For smart51 pins, we need to preserve the input pin-control bits at
auto-mute controls instead of overwriting zero or pin-out-only.
Otherwise the VREF won't be set properly when smart51 is disabled
again.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When Independent-HP mode is changed for VIA, the driver needs to
re-issue the auto-mute check so that the line-out pins are set properly
without influence of HP pin state.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the line-jack is plugged/unplugged, the driver must check also
the headphone jack state in addition to the line-out jack. Currently
it checks only the line-out state and ignores the headphone.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of checking the model quirk, use a fixup table for workaround
of 44kHz-fixed PCM for Lenovo IdeaPad with ALC269.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
it shouldn't contain space letters and
special letters like parentheses.
aplay will be "Segmentation fault" without this patch
special thanks to Takashi.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
it shouldn't contain space letters and
special letters like parentheses.
aplay will be "Segmentation fault" without this patch.
special thanks to Takashi.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
it shouldn't contain space letters and
special letters like parentheses.
aplay will be "Segmentation fault" without this patch.
special thanks to Takashi.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
It's harmless but annyoing.
sound/pci/hda/patch_realtek.c: In function ‘alc_cap_getput_caller’:
sound/pci/hda/patch_realtek.c:2722:9: warning: ‘err’ may be used uninitialized in this function
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now all alc*_parse_auto_config() do almost same thing except for the
NID list to ignore and the PINs for SSID-check, we can merge all these
to a single function. A good amount of code reduction.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
One more code reduction. This codec has less DACs, thus the wiring
to DAC can't be filled uniquely for all output pins, i.e. some outputs
share the same volume control.
Except for that, all seems working fine.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Merge more auto-parser code in patch_realtek.c, now for ALC861.
The topology of this codec is pretty simple, and can be parsed well
by the current starndard parser.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
query_amp_caps() may return non-zero if the amp cap isn't supported
by the codec. Thus one needs to check widget-caps first, then check
the corresponding amp-caps.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A regression fix from commit 21268961d3
ALSA: hda - More flexible dynamic-ADC switching for Realtek codecs
The auto-mic wasn't detected properly when no ADC-switch is needed.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For VT2002P, VT1802 and VT1812 codecs, to create Independent HP
control.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For VT2002P, VT1802 and VT1812 codecs, there're only two DACs. So smart51
control shouldn't be created.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For VT2002P, VT1802 and VT1812 codecs, the original activate_output_path()
function can't initialize output and hp path correctly, since mixers connected to
output pin widgets are not considered. So modify the activate_output_path()
function to satisify this kind of codec.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove unused variable 'dai' to eliminate below warning.
CC sound/soc/pxa/pxa2xx-pcm.o
sound/soc/pxa/pxa2xx-pcm.c: In function 'pxa2xx_soc_pcm_new':
sound/soc/pxa/pxa2xx-pcm.c:91: warning: unused variable 'dai'
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Put the all static quirk codes out of patch_realtek.c, split into the
file for each codec model. For controlling the build of quirk codes,
a new Kconfig, CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS is introduced.
By setting this off, all quirk codes won't be built, thus you can save
lots of memory.
The codes in patch_realtek.c are also shuffled and more comments are
given, but the contents aren't changed. This is just a refactoring.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch changes the auto-parser and the auto-mic handling codes to
allow more flexible dynamic ADC-switching with Realtek codecs.
In the new code, the following strategy is taken:
- When a cap-src can't handle all input-sources, either skip it, or
switch to the ADC-switching mode. In ADC-switching mode, like the
former dual-ADC mode for ALC275, it changes ADC on the fly according
to the current input source.
- When auto-mic is possible, always assign imux. If the mic pins are
set statically via a quirk, rebuild imux according to the pins.
In the auto-mic mode, the driver always changes the imux (although
the imux isn't exposed as a mixer element).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The legacy mode has been accidentaly removed by commit:
ASoC: twl6040: add all ABE DAIs
Add back the twl6040-hifi dai.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since the PLL handling has been simplified, and
rebased on 0, there is no longer need for converting
the PLL ID.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Avoid configuring the PLL several times during audio startup.
We can configure the PLL at prepare time with parameters collected
earlier hw_param, and set_dai_sysclk calls.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We can manage the sample rate constraints without the need
to maintain a variable and a pointer.
This simplifies the handling of the constraint, and makes it
more robust.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
It is better if the selection between the Low power,
and High performance PLL is handled within the codec
driver, not in machine driver(s) to avoid duplicated
code, and also to have consistent tracking of the selected
PLL, and the resulting differences in supported sample
rates.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Change the variable names to be neutral (not refering to HS).
