif.c:676:2: warning: comparison of unsigned expression < 0 is always false [-Wtype-limits]
sound/soc/mxs/mxs-saif.c:688:3: warning: comparison of unsigned expression < 0 is always false [-Wtype-limits]
sound/soc/mxs/mxs-saif.c:692:2: warning: comparison of unsigned expression < 0 is always false [-Wtype-limits]
Use a signed variable 'ret' to handle the error values.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
All MXS users have been converted to device tree and the board files have been
removed.
No need to keep platform data in the driver.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Acked-by: Dong Aisheng <dong.aisheng@linaro.org>
Acked-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
Otherwise we won't run correctly on systems that require this for larger
data transfers.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
In preparation for more advanced sample rate managment move the existing
code out of the main hw_params() function.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into asoc-arizona
|
|
Some systems use external analogue switches to connect more analogue
devices to the CODEC than are supported by the device. In some systems
this requires changing the switched output from single ended to
differential mode dynamically at runtime. Add a new function
arizona_set_output_mode() to support this.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
|
|
These are not supported
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
|
|
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
These are not supported.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
|
|
These are not supported.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
|
|
The bestcomm dma hardware, and some of its users like the FEC ethernet
component, is used in different FreeScale parts, including non-powerpc
parts like the ColdFire MCF547x & MCF548x families. Don't keep the
driver hidden in arch/powerpc where it is inaccessible for other arches.
.c files are moved to drivers/dma/bestcomm, while .h files are moved to
include/linux/fsl/bestcomm. Makefiles, Kconfigs and #include directives
are updated for the new file locations.
Tested by recompiling for MPC5200 with all bestcomm users enabled.
Signed-off-by: Philippe De Muyter <phdm@macqel.be>
Signed-off-by: Anatolij Gustschin <agust@denx.de>
|
|
Current FSI driver is using platform info pointer,
but it is not good design for DT support.
This patch made it not to use platform info pointer.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
Remove creation of an soc-audio device from the machine platform_driver probe
function, and add a call to snd_soc_register_card() instead.
The current code still works, but this mechanism has been deprecated, if I'm not
mistaken. The ASoC core code produces the warning "ASoC: machine Palm/PXA27x
should use snd_soc_register_card()"
Signed-off-by: Mike Dunn <mikedunn@newsguy.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
ASoC core code now handles creation of controls and routing based on contents of
struct snd_soc_card, so remove calls to snd_soc_dapm_new_controls() and
snd_soc_dapm_add_routes() from the snd_soc_dai_link init function, and add
widget and route definitions to struct snd_soc_card.
Signed-off-by: Mike Dunn <mikedunn@newsguy.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
These boards are using the common omap-twl4030 machine driver, no need for
separate machine drivers anymore.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
Update the common machine driver to support more boards including Zoom2 and
SDP3430.
- Support for voice port of twl4030
- HS jack plug detection support
- The audio routing can be fine tuned via pdata or via provided routing
table from DT.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
The codec driver takes care of this.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
When HS extmute is enabled without custom GPIO we should configure the mux
to allow the pin to be used as extmute signal.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
In order to avoid breakage update the machine drivers at the same time using
twl4030: omap3pandora, sdp3430 and zoom2
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
The codec driver takes care of these bits.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
The codec driver takes care of these bits.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
In order to be able to use the Voice port of twl4030 three bits need to be
handled in VOICE_IF register:
VIF_EN: to enable the voice port (needed for both playback and capture)
VIF_DIN_EN: Need to be enabled for playback only (input to the codec)
VIF_DOUT_EN: Need to be enabled for capture only (output from codec)
Use DAPM_SUPPLY for the VIF_EN bit and add DAPM_AIF_IO/OUT widget to handle
the playback/capture bit.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
Refactor set_pll code to avoid the following warnings:
sound/soc/codecs/wm8985.c:852:50: warning: 'pll_div.k' may be used uninitialized in this function
sound/soc/codecs/wm8985.c:849:9: warning: 'pll_div.n' may be used uninitialized in this function
sound/soc/codecs/wm8985.c:848:23: warning: 'pll_div.div2' may be used uninitialized in this function
Do the same as in commit 86ce6c9a (ASoC: WM8804: Refactor set_pll code to avoid
GCC warnings).
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
When probing aux_dev, initializing is as below:
device_initialize()
device_add()
So when remove aux_dev, we need do as below:
device_del()
device_put()
Otherwise, the rtd_release() will not be called.
So here using device_unregister() to replace device_del(),
like the action in soc_remove_link_dais().
