Allwinner DAC seems to have a delay in the Speaker audio routing. When
playing a sound for the first time, the sound gets chopped. On a second
play the sound is played correctly. After some time (~5s) the issue gets
back.
This commit seems to be fixing the same issue as bf14da7 but
for another codepath.
This is the DTS that was used to debug the problem.
&codec {
allwinner,pa-gpios = <&r_pio 0 11 GPIO_ACTIVE_HIGH>; /* PL11 */
allwinner,audio-routing =
"Speaker", "LINEOUT";
status = "okay";
}
Signed-off-by: Georgii Staroselskii <georgii.staroselskii@emlid.com>
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Allwinner H6 has a different mapping for the fifo register controller.
Actually only the fifo TX bit is used in the drivers.
Use the freshly introduced quirks to make this drivers compatible with
the Allwinner H6.
Signed-off-by: Clément Péron <peron.clem@gmail.com>
Acked-by: Maxime Ripard <maxime.ripard@bootlin.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Allwinner H6 has a different bit to flush the TX FIFO.
Add a quirks to prepare introduction of H6 SoC.
Signed-off-by: Clément Péron <peron.clem@gmail.com>
Acked-by: Maxime Ripard <maxime.ripard@bootlin.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The quirks are actually defines in the middle of the file with
short explanation.
Move this at the top and add a section to have coherency with
sun4i-i2s.
Signed-off-by: Clément Péron <peron.clem@gmail.com>
Acked-by: Maxime Ripard <maxime.ripard@bootlin.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Based on 1 normalized pattern(s):
released under the gpl this program is free software you can
redistribute it and or modify it under the terms of the gnu general
public license as published by the free software foundation either
version 2 of the license or at your option any later version this
program is distributed in the hope that it will be useful but
without any warranty without even the implied warranty of
merchantability or fitness for a particular purpose see the gnu
general public license for more details
extracted by the scancode license scanner the SPDX license identifier
GPL-2.0-or-later
has been chosen to replace the boilerplate/reference in 2 file(s).
Signed-off-by: Thomas Gleixner <tglx@linutronix.de>
Reviewed-by: Allison Randal <allison@lohutok.net>
Reviewed-by: Richard Fontana <rfontana@redhat.com>
Cc: linux-spdx@vger.kernel.org
Link: https://lkml.kernel.org/r/20190523091651.124582774@linutronix.de
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Add SPDX license identifiers to all Make/Kconfig files which:
- Have no license information of any form
These files fall under the project license, GPL v2 only. The resulting SPDX
license identifier is:
GPL-2.0-only
Signed-off-by: Thomas Gleixner <tglx@linutronix.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
The A64 datasheet lists the supply rail for the headphone amp's charge
pump as "CPVDD". cpvdd-supply is the name of the property for this power
rail specified in the device tree bindings. "HPVCC" was the name used in
the A33 datasheet for the same function.
Rename the supply so it matches the datasheet, bindings, and the subject
from the original commit.
Fixes: ca0412a057 ("ASoC: sunxi: sun50i-codec-analog: Add support for cpvdd regulator supply")
Signed-off-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add Line Playback Volume for Allwinner A10 and Allwinner A20.
Add Line Boost Volume for Allwinner A10 and Allwinner A20.
Add Line Right, Line Left, Line Playback Switch for Allwinner A10 and
Allwinner A20.
Signed-off-by: Danny Milosavljevic <dannym@scratchpost.org>
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add FM Playback Volume for Allwinner A10 and Allwinner A20.
Add FM Left, FM Right, FM Playback Switch for Allwinner A10 and
Allwinner A20.
Signed-off-by: Danny Milosavljevic <dannym@scratchpost.org>
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add Mic1 Playback Switch and Mic2 Playback Switch for Allwinner A10 and
Allwinner A20.
Signed-off-by: Danny Milosavljevic <dannym@scratchpost.org>
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Since it's now possible to have a DAPM mixer control with multiple
channels, use it to cut down the total number of controls.
Keep "Left Mixer Left DAC Playback Switch" and "Right Mixer Right DAC
Playback Switch" name & layout the same as before for compatibility.
Signed-off-by: Danny Milosavljevic <dannym@scratchpost.org>
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add Mic1 Boost Volume and Mic2 Boost Volume for Allwinner A10 and for
Allwinner A20.
Those controls are in different registers per chip model, so put the
Allwinner A10 controls and the Allwinner A20 controls into the newly
split sun4i_codec_controls and sun7i_codec_controls, respectively.
