- Alsa-devel - mailman.alsa-project.org

Re: [PATCH 6/6] i2c: Make remove callback return void
by Pavel Machek 17 Jul '22

17 Jul '22

Hi! > From: Uwe Kleine-König <uwe(a)kleine-koenig.org> > > The value returned by an i2c driver's remove function is mostly ignored. > (Only an error message is printed if the value is non-zero that the > error is ignored.) > > So change the prototype of the remove function to return no value. This > way driver authors are not tempted to assume that passing an error to > the upper layer is a good idea. All drivers are adapted accordingly. > There is no intended change of behaviour, all callbacks were prepared to > return 0 before. > > Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> 2-4: Acked-by: Pavel Machek <pavel(a)ucw.cz> Best regards, Pavel -- People of Russia, stop Putin before his war on Ukraine escalates.

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Inappropriate Envy24 rate locking/XRUNS for playback and/or recording
by GitHub issues - opened 17 Jul '22

17 Jul '22

alsa-project/alsa-tools issue #11 was opened from foobers: On Debian 'oldstable', there's a rate-locking and 'low pitch' problem with following setup (same on Mint/Unstable): - dedicated audio card: Multimedia audio controller: VIA Technologies Inc. ICE1712 [Envy24] PCI Multi-Channel I/O Controller (rev 02) based - integrated audio in graphics card: Audio device: Advanced Micro Devices, Inc. [AMD/ATI] Device xxxx - onboard audio chip: Audio device: Advanced Micro Devices, Inc. [AMD/ATI] xxx (Intel HDA) (xxxx) Initially, in 'alsamixer', there are two values shown for 'Multi Track Internal Clock': '[44100]' and 'Default [41000]' for the dedicated AC. In alsamixer, it's only possible, to change the first value, but without any noticable effect (only using 'Envy24 Control' or 'Mudita24' seem to have). Now, using the audio card, sometimes, right from the start, playback of internet audio streams (in Firefox) seems to have a wrong sampling rate set (i. e. probably 48 kHz), resulting in audibly lower 'pitch' and sound distortion. Even worse, playback isn't always affected from start, but recordings from 'gnome-sound-recorder' will be in either case. This can result in several hours of recorded audio being borked, since there's no controlling of the outcome, until recording is stopped, which is very undesirable, especially in ad-hoc recording situations. I was able to track down the problem being linked to 'Master clock', 'Rate state' settings 'Locked' and 'Reset' in 'Envy24 Control Utility': checking and unchecking / switching forth to 48kHz and back to 44100Hz, helped resetting both playback and recording sampling rate to 44100Hz, to perform regular audio recordings with 'gnome-sound-recorder'. The rate locking problem can also occur, after playing MP4 videos with Mplayer (media info: sampling rate 44100Hz); then, audio files (media info: bit rate 112Kbit/s, sampling rate 44100Hz), previously kown-good audio records will be played at a noticably lower 'pitch', also recording (with gnome-recorder) now will record at a lower pitch, i. e wrong rate. In alsamixer, there are now two different values shown for 'Multi Track Internal Clock': '[44100]' and 'Default [48000]'. Also, when trying to play a regular film DVD with VLC now, it very likely shows short dropouts of black screen and audio, at steady intervals of ~2s, requiring performing a logoff and login, to reset rates again. Interestingly (and unexpectedly), setting 'Master clock' to 'S/PDIF in', seemed to internally freeze the clock / source to 'S/PDIF', in effect preventing any more sound output or recording from the device; trying to switch back to 44110Hz or 48000Hz would fail and the could only be reset, after until doing a complete reboot. I've seen references, pointing to the rate locking issue, which suggest, that unchecking 'locked' and 'reset' for 'Rate state' in 'Envy24' would be the key here - only, on my system, those values didn't seem to be initially checked (at least not visually): https://discourse.ardour.org/t/all-sound-is-low-pitched/82149 So the solution here would seem to be, to ensure unset 'Rate state' options at boot time; in contrast to this suggestion, the documentation says, 'locked' might cause 'errors' and 'XRUNS', and 'reset' only would be the recommended setting by default, to allow applications alter the clock sample rate to their needs, resetting them to the selected 'Master clock' default afterwards: https://wiki.archlinux.org/title/Envy24control#Rate_State While this seems to make sense, 'reset' not being set appropriatly, might also explain some of the weird behavior described above, namely, when some application sets and locks 48kHz for playback and/or input channels, which then don't match the input and/or output of another application, started after that. The effect of VLC/MPlayer showing 'dropouts' may also have its cause in the mentioned 'XRUNS', in this case, supposedly buffer underruns, when applying an 48kHz sampling rate to video files with 44,1kHz audio streams, or buffer overflows, when applying 44,1kHz sampling to files with 48kHz audio streams: https://unix.stackexchange.com/questions/199498/what-are-xruns What (combination) of 'Rate state' settings (if any) would be a 'bullet-proof' remedy here, I cannot tell, since my settings obviously both empty. So, either this was a false visual feedback, or 'reset' does not work as intended. So far, I can't confirm, if these settings differ in Mint, where the problem shows, too, as said (or if the defaults have been altered). Currently I can't try perma-setting 'reset' only (the recommended default), but I think it didn't work out before for recording. Issue URL : https://github.com/alsa-project/alsa-tools/issues/11 Repository URL: https://github.com/alsa-project/alsa-tools

1 0

[PATCH] ASoC: DPCM: Don't pick up BE without substream
by Takashi Iwai 17 Jul '22

17 Jul '22

When DPCM tries to add valid BE connections at dpcm_add_paths(), it doesn't check whether the picked BE actually supports for the given stream direction. Due to that, when an asymmetric BE stream is present, it picks up wrongly and this may result in a NULL dereference at a later point where the code assumes the existence of a corresponding BE substream. This patch adds the check for the presence of the substream for the target BE for avoiding the problem above. Note that we have already some fix for non-existing BE substream at commit 6246f283d5e0 ("ASoC: dpcm: skip missing substream while applying symmetry"). But the code path we've hit recently is rather happening before the previous fix. So this patch tries to fix at picking up a BE instead of parsing BE lists. Fixes: bbf7d3b1c4f4 ("ASoC: soc-pcm: align BE 'atomicity' with that of the FE") Reported-by: Alex Natalsson <harmoniesworlds(a)gmail.com> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Link: https://lore.kernel.org/r/CADs9LoPZH_D+eJ9qjTxSLE5jGyhKsjMN7g2NighZ16biVxsy… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/soc-pcm.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index a827cc3c158a..0c1de5624842 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1318,6 +1318,9 @@ static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card, if (!be->dai_link->no_pcm) continue; + if (!snd_soc_dpcm_get_substream(be, stream)) + continue; + for_each_rtd_dais(be, i, dai) { w = snd_soc_dai_get_widget(dai, stream); -- 2.35.3

1 0

Audio interface unable to be loaded in 1.2.7.2
by GitHub issues - opened 17 Jul '22

17 Jul '22

alsa-project/alsa-lib issue #254 was opened from KevHg: I upgraded from 1.2.6.1 to 1.2.7.2 and found that pulseaudio no longer detects my Motu M2 interface. Main error log seems to be: `module-alsa-card.c: Failed to find a working profile.` `failed to load module module-alsa-card` I'm new to debugging Linux issues, let me know what other info I can provide to help out on this topic. Issue URL : https://github.com/alsa-project/alsa-lib/issues/254 Repository URL: https://github.com/alsa-project/alsa-lib

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[PATCH 0/5] ALSA: Drop async signal support
by Takashi Iwai 16 Jul '22