This will ease up the introduction of PLL selection, which
going to use the same enum strings.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The twl6040_request_irq/free_irq inline functions are going
to be removed, so replace them with direct calls.
The irq number is provided by the core driver via resource.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Reviewed-by: Felipe Balbi <balbi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
dmaengine expects the maxburst parameter in words, not bytes.
The imxdma driver and its users do this wrong. Fix this.
As a side note the imx-pcm-dma-mx2 driver was 'fixed' to work
with imx-dma. This broke the driver with imx-sdma support which
correctly takes the maxburst parameter in words. This patch
puts the sdma based sound back to work.
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Instead of assigning each default hda_pcm_stream pointers, do NULL-checks
and assign default values in alc_build_pcms().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The only different implmentation was alc880_auto_init_input_src(),
and now it covers this variant, and we can use the single function
for all codecs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now with the new code for looking for ADCs and MUXs, we can replace
the whole ADC assignment with the parsed results.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
All alc*_auto_init_analog_input() calls are identical, so let's use
the same function more clearly without aliases.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Parse ADCs and cap-srcs in alc_auto_create_input_ctls() by itself
instead of passing explicitly from the caller. By this change, all
alc*_auto_create_input_ctls() can be unified to the same calls.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use the same common code for auto-parsing the output paths and their
initializations, based on the existing ALC662 code, which is smarter
than the old ALC880/2 code.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When multiple inputs are present on the mixer widget (typically a DAC
and a loopback), mute/unmute both inputs with the corresponding mixer
element.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In alc662_auto_fill_dac_nids(), the HP and speaker DACs aren't parsed
when the corresponding pins aren't fixed with single DACs.
Now check these DACs even for non-fixed pins.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the dual-adc switching mode is active in Realtek auto-parser,
we need to couple all ADCs as a single capture-volume. Currently, the
volume control changes only the first ADC, thus others may remain silent.
This patch fixes the problem.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add the support of ALC269VC codec.
Also delete the unnecessary codec_variant type enum list:
now only three variants (ALC269VA ALC269VB ALC269VC) are needed.
In addition, added some aliases:
- Add ALC269VB alias name ALC277
- Add ALC269VC alias name ALC259 ALC281X
- Add ALC269VC for Lenovo device 0x21f3 name ALC3202
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is a minimal driver for the Tegra SPDIF controller.
In hardware, the SPDIF output signal is always routed to any active HDMI
display controllers, and may also be routed to external pins on Tegra
using the pinmux.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Allow platform probe to register platform kcontrols and DAPM just like
the CODEC probe().
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Allow platform driver widgets to perform any IO required for DAPM.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In preparation for Dynamic PCM (AKA DSP) support.
Allow platform drivers to register kcontrols.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Now we have supply widgets there's no need to open code the handling of
the ACTIVE bit.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Tested-by: Nicolas Ferre <nicolas.ferre@atmel.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Now we have supply widgets there's no need to open code the handling of
the ACTIVE bit.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Tested-by: Nicolas Ferre <nicolas.ferre@atmel.com>
Acked-by: Liam Girdwood <lrg@ti.com>
The snd_card->driver field contains a driver name string, and in
general it shouldn't contain space or special letters. The commit
2b39535b9e changed the string copy from
card->name, but the long name string may contain such letters, thus
it may still lead to a segfault.
A temporary fix is not to copy the long name string but just keep it
empty as the earlier version did.
Reported-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In preparation for ASoC Dynamic PCM (AKA DSP) support.
Allow platform driver to perform IO. Intended for platform DAPM.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Commit af46800 ("ASoC: Implement mux control sharing") revealed that
"Left Line1[L | R] Mux" and "Right Line1[L | R] Mux" widgets were pointing
to the same kcontrols and codec registers and thus soc-core falsely detected
them as shared controls. This is actually wrong since there are separate
registers in hardware that configure Line1L to RADC and Line1R to LADC cross
connects so these muxes should not be shared.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
Enable ramp down/up step to be configured based on
platform.
Signed-off-by: Axel Castaneda Gonzalez <x0055901@ti.com>
Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Set default sysclk constraints to high performance mode.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Remove dependency between pll (hppll, lppll) and headset power
mode (low-power, high-performance), as headset power mode can
be used with any pll.