Signed-off-by: Danny Milosavljevic <dannym@scratchpost.org>
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Introduce sun7i_codec_controls because some of the controls are different
on Allwinner A20 compared to Allwinner A10.
Also introduce sun7i_codec_codec in order to use sun7i_codec_controls and
make sun7i_codec_quirks use sun7i_codec_codec.
Signed-off-by: Danny Milosavljevic <dannym@scratchpost.org>
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add a control "Mic Playback Volume" that allows the user to control the
MIC gain stage (common for Mic1 and Mic2) leading to the output mixer.
Signed-off-by: Danny Milosavljevic <dannym@scratchpost.org>
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add MIC2 Pre-Amplifier, Mic2 input for Allwinner A10 and Allwinner A20.
Previously, there only the Mic1 input and MIC1 Pre-Amplifier was exposed.
This exposes the Mic2 input and MIC2 Pre-Amplifier.
Signed-off-by: Danny Milosavljevic <dannym@scratchpost.org>
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
On the Allwinner A64 SoCs, the audio codec has a built-in headphone
amplifier. This amplifier has a power supply separate from the rest of
the analog audio circuitry, labeled cpvdd.
This patch adds a DAPM widget for this supply, and ties it to the
headphone amp widget.
Signed-off-by: Chen-Yu Tsai <wens@csie.org>
Acked-by: Maxime Ripard <maxime.ripard@bootlin.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
sun8i-codec misses a route from ADC to AIF1 Slot 0 ADC. Add it
to the driver to avoid adding it to every dts.
Fixes: eda85d1fee ("ASoC: sun8i-codec: Add ADC support for a33")
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
drvdata is actually sun8i_codec, not snd_soc_card, so it crashes
when calling snd_soc_card_get_drvdata().
Drop card and scodec vars anyway since we don't need to
disable/unprepare clocks - it's already done by calling
runtime_suspend()
Drop clk_disable_unprepare() calls for the same reason.
Fixes: 36c684936f ("ASoC: Add sun8i digital audio codec")
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
Acked-by: Maxime Ripard <maxime.ripard@bootlin.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
SND_SUN50I_CODEC_ANALOG selects SND_SUNXI_ADDA_PR_REGMAP which is leftover
of renaming SND_SUNXI_ADDA_PR_REGMAP to SND_SUN8I_ADDA_PR_REGMAP. Replace
it with SND_SUN8I_ADDA_PR_REGMAP to fix possible link errors for some
configurations:
sound/soc/sunxi/sun50i-codec-analog.o: In function `sun50i_codec_analog_probe':
sun50i-codec-analog.c:(.text+0x62): undefined reference to `sun8i_adda_pr_regmap_init'
Fixes: 42371f327d ("ASoC: sunxi: Add new driver for Allwinner A64 codec's analog path controls")
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
Reviewed-by: Andre Przywara <andre.przywara@arm.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
startup() and shutdown() hooks are called for both substreams,
so stopping either substream when another is running breaks the
latter.
E.g. playback breaks if capture is stopped when playback is running.
Move code from startup() and shutdown() to resume() and suspend()
hooks respectively to fix this issue
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
Acked-by: Maxime Ripard <maxime.ripard@bootlin.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Allwinner A64 uses the same digital codec part as in A33, so we need
to build this driver on ARM64 as well.
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
Acked-by: Maxime Ripard <maxime.ripard@bootlin.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The internal codec on A64 is split into 2 parts. The analog path controls
are routed through an embedded custom register bus accessed through
the PRCM block.
Add an ASoC component driver for it. This should be tied to the codec
audio card as an auxiliary device.
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
Acked-by: Maxime Ripard <maxime.ripard@bootlin.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
It will be reused by sun50i-codec-analog later.
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
Acked-by: Maxime Ripard <maxime.ripard@bootlin.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
BCLK / LRCK ratio should be sample size * channels, but it was
hardcoded to 32 (0x1 is 32 as per A33 and A64 datasheets).
Calculate it basing on sample size and number of channels.
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
Acked-by: Maxime Ripard <maxime.ripard@bootlin.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The I2S block used for the audio codec in the A64 differs from other 3
I2S modules in A64 and isn't compatible with H3. But it is very similar
to what is found in A10(sun4i). However, its TX FIFO is
located at a different address.
Signed-off-by: Marcus Cooper <codekipper@gmail.com>
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
Acked-by: Maxime Ripard <maxime.ripard@bootlin.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
I2S's RX slot number of SUN8I should be shifted 4 bit to left.
Fixes: 7d2993811a ("ASoC: sun4i-i2s: Add support for H3")
Signed-off-by: Yong Deng <yong.deng@magewell.com>
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
Add ADC support for the sun8i-codec driver.