16 Jul '22

Hi, this is a patch series to drop one of unused features, the async signal support, from ALSA core. The async signal itself is very difficult to use properly due to various restrictions (e.g. you cannot perform any I/O in the context), hence it's a feature that has been never used by real applications. OTOH, the real problem is that there have been quite a few syzcaller reports indicating that fasync code path may lead to some potential deadlocks for long time. Dropping the feature is the easiest solution, obviously. The corresponding update for alsa-lib will follow once when we agree with this approach. thanks, Takashi === Takashi Iwai (5): ALSA: timer: Drop async signal support ALSA: pcm: Drop async signal support ALSA: control: Drop async signal support ALSA: core: Drop async signal support ALSA: doc: Drop stale fasync entry .../kernel-api/writing-an-alsa-driver.rst | 1 - include/sound/control.h | 1 - include/sound/pcm.h | 1 - sound/core/control.c | 11 ----------- sound/core/init.c | 11 +---------- sound/core/pcm_lib.c | 4 +--- sound/core/pcm_native.c | 18 ------------------ sound/core/timer.c | 13 ------------- 8 files changed, 2 insertions(+), 58 deletions(-) -- 2.35.3

2 7

[PATCH 0/4] ASoC: SOF: Intel: add support for SoundWire-based HP Omen16
by Pierre-Louis Bossart 15 Jul '22

15 Jul '22

This device exposes a headset codec on link0 and an amplifier on link3. This is a very unusual pin-muxing, usually the microphones are pin-muxed with link2/link3. This resulted in a problematic error handling leading to a kernel oops, and invalidated a hard-coded assumption. Full support for this device requires a DMI quirk shared separately ("soundwire: dmi-quirks: add remapping for HP Omen 16-k0005TX"). Pierre-Louis Bossart (4): ASoC: Intel: sof_sdw: avoid oops in error handling ASoC: Intel: soc-acpi: add table for HP Omen 16-k0005TX ASoC: Intel: sof_sdw: add quirk for HP Omen 16-k0005TX ASoC: SOF: Intel: enable dmic handling with 2 or fewer SoundWire links sound/soc/intel/boards/sof_sdw.c | 9 ++++++ sound/soc/intel/boards/sof_sdw_rt711.c | 3 ++ sound/soc/intel/boards/sof_sdw_rt711_sdca.c | 3 ++ .../intel/common/soc-acpi-intel-adl-match.c | 29 +++++++++++++++++++ sound/soc/sof/intel/hda.c | 8 ++--- 5 files changed, 48 insertions(+), 4 deletions(-) -- 2.34.1