A new control is created to allow headset power mode configuration
from userspace. Changing headset power mode during earpiece related
usecases is not propagated down to the codec as earpiece requires
HS DAC in HP mode.
Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add other supported sample rates to LP and HP modes.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add all DAIs to fully support OMAP4 ABE.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Convert TWL6040 CODEC driver into a TWL6040 MFD child, it implies
that MFD-level operations like register accesses, clock setting
and power management are done through MFD APIs, not directly by
CODEC driver anymore. To avoid conflicts with the other MFD child,
vibrator registers are skipped in CODEC driver.
Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This delay is very conservative.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Cc: stable@kernel.org
The clock needed by the I2S driver is associated with the I2S device name
in the standard fashion. Hence, use clk_get(dev) instead of clk_get_sys(clk_name).
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The I2S controller needs a clock to respond to register writes. Without
this, register writes will at worst hang the CPU. In practice, I've only
observed writes being dropped.
Luckily, the dropped register writes historically had no effect:
TEGRA_I2S_TIMING: The value we wrote was the reset default.
TEGRA_I2S_FIFO_SCR: The default was for the FIFOs to request more data
when one slot was empty. The requested value was for the FIFOs to request
when four slots were empty. The DMA controller in the mainline kernel is
configured to burst a single entry at a time into the FIFO, hence there
was no issue. The only negative effect was on bus efficiency losses due
to an increased number of arbitration attempts.
However, in various non-upstream changes, the DMA controller now bursts
four entries at a time into the FIFO. If there is only space for one
entry, the data is simply dropped. In practice, this resulted in 3/4 of
samples being dropped, and playback at 4x the expected rate and pitch.
By fixing the clocking issue, this is solved.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Allign the platform data names for twl4030 audio submodule:
twl4030_audio_data: for the core MFD driver
twl4030_codec_data: for ASoC codec driver
twl4030_vibra_data: for the input/ForceFeedback driver
To avoid breakage, change all depending drivers, files
to use the new types.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Samuel Ortiz <sameo@linux.intel.com>
Rename the driver, and header file from twl4030-codec to
twl4030-audio.
To avoid breakage change depending drivers at the same time.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
CC: Misael Lopez Cruz <misael.lopez@ti.com>
Acked-by: Samuel Ortiz <sameo@linux.intel.com>
Since we now route the front DAC via aa-mix widget, adding the aa-mix
to surrounds will result in a mix-up of both front and surround PCM
signals. For avoiding this, the aa-mix routes have to be disabled
for surround paths.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the individual DAC is available for the headphone output, the driver
should create the DAC for its volume control.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For VT1718S, the multi-channel path should be like following:
DAC 0-->Mixer 9(index 5)-->Mixer 0(index 1)-->Front Pin;
DAC 1-->Mixer 1(index 0)-->Surround Pin;
DAC 2-->C/LFE Pin;
DAC 3-->Mixer 2(index 0)-->Side Pin;
But current code built Surround and Side path through index 1 of
Mixer 1 and 2. So Adjusting Surround and Side channel amplifier is
invalid. This patch fixes the issue.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For VT1718S, Mixer 9 doesn't expose the connection to DAC 0. So when
building up a 'PCM Playback' amplifier control, it will fail since
getting DAC 0 index of Mixer 9 returned -1. So I added a dac_mixer_idx
to indicated the actual index of DAC 0 to Mixer 9. Following is the
patch and next mail is another.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In preparation for ASoC dynamic PCM support (AKA ASoC DSP)
Platform will also support DAPM so separate out the probe function
to simplify the code (just like the codec probe).
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Unmute DAC on front speaker path when Independent HP is enabled.
When to enable Independent HP, the front speaker won't output any sound
for VT1708, VT1708B, VT1708S and VT1702.
I find the via_independent_hp_put() routine will mute DAC 0 path in Mixer 0.
For these codecs, when using Independent HP, there could have two
independent streams, one is from DAC0-->Mixer0-->Front Pin, the other is
from DAC3-->GainSW3-->Side Pin.
So I added a check for DAC-->Mixer path in activate_output_path().
If current path is DAC-->Mixer, no need to mute DAC index in Mixer.
In fact, to change connection of Headphone pin or Mux connected with HP
is enough.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the re-implementation of Independent-HP mode in the
case where the DAC is shared between HP and side-channel streams.
Now the driver tries to parse the output-path using the pre-parsed
side-channel DAC for the independent HP output, too.