This driver uses microphones widgets and routes provided by the
analog part (sun8i-codec-analog).
Some digital configurations are needed by creating new ADC widgets
and routes.
Signed-off-by: Mylène Josserand <mylene.josserand@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When any of the DAI hardware configuration callbacks (.hw_param,
.set_fmt, .set_sysclk) fails, there is no explanation about why it
failed. This is particularly confusing for .hw_param, which covers
many parameters of the DAI. Telling the users what parameter isn't
supported, and what the requested value was goes a long way for
developers trying to combine sun4i-i2s with external codecs.
This patch adds dev_err calls explaining what isn't supported or
failed, and what the value was. sun4i_i2s_set_clk_rate()'s first
parameter was changed to a struct snd_soc_dai *dai, so we can
get the underlying device.
Signed-off-by: Chen-Yu Tsai <wens@csie.org>
Acked-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Acked-by: Marcus Cooper <codekipper@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
H3 ASoC supports 12Khz and 24Khz audio sample rates but the current
drivers doesn't advertise these rates properly and they cannot be used.
For example attempt to capture at 12Khz uses 11Khz (same applies to
audio playback):
Recording raw data '/tmp/testS16_LE.raw' : Signed 16 bit Little Endian, Rate 12000 Hz, Stereo
Warning: rate is not accurate (requested = 12000Hz, got = 11025Hz)
This patch fixes the audio sample rates declared and supported by the
driver according to the H3 data sheet. Specifically for audio playback:
8000, 11050, 12000, 16000, 22050, 24000, 32000, 44100, 48000, 96000, 192000
and for audio capture:
8000, 11050, 12000, 16000, 22050, 24000, 32000, 44100, 48000
Signed-off-by: Andrea Bondavalli <andrea.bondavalli74@gmail.com>
Acked-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
The I2S controller in the A83T is mostly compatible with the one found
in earlier SoCs such as the A20 and A31. While the documents publicly
available for the A83T do not cover this hardware, the officially
released BSP kernel does have register definitions for it. These were
matched against the A20 user manual. The only difference is the TX FIFO
and interrupt status registers have been swapped around, like what we
have seen with the SPDIF controller.
This patch adds support for this hardware.
Signed-off-by: Chen-Yu Tsai <wens@csie.org>
Acked-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
32bit and 24bit audio capture formats for H3/H2+ are broken because the
RX_SAMPLE_BITS and the RX_FIFO_MODE bits of AC_ADC_FIFOC register of the audio
codec are not set to operate in 24bit mode but in 16bit mode only.
The following patch sets the H3 audio codec registers and the DMA bus width
properly when a 24/32bit capture is requested.
Signed-off-by: Andrea Bondavalli <andrea.bondavalli74@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The current code might be a bit intriguing without having experienced the
issue before, and might come up as a mistake.
Make explicit what's going on by adding a comment.
Signed-off-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
While the current code was reporting to be able to work in master mode, it
failed to do so because the BCLK divider wasn't programmed, meaning that
the BCLK would run at the PLL's frequency no matter the sample rate.
It was obviously a bit too fast.
Add support to retrieve the divider to use, and set it. Since our PLL is
not always able to generate a perfect multiple of the sample rate, we'll
have to choose the closest divider that matches our setup.
Fixes: 36c684936f ("ASoC: Add sun8i digital audio codec")
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: <stable@vger.kernel.org>
Since its introduction, the codec had an inversion of the left and right
channels. It turned out to be pretty simple as it appears that the codec
doesn't have the same polarity on the LRCK signal than the I2S block.
Fix this by inverting our bit value for the LRCK inversion.
Fixes: 36c684936f ("ASoC: Add sun8i digital audio codec")
Cc: <stable@vger.kernel.org>
Signed-off-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
The current code had the condition backward when checking if the codec
should be running in slave or master mode.
Fix it, and make the comment a bit more readable.
Fixes: 36c684936f ("ASoC: Add sun8i digital audio codec")
Signed-off-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: <stable@vger.kernel.org>
Many source files in the tree are missing licensing information, which
makes it harder for compliance tools to determine the correct license.
By default all files without license information are under the default
license of the kernel, which is GPL version 2.
Update the files which contain no license information with the 'GPL-2.0'
SPDX license identifier. The SPDX identifier is a legally binding
shorthand, which can be used instead of the full boiler plate text.
This patch is based on work done by Thomas Gleixner and Kate Stewart and
Philippe Ombredanne.
How this work was done:
Patches were generated and checked against linux-4.14-rc6 for a subset of
the use cases:
- file had no licensing information it it.