2 5

[PATCH] ASoC: codecs: add support for ES8326
by Zhu Ning 15 Jul '22

15 Jul '22

The ES8326 codec is not compatible with ES8316 and requires a dedicated driver. Signed-off-by: David Yang <yangxiaohua(a)everest-semi.com> Signed-off-by: Zhu Ning <zhuning(a)everest-semi.com> --- sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/es8326.c | 794 ++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/es8326.h | 187 +++++++++ 4 files changed, 988 insertions(+) mode change 100644 => 100755 sound/soc/codecs/Kconfig mode change 100644 => 100755 sound/soc/codecs/Makefile create mode 100755 sound/soc/codecs/es8326.c create mode 100755 sound/soc/codecs/es8326.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig old mode 100644 new mode 100755 index 6165db92a629..b0a15cce416d --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -98,6 +98,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_DA9055 imply SND_SOC_DMIC imply SND_SOC_ES8316 + imply SND_SOC_ES8326 imply SND_SOC_ES8328_SPI imply SND_SOC_ES8328_I2C imply SND_SOC_ES7134 @@ -911,6 +912,10 @@ config SND_SOC_ES8316 tristate "Everest Semi ES8316 CODEC" depends on I2C +config SND_SOC_ES8326 + tristate "Everest Semi ES8326 CODEC" + depends on I2C + config SND_SOC_ES8328 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile old mode 100644 new mode 100755 index 28dc4edfd01f..ae9d789b5381 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -100,6 +100,7 @@ snd-soc-dmic-objs := dmic.o snd-soc-es7134-objs := es7134.o snd-soc-es7241-objs := es7241.o snd-soc-es8316-objs := es8316.o +snd-soc-es8326-objs := es8326.o snd-soc-es8328-objs := es8328.o snd-soc-es8328-i2c-objs := es8328-i2c.o snd-soc-es8328-spi-objs := es8328-spi.o @@ -452,6 +453,7 @@ obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_ES7134) += snd-soc-es7134.o obj-$(CONFIG_SND_SOC_ES7241) += snd-soc-es7241.o obj-$(CONFIG_SND_SOC_ES8316) += snd-soc-es8316.o +obj-$(CONFIG_SND_SOC_ES8326) += snd-soc-es8326.o obj-$(CONFIG_SND_SOC_ES8328) += snd-soc-es8328.o obj-$(CONFIG_SND_SOC_ES8328_I2C)+= snd-soc-es8328-i2c.o obj-$(CONFIG_SND_SOC_ES8328_SPI)+= snd-soc-es8328-spi.o diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c new file mode 100755 index 000000000000..d62c596ac878 --- /dev/null +++ b/sound/soc/codecs/es8326.c @@ -0,0 +1,794 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// es8326.c -- es8326 ALSA SoC audio driver +// Copyright Everest Semiconductor Co., Ltd +// +// Authors: David Yang <yangxiaohua(a)everest-semi.com> +// + +#include <linux/clk.h> +#include <linux/i2c.h> +#include <linux/interrupt.h> +#include <linux/irq.h> +#include <linux/module.h> +#include <sound/jack.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> +#include "es8326.h" + +struct es8326_priv { + struct clk *mclk; + struct snd_pcm_hw_constraint_list *sysclk_constraints; + struct i2c_client *i2c; + struct regmap *regmap; + struct snd_soc_component *component; + struct delayed_work jack_detect_work; + struct snd_soc_jack *jack; + int irq; + /* The lock protects the situation that an irq is generated + * while enabling or disabling irq. + */ + struct mutex lock; + u8 mic1_src; + u8 mic2_src; + u8 jack_pol; + bool jd_inverted; + unsigned int sysclk; +}; + +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(dac_vol_tlv, -9550, 50, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(adc_vol_tlv, -9550, 50, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(adc_analog_pga_tlv, 0, 300, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(adc_pga_tlv, 0, 600, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(softramp_rate, 0, 100, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(drc_target_tlv, -3200, 200, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(drc_recovery_tlv, -125, 250, 0); + +static const char *const winsize[] = { + "0.25db/2 LRCK", + "0.25db/4 LRCK", + "0.25db/8 LRCK", + "0.25db/16 LRCK", + "0.25db/32 LRCK", + "0.25db/64 LRCK", + "0.25db/128 LRCK", + "0.25db/256 LRCK", + "0.25db/512 LRCK", + "0.25db/1024 LRCK", + "0.25db/2048 LRCK", + "0.25db/4096 LRCK", + "0.25db/8192 LRCK", + "0.25db/16384 LRCK", + "0.25db/32768 LRCK", + "0.25db/65536 LRCK", +}; + +static const char *const dacpol_txt[] = { + "Normal", "R Invert", "L Invert", "L + R Invert" }; + +static const struct soc_enum dacpol = + SOC_ENUM_SINGLE(ES8326_DAC_DSM_4D, 4, 4, dacpol_txt); +static const struct soc_enum alc_winsize = + SOC_ENUM_SINGLE(ES8326_ADC_RAMPRATE_2E, 4, 16, winsize); +static const struct soc_enum drc_winsize = + SOC_ENUM_SINGLE(ES8326_DRC_WINSIZE_54, 4, 16, winsize); + +static const struct snd_kcontrol_new es8326_snd_controls[] = { + SOC_SINGLE_TLV("DAC Playback Volume", ES8326_DAC_VOL_50, 0, 0xbf, 0, dac_vol_tlv), + SOC_ENUM("Playback Polarity", dacpol), + SOC_SINGLE_TLV("DAC Ramp Rate", ES8326_DAC_RAMPRATE_4E, 0, 0x0f, 0, softramp_rate), + SOC_SINGLE_TLV("DRC Recovery Level", ES8326_DRC_RECOVERY_53, 0, 4, 0, drc_recovery_tlv), + SOC_ENUM("DRC Winsize", drc_winsize), + SOC_SINGLE_TLV("DRC Target Level", ES8326_DRC_WINSIZE_54, 0, 0x0f, 0, drc_target_tlv), + + SOC_DOUBLE_R_TLV("ADC Capture Volume", ES8326_ADC1_VOL_2C, ES8326_ADC2_VOL_2D, 0, 0xff, 0, + adc_vol_tlv), + SOC_DOUBLE_TLV("ADC PGA Volume", ES8326_ADC_SCALE_29, 4, 0, 5, 0, adc_pga_tlv), + SOC_SINGLE_TLV("ADC PGA Gain Volume", ES8326_PGAGAIN_23, 0, 10, 0, adc_analog_pga_tlv), + SOC_SINGLE_TLV("ADC Ramp Rate", ES8326_ADC_RAMPRATE_2E, 0, 0x0f, 0, softramp_rate), + SOC_SINGLE("ALC Capture Switch", ES8326_ALC_RECOVERY_32, 3, 1, 0), + SOC_SINGLE_TLV("ALC Capture Recovery Level", ES8326_ALC_LEVEL_33, + 0, 4, 0, drc_recovery_tlv), + SOC_ENUM("ALC Capture Winsize", alc_winsize), + SOC_SINGLE_TLV("ALC Capture Target Level", ES8326_ALC_LEVEL_33, + 0, 0x0f, 0, drc_target_tlv), + +}; + +static const struct snd_soc_dapm_widget es8326_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("MIC2"), + SND_SOC_DAPM_INPUT("MIC3"), + SND_SOC_DAPM_INPUT("MIC4"), + + SND_SOC_DAPM_ADC("ADC L", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC R", NULL, SND_SOC_NOPM, 0, 0), + + /* Digital Interface */ + SND_SOC_DAPM_AIF_OUT("I2S OUT", "I2S1 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("I2S IN", "I2S1 Playback", 0, SND_SOC_NOPM, 0, 0), + + /* ADC Digital Mute */ + SND_SOC_DAPM_PGA("ADC L1", ES8326_ADC_MUTE_15, 0, 1, NULL, 0), + SND_SOC_DAPM_PGA("ADC R1", ES8326_ADC_MUTE_15, 1, 1, NULL, 0), + SND_SOC_DAPM_PGA("ADC L2", ES8326_ADC_MUTE_15, 2, 1, NULL, 0), + SND_SOC_DAPM_PGA("ADC R2", ES8326_ADC_MUTE_15, 3, 1, NULL, 0), + + /* Analog Power Supply*/ + SND_SOC_DAPM_DAC("Right DAC", NULL, ES8326_ANA_PDN_16, 0, 1), + SND_SOC_DAPM_DAC("Left DAC", NULL, ES8326_ANA_PDN_16, 1, 1), + SND_SOC_DAPM_SUPPLY("Analog Power", ES8326_ANA_PDN_16, 7, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("IBias Power", ES8326_ANA_PDN_16, 6, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC Vref", ES8326_ANA_PDN_16, 5, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC Vref", ES8326_ANA_PDN_16, 4, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Vref Power", ES8326_ANA_PDN_16, 3, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS1", ES8326_ANA_MICBIAS_1B, 2, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS2", ES8326_ANA_MICBIAS_1B, 3, 0, NULL, 0), + + SND_SOC_DAPM_PGA("LHPMIX", ES8326_DAC2HPMIX_25, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("RHPMIX", ES8326_DAC2HPMIX_25, 3, 0, NULL, 0), + + /* Headphone Charge Pump and Output */ + SND_SOC_DAPM_SUPPLY("HPOR Cal", ES8326_HP_CAL_27, 7, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("HPOL Cal", ES8326_HP_CAL_27, 3, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Headphone Charge Pump", ES8326_HP_DRIVER_24, + 3, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Headphone Driver Bias", ES8326_HP_DRIVER_24, + 2, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Headphone LDO", ES8326_HP_DRIVER_24, + 1, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Headphone Reference", ES8326_HP_DRIVER_24, + 0, 1, NULL, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_supply, "HPOR Supply", ES8326_HP_CAL_27, + ES8326_HPOR_SHIFT, 7, 7, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_supply, "HPOL Supply", ES8326_HP_CAL_27, + 0, 7, 7, 0), + + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), +}; + +static const struct snd_soc_dapm_route es8326_dapm_routes[] = { + {"ADC L1", NULL, "MIC1"}, + {"ADC R1", NULL, "MIC2"}, + {"ADC L2", NULL, "MIC3"}, + {"ADC R2", NULL, "MIC4"}, + + + {"ADC L", NULL, "ADC L1"}, + {"ADC R", NULL, "ADC R1"}, + {"ADC L", NULL, "ADC L2"}, + {"ADC R", NULL, "ADC R2"}, + + {"I2S OUT", NULL, "ADC L"}, + {"I2S OUT", NULL, "ADC R"}, + + {"I2S OUT", NULL, "Analog Power"}, + {"I2S OUT", NULL, "ADC Vref"}, + {"I2S OUT", NULL, "Vref Power"}, + {"I2S OUT", NULL, "IBias Power"}, + {"I2S IN", NULL, "Analog Power"}, + {"I2S IN", NULL, "DAC Vref"}, + {"I2S IN", NULL, "Vref Power"}, + {"I2S IN", NULL, "IBias Power"}, + + {"Right DAC", NULL, "I2S IN"}, + {"Left DAC", NULL, "I2S IN"}, + + {"LHPMIX", NULL, "Left DAC"}, + {"RHPMIX", NULL, "Right DAC"}, + + {"HPOR", NULL, "HPOR Cal"}, + {"HPOL", NULL, "HPOL Cal"}, + {"HPOR", NULL, "HPOR Supply"}, + {"HPOL", NULL, "HPOL Supply"}, + {"HPOL", NULL, "Headphone Charge Pump"}, + {"HPOR", NULL, "Headphone Charge Pump"}, + {"HPOL", NULL, "Headphone Driver Bias"}, + {"HPOR", NULL, "Headphone Driver Bias"}, + {"HPOL", NULL, "Headphone LDO"}, + {"HPOR", NULL, "Headphone LDO"}, + {"HPOL", NULL, "Headphone Reference"}, + {"HPOR", NULL, "Headphone Reference"}, + + {"HPOL", NULL, "LHPMIX"}, + {"HPOR", NULL, "RHPMIX"}, +}; + +static const struct regmap_range es8326_volatile_ranges[] = { + regmap_reg_range(ES8326_HP_DETECT_FB, ES8326_HP_DETECT_FB), +}; + +static const struct regmap_access_table es8326_volatile_table = { + .yes_ranges = es8326_volatile_ranges, + .n_yes_ranges = ARRAY_SIZE(es8326_volatile_ranges), +}; + +const struct regmap_config es8326_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + .max_register = 0xff, + .volatile_table = &es8326_volatile_table, + .cache_type = REGCACHE_RBTREE, +}; + +struct _coeff_div { + u16 fs; + u32 rate; + u32 mclk; + u8 reg4; + u8 reg5; + u8 reg6; + u8 reg7; + u8 reg8; + u8 reg9; + u8 rega; + u8 regb; +}; + +/* codec hifi mclk clock divider coefficients */ +/* {ratio, LRCK, MCLK, REG04, REG05, REG06, REG07, REG08, REG09, REG10, REG11} */ +static const struct _coeff_div coeff_div[] = { + {32, 8000, 256000, 0x60, 0x00, 0x0F, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {32, 16000, 512000, 0x20, 0x00, 0x0D, 0x75, 0x0A, 0x1B, 0x1F, 0x3F}, + {32, 44100, 1411200, 0x00, 0x00, 0x13, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {32, 48000, 1536000, 0x00, 0x00, 0x13, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {36, 8000, 288000, 0x20, 0x00, 0x0D, 0x75, 0x0A, 0x1B, 0x23, 0x47}, + {36, 16000, 576000, 0x20, 0x00, 0x0D, 0x75, 0x0A, 0x1B, 0x23, 0x47}, + {48, 8000, 384000, 0x60, 0x02, 0x1F, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {48, 16000, 768000, 0x20, 0x02, 0x0F, 0x75, 0x0A, 0x1B, 0x1F, 0x3F}, + {48, 48000, 2304000, 0x00, 0x02, 0x0D, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {64, 8000, 512000, 0x60, 0x00, 0x0D, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {64, 16000, 1024000, 0x20, 0x00, 0x05, 0x75, 0x0A, 0x1B, 0x1F, 0x3F}, + + {64, 44100, 2822400, 0x00, 0x00, 0x11, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {64, 48000, 3072000, 0x00, 0x00, 0x11, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {72, 8000, 576000, 0x20, 0x00, 0x13, 0x35, 0x0A, 0x1B, 0x23, 0x47}, + {72, 16000, 1152000, 0x20, 0x00, 0x05, 0x75, 0x0A, 0x1B, 0x23, 0x47}, + {96, 8000, 768000, 0x60, 0x02, 0x1D, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {96, 16000, 1536000, 0x20, 0x02, 0x0D, 0x75, 0x0A, 0x1B, 0x1F, 0x3F}, + {100, 48000, 4800000, 0x04, 0x04, 0x3F, 0x6D, 0x38, 0x08, 0x4f, 0x1f}, + {125, 48000, 6000000, 0x04, 0x04, 0x1F, 0x2D, 0x0A, 0x0A, 0x27, 0x27}, + {128, 8000, 1024000, 0x60, 0x00, 0x13, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {128, 16000, 2048000, 0x20, 0x00, 0x11, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + + {128, 44100, 5644800, 0x00, 0x00, 0x01, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {128, 48000, 6144000, 0x00, 0x00, 0x01, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {144, 8000, 1152000, 0x20, 0x00, 0x03, 0x35, 0x0A, 0x1B, 0x23, 0x47}, + {144, 16000, 2304000, 0x20, 0x00, 0x11, 0x35, 0x0A, 0x1B, 0x23, 0x47}, + {192, 8000, 1536000, 0x60, 0x02, 0x0D, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {192, 16000, 3072000, 0x20, 0x02, 0x05, 0x75, 0x0A, 0x1B, 0x1F, 0x3F}, + {200, 48000, 9600000, 0x04, 0x04, 0x0F, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {250, 48000, 12000000, 0x04, 0x04, 0x0F, 0x2D, 0x0A, 0x0A, 0x27, 0x27}, + {256, 8000, 2048000, 0x60, 0x00, 0x11, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {256, 16000, 4096000, 0x20, 0x00, 0x01, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + + {256, 44100, 11289600, 0x00, 0x00, 0x10, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {256, 48000, 12288000, 0x00, 0x00, 0x30, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {288, 8000, 2304000, 0x20, 0x00, 0x01, 0x35, 0x0A, 0x1B, 0x23, 0x47}, + {384, 8000, 3072000, 0x60, 0x02, 0x05, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {384, 16000, 6144000, 0x20, 0x02, 0x03, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + {384, 48000, 18432000, 0x00, 0x02, 0x01, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {400, 48000, 19200000, 0x09, 0x04, 0x0f, 0x6d, 0x3a, 0x0A, 0x4F, 0x1F}, + {500, 48000, 24000000, 0x18, 0x04, 0x1F, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {512, 8000, 4096000, 0x60, 0x00, 0x01, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {512, 16000, 8192000, 0x20, 0x00, 0x10, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + + {512, 44100, 22579200, 0x00, 0x00, 0x00, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {512, 48000, 24576000, 0x00, 0x00, 0x00, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {768, 8000, 6144000, 0x60, 0x02, 0x11, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {768, 16000, 12288000, 0x20, 0x02, 0x01, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + {800, 48000, 38400000, 0x00, 0x18, 0x13, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {1024, 8000, 8192000, 0x60, 0x00, 0x10, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {1024, 16000, 16384000, 0x20, 0x00, 0x00, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + {1152, 16000, 18432000, 0x20, 0x08, 0x11, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + {1536, 8000, 12288000, 0x60, 0x02, 0x01, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + + {1536, 16000, 24576000, 0x20, 0x02, 0x10, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + {1625, 8000, 13000000, 0x0C, 0x18, 0x1F, 0x2D, 0x0A, 0x0A, 0x27, 0x27}, + {1625, 16000, 26000000, 0x0C, 0x18, 0x1F, 0x2D, 0x0A, 0x0A, 0x27, 0x27}, + {2048, 8000, 16384000, 0x60, 0x00, 0x00, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {2304, 8000, 18432000, 0x40, 0x02, 0x10, 0x35, 0x0A, 0x1B, 0x1F, 0x5F}, + {3072, 8000, 24576000, 0x60, 0x02, 0x10, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {3250, 8000, 26000000, 0x0C, 0x18, 0x0F, 0x2D, 0x0A, 0x0A, 0x27, 0x27}, + +}; + +static inline int get_coeff(int mclk, int rate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(coeff_div); i++) { + if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk) + return i; + } + + return -EINVAL; +} + +static unsigned int es8326_rates[] = { + 8000, 12000, 16000, 24000, 32000, 48000, 96000 +}; + +static struct snd_pcm_hw_constraint_list es8326_constraints = { + .count = ARRAY_SIZE(es8326_rates), + .list = es8326_rates, +}; + +static int es8326_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_component *codec = codec_dai->component; + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(codec); + + es8326->sysclk = freq; + + if (freq == 0) { + es8326->sysclk_constraints->list = NULL; + es8326->sysclk_constraints->count = 0; + return 0; + } + + es8326->sysclk_constraints = &es8326_constraints; + return 0; +} + +static int es8326_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_component *component = codec_dai->component; + u8 iface = 0; + + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: + snd_soc_component_update_bits(component, ES8326_RESET_00, + ES8326_MASTER_MODE_EN, ES8326_MASTER_MODE_EN); + break; + case SND_SOC_DAIFMT_CBC_CFC: + break; + default: + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_RIGHT_J: + dev_err(component->dev, "Codec driver does not support right justified\n"); + return -EINVAL; + case SND_SOC_DAIFMT_LEFT_J: + iface |= ES8326_DAIFMT_LEFT_J; + break; + case SND_SOC_DAIFMT_DSP_A: + iface |= ES8326_DAIFMT_DSP_A; + break; + case SND_SOC_DAIFMT_DSP_B: + iface |= ES8326_DAIFMT_DSP_B; + break; + default: + return -EINVAL; + } + + snd_soc_component_update_bits(component, ES8326_FMT_13, ES8326_DAIFMT_MASK, iface); + + return 0; +} + +static int es8326_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *codec = dai->component; + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(codec); + u8 srate = 0; + int coeff; + + coeff = get_coeff(es8326->sysclk, params_rate(params)); + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + srate |= ES8326_S16_LE; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + srate |= ES8326_S20_3_LE; + break; + case SNDRV_PCM_FORMAT_S18_3LE: + srate |= ES8326_S18_LE; + break; + case SNDRV_PCM_FORMAT_S24_LE: + srate |= ES8326_S24_LE; + break; + case SNDRV_PCM_FORMAT_S32_LE: + srate |= ES8326_S32_LE; + break; + default: + return -EINVAL; + } + + /* set iface & srate */ + snd_soc_component_update_bits(codec, ES8326_FMT_13, ES8326_DATA_LEN_MASK, srate); + + if (coeff >= 0) { + regmap_write(es8326->regmap, ES8326_CLK_DIV1_04, + coeff_div[coeff].reg4); + regmap_write(es8326->regmap, ES8326_CLK_DIV2_05, + coeff_div[coeff].reg5); + regmap_write(es8326->regmap, ES8326_CLK_DLL_06, + coeff_div[coeff].reg6); + regmap_write(es8326->regmap, ES8326_CLK_MUX_07, + coeff_div[coeff].reg7); + regmap_write(es8326->regmap, ES8326_CLK_ADC_SEL_08, + coeff_div[coeff].reg8); + regmap_write(es8326->regmap, ES8326_CLK_DAC_SEL_09, + coeff_div[coeff].reg9); + regmap_write(es8326->regmap, ES8326_CLK_ADC_OSR_0A, + coeff_div[coeff].rega); + regmap_write(es8326->regmap, ES8326_CLK_DAC_OSR_0B, + coeff_div[coeff].regb); + } + + return 0; +} + +static int es8326_set_bias_level(struct snd_soc_component *codec, + enum snd_soc_bias_level level) +{ + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(codec); + int ret; + + switch (level) { + case SND_SOC_BIAS_ON: + if (!IS_ERR(es8326->mclk)) { + ret = clk_prepare_enable(es8326->mclk); + if (ret) + return ret; + } + regmap_write(es8326->regmap, ES8326_RESET_00, ES8326_PWRUP_SEQ_EN); + regmap_write(es8326->regmap, ES8326_INTOUT_IO_59, 0x45); + regmap_write(es8326->regmap, ES8326_SDINOUT1_IO_5A, + (ES8326_IO_DMIC_CLK << ES8326_SDINOUT1_SHIFT)); + regmap_write(es8326->regmap, ES8326_SDINOUT23_IO_5B, ES8326_IO_INPUT); + regmap_write(es8326->regmap, ES8326_CLK_RESAMPLE_03, 0x05); + regmap_write(es8326->regmap, ES8326_VMIDSEL_18, 0x02); + regmap_write(es8326->regmap, ES8326_PGA_PDN_17, 0x40); + regmap_write(es8326->regmap, ES8326_DAC2HPMIX_25, 0xAA); + regmap_write(es8326->regmap, ES8326_RESET_00, ES8326_CSM_ON); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + break; + case SND_SOC_BIAS_OFF: + if (es8326->mclk) + clk_disable_unprepare(es8326->mclk); + regmap_write(es8326->regmap, ES8326_DAC2HPMIX_25, 0x11); + regmap_write(es8326->regmap, ES8326_RESET_00, ES8326_CSM_OFF); + regmap_write(es8326->regmap, ES8326_PGA_PDN_17, 0xF8); + regmap_write(es8326->regmap, ES8326_VMIDSEL_18, 0x00); + regmap_write(es8326->regmap, ES8326_INT_SOURCE_58, 0x08); + regmap_write(es8326->regmap, ES8326_SDINOUT1_IO_5A, ES8326_IO_INPUT); + regmap_write(es8326->regmap, ES8326_SDINOUT23_IO_5B, ES8326_IO_INPUT); + regmap_write(es8326->regmap, ES8326_RESET_00, + ES8326_CODEC_RESET | ES8326_PWRUP_SEQ_EN); + break; + } + + return 0; +} + +#define es8326_RATES SNDRV_PCM_RATE_8000_96000 + +#define es8326_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +static const struct snd_soc_dai_ops es8326_ops = { + .hw_params = es8326_pcm_hw_params, + .set_fmt = es8326_set_dai_fmt, + .set_sysclk = es8326_set_dai_sysclk, +}; + +static struct snd_soc_dai_driver es8326_dai = { + .name = "ES8326 HiFi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = es8326_RATES, + .formats = es8326_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = es8326_RATES, + .formats = es8326_FORMATS, + }, + .ops = &es8326_ops, + .symmetric_rate = 1, +}; + +static void es8326_enable_micbias(struct snd_soc_component *component) +{ + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + + snd_soc_dapm_mutex_lock(dapm); + snd_soc_dapm_force_enable_pin_unlocked(dapm, "MICBIAS1"); + snd_soc_dapm_force_enable_pin_unlocked(dapm, "MICBIAS2"); + snd_soc_dapm_sync_unlocked(dapm); + snd_soc_dapm_mutex_unlock(dapm); +} + +static void es8326_disable_micbias(struct snd_soc_component *component) +{ + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + + snd_soc_dapm_mutex_lock(dapm); + snd_soc_dapm_disable_pin_unlocked(dapm, "MICBIAS1"); + snd_soc_dapm_disable_pin_unlocked(dapm, "MICBIAS2"); + snd_soc_dapm_sync_unlocked(dapm); + snd_soc_dapm_mutex_unlock(dapm); +} +static