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The .remove() callback for a platform driver returns an int which makes
many driver authors wrongly assume it's possible to do error handling by
returning an error code. However the value returned is (mostly) ignored
and this typically results in resource leaks. To improve here there is a
quest to make the remove callback return void. In the first step of this
quest all drivers are converted to .remove_new() which already returns
void.
Trivially convert this driver from always returning zero in the remove
callback to the void returning variant.
Signed-off-by: Uwe Kleine-König <u.kleine-koenig@pengutronix.de>
Acked-by: Jernej Skrabec <jernej.skrabec@gmail.com>
Acked-by: Takashi Iwai <tiwai@suse.de>
Acked-by: Nicolas Ferre <nicolas.ferre@microchip.com>
Link: https://lore.kernel.org/r/20230315150745.67084-141-u.kleine-koenig@pengutronix.de
Signed-off-by: Mark Brown <broonie@kernel.org>
At present, succesfull probing of H3 Codec results in an error
debugfs: Directory '1c22c00.codec' with parent 'H3 Audio Codec' already present!
This is caused by a directory name conflict between codec
components. Fix it by setting debugfs_prefix for the CPU DAI
component.
Signed-off-by: Mikhail Rudenko <mike.rudenko@gmail.com>
Link: https://lore.kernel.org/r/20220913212256.151799-2-mike.rudenko@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
In the case when a codec device is probed before codec analog
controls, snd_soc_register_card() returns -EPROBE_DEFER, resulting in
a misleading error message
sun4i-codec 1c22c00.codec: Failed to register our card
even if the device is probed successfully later. Use dev_err_probe()
to demote the above error to a debug message.
Signed-off-by: Mikhail Rudenko <mike.rudenko@gmail.com>
Acked-by: Jernej Skrabec <jernej.skrabec@gmail.com>
Link: https://lore.kernel.org/r/20220911145713.55199-1-mike.rudenko@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The ASoC core has now been changed to default to the non-legacy DAI
naming, as such drivers using the new scheme no longer need to specify
the non_legacy_dai_naming flag.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20220623125250.2355471-42-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Change the legacy DAI naming flag from opting in to the new scheme
(non_legacy_dai_naming), to opting out of it (legacy_dai_naming).
These drivers appear to be on the CPU side of the DAI link and
currently uses the legacy naming, so add the new flag.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20220623125250.2355471-15-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Appropriately change calls to {regmap/regmap_field}_update_bits()
with {regmap/regmap_field}_set_bits()
and {regmap/regmap_field}_clear_bits() for improved readability.
Signed-off-by: Li Chen <lchen@ambarella.com>
Link: https://lore.kernel.org/r/180eef50e96.cb7c34db60740.8898768158778553647@zohomail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
devm_ioremap_resource() prints error message in itself. Remove the
dev_err call to avoid redundant error message.
Signed-off-by: Muhammad Usama Anjum <musamaanjum@gmail.com>
Link: https://lore.kernel.org/r/20210407095634.GA1379642@LEGION
Signed-off-by: Mark Brown <broonie@kernel.org>
card->owner is a required property and since commit 81033c6b58 ("ALSA:
core: Warn on empty module") a warning is issued if it is empty. Add it.
This fixes following warning observed on Lamobo R1:
WARNING: CPU: 1 PID: 190 at sound/core/init.c:207 snd_card_new+0x430/0x480 [snd]
Modules linked in: sun4i_codec(E+) sun4i_backend(E+) snd_soc_core(E) ...