- file was a */uapi/* one with no licensing information in it,
- file was a */uapi/* one with existing licensing information,
Further patches will be generated in subsequent months to fix up cases
where non-standard license headers were used, and references to license
had to be inferred by heuristics based on keywords.
The analysis to determine which SPDX License Identifier to be applied to
a file was done in a spreadsheet of side by side results from of the
output of two independent scanners (ScanCode & Windriver) producing SPDX
tag:value files created by Philippe Ombredanne. Philippe prepared the
base worksheet, and did an initial spot review of a few 1000 files.
The 4.13 kernel was the starting point of the analysis with 60,537 files
assessed. Kate Stewart did a file by file comparison of the scanner
results in the spreadsheet to determine which SPDX license identifier(s)
to be applied to the file. She confirmed any determination that was not
immediately clear with lawyers working with the Linux Foundation.
Criteria used to select files for SPDX license identifier tagging was:
- Files considered eligible had to be source code files.
- Make and config files were included as candidates if they contained >5
lines of source
- File already had some variant of a license header in it (even if <5
lines).
All documentation files were explicitly excluded.
The following heuristics were used to determine which SPDX license
identifiers to apply.
- when both scanners couldn't find any license traces, file was
considered to have no license information in it, and the top level
COPYING file license applied.
For non */uapi/* files that summary was:
SPDX license identifier # files
---------------------------------------------------|-------
GPL-2.0 11139
and resulted in the first patch in this series.
If that file was a */uapi/* path one, it was "GPL-2.0 WITH
Linux-syscall-note" otherwise it was "GPL-2.0". Results of that was:
SPDX license identifier # files
---------------------------------------------------|-------
GPL-2.0 WITH Linux-syscall-note 930
and resulted in the second patch in this series.
- if a file had some form of licensing information in it, and was one
of the */uapi/* ones, it was denoted with the Linux-syscall-note if
any GPL family license was found in the file or had no licensing in
it (per prior point). Results summary:
SPDX license identifier # files
---------------------------------------------------|------
GPL-2.0 WITH Linux-syscall-note 270
GPL-2.0+ WITH Linux-syscall-note 169
((GPL-2.0 WITH Linux-syscall-note) OR BSD-2-Clause) 21
((GPL-2.0 WITH Linux-syscall-note) OR BSD-3-Clause) 17
LGPL-2.1+ WITH Linux-syscall-note 15
GPL-1.0+ WITH Linux-syscall-note 14
((GPL-2.0+ WITH Linux-syscall-note) OR BSD-3-Clause) 5
LGPL-2.0+ WITH Linux-syscall-note 4
LGPL-2.1 WITH Linux-syscall-note 3
((GPL-2.0 WITH Linux-syscall-note) OR MIT) 3
((GPL-2.0 WITH Linux-syscall-note) AND MIT) 1
and that resulted in the third patch in this series.
- when the two scanners agreed on the detected license(s), that became
the concluded license(s).
- when there was disagreement between the two scanners (one detected a
license but the other didn't, or they both detected different
licenses) a manual inspection of the file occurred.
- In most cases a manual inspection of the information in the file
resulted in a clear resolution of the license that should apply (and
which scanner probably needed to revisit its heuristics).
- When it was not immediately clear, the license identifier was
confirmed with lawyers working with the Linux Foundation.
- If there was any question as to the appropriate license identifier,
the file was flagged for further research and to be revisited later
in time.
In total, over 70 hours of logged manual review was done on the
spreadsheet to determine the SPDX license identifiers to apply to the
source files by Kate, Philippe, Thomas and, in some cases, confirmation
by lawyers working with the Linux Foundation.
Kate also obtained a third independent scan of the 4.13 code base from
FOSSology, and compared selected files where the other two scanners
disagreed against that SPDX file, to see if there was new insights. The
Windriver scanner is based on an older version of FOSSology in part, so
they are related.
Thomas did random spot checks in about 500 files from the spreadsheets
for the uapi headers and agreed with SPDX license identifier in the
files he inspected. For the non-uapi files Thomas did random spot checks
in about 15000 files.
In initial set of patches against 4.14-rc6, 3 files were found to have
copy/paste license identifier errors, and have been fixed to reflect the
correct identifier.
Additionally Philippe spent 10 hours this week doing a detailed manual
inspection and review of the 12,461 patched files from the initial patch
version early this week with:
- a full scancode scan run, collecting the matched texts, detected
license ids and scores
- reviewing anything where there was a license detected (about 500+
files) to ensure that the applied SPDX license was correct
- reviewing anything where there was no detection but the patch license
was not GPL-2.0 WITH Linux-syscall-note to ensure that the applied
SPDX license was correct
This produced a worksheet with 20 files needing minor correction. This
worksheet was then exported into 3 different .csv files for the
different types of files to be modified.
