| Commit message (Collapse) | Author | Age |
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'asoc/topic/rockchip' into asoc-next
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The spdif mclk should be 128 times of sample rate, and there is a
internal divider, the real rate of spdif mclk is mclk / (div + 1).
Hence, the original driver always get the good frequency for
48000/96000/44100/192000. But for 32000, the mclk is incorrect,
it should be 32000*128, but get 48000*128. Do not use the internal
divider here, just set all mclk to 128 * sample rate directly.
Signed-off-by: Chris Zhong <zyw@rock-chips.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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this patch make it more reasonable and readable, because when we chose
I2S_CKR_TRCM_TXONLY, we only output clk_lrck_tx, and hardware need to
confirm this signal is wired to external codec lrck_tx/rx at the same time.
for convenience, we just handle lrck_txonly if we enable symmetric_rates
in driver and dai_link. otherwise, we use the separate lrck_tx/rx.
Signed-off-by: Sugar Zhang <sugar.zhang@rock-chips.com>
Signed-off-by: Xing Zheng <zhengxing@rock-chips.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Physically there is a jackset which includes a Headphone and a Jackset
Mic pin. The patch add thw two pins with the correct pin name so the
DAPM management can find the pin and make the jack detection and event
reporting work again.
The patch also shut up the following error:
rockchip-snd-max98090 sound: ASoC: DAPM unknown pin Headset Jack
Signed-off-by: Enric Balletbo i Serra <enric.balletbo@collabora.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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The path Headset Mic --> MICBIAS is wrong because connects a non-supply
widget as a source with a supply widget as a sink. It's the other way
around:
MICBIAS (source) --> Headset Mic (sink).
This patch also shut up the following error message:
rockchip-snd-max98090 sound: Connecting non-supply widget to supply widget is not supported (Headset Mic -> MICBIAS)
rockchip-snd-max98090 sound: ASoC: no dapm match for Headset Mic --> (null) --> MICBIAS
rockchip-snd-max98090 sound: ASoC: Failed to add route Headset Mic -> direct -> MICBIAS
Signed-off-by: Enric Balletbo i Serra <enric.balletbo@collabora.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Commit f2ed6b07645e ("ASoC: Make aux_dev more like a generic
component") caused a regression on this driver, since now a
kernel oops is seen when rockchip-mac98090 driver is loaded.
That commit changed the probing of aux_devs before checking
new DAI links, so for this driver rk_98090_headset_init is
called before rk_init and then the kernel oops due a NULL
pointer dereference inside rk_98090_headset_init function
since there is a call that tries to access the jack pointer
which has not been allocated yet.
This is the call chain that causes the crash:
rk_98090_headset_init
-> ts3a227e_enable_jack_detect
-> snd_jack_set_key
rk_init
-> snd_soc_card_jack_new
This patch moves the new jack object creation from rk_init
to rk_98090_headset_init function making sure the jack is
created before is accessed.
Signed-off-by: Enric Balletbo i Serra <enric.balletbo@collabora.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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There are 3 i2s sdio pins, which iomux mode is as follows:
- sdi3_sdo1
- sdi2_sdo2
- sdi1_sdo3
we need to configure these pins' iomux mode via the GRF register
when use multi channel playback/capture.
Signed-off-by: Sugar Zhang <sugar.zhang@rock-chips.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into asoc-rcar
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Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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simple-card needs to get its card name.
This patch makes this method simple style standard.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Current simple-card is creating dai_link->name / dai_link->stream_name.
These are based on CPU + Codec name, or "fe.CPU" or "be.Codec" if it
was DPCM.
This patch adds asoc_simple_card_set_dailink_name() and set dailink
name as common method.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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We can use simpel utils asoc_simple_card_parse_daifmt().
Let's use it
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Current ALSA SoC has simple-card driver which is supporting both
platform and DT probe.
Now, some sound cards driver are created based on simple-card.
They have similar feature or function, but implemented separately
on each drivers. This is a waste of code.
OTOH, merging these driver into same driver is highly risk,
because it will be very difficult to keep compatibility.
