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Take the return value from mcasp_common_hw_param() and use that in case of
error.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into HEAD
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Set snd_soc_pm_ops for the pm ops.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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The use of pm_runtime in trigger() callback is not correct and it will lead
to unbalanced power.usage_count.
The only place which might need to call pm_runtime is the set_fmt callback.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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These registers can be configured synchronously for playback and capture.
Furthermore when McASP is in master and sync mode the capture operation
needs the TX path to be configured in order to be able to provide the needed
clocks for the bus.
xxFMT and xxFMCT registers has been already configured for both TX and RX
other places in the driver.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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Instead of
davinci_hw_common_param - for common, I2S/DIT mode settings
davinci_hw_dit_param - for DIT protocol configuration
davinci_hw_param - for I2S (and compatible protocols)
Use the following names:
mcasp_common_hw_param, mcasp_dit_hw_param and mcasp_i2s_hw_param.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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Changed Sat -> Say.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
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These symbols got eliminated when non-DT support for Exynos was
removed. Remove them.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
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While looking into some spurious responses, I found that the addr value was
treated a bit inconsistent: values 8..0xf will be treated as codec 0 and
values 0..7 will be treated as no error regardless of whether there is a codec
there, or not.
With this patch, all non-existing codecs will be treated equally.
In addition, printing rp and wp could help figuring out if the wp value is
reported wrongly from the controller or if something else is wrong.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Now all weird setups have been converted to fixups for the generic
parser, and we can disable the static quirks. This commit just turns
the build off. The bulky static quirk code still remains for a while,
in case we get an overlooked regression. It'll be removed at the next
kernel version.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Both CX20549 and CX20551 codecs have a mixer widget and it can be
connected as the ADC source. Like AD and VIA codecs, enable the
add_stereo_mix_input flag for these codecs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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CX20549 has an aamixer widget at NID 0x17. Let's enable it.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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For the generic parser, use the standard fixup matching.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This laptop with CX20549 codec misses the internal mic at NID 0x12.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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We need to fix bogus pincfgs on this machine, but it works well else.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Apply the amp cap override for CX20549 mixer widget in case where the
generic parser is used, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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OLPC XO needs a few special handling. Now these are implemented as a
fixup to the generic parser.
Obviously, the DC BIAS mode had to be added manually. This is mainly
implemented in the mic_autoswitch hook, where the mic pins are
overwritten depending on the DC bias mode. This also required the
override of the mic boost control, since the mic boost is applied only
when the DC mode is disabled.
In addition, the mic pins must be set dynamically at recording time
because these also control the LED.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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... by using snd_Hda_codec_update_cache() instead of *_write_cache().
Since all path elements should have been updated by this function,
we are safe to assume that the cache contents are consistent.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Like other codecs, apply a specific fixup given by a model string.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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When we plug a 3-ring headset on the Dell machine (Vendor ID:
0x10ec0255, Subsystem ID: 0x1028064d), the headset mic can't be
detected, after apply this patch, the headset mic can work well.
BugLink: https://bugs.launchpad.net/bugs/1260303
Cc: David Henningsson <david.henningsson@canonical.com>
Tested-by: Doro Wu <fan-cheng.wu@canonical.com>
Cc: stable@vger.kernel.org
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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for-next
This completes the hardware support for the Asus Xonar DG/DGX cards,
and makes them actually usable.
This is v4 of Roman's patch set with some small formatting changes.
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Pull slave-dma updates from Vinod Koul:
- new driver for BCM2835 used in R-pi
- new driver for MOXA ART
- dma_get_any_slave_channel API for DT based systems
- minor fixes and updates spread acrooss driver
[ The fsl-ssi dual fifo mode support addition clashed badly with the
other changes to fsl-ssi that came in through the sound merge. I did
a very rough cut at fixing up the conflict, but Nicolin Chen (author
of both sides) will need to verify and check things ]
* 'for-linus' of git://git.infradead.org/users/vkoul/slave-dma: (36 commits)
dmaengine: mmp_pdma: fix mismerge
dma: pl08x: Export pl08x_filter_id
acpi-dma: align documentation with kernel-doc format
dma: fix vchan_cookie_complete() debug print
DMA: dmatest: extend the "device" module parameter to 32 characters
drivers/dma: fix error return code
dma: omap: Set debug level to debugging messages
dmaengine: fix kernel-doc style typos for few comments
dma: tegra: add support for Tegra148/124
dma: dw: use %pad instead of casting dma_addr_t
dma: dw: join split up messages
dma: dw: fix style of multiline comment
dmaengine: k3dma: fix sparse warnings
dma: pl330: Use dma_get_slave_channel() in the of xlate callback
dma: pl330: Differentiate between submitted and issued descriptors
dmaengine: sirf: Add device_slave_caps interface
DMA: Freescale: change BWC from 256 bytes to 1024 bytes
dmaengine: Add MOXA ART DMA engine driver
dmaengine: Add DMA_PRIVATE to BCM2835 driver
dma: imx-sdma: Assign a default script number for ROM firmware cases
...