These .csv files were then reviewed by Greg. Thomas wrote a script to
parse the csv files and add the proper SPDX tag to the file, in the
format that the file expected. This script was further refined by Greg
based on the output to detect more types of files automatically and to
distinguish between header and source .c files (which need different
comment types.) Finally Greg ran the script using the .csv files to
generate the patches.
Reviewed-by: Kate Stewart <kstewart@linuxfoundation.org>
Reviewed-by: Philippe Ombredanne <pombredanne@nexb.com>
Reviewed-by: Thomas Gleixner <tglx@linutronix.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
The sun8i-h3 introduces a lot of changes to the i2s block such
as different register locations, extended clock division and
more operational modes. As we have to consider the earlier
implementation then these changes need to be isolated.
None of the new functionality has been implemented yet, the
driver has just been expanded to allow it work on the H3 SoC.
Signed-off-by: Marcus Cooper <codekipper@gmail.com>
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
The default value of the config register is different on newer
SoCs and therefore enabling/disabling with a register write
will clear bits used to set the direction of the clock and frame
pins.
Signed-off-by: Marcus Cooper <codekipper@gmail.com>
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
The newer SoCs do not have this setting. Instead they set the pin
direction. Add a check to see if the bit is valid and if so set
it accordingly.
Signed-off-by: Marcus Cooper <codekipper@gmail.com>
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
On the newer SoCs the bits to configure the operational mode are
located in a different register. Add a regmap field so that this
location can be configured.
Signed-off-by: Marcus Cooper <codekipper@gmail.com>
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
The location of the mclk output enable bit is different on newer
SoCs. Use a regmap field to enable it.
Signed-off-by: Marcus Cooper <codekipper@gmail.com>
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
On newer SoCs the bit fields for the blck and lrclk polarity are in
a different locations. Use regmap fields to set the polarity bits
as intended.
Signed-off-by: Marcus Cooper <codekipper@gmail.com>
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
On newer SoCs the location of the slot width select and sample
resolution are different and also there is a bigger range of
support.
For the current supported rates then an offset is required.
Signed-off-by: Marcus Cooper <codekipper@gmail.com>
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
On the original i2s block the channel mapping and selection were
configured for stereo audio by default: This is not the case with
the newer SoCs and they are also located at different offsets.
To support the newer SoC then regmap fields have been added to the
quirks and these are initialised to their correct settings during
probing.
Signed-off-by: Marcus Cooper <codekipper@gmail.com>
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
First of all,the address of pdev->dev is assigned to card->dev in
create_card,then the function platform_set_drvdata copies the value
the variable card to pdev->dev.driver_data, but when calling
snd_soc_register_card,the function dev_set_drvdata(card->dev, card)
will also do the same copy operation,so i think that the former copy
operation can be removed.
Signed-off-by: Peng Donglin <dolinux.peng@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
It has been seen that the newer SoCs have a different TX FIFO
address. Add this to the quirks structure.
Signed-off-by: Marcus Cooper <codekipper@gmail.com>
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
The newer SoCs have a larger range than the original SoC that this
driver was developed for. By adding the regmap config to the quirks
then the driver can initialise the managed register map correctly.
Signed-off-by: Marcus Cooper <codekipper@gmail.com>
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
The BCLKDIV and MCLKDIV found on newer SoCs start from an offset of 1.
Add the functionality to adjust the division values according to the
needs to the device being used.
Signed-off-by: Marcus Cooper <codekipper@gmail.com>
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Declare snd_soc_codec_driver structures as const as they are either
passed as an argument to the function snd_soc_register_codec or stored as
reference in field codec of type sun4i_codec_quirks. Both the fucntion
argument and the codec field are of type const, so declare the
structures with this property as const.
Signed-off-by: Bhumika Goyal <bhumirks@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In preparation for the changes required to support newer SoCs then
quirks has been moved and also added to the device structure.
Signed-off-by: Marcus Cooper <codekipper@gmail.com>
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
clk_prepare_enable() can fail here and we must check its return value.
Signed-off-by: Arvind Yadav <arvind.yadav.cs@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Commit a53e35db70 ("reset: Ensure drivers are explicit when requesting
reset lines") started to transition the reset control request API calls
to explicitly state whether the driver needs exclusive or shared reset
control behavior. Convert all drivers requesting exclusive resets to the
explicit API call so the temporary transition helpers can be removed.