CPU: 1 PID: 190 Comm: systemd-udevd Tainted: G C E 5.10.0-1-armmp #1 Debian 5.10.4-1
Hardware name: Allwinner sun7i (A20) Family
Call trace:
(snd_card_new [snd])
(snd_soc_bind_card [snd_soc_core])
(snd_soc_register_card [snd_soc_core])
(sun4i_codec_probe [sun4i_codec])
Fixes: 45fb6b6f2a ("ASoC: sunxi: add support for the on-chip codec on early Allwinner SoCs")
Related: commit 3c27ea23ff ("ASoC: qcom: Set card->owner to avoid warnings")
Related: commit ec653df2a0 ("drm/vc4/vc4_hdmi: fill ASoC card owner")
Cc: linux-arm-kernel@lists.infradead.org
Cc: alsa-devel@alsa-project.org
Signed-off-by: Bastian Germann <bage@linutronix.de>
Link: https://lore.kernel.org/r/20210331151843.30583-1-bage@linutronix.de
Signed-off-by: Mark Brown <broonie@kernel.org>
commit 3f780533ba ("ASoC: sunxi: sun4i-codec: don't select unnecessary
Platform")
Current ALSA SoC avoid to add duplicate component to rtd,
and this driver was selecting CPU component as Platform component.
Thus, above patch removed Platform settings from this driver,
because it assumed these are same component.
But, some CPU driver is using generic DMAEngine, in such case, both
CPU component and Platform component will have same of_node/name.
In other words, there are some components which are different but
have same of_node/name.
In such case, Card driver definitely need to select Platform even
though it is same as CPU.
It is depends on CPU driver, but is difficult to know it from Card driver.
This patch reverts above patch.
Fixes: commit 3f780533ba ("ASoC: sunxi: sun4i-codec: don't select unnecessary Platform")
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
ALSA SoC is now supporting "no Platform". Sound card doesn't need to
select "CPU component" as "Platform" anymore if it doesn't need
special Platform.
This patch removes such settings.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
you might feel like a deja vu to receive a bulk of changes at rc5,
and it happens again; we've got a collection of fixes for ASoC.
Most of fixes are targeted for the newly merged SOF (Sound Open
Firmware) stuff and the relevant fixes for Intel platforms.
Other than that, there are a few regression fixes for the recent
ASoC core changes and HD-audio quirk, as well as a couple of
FireWire fixes and for other ASoC codecs.
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Merge tag 'sound-5.2-rc5' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"It might feel like deja vu to receive a bulk of changes at rc5, and it
happens again; we've got a collection of fixes for ASoC. Most of fixes
are targeted for the newly merged SOF (Sound Open Firmware) stuff and
the relevant fixes for Intel platforms.
Other than that, there are a few regression fixes for the recent ASoC
core changes and HD-audio quirk, as well as a couple of FireWire fixes
and for other ASoC codecs"
* tag 'sound-5.2-rc5' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (54 commits)
Revert "ALSA: hda/realtek - Improve the headset mic for Acer Aspire laptops"
ALSA: ice1712: Check correct return value to snd_i2c_sendbytes (EWS/DMX 6Fire)
ALSA: oxfw: allow PCM capture for Stanton SCS.1m
ALSA: firewire-motu: fix destruction of data for isochronous resources
ASoC: Intel: sst: fix kmalloc call with wrong flags
ASoC: core: Fix deadlock in snd_soc_instantiate_card()
SoC: rt274: Fix internal jack assignment in set_jack callback
ALSA: hdac: fix memory release for SST and SOF drivers
ASoC: SOF: Intel: hda: use the defined ppcap functions
ASoC: core: move DAI pre-links initiation to snd_soc_instantiate_card
ASoC: Intel: cht_bsw_rt5672: fix kernel oops with platform_name override
ASoC: Intel: cht_bsw_nau8824: fix kernel oops with platform_name override
ASoC: Intel: bytcht_es8316: fix kernel oops with platform_name override
ASoC: Intel: cht_bsw_max98090: fix kernel oops with platform_name override
ASoC: sun4i-i2s: Add offset to RX channel select
ASoC: sun4i-i2s: Fix sun8i tx channel offset mask
ASoC: max98090: remove 24-bit format support if RJ is 0
ASoC: da7219: Fix build error without CONFIG_I2C
ASoC: SOF: Intel: hda: Fix COMPILE_TEST build error
ASoC: SOF: fix DSP oops definitions in FW ABI
...
ASoC is now supporting modern style dai_link
(= snd_soc_dai_link_component) for CPU/Codec/Platform.
This patch switches to use it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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>
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 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>
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>
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>
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>
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>
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>
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 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>
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>
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>
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>