When a playback PCM stream is opened with this shared mode, the
Independent-HP mixer switch can't be changed for avoiding the conflict,
thus it returns -EBUSY error.
One remaining unintuitive issue is that the DAC volume is still
controlled as "Side" volume although it's shared by both independent-HP
and side streams.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The char can be unsigned on some architectures. Since the code checks
the negative values, they should be declared as signed char explicitly.
sound/pci/rme9652/hdspm.c:5449: warning: comparison is always false due to limited range of data type
sound/pci/rme9652/hdspm.c:5462: warning: comparison is always false due to limited range of data type
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the wrongly converted short values:
sound/pci/cs5535audio/cs5535audio_pcm.c:152: warning: large integer implicitly truncated to unsigned type
sound/pci/cs5535audio/cs5535audio_pcm.c:160: warning: large integer implicitly truncated to unsigned type
Signed-off-by: Takashi Iwai <tiwai@suse.de>
More for neatness than any actual performance improvement.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
It was consistently wrong for everything except WM8993 so should be no
functional change.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.comm>
The Blackfin DMA controller can report one frame beyond the end of the
buffer in the wraparound case but ALSA requires that the pointer always
be in the buffer. Do the wraparound to handle this. A similar bug is
likely to apply to the other Blackfin PCM drivers but the code is less
obvious to inspection and I don't have a user to test.
Reported-by: Kieran O'Leary <Kieran.O'Leary@wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
I noticed that the last character of the ELD monitor name is lost,
this fixes the issue.
This fix should be confirming to the HDA spec, and works together with
the DRM part of the ELD patch.
The HDA spec does not mention that Monitor_Name_String is an '\0'
ending string, and it allows NML to be 1, which is only valid when MNL
does not count the possible ending '\0'.
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move codec init verb which is only applicatable for VT1708.
I've found the root cause that jack plugged in can't be detected.
The verb in vt1708_init_verbs is used to power down jack detect circuit.
This verb is only applicable to VT1708. vt1708 didn't implement jack
detect function in hardware, so we should shut down this function to
avoid noise. But for other codecs, hardware implement jack detect
function. If sending this verb during initialization, jack detect will
be invalid. So I move this verb from via_parse_auto_config() to
patch_vt1708().
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Create snd_hda_get_conn_index() helper function for obtaining the
connection index of the widget. Replaced the similar codes used in
several codec-drivers with this common helper.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix a issue to enable unsolicited response to line-out pins.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a machine driver to support the EVAL-ADAV801 and EVAL-ADAV803 boards
connected to a Analog Devices BF5XX evaluation board.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Mike Frysinger <vapier@gentoo.org>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds support for the Analog Devices ADAV801 and ADAV803 audio codec.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch updates the email address of the sound drivers supported by me to an
email account I will use on a more regular basis in the future.
Signed-off-by: Hans-Christian Egtvedt <hans-christian.egtvedt@atmel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since we're not using the new auto parser as a fallback yet,
add it manually as a quirk.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Conexant ID 506c was found on Acer Aspire 3830TG. As users report
no playback, sending to stable should be safe.
Cc: stable@kernel.org
BugLink: https://bugs.launchpad.net/bugs/783582
Reported-by: andROOM
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use static inline for dummy function to fix the warnings like below
sound/pci/hda/patch_sigmatel.c: In function ‘stac92xx_init’:
sound/pci/hda/patch_sigmatel.c:4387:3: warning: statement with no effect
sound/pci/hda/patch_sigmatel.c: In function ‘stac92xx_resume’:
sound/pci/hda/patch_sigmatel.c:4927:3: warning: statement with no effect
Signed-off-by: Takashi Iwai <tiwai@suse.de>
More similar fixes like previous commits: handle the exceptional case
like ALC267 where no volume amp is found in ADC widget but in the
capsrc widget instead.
Also minor checks for avoiding possible erros: no connection-select
when the pin has a single selection, and add beep verbs only when the
0x1d is used for beep.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALC259 seems to provide an invalid widget capability for the input-src
selector widget. The widget shows the input-amp while it's a selector,
and this confuses the current ALC882 initialization code that is used
for ALC259, too. For fixing this, check the amp capability and handle
the connection selection individually.
Also, ALC259 has no mute bit in DAC volume, so we need to initialize
it as ZERO instead of MUTE.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALC269 and compatible codecs have the output volume in DACs, thus we
can't use the ALC880's code as is. Fixed by checking the amp caps and
picking up the right widget for initialization.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Look through the known NIDs that may have EAPD capabilities and turn
on/off them appropriately instead of checking the individual vendor ids.