No functional changes.
Cc: Liam Girdwood <lgirdwood@gmail.com>
Cc: Mark Brown <broonie@kernel.org>
Cc: Maxime Ripard <maxime.ripard@free-electrons.com>
Cc: Chen-Yu Tsai <wens@csie.org>
Cc: alsa-devel@alsa-project.org
Signed-off-by: Philipp Zabel <p.zabel@pengutronix.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
This structure is only stored in the ops field of a snd_soc_dai_driver
structure. That field is declared const, so snd_soc_dai_ops structures
that have this property can be declared as const also.
Signed-off-by: Gustavo A. R. Silva <garsilva@embeddedor.com>
Acked-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The gpiod API checks for NULL descriptors, so there is no need to
duplicate the check in the driver.
Signed-off-by: Fabio Estevam <fabio.estevam@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The codec in the V3s is similar to the one found on the A31. One key
difference is the analog path controls are routed through the PRCM
block. This is supported by the sun8i-codec-analog driver, and tied
into this codec driver with the audio card's aux_dev.
In addition, the V3s does not have LINEIN, LINEOUT, MBIAS and MIC2,
MIC3, and the FIFO related registers are like H3.
Signed-off-by: Icenowy Zheng <icenowy@aosc.xyz>
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Acked-by: Rob Herring <robh@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
The V3s SoC features an analog codec with headphone support but without
mic2 and linein.
Add support for it.
Signed-off-by: Icenowy Zheng <icenowy@aosc.xyz>
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Acked-by: Rob Herring <robh@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Allwinner V3s has an analog codec without MIC2 and Line In, which will
need a special set of mixer controls/widgets/routes, otherwise meaningless
controls will be exported to userspace and confuse the user.
Add the special set, and use it when the SoC has no MIC2 and Line In.
Signed-off-by: Icenowy Zheng <icenowy@aosc.io>
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Allwinner V3s features an analog codec without MBIAS pin.
Split out this part, in order to prepare for the V3s analog codec.
Signed-off-by: Icenowy Zheng <icenowy@aosc.xyz>
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Update the driver to use SND_SOC_DAPM_AIF_IN instead of
SND_SOC_DAPM_DAC.
Rename the interface's widgets to be more precise on which slot
the interface is connected.
Signed-off-by: Mylène Josserand <mylene.josserand@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
An unwanted space is present in an audio widget's name on the dapm
routing. It causes an error on the recognition of this widget (error:
("no dapm match for AIF1 Slot 0 Right").
Remove the space fixes it.
Signed-off-by: Mylène Josserand <mylene.josserand@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Allwinner V3s features an analog codec without LINEIN.
Split out this part, in order to prepare for the V3s analog codec.
Signed-off-by: Icenowy Zheng <icenowy@aosc.xyz>
Acked-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Allwinner V3s features an analog codec without MIC2.
Split out this part, in order to prepare for the V3s analog codec.
Signed-off-by: Icenowy Zheng <icenowy@aosc.xyz>
Acked-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
SOC_MIXER_ARRAY is a simplified function of SND_SOC_DAPM_MIXER
which handles automatically the ARRAY_SIZE of controls.
Update the driver to use SOC_MIXER_ARRAY.
Signed-off-by: Mylène Josserand <mylene.josserand@free-electrons.com>
Acked-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Update the driver to use the new SOC_DAPM_DOUBLE definition
on the digital DAC mixer.
Update the names accordingly as, when they are shared, the
controls are not prefixed with the widget's name anymore.
Signed-off-by: Mylène Josserand <mylene.josserand@free-electrons.com>
Acked-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
The "HP" widget is already present and take part to
the analog part (sun8i-codec-analog).
Remove it from the digital part as it is unnecessary.
Signed-off-by: Mylène Josserand <mylene.josserand@free-electrons.com>
Acked-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
It's not necessary to unregister a component registered
with devm_snd_soc_register_component(). Also removed
pointness clk_disable_unprepare() from error path and
snd_soc_unregister_platform() from the remove.
Fixes: f8260afa44 ("ASoC: sunxi: Add support for the SPDIF block")
Signed-off-by: Wei Yongjun <weiyongjun1@huawei.com>
Acked-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
As the 64-bit Allwinner H5 SoC has the same analog codec part (also the
same digital part) as H3, enable the driver to be built on ARM64
Allwinner platform, so that it can be used on H5.
Signed-off-by: Icenowy Zheng <icenowy@aosc.xyz>
Acked-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Acked-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Some SoCs have a reset line that must be asserted/deasserted.