void es8326_jack_detect_handler(struct work_struct *work) +{ + struct es8326_priv *es8326 = + container_of(work, struct es8326_priv, jack_detect_work.work); + struct snd_soc_component *comp = es8326->component; + unsigned int iface; + + iface = snd_soc_component_read(comp, ES8326_HP_DETECT_FB); + dev_dbg(comp->dev, "gpio flag %#04x", iface); + if ((iface & ES8326_HPINSERT_FLAG) == 0) { + dev_dbg(comp->dev, "No headset detected"); + snd_soc_jack_report(es8326->jack, 0, SND_JACK_HEADSET); + snd_soc_component_write(comp, ES8326_ADC1_SRC_2A, es8326->mic2_src); + es8326_disable_micbias(comp); + } else if ((iface & ES8326_HPINSERT_FLAG) == ES8326_HPINSERT_FLAG) { + if ((iface & ES8326_HPBUTTON_FLAG) == 0x00) { + dev_dbg(comp->dev, "Headset detected"); + snd_soc_jack_report(es8326->jack, SND_JACK_HEADSET, SND_JACK_HEADSET); + snd_soc_component_write(comp, ES8326_ADC1_SRC_2A, es8326->mic1_src); + } else { + dev_dbg(comp->dev, "Headphone detected"); + snd_soc_jack_report(es8326->jack, SND_JACK_HEADPHONE, SND_JACK_HEADSET); + } + } +} + +static irqreturn_t es8326_irq(int irq, void *dev_id) +{ + struct es8326_priv *es8326 = dev_id; + struct snd_soc_component *comp = es8326->component; + + if(!es8326->jack) + goto out; + es8326_enable_micbias(comp); + + queue_delayed_work(system_wq, &es8326->jack_detect_work, + msecs_to_jiffies(300)); + +out: + return IRQ_HANDLED; +} + +static int es8326_resume(struct snd_soc_component *component) +{ + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); + + regmap_write(es8326->regmap, ES8326_CLK_CTL_01, ES8326_CLK_ON); + /* Two channel ADC */ + regmap_write(es8326->regmap, ES8326_PULLUP_CTL_F9, 0x02); + regmap_write(es8326->regmap, ES8326_CLK_INV_02, 0x00); + regmap_write(es8326->regmap, ES8326_CLK_DIV_CPC_0C, 0x1F); + regmap_write(es8326->regmap, ES8326_CLK_VMIDS1_10, 0xC8); + regmap_write(es8326->regmap, ES8326_CLK_VMIDS2_11, 0x88); + regmap_write(es8326->regmap, ES8326_CLK_CAL_TIME_12, 0x20); + regmap_write(es8326->regmap, ES8326_SYS_BIAS_1D, 0x08); + regmap_write(es8326->regmap, ES8326_DAC2HPMIX_25, 0x22); + regmap_write(es8326->regmap, ES8326_ADC1_SRC_2A, es8326->mic1_src); + regmap_write(es8326->regmap, ES8326_ADC2_SRC_2B, es8326->mic2_src); + regmap_write(es8326->regmap, ES8326_HPJACK_TIMER_56, 0x88); + regmap_write(es8326->regmap, ES8326_HP_DET_57, + ES8326_HP_DET_SRC_PIN9 | es8326->jack_pol); + regmap_write(es8326->regmap, ES8326_INT_SOURCE_58, 0x08); + regmap_write(es8326->regmap, ES8326_INTOUT_IO_59, 0x45); + regmap_write(es8326->regmap, ES8326_RESET_00, ES8326_CSM_ON); + snd_soc_component_update_bits(component, ES8326_PGAGAIN_23, + ES8326_MIC_SEL_MASK, ES8326_MIC1_SEL); + + es8326_irq(es8326->irq, es8326); + return 0; +} + +static int es8326_suspend(struct snd_soc_component *component) +{ + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); + + regmap_write(es8326->regmap, ES8326_CLK_CTL_01, ES8326_CLK_OFF); + return 0; +} + +static int es8326_probe(struct snd_soc_component *component) +{ + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); + int ret; + + es8326->component = component; + es8326->jd_inverted = device_property_read_bool(component->dev, + "everest,jack-detect-inverted"); + + ret = device_property_read_u8(component->dev, "everest,mic1-src", &es8326->mic1_src); + if (ret != 0) { + dev_dbg(component->dev, "mic1-src return %d", ret); + es8326->mic1_src = ES8326_ADC_AMIC; + } + dev_dbg(component->dev, "mic1-src %x", es8326->mic1_src); + + ret = device_property_read_u8(component->dev, "everest,mic2-src", &es8326->mic2_src); + if (ret != 0) { + dev_dbg(component->dev, "mic2-src return %d", ret); + es8326->mic2_src = ES8326_ADC_DMIC; + } + dev_dbg(component->dev, "mic2-src %x", es8326->mic2_src); + + ret = device_property_read_u8(component->dev, "everest,jack-pol", &es8326->jack_pol); + if (ret != 0) { + dev_dbg(component->dev, "jack-pol return %d", ret); + es8326->jack_pol = ES8326_HP_DET_BUTTON_POL | ES8326_HP_TYPE_OMTP; + } + dev_dbg(component->dev, "jack-pol %x", es8326->jack_pol); + + es8326_resume(component); + return 0; +} + +static void es8326_enable_jack_detect(struct snd_soc_component *component, + struct snd_soc_jack *jack) +{ + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); + + mutex_lock(&es8326->lock); + if (es8326->jd_inverted) + snd_soc_component_update_bits(component, ES8326_HP_DET_57, + ES8326_HP_DET_JACK_POL, ~es8326->jack_pol); + es8326->jack = jack; + + mutex_unlock(&es8326->lock); + es8326_irq(es8326->irq, es8326); +} + +static void es8326_disable_jack_detect(struct snd_soc_component *component) +{ + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); + + dev_dbg(component->dev, "Enter into %s\n", __func__); + if (!es8326->jack) + return; /* Already disabled (or never enabled) */ + + cancel_delayed_work_sync(&es8326->jack_detect_work); + + mutex_lock(&es8326->lock); + if (es8326->jack->status & SND_JACK_MICROPHONE) + snd_soc_jack_report(es8326->jack, 0, SND_JACK_BTN_0); + + es8326->jack = NULL; + mutex_unlock(&es8326->lock); +} + +static int es8326_set_jack(struct snd_soc_component *component, + struct snd_soc_jack *jack, void *data) +{ + if (jack) + es8326_enable_jack_detect(component, jack); + else + es8326_disable_jack_detect(component); + + return 0; +} + +static void es8326_remove(struct snd_soc_component *component) +{ + es8326_disable_jack_detect(component); + es8326_set_bias_level(component, SND_SOC_BIAS_OFF); +} + +static const struct snd_soc_component_driver soc_component_dev_es8326 = { + .probe = es8326_probe, + .remove = es8326_remove, + .resume = es8326_resume, + .suspend = es8326_suspend, + .set_bias_level = es8326_set_bias_level, + .set_jack = es8326_set_jack, + + .dapm_widgets = es8326_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(es8326_dapm_widgets), + .dapm_routes = es8326_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(es8326_dapm_routes), + .controls = es8326_snd_controls, + .num_controls = ARRAY_SIZE(es8326_snd_controls), +}; + +static int es8326_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct es8326_priv *es8326; + int ret; + + es8326 = devm_kzalloc(&i2c->dev, sizeof(struct es8326_priv), GFP_KERNEL); + if (!es8326) + return -ENOMEM; + + i2c_set_clientdata(i2c, es8326); + es8326->i2c = i2c; + mutex_init(&es8326->lock); + es8326->regmap = devm_regmap_init_i2c(i2c, &es8326_regmap_config); + if (IS_ERR(es8326->regmap)) { + ret = PTR_ERR(es8326->regmap); + dev_err(&i2c->dev, "Failed to init regmap: %d\n", ret); + return ret; + } + + es8326->irq = i2c->irq; + /* ES8316 is level-based while ES8326 is edge-based */ + ret = devm_request_threaded_irq(&i2c->dev, es8326->irq, NULL, es8326_irq, + IRQF_TRIGGER_RISING | IRQF_ONESHOT, + "es8326", es8326); + if (ret) + dev_err(&i2c->dev, "Failed to reguest IRQ: %d\n", ret); + INIT_DELAYED_WORK(&es8326->jack_detect_work, + es8326_jack_detect_handler); + + es8326->mclk = devm_clk_get_optional(&i2c->dev, "mclk"); + if (IS_ERR(es8326->mclk)) { + dev_err(&i2c->dev, "unable to get mclk\n"); + return