Signed-off-by: liu chuansheng <chuansheng.liu@intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
After called device_initialize(), even device_add() returns
error, we still need use the put_device() to release the reference
to call rtd_release(), which will do the free() action.
Signed-off-by: liu chuansheng <chuansheng.liu@intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
There's no need to test whether a (delayed) work item in pending
before queueing, flushing or cancelling it. Most uses are unnecessary
and quite a few of them are buggy.
Remove unnecessary pending tests from wm8350. Only compile tested.
Signed-off-by: Tejun Heo <tj@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
In the case of no-match, return -EINVAL instead of 0.
Since we assign i to ret in the for loop, ret always less than
ARRAY_SIZE(clk_map_table). Thus remove the boundary checking for ret.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
This connection is used by the AC97 controller.
Signed-off-by: Lucas Stach <dev@lynxeye.de>
Acked-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
AC97 uses a fixed rate, unrelated to the sample rate. Add a function to
make the setup more trivial.
Signed-off-by: Lucas Stach <dev@lynxeye.de>
Acked-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
3449f5fab8c51e37a8a48bc2516588c615373191
(ASoC: fsi: add SND_SOC_DAIFMT_INV_xxx support)
added clock inversion support via snd_soc_dai_set_fmt().
Thus, this patch removed SH_FSI_xxx_INV and fsi_get_info()
from fsi driver, and modified platform settings to use new style.
Then, it cleaned up meaningless settings from platform.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Simon Horman <horms+renesas@verge.net.au>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
ab6f6d85210c4d0265cf48e9958c04e08595055a
(ASoC: fsi: add master clock control functions)
added driver level clock control functions.
And now, platform depended .set_rate() is no longer needed.
This patch removed unnecessary .set_rate() platform callback support.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
Refactor set_pll code to avoid the following warnings:
sound/soc/codecs/wm8983.c:873:40: warning: 'pll_div.k' may be used uninitialized in this function [-Wuninitialized]
sound/soc/codecs/wm8983.c:870:9: warning: 'pll_div.n' may be used uninitialized in this function [-Wuninitialized]
sound/soc/codecs/wm8983.c:869:23: warning: 'pll_div.div2' may be used uninitialized in this function [-Wuninitialized]
Do the same as in commit 86ce6c9a (ASoC: WM8804: Refactor set_pll code to avoid
GCC warnings).
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
The condition "if (!freq_in || !freq_out)" has already been tested previously,
so no need to do it again.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
The CS4271 requires its LRCLK and MCLK to be stable before its RESET
line is de-asserted. That also means that clocks cannot be changed
without putting the chip back into hardware reset, which also requires
a complete re-initialization of all registers.
One (undocumented) workaround is to assert and de-assert the PDN bit
in the MODE2 register.
This patch adds a new flag to both the DT bindings as well as to the
platform data to enable that workaround.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Acked-by: Alexander Sverdlin <subaparts@yandex.ru>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
|
|
Not very useful on non AT91/AVR32 platforms but it provides
more build coverage and prepares for ARM multiplatform.
Also fixes a couple of format type warnings.
Signed-off-by: Joachim Eastwood <manabian@gmail.com>
Acked-by: Nicolas Ferre <nicolas.ferre@atmel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
Update lm49453_reg_defs values as per LM49453 HW revision-B
Signed-off-by: M R Swami Reddy <mr.swami.reddy@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
Add adc, mic, sidetone volume ranges and appropriately added the controls.
Fix the DAC HP/EP/LS/LO/HA maximum gain values.
Signed-off-by: MR Swami Reddy <mr.swami.reddy@ti.com>
Tested-by: Vinod Koul <vinod.koul@intel.com>
--
sound/soc/codecs/lm49453.c | 43 ++++++++++++++++++++++++-------------------
1 files changed, 24 insertions(+), 19 deletions(-)
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
Since we are now using the clock API integration to manage MCLK we can now
use clk_get_rate() to determine if we need to divide MCLK without relying
on platform data.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
Request MCLK as a clock and then enable it when carrying out a state
transtion and while ANC is active, minimising system power consumption
in idle modes.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
The FLL source constants were numbered as a simple enumeration but were
being used in the code as direct values to be written to the registers.
Renumber the constants to reflect the usage.
Reported-by: Ryo Tsutsui <Ryo.Tsutsui@wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
|
|
ARIZONA_AIF1_RATE_MASK is 0x7800 /* AIF1_RATE - [14:11] */
Thus we need left shift ARIZONA_AIF1_RATE_SHIFT when setting aif1 rate.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
|
|
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
The references used for the noise gates and parameters for their triggering
are configurable, expose that to users.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