This patch adds a quirk to handle the new compatible
"allwinner,sun6i-a31-i2s" which will deassert the reset
line on probe function and assert it on remove's one.
This new compatible is useful in case of A33 codec driver, for example.
Signed-off-by: Mylène Josserand <mylene.josserand@free-electrons.com>
Acked-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add the sun8i audio codec which handles the digital register of
A33 codec.
The driver handles only the basic playback from the DAC to headphones.
All other features (microphone, capture, etc) will be added later.
Signed-off-by: Mylène Josserand <mylene.josserand@free-electrons.com>
Acked-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When playing a sound for the first time, a short delay, where the audio
file is not played, can be noticed.
On a second play (right after), the sound is played correctly.
If we wait a short time (~5 sec which corresponds to the aplay
timeout), the delay is back.
This patch fixes it by using an event on headphone amplifier.
It allows to keep the amplifier enable while playing a sound.
A delay of 700ms allows to wait that the amplifier is powered-up
before playing the sound.
Signed-off-by: Mylène Josserand <mylene.josserand@free-electrons.com>
Acked-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The H3 SoC uses the same SPDIF block as found in earlier SoCs, but its
TXFIFO is mapped to another address.
Signed-off-by: Marcus Cooper <codekipper@gmail.com>
Acked-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
As done previously for sun4i-codec, the DMA maxburst of 4
is not supported by every SoCs so the DMA controller engine
returns "unsupported value".
As a maxburst of 8 is supported by all variants, this patch
increases it to 8.
For more details, see commit from Chen-Yu Tsai:
commit 730e2dd0cb ("ASoC: sun4i-codec: Increase DMA max burst to 8")
Signed-off-by: Mylène Josserand <mylene.josserand@free-electrons.com>
Acked-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The mono differential output for "Line Out" downmixes the stereo audio
from the mixer, instead of just taking the left channel.
Add a route from the "Right Mixer" to "Line Out Source Playback Route"
through the "Mono Differential" path, so DAPM doesn't shut down
everything if the left channel is muted.
Fixes: 0f909f98d7 ("ASoC: sun4i-codec: Add support for A31 Line Out
playback")
Signed-off-by: Chen-Yu Tsai <wens@csie.org>
Acked-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
It has been seen that some newer SoCs have a different TX FIFO
address and we already have the difference with the A31 requiring
a reset. Add a quirks structure so that these can be managed
easily.
Signed-off-by: Marcus Cooper <codekipper@gmail.com>
Acked-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The dapm components are now handled by the ALSA SoC SPDIF DIT driver
so can be removed.
Signed-off-by: Marcus Cooper <codekipper@gmail.com>
Acked-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The codec on the H3 is similar to the one found on the A31. One key
difference is the analog path controls are routed through the PRCM
block. This is supported by the sun8i-codec-analog driver, and tied
into this codec driver with the audio card's aux_dev.
In addition, the H3 has no HP (headphone) and HBIAS support, and no
MIC3 input. The FIFO related registers are slightly rearranged.
Signed-off-by: Chen-Yu Tsai <wens@csie.org>
Acked-by: Rob Herring <robh@kernel.org>
Acked-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The codec in the A23 is similar to the one found on the A31. One key
difference is the analog path controls are routed through the PRCM
block. This is supported by the sun8i-codec-analog driver, and tied
into this codec driver with the audio card's aux_dev.
In addition, the A23 does not have LINEOUT, and it does not support
headset jack detection or buttons.
Signed-off-by: Chen-Yu Tsai <wens@csie.org>
Acked-by: Rob Herring <robh@kernel.org>
Acked-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Oddly enough, my version of GCC misses this uninitialized variable.
Fixes: ba2ff3027b ("ASoC: sunxi: Add support for A23/A33/H3 codec's analog path controls")
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Acked-by: Chen-Yu Tsai <wens@csie.org>
Acked-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The internal codec on A23/A33/H3 is split into 2 parts. The
analog path controls are routed through an embedded custom register
bus accessed through the PRCM block.
The SoCs share a common set of inputs, outputs, and audio paths.
The following table lists the differences.
----------------------------------------
| Feature \ SoC | A23 | A33 | H3 |
----------------------------------------
| Headphone | v | v | |
----------------------------------------
| Line Out | | | v |
----------------------------------------
| Phone In/Out | v | v | |
----------------------------------------
Add an ASoC component driver for it. This should be tied to the codec
audio card as an auxiliary device. This patch adds the commont paths
and controls, and variant specific headphone out and line out.