More over, ALSA SoC want to have graph base of DT feature in the
future. Maybe it want to use simple-card like feature / function.
Because of these background, this patch creates simple-card
helper utils, and provides common function to each drivers.
1st is asoc_simple_card_parse_daifmt()
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Current simple-card supports snd_soc_jack/pin/gpio.
These code are very similar, but driver has verbosity code.
So, this patch adds new snd_soc_jack and cleanups code
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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In current simple-card, platform is handled as special case,
but, the code is not readable.
This patch makes platform to use asoc_simple_card_sub_parse_of()
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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When building with 'make W=1', we get a harmless warning about
slightly incorrect prototypes in the rcar audio driver:
sound/soc/sh/rcar/gen.c: In function 'rsnd_gen2_probe':
sound/soc/sh/rcar/gen.c:209:2: error: 'static' is not at beginning of declaration [-Werror=old-style-declaration]
This changes the 'const static' to 'static const' as it should be.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
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AUDIO-CLKOUTn can asynchronizes with L/R clock.
AUDIO-CLKOUTn synchronizes with L/R clock is now default behavior.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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McPDM module receives it's functional clock from external source. This
clock is the pdmclk provided by the twl6040 audio IC. If the clock is not
available all register accesses to McPDM fails and the module is not
operational.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Rob Herring <robh@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Implement ASoC's suspend and resume callbacks.
Since McPDM does not use pcm_trigger for start and stop of the stream due
to strict sequencing needs with the twl6040, the callbacks will stop and
restart the McPDM in case the board suspended during audio activity.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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The DS4_WD_EN bit is only touched before calling omap_mcpdm_open_streams().
Move it inside of that function for simplicity.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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The pdmclk is needed for McPDM. It is generated by twl6040.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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'asoc/topic/max9877', 'asoc/topic/mtk' and 'asoc/topic/nau8825' into asoc-next
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Thanks Mark and Anatol for the discussion. According to the result,
the standard C will translate any non-zero value into true, or
false otherwise.
QUOTE:
"6.3.1.2 Boolean type
When any scalar value is converted to _Bool, the result is 0 if the
value compares equal to 0; otherwise, the result is 1
"
Thus, the "!!" idiom is removed.
Signed-off-by: John Hsu <KCHSU0@nuvoton.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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The original design only covers the jack insertion logic is active low.
Add more condition to cover no matter the logic is active low and high.
Signed-off-by: John Hsu <KCHSU0@nuvoton.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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The newly added nau8825_dai_is_active() function is only called from
the PM logic that is build-time conditional in this driver, so we get
a warning when CONFIG_PM is disabled:
sound/soc/codecs/nau8825.c:229:13: error: 'nau8825_dai_is_active' defined but not used [-Werror=unused-function]
static bool nau8825_dai_is_active(struct nau8825 *nau8825)
By replacing the #ifdef around the functions with a __maybe_unused
annotation, the code becomes more robust to this kind of problem
and we no longer get the warning while also slightly improving
readability.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Fixes: b50455fab459 ("ASoC: nau8825: cross talk suppression measurement function")
Signed-off-by: Mark Brown <broonie@kernel.org>
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The cross talk measurement function can reduce cross talk across the JKTIP
HPL) and JKR1(HPR) outputs which measures the cross talk signal level to
determine what is the cross talk reduction gain. This system works by
sending a 23Hz -24dBV sine wave into the headset output DAC and through
the PGA. The output of the PGA is then connected to an internal current
sense which measures the attenuated 23Hz signal and passing the output to
an ADC which converts the measurement to a binary code. With two separated
measurement, one for JKR1(HPR) and the other JKTIP(HPL), measurement data
can be separated read in IMM_RMS_L for HSR and HSL after each measurement.
Thus, the measurement function has four states to complete whole sequence.
(1)Prepare state : Prepare the resource for detection and transfer to HPR
IMM stat to make JKR1(HPR) impedance measure.
(2)HPR IMM state : Read out orignal signal level of JKR1(HPR) and transfer
to HPL IMM state to make JKTIP(HPL) impedance measure.