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Remove old SPI control functions, change anti-pop init
sequence, remove some garbage from structures. The 'Apply' functions
must be called at the mixer initialization, otherwise
mixer settings sometimes will not be applied at startup.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
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Change the 'put' function of the high-pass filter control to use the new
SPI functions.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
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First of all, we should not touch the GPIOs. They are not
for selecting the capture source, but they seems just enable
the whole audio input curcuit. The 'put' function calls the
'apply' functions to change register values. Change the order
of capture sources.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
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Modify the input_vol_* functions to use the new SPI routines,
There is a new applying function that will be called when
the capture source changed.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
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I tried both variants: volume control and impedance selector.
In the first case one minus is that we can't change the
volume of multichannel output without additional software
volume control. However, I am using this variant for the
last three months and this seems good. All multichannel
speaker systems have internal amplifier with the
volume control included, but not all headphones have
this regulator. In the second case, my software volume
control does not save the value after reboot.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
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Change the order of elements in the output select control. This will
reduce the number of relay switches. Change 'put' function to call the
oxygen_update_dac_routing() function. Otherwise multichannel playback
does not work. Also there is a new function to apply settings, this
prevents from duplicating the code.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
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Actually CS4245 connected to the I2S channel 1 for
capture, not channel 2. Otherwise capturing and
playback does not work for CS4245.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
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Moving the mixer code away makes things easier. The mixer
will control the driver, so the functions of the
driver need to be non-static.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
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Change the function to read the data from the new shadow buffer.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
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When selecting the audio output destinations (headphones,
FP headphones, multichannel output), the channel routing
should be changed depending on what destination selected.
Also unnecessary I2S channels are digitally muted. This
function called when the user selects the destination
in the ALSA mixer.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
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When selecting the audio sample rate for CS4245,
the MCLK divider should also be changed.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
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Change CS4245 initialization: different sequence and GPIO values,
according to datasheets and reverse-engineering information.
Change cleanup/resume/suspend functions, since they use
initialization.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
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Add the new SPI write and read functions. The SPI read function
is used for creating initial registers dump and may be used for
debugging purposes. SPI operations are cached, so there is a new
function to manage the cache (shadow). I have to remove
the shift from the CS4245_SPI_* constants, since when
we are performing the reading, we need to shift by 8 instead
of 16.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
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Add additional constants to the xonar_dg.h file:
capture and playback sources. Move GPIO_* constants and the
dg struct to the header file from the xonar_dg.c file.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
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Add some additional information in comments and my copyright.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
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When the user switches the output from stereo to multichannel
or vice versa, the driver needs to update the channel routing.
Instead of creating additional subroutines, I better export existing
oxygen_update_dac_routing symbol from the oxygen mixer
and call this function. It calls model.adjust_dac_routing()
and my function does the work.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
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The Xonar DG/DGX driver needs this mask to mute unnecessary
channels.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
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Modify the oxygen_write_spi() function to use the newly
introduced oxygen_wait_spi() function. Change return value
from void to int, so it can return error codes. Older
drivers just ignore that return value, new drivers can
check this value. We need to wait AFTER
initiating the SPI transaction, otherwise read
operation will not work.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
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The oxygen_wait_spi() function now performs waiting when the
SPI bus completes a transaction. Introduce the timeout error
checking and increase timeout to 200 from 40.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
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Processing coefficients are often a vital part of the codec's configuration,
so dumping them can be important. However, because they are undocumented and
secret, we do not want to enable this for all codecs by default.
Therefore instead add this as a debugging parameter.
I have prepared for codecs that want to enable this by default by the extra
dump_coef bitfield, but unsure if we want to do that as long as the
(unlikely, but still) race remains.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v3.14
A few fixes, all in drivers. Nothing stands out particularly, the
biggest set of fixes is some build coverage issues from Sachin.
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Make BCLK divider setting implicite in hw_params call if McASP device
is the bit clock master on the audio serial bus.
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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'asoc/fix/omap', 'asoc/fix/samsung', 'asoc/fix/simple', 'asoc/fix/tlv320aic32x4' and 'asoc/fix/wm5100' into asoc-linus
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Currently the Negative Terminal Input Routing Configuration is only set
when there is a special routing configuration. If we don't use one of
the inputs IN1 or IN2 as negative terminal input, the PGA and recording
does not work.
This patch adds a route from CM1L/CM1R to the PGA as negative input by
default. With this configuration the PGA can amplify all input signals
and line-in/mic works again.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
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Playback of a mono stream should output the same stream on both
channels. At the moment only the left analog signal is valid, the right
one is just noise.
This patch maps the left digital channel onto both DACs when receiving a
mono stream.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
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The referenced clock is used to get codec clock rate and the clock is
disabled and enabled in startup and shutdown snd_soc_ops call
backs. The change is also documented in DT bindigs document.
Signed-off-by: Jyri Sarha <jsarha@ti.com>
cc: bcousson@baylibre.com
Signed-off-by: Mark Brown <broonie@linaro.org>
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Similarly to other Apple products, MBA 1,1 needs a specific quirk.
Pin 0x18 must be set to VREF_50 to have sound output. This was no
longer done since commit 1a97b7f, resulting in a mute built-in speaker.
This patch corrects the regression by creating a fixup for the MBA 1,1.
Fixes: 1a97b7f22774 ("ALSA: hda/realtek - Remove the last static quirks for ALC882")
Cc: <stable@vger.kernel.org> [v3.4+]
Tested-by: Adrien Vergé <adrienverge@gmail.com>
Signed-off-by: Adrien Vergé <adrienverge@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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mach/dma.h is not referenced by this file. Remove it.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
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