Signed-off-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
The later Allwinner SoCs have a dedicated reset controller, and
peripherals have dedicated reset controls which need to be deasserted
before the associated peripheral can be used.
Add support for this to the quirks structure and probe/remove functions.
Signed-off-by: Chen-Yu Tsai <wens@csie.org>
Acked-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The A31's internal codec capture path has a mixer in front of the ADC
for each channel, capable of selecting various inputs, including
microphones, line in, phone in, and the main output mixer.
This patch adds the various controls, widgets and routes needed for
audio capture from the already supported inputs on the A31.
Signed-off-by: Chen-Yu Tsai <wens@csie.org>
Acked-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In our i2s driver, we were previously trying to guess which oversample the
user wanted to use by looking at the rate and trying to max it.
However, the cards, and especially simple-card with its mclk-fs property
will already provide the expected oversample ratio by using the set_sysclk
callback.
We can thus implement it and remove the logic to deal with the runtime
guess.
Signed-off-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The A31 SoC's codec has various inputs, outputs and microphone bias
supplies. These can be routed on the board in different ways, such as:
- HPCOM may be connected to have the headphone DC coupled.
- Microphones all use the MBIAS main microphone supply or one mic may
use the HBIAS supply, which supports headset detection and buttons.
- Line Out may be routed to an audio jack, or an onboard speaker amp
with power controls.
Add support for specifying the audio routes in the device tree.
Signed-off-by: Chen-Yu Tsai <wens@csie.org>
Acked-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The A31 internal codec has 3 microphone outputs, of which MIC2 and MIC3
are muxed internally. The resulting two microphone inputs have separate
gain controls and mixer inputs.
The codec also has 2 microphone bias pins. HBIAS is specifically for the
headphone jack, which also supports headphone detection and control
buttons. These extra functions are not supported yet. The other, MBIAS,
is for all other analog microphones.
There is also mention of digital microphone support, but documentation
is scarce, and no hardware with it is available.
Signed-off-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
The A31 integrated codec has a second "Line Out" output which does not
include an integrated amplifier in its path. This path does have a
separate volume control.
This patch adds support for the playback path from the DAC to the Line
Out pins.
Signed-off-by: Chen-Yu Tsai <wens@csie.org>
Acked-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The A31 integrated codec has a stereo "Line In" input. Add support for
it to the playback paths.
Signed-off-by: Chen-Yu Tsai <wens@csie.org>
Acked-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The A31 has a similar codec to the A10/A20. The PCM parts are very
similar, with different register offsets. The analog paths are very
different. There are more inputs and outputs. The ADC mux has been
replaced with a proper mixer.
This patch adds support for the basic playback path of the A31 codec,
from the DAC to the headphones. Headphone detection, microphone,
signaling, other inputs/outputs and capture will be added later.
Signed-off-by: Chen-Yu Tsai <wens@csie.org>
Acked-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
According to the DMA engine API documentation, maxburst denotes the
largest possible size of a single transfer, so as not to overflow
destination FIFOs as explained in this excerpt from dmaengine.h
* @src_maxburst: the maximum number of words (note: words, as in
* units of the src_addr_width member, not bytes) that can be sent
* in one burst to the device. Typically something like half the
* FIFO depth on I/O peripherals so you don't overflow it. This
* may or may not be applicable on memory sources.
* @dst_maxburst: same as src_maxburst but for destination target
* mutatis mutandis.
The TX FIFO is 64 samples deep for stereo, and the RX FIFO is 16
samples deep. So maxburst could be 32 and 8 for TX and RX respectively.
Unfortunately the sunxi DMA controller driver takes maxburst as
the requested burst size, rather than a limit, and returns an error
for unsupported values. The original value was 4, but some later
SoCs do not officially support this burst size.
This patch increases maxburst on the TX side to 8, which is supported
by all variants of the sunxi DMA controller.
Signed-off-by: Chen-Yu Tsai <wens@csie.org>
Acked-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The A31 has a similar codec to the A10/A20. The PCM parts are very
similar, with just different register offsets. The analog paths are
very different. There are more inputs and outputs.
The A31s, A23, and H3 have a similar PCM interface, again with register
offsets slightly rearranged. The analog path controls, while very
similar between them and the A31, have been moved a separate bus which
is accessed through a message box like interface in the PRCM address
range. This would be handled by a separate auxiliary device tied in
through the device tree in its supporting create_card function.
The quirks structure is expanded to include different register offsets
and separate callbacks for creating the ASoC card. The regmap_config,
quirks, and of_device_match tables have been moved to facilitate this.
Signed-off-by: Chen-Yu Tsai <wens@csie.org>
Acked-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