(3)HPL IMM state : Read out cross talk signal level of JKTIP(HPL) and
transfer to IMM state to determine suppression sidetone gain.
(4)IMM state : Computes cross talk suppression sidetone gain with orignal
and cross talk signal level. Apply this gain and then restore codec con-
figuration. Then transfer to Done state for ending.
In order to get the cross talk suppression sidetone gain, we need the
function to compute log10 value and the result is round off to 3 decimal.
This function takes reference to dvb-math. The source code locates as the
following. "Linux/drivers/media/dvb-core/dvb_math.c"
Then, the orignal and cross talk signal vlues need to be characterized.
The sidetone value can be converted to decibel with the equation below.
sidetone = 20 * log (original signal level / crosstalk signal level)
Besides, the state machine for cross talk process needs interruptions to
trigger worked. We have the RMS intrruption enabled with the internal VCO
clock when headset connected. In the interrupt handler, the driver will
judge the headset is high impedance or not. If yes, the cross talk supp-
ression shouldn't apply and do nothing but relieve the protection raised
before. Otherwise, apply the cross talk suppression in the headset and
start the process.
Because the process spends a lot of time, there is an resource race issue
easily between the application and interruption. They will control codec
power and clock concurrently. In one situaiton, the jack is inserted when
playback, and then the application changes to headset device. The applica-
tion prepares the playback and interrupt handler raises work for cross
talk process together. For this case, the solution is that driver delays
soc jack report until cross talk process completes. The mechanism can
avoid application to do playback preparation before cross talk detection
is still working.
In another situaiton, the system suspends when playback. After resume, the
system restarts playback, and meanwhile jack detection restarts. The play-
back preparation and cross talk process triggered by interruptions happens
concurrently. For the case, the driver provides the semaphone to syn-
chronize the playback and interrupt handler. In order to avoid the play-
back interfered by cross talk process, the driver make the playback prepa-
ration halted until cross talk process finish. After codec resume, the
driver finds the codec dai is active, and then the driver raises the pro-
tection for cross talk function to avoid the playback recovers before
cross talk process finish.
The driver also provides cancel method to forcely cancel the cross talk
task and restores the configuration to original status. Before the codec
remove, ejection, or suspend, the driver is obliged to cancel the cross
talk detection process. It can reduce the risk of failure when quickly and
continually doing jack insertion and ejection.
Signed-off-by: John Hsu <KCHSU0@nuvoton.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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There is typo in the name of biquad filter coefficients control.
The patch is to fix the typo.
Signed-off-by: John Hsu <KCHSU0@nuvoton.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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The driver changes jack type detection interruption to non-clock archi-
tecture for less 1mW power saving. The architecture is called manual mode
jack detection. It has no hardware debounce, no jack type detection, but
only detecting jack insertion. After jack insertion, the driver will
switch to auto mode jack detection with internal clock which can detect
microphone, jack type and do hardware debounce.
The manual architecture has these main changes including codec initiation,
interruption, clock control, and power management. When codec initiation
or system resume, the clock is closed as jack insertion detection at man-
ual mode, and bypass debounce circuit. These configurations move to resume
setup function when setup bias level after resume.
When jack insertion detection happens, the manual mode turns off and make
configuration about jack type detection interruption at auto mode in auto
irq setup function which can detect microphone and jack type. The inter-
ruption will switch to manual mode again with clock free until jack ejec-
tion happens.
The system clock configuration adds clock disable option which can disable
internal VCO clock. Before the system clock change, there is an restric-
tion added to make sure clock disabled and not config any clock when no
headset connected.
In power management, we involve the solution about races and jack detec-
tion in resume from Ben Zhang in the following patch and list his comment.
[PATCH] ASoC: nau8825: Fix jack detection across suspend
"Jack plug status is rechecked at resume to handle plug/unplug
in S3 when the chip has no power."