This will also avoid the forgotten entries of newly added codec ids
in future.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the regression introduced by the commit
1f0f4b8036
ALSA: hda - Reduce static init verbs for Realtek auto-parsers
The input amps of mixer widgets should be unmuted as default (as
usually they have no assigned mixer switches).
More fixes in this commit are, however, for ALC260: ALC260 codec can
have multiple output mixers connnected to a single DAC althouh the
driver didn't pick up them properly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some Realtek codecs have the output pins hardwired with certain DACs.
These DACs have to be assigned at first and assign the rest for
multi-DAC pins so that all DACs can be assigned properly.
Without such an optimization, speaker outputs may be assigned to the
same DAC as the headphone or others.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In alc_auto_add_multi_channel_mode(), when the primary HP workaround
is enabled, it re-initializes the DAC list but calls alc662's function
in a fixed way. This isn't pretty suitable for other codecs, of course.
Now we call it with fill_dac function pointer so that the proper
function can be called at that point.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of using fixed init verbs, initialize DACs, ADCs and mixers
more dynamically for Realtek auto-parsers.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch add idma related register definitions to support idma.
Signed-off-by: Sangbeom Kim <sbkim73@samsung.com>
Acked-by: Jassi Brar <jaswinder.singh@linaro.org>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
I2S registers can be used for control idma.
Previously, register is defined in i2s.c.
For sharing the registers, It is moved to i2s-regs.h
Signed-off-by: Sangbeom Kim <sbkim73@samsung.com>
Acked-by: Jassi Brar <jaswinder.singh@linaro.org>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch add WM8994 PCM machine driver to support PCM audio
on SMDKV310, SMDKC210 boards.
Playback and Capture supports 8kHz sampling rates.
and It is tested on SMDKV310, SMDKC210.
Signed-off-by: Sangbeom Kim <sbkim73@samsung.com>
Acked-by: Jassi Brar <jaswinder.singh@linaro.org>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add a new helper function snd_hda_get_conn_list().
Unlike snd_hda_get_connections(), this function doesn't copy the
connection-list but gives the raw pointer for the cached list.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
each device has a unique mac address, which can be used to distinguish
multiple devices in the same machine. we therefore include the full mac
address in the device shortname and the last 6 bytes in the device id.
Signed-off-by: Tim Blechmann <tim@klingt.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the sound/ directory there are two files (flagged by 'make
versioncheck'); sound/pci/asihpi/asihpi.c and
sound/soc/codecs/wm8991.c that include linux/version.h although they
don't need it. This patch removes the unneeded includes.
Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Change alc_get_line_out_pfx() in patch_realtek.c to provide the channel
specific name and assign the index so that each caller doesn't have to
set the channel name by itself.
Also, check the multi-io case with the primary hp-out; for the multi-io
channels, assign the channel name instead of "Headphone" with indices.
This makes the mixer names more intuitive and reduces confusion.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some BIOS set up the pin config wrongly as line-out although it's
supposed to be a speaker out. In most cases, though, we can judge
the validity by checking the connection type -- when it's FIXED,
mostly it's an invalid line-out but a speaker.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Even if the machine has no line-out but only HP-out, try to detect the
multi-io. It'll allow more possibilities for 5.1 outputs on laptops.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Don't rely on the codec's channels_min information to decide wheter or
not allocate a substream's DMA buffer. Rather check if the substream
itself was allocated previously.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
SND_MXC_SOC_SSI looks to be unused, so kill it.
Signed-off-by: Arnaud Patard <arnaud.patard@rtp-net.org>
Acked-by: Sascha Hauer <s.hauer@pengutronix.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM8918 is register compatible with the WM8904 with a subset of the
functionality. Add the device ID, a subsequent patch will ensure that only
the relevant functionality is exported to userspace.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Some devices do not support manual updates of the DC servo.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
BIOS lists the internal speaker as an internal line-out. Change to
internal speaker + model=auto for better auto-mute capabilities.
BugLink: http://bugs.launchpad.net/bugs/754964
Reported-by: Marc Legris <marc.legris@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Johannes Stezenbach <js@sig21.net>
[zonque@gmail.com: transform to new ASoC structure]
Signed-off-by: Daniel Mack <zonque@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Johannes Stezenbach <js@sig21.net>
[zonque@gmail.com: transform to new ASoC structure]
Signed-off-by: Daniel Mack <zonque@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