PTR_ERR(es8326->mclk); + } + if (!es8326->mclk) + dev_warn(&i2c->dev, "assuming static mclk\n"); + + ret = clk_prepare_enable(es8326->mclk); + if (ret) { + dev_err(&i2c->dev, "unable to enable mclk\n"); + return ret; + } + return devm_snd_soc_register_component(&i2c->dev, + &soc_component_dev_es8326, + &es8326_dai, 1); +} + +static const struct i2c_device_id es8326_i2c_id[] = { + {"es8326", 0 }, + {} +}; +MODULE_DEVICE_TABLE(i2c, es8326_i2c_id); + +#ifdef CONFIG_OF +static const struct of_device_id es8326_of_match[] = { + { .compatible = "everest,es8326", }, + {} +}; +MODULE_DEVICE_TABLE(of, es8326_of_match); +#endif + +#ifdef CONFIG_ACPI +static const struct acpi_device_id es8326_acpi_match[] = { + {"ESSX8326", 0}, + {}, +}; +MODULE_DEVICE_TABLE(acpi, es8326_acpi_match); +#endif + +static struct i2c_driver es8326_i2c_driver = { + .driver = { + .name = "es8326", + .acpi_match_table = ACPI_PTR(es8326_acpi_match), + .of_match_table = of_match_ptr(es8326_of_match), + }, + .probe = es8326_i2c_probe, +}; +module_i2c_driver(es8326_i2c_driver); + +MODULE_DESCRIPTION("ASoC es8326 driver"); +MODULE_AUTHOR("David Yang <yangxiaohua(a)everest-semi.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/es8326.h b/sound/soc/codecs/es8326.h new file mode 100755 index 000000000000..9affd5c14407 --- /dev/null +++ b/sound/soc/codecs/es8326.h @@ -0,0 +1,187 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * es8326.c -- es8326 ALSA SoC audio driver + * Copyright Everest Semiconductor Co.,Ltd + * + * Authors: David Yang <yangxiaohua(a)everest-semi.com> + */ + +#ifndef _ES8326_H +#define _ES8326_H + +#define CONFIG_HHTECH_MINIPMP 1 + +/* ES8326 register space */ + +#define ES8326_RESET_00 0x00 +#define ES8326_CLK_CTL_01 0x01 +#define ES8326_CLK_INV_02 0x02 +#define ES8326_CLK_RESAMPLE_03 0x03 +#define ES8326_CLK_DIV1_04 0x04 +#define ES8326_CLK_DIV2_05 0x05 +#define ES8326_CLK_DLL_06 0x06 +#define ES8326_CLK_MUX_07 0x07 +#define ES8326_CLK_ADC_SEL_08 0x08 +#define ES8326_CLK_DAC_SEL_09 0x09 +#define ES8326_CLK_ADC_OSR_0A 0x0a +#define ES8326_CLK_DAC_OSR_0B 0x0b +#define ES8326_CLK_DIV_CPC_0C 0x0c +#define ES8326_CLK_DIV_BCLK_0D 0x0d +#define ES8326_CLK_TRI_0E 0x0e +#define ES8326_CLK_DIV_LRCK_0F 0x0f +#define ES8326_CLK_VMIDS1_10 0x10 +#define ES8326_CLK_VMIDS2_11 0x11 +#define ES8326_CLK_CAL_TIME_12 0x12 +#define ES8326_FMT_13 0x13 + +#define ES8326_DAC_MUTE_14 0x14 +#define ES8326_ADC_MUTE_15 0x15 +#define ES8326_ANA_PDN_16 0x16 +#define ES8326_PGA_PDN_17 0x17 +#define ES8326_VMIDSEL_18 0x18 +#define ES8326_ANA_LOWPOWER_19 0x19 +#define ES8326_ANA_DMS_1A 0x1a +#define ES8326_ANA_MICBIAS_1B 0x1b +#define ES8326_ANA_VSEL_1C 0x1c +#define ES8326_SYS_BIAS_1D 0x1d +#define ES8326_BIAS_SW1_1E 0x1e +#define ES8326_BIAS_SW2_1F 0x1f +#define ES8326_BIAS_SW3_20 0x20 +#define ES8326_BIAS_SW4_21 0x21 +#define ES8326_VMIDLOW_22 0x22 + +#define ES8326_PGAGAIN_23 0x23 +#define ES8326_HP_DRIVER_24 0x24 +#define ES8326_DAC2HPMIX_25 0x25 +#define ES8326_HP_VOL_26 0x26 +#define ES8326_HP_CAL_27 0x27 +#define ES8326_HP_DRIVER_REF_28 0x28 +#define ES8326_ADC_SCALE_29 0x29 +#define ES8326_ADC1_SRC_2A 0x2a +#define ES8326_ADC2_SRC_2B 0x2b +#define ES8326_ADC1_VOL_2C 0x2c +#define ES8326_ADC2_VOL_2D 0x2d +#define ES8326_ADC_RAMPRATE_2E 0x2e +#define ES8326_2F 0x2f +#define ES8326_30 0x30 +#define ES8326_31 0x31 +#define ES8326_ALC_RECOVERY_32 0x32 +#define ES8326_ALC_LEVEL_33 0x33 +#define ES8326_ADC_HPFS1_34 0x34 +#define ES8326_ADC_HPFS2_35 0x35 +#define ES8326_ADC_EQ_36 0x36 +#define ES8326_HP_CAL_4A 0x4A +#define ES8326_HPL_OFFSET_INI_4B 0x4B +#define ES8326_HPR_OFFSET_INI_4C 0x4C +#define ES8326_DAC_DSM_4D 0x4D +#define ES8326_DAC_RAMPRATE_4E 0x4E +#define ES8326_DAC_VPPSCALE_4F 0x4F +#define ES8326_DAC_VOL_50 0x50 +#define ES8326_DRC_RECOVERY_53 0x53 +#define ES8326_DRC_WINSIZE_54 0x54 +#define ES8326_HPJACK_TIMER_56 0x56 +#define ES8326_HP_DET_57 0x57 +#define ES8326_INT_SOURCE_58 0x58 +#define ES8326_INTOUT_IO_59 0x59 +#define ES8326_SDINOUT1_IO_5A 0x5A +#define ES8326_SDINOUT23_IO_5B 0x5B +#define ES8326_JACK_PULSE_5C 0x5C + +#define ES8326_PULLUP_CTL_F9 0xF9 +#define ES8326_HP_DETECT_FB 0xFB +#define ES8326_CHIP_ID1_FD 0xFD +#define ES8326_CHIP_ID2_FE 0xFE +#define ES8326_CHIP_VERSION_FF 0xFF + +/* ES8326_RESET_00 */ +#define ES8326_CSM_ON (1 << 7) +#define ES8326_MASTER_MODE_EN (1 << 6) +#define ES8326_PWRUP_SEQ_EN (1 << 5) +#define ES8326_CODEC_RESET (0x0f << 0) +#define ES8326_CSM_OFF (0 << 7) + +/* ES8326_CLK_CTL_01 */ +#define ES8326_CLK_ON (0x7f << 0) +#define ES8326_CLK_OFF (0 << 0) + +/* ES8326_CLK_INV_02 */ +#define ES8326_BCLK_AS_MCLK (1 << 3) + +/* ES8326_FMT_13 */ +#define ES8326_S24_LE (0 << 2) +#define ES8326_S20_3_LE (1 << 2) +#define ES8326_S18_LE (2 << 2) +#define ES8326_S16_LE (3 << 2) +#define ES8326_S32_LE (4 << 2) +#define ES8326_DATA_LEN_MASK (7 << 2) + +#define ES8326_DAIFMT_MASK ((1 << 5) | (3 << 0)) +#define ES8326_DAIFMT_I2S 0 +#define ES8326_DAIFMT_LEFT_J (1 << 0) +#define ES8326_DAIFMT_DSP_A (3 << 0) +#define ES8326_DAIFMT_DSP_B ((1 << 5) | (3 << 0)) + +/* ES8326_PGAGAIN_23 */ +#define ES8326_MIC_SEL_MASK (3 << 4) +#define ES8326_MIC1_SEL (1 << 4) +#define ES8326_MIC2_SEL (1 << 5) + +/* ES8326_HP_CAL_27 */ +#define ES8326_HPOR_SHIFT 4 + +/* ES8326_ADC1_SRC_2A */ +#define ES8326_ADC1_SHIFT 0 +#define ES8326_ADC2_SHIFT 4 +#define ES8326_ADC_SRC_ANA 0 +#define ES8326_ADC_SRC_ANA_INV_SW0 1 +#define ES8326_ADC_SRC_ANA_INV_SW1 2 +#define ES8326_ADC_SRC_DMIC_MCLK 3 +#define ES8326_ADC_SRC_DMIC_SDIN2 4 +#define ES8326_ADC_SRC_DMIC_SDIN2_INV 5 +#define ES8326_ADC_SRC_DMIC_SDIN3 6 +#define ES8326_ADC_SRC_DMIC_SDIN3_INV 7 + +#define ES8326_ADC_AMIC ((ES8326_ADC_SRC_ANA_INV_SW1 << ES8326_ADC2_SHIFT) \ + | (ES8326_ADC_SRC_ANA_INV_SW1 << ES8326_ADC1_SHIFT)) +#define ES8326_ADC_DMIC ((ES8326_ADC_SRC_DMIC_SDIN2 << ES8326_ADC2_SHIFT) \ + | (ES8326_ADC_SRC_DMIC_SDIN2 << ES8326_ADC1_SHIFT)) +/* ES8326_ADC2_SRC_2B */ +#define ES8326_ADC3_SHIFT 0 +#define ES8326_ADC4_SHIFT 3 + +/* ES8326_HP_DET_57 */ +#define ES8326_HP_DET_SRC_PIN27 (1 << 5) +#define ES8326_HP_DET_SRC_PIN9 (1 << 4) +#define ES8326_HP_DET_JACK_POL (1 << 3) +#define ES8326_HP_DET_BUTTON_POL (1 << 2) +#define ES8326_HP_TYPE_OMTP (3 << 0) +#define ES8326_HP_TYPE_CTIA (2 << 0) +#define ES8326_HP_TYPE_AUTO (1 << 0) +#define ES8326_HP_TYPE_AUTO_INV (0 << 0) + +/* ES8326_SDINOUT1_IO_5A */ +#define ES8326_IO_INPUT (0 << 0) +#define ES8326_IO_SDIN_SLOT0 (1 << 0) +#define ES8326_IO_SDIN_SLOT1 (2 << 0) +#define ES8326_IO_SDIN_SLOT2 (3 << 0) +#define ES8326_IO_SDIN_SLOT7 (8 << 0) +#define ES8326_IO_DMIC_CLK (9 << 0) +#define ES8326_IO_DMIC_CLK_INV (0x0a << 0) +#define ES8326_IO_SDOUT2 (0x0b << 0) +#define ES8326_IO_LOW (0x0e << 0) +#define ES8326_IO_HIGH (0x0f << 0) +#define ES8326_ADC2DAC (1 << 3) +#define ES8326_SDINOUT1_SHIFT 4 + +/* ES8326_SDINOUT23_IO_5B */ +#define ES8326_SDINOUT2_SHIFT 4 +#define ES8326_SDINOUT3_SHIFT 0 + +/* ES8326_HP_DETECT_FB */ +#define ES8326_HPINSERT_FLAG (1 << 1) +#define ES8326_HPBUTTON_FLAG (1 << 0) + +/* ES8326_CHIP_VERSION_FF 0xFF */ +#define ES8326_VERSION_B (1 << 0) + +#endif -- 2.36.1