"Suspend/resume callbacks are moved from the i2c dev_pm_ops to
snd_soc_codec_driver. soc_resume_deferred is a delayed work
which may trigger nau8825_set_bias_level. The bias change races
against dev_pm_ops, causing jack detection issues.
soc_resume_deferred ensures bias change and snd_soc_codec_driver
suspend/resume are sequenced correctly."
Change SAR widget to supply type which can prevent the codec keeping at
SND_SOC_BIAS_ON during suspend. The codec suspend function can just invoke
normally.
Before the system suspends, the driver turns off all interruptions. Keep
the interruption quiet before resume setup completes. The ADC channel will
be disabled which is needed for interruptions at audo mode.
Signed-off-by: John Hsu <KCHSU0@nuvoton.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Add programmable biquad filter configuration control for user space.
The filter is configurable for low pass filters, high pass filters,
Notch filter, etc in the ADC and DAC path.
Signed-off-by: John Hsu <KCHSU0@nuvoton.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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The I2C driver has an i2c_device_id array but that information isn't
exported to the module using the MODULE_DEVICE_TABLE() macro. So the
module autoloading won't work if the I2C device is registered using
OF or legacy board files due missing alias information in the module.
The issue was found using Kieran Bingham's coccinelle semantic patch:
https://lkml.org/lkml/2016/5/10/520
Signed-off-by: Javier Martinez Canillas <javier@osg.samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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The default value of DAC channel select is reverse in codec.
For normal usage, switch the channel select when codec bootup.
Signed-off-by: John Hsu <KCHSU0@nuvoton.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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The interrupt clock is gated by x1[10:8], one of them needs to be enabled
all the time for interrupts to happen. We change codec to enable ADC
because it's helpful to reduce playback pop noise.
Don't use force enable pin to enable ADC instead of ADC widget event.
That won't interfere DAPM operation and let bias work normally.
Signed-off-by: John Hsu <KCHSU0@nuvoton.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Decrease internal clock frequency for power saving when standby.
But clock divider needs restore when MCLK as system clock in playback.
Signed-off-by: John Hsu <KCHSU0@nuvoton.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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In FLL calculation, increase VCO/DCO frequency for better performance.
Besides, have different register configuration according to fraction or not
when apply FLL parameters.
Signed-off-by: John Hsu <KCHSU0@nuvoton.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Extend FLL clock source selection. The source can be from MCLK, BCLK or FS.
Signed-off-by: John Hsu <KCHSU0@nuvoton.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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This patch adds HDMI audio output support to the MT8173 RT5650 machine
driver.
Signed-off-by: PC Liao <pc.liao@mediatek.com>
Signed-off-by: Philipp Zabel <p.zabel@pengutronix.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Fixes the following sparse warning:
sound/soc/mediatek/mt2701/mt2701-afe-pcm.c:72:5: warning:
symbol 'mt2701_dai_num_to_i2s' was not declared. Should it be static?
Signed-off-by: Wei Yongjun <yongjun_wei@trendmicro.com.cn>
Acked-by: Garlic Tseng <garlic.tseng@mediatek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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The check for if the "afe" allocation failed was too late and there
wasn't a check for "afe->platform_priv".
Fixes: 43a6a7e71063 ('ASoC: mediatek: add mt2701 platform driver implementation.')
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Acked-by: Garlic Tseng <garlic.tseng@mediatek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Add BT implementation for mt2701 platform driver.
Signed-off-by: Garlic Tseng <garlic.tseng@mediatek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Add machine driver and config option for MT2701.
Signed-off-by: Garlic Tseng <garlic.tseng@mediatek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Add mt2701 platform driver implementation for playback and capture.
The implement follow DAPM structure (memory interface as FE and I2S
as BE).
Because of the hardware design, i2s out required to be enabled when
we need to enable i2s in. This patch includes the implementation.
Signed-off-by: Garlic Tseng <garlic.tseng@mediatek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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add structure define and clock control function for 2701.
Signed-off-by: Garlic Tseng <garlic.tseng@mediatek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Modify mt8173 driver implementation to use common structure.
Signed-off-by: Garlic Tseng <garlic.tseng@mediatek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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