3 2

[PATCH 0/4] ASoC: Intel: Mark BE DAIs as nonatomic for hsw and
by Cezary Rojewski 15 Jul '22

15 Jul '22

Address the warning: "Codec: dpcm_be_connect: FE is nonatomic but BE is not, forcing BE as nonatomic" by marking BE DAI as nonatomic. Aligns with what is already done for FE DAIs. This patchset iterates the change over all HSW and BDW related machine board drivers. Cezary Rojewski (4): ASoC: Intel: hsw_rt5640: Mark BE DAI as nonatomic ASoC: Intel: bdw_rt286: Mark BE DAI as nonatomic ASoC: Intel: bdw_rt5650: Mark BE DAI as nonatomic ASoC: Intel: bdw_rt5677: Mark BE DAI as nonatomic sound/soc/intel/boards/bdw-rt5650.c | 1 + sound/soc/intel/boards/bdw-rt5677.c | 1 + sound/soc/intel/boards/bdw_rt286.c | 1 + sound/soc/intel/boards/hsw_rt5640.c | 1 + 4 files changed, 4 insertions(+) -- 2.25.1

3 13

[PATCH] Revert "ALSA: hda: Fix page fault in snd_hda_codec_shutdown()"
by Takashi Iwai 15 Jul '22

15 Jul '22

This reverts commit 980b3a8790b402e959a6d773b38b771019682be1. The commit didn't consider the fact that ASoC hdac-hda driver initializes the HD-audio stuff without calling snd_hda_codec_device_init(). Hence this caused a regression leading to Oops. Revert the commit to restore the behavior. Fixes: 980b3a8790b4 ("ALSA: hda: Fix page fault in snd_hda_codec_shutdown()") Link: https://lore.kernel.org/r/3c40df55-3aee-1e08-493b-7b30cd84dc00@linux.intel.… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_codec.c | 41 ++++++++++++++++++++------------------- 1 file changed, 21 insertions(+), 20 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 7be74227bf19..7b2e62fa82d5 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -931,28 +931,8 @@ snd_hda_codec_device_init(struct hda_bus *bus, unsigned int codec_addr, } codec->bus = bus; - codec->depop_delay = -1; - codec->fixup_id = HDA_FIXUP_ID_NOT_SET; - codec->core.dev.release = snd_hda_codec_dev_release; - codec->core.exec_verb = codec_exec_verb; codec->core.type = HDA_DEV_LEGACY; - mutex_init(&codec->spdif_mutex); - mutex_init(&codec->control_mutex); - snd_array_init(&codec->mixers, sizeof(struct hda_nid_item), 32); - snd_array_init(&codec->nids, sizeof(struct hda_nid_item), 32); - snd_array_init(&codec->init_pins, sizeof(struct hda_pincfg), 16); - snd_array_init(&codec->driver_pins, sizeof(struct hda_pincfg), 16); - snd_array_init(&codec->cvt_setups, sizeof(struct hda_cvt_setup), 8); - snd_array_init(&codec->spdif_out, sizeof(struct hda_spdif_out), 16); - snd_array_init(&codec->jacktbl, sizeof(struct hda_jack_tbl), 16); - snd_array_init(&codec->verbs, sizeof(struct hda_verb *), 8); - INIT_LIST_HEAD(&codec->conn_list); - INIT_LIST_HEAD(&codec->pcm_list_head); - INIT_DELAYED_WORK(&codec->jackpoll_work, hda_jackpoll_work); - refcount_set(&codec->pcm_ref, 1); - init_waitqueue_head(&codec->remove_sleep); - return codec; } EXPORT_SYMBOL_GPL(snd_hda_codec_device_init); @@ -1005,8 +985,29 @@ int snd_hda_codec_device_new(struct hda_bus *bus, struct snd_card *card, if (snd_BUG_ON(codec_addr > HDA_MAX_CODEC_ADDRESS)) return -EINVAL; + codec->core.dev.release = snd_hda_codec_dev_release; + codec->core.exec_verb = codec_exec_verb; + codec->card = card; codec->addr = codec_addr; + mutex_init(&codec->spdif_mutex); + mutex_init(&codec->control_mutex); + snd_array_init(&codec->mixers, sizeof(struct hda_nid_item), 32); + snd_array_init(&codec->nids, sizeof(struct hda_nid_item), 32); + snd_array_init(&codec->init_pins, sizeof(struct hda_pincfg), 16); + snd_array_init(&codec->driver_pins, sizeof(struct hda_pincfg), 16); + snd_array_init(&codec->cvt_setups, sizeof(struct hda_cvt_setup), 8); + snd_array_init(&codec->spdif_out, sizeof(struct hda_spdif_out), 16); + snd_array_init(&codec->jacktbl, sizeof(struct hda_jack_tbl), 16); + snd_array_init(&codec->verbs, sizeof(struct hda_verb *), 8); + INIT_LIST_HEAD(&codec->conn_list); + INIT_LIST_HEAD(&codec->pcm_list_head); + refcount_set(&codec->pcm_ref, 1); + init_waitqueue_head(&codec->remove_sleep); + + INIT_DELAYED_WORK(&codec->jackpoll_work, hda_jackpoll_work); + codec->depop_delay = -1; + codec->fixup_id = HDA_FIXUP_ID_NOT_SET; #ifdef CONFIG_PM codec->power_jiffies = jiffies; -- 2.35.3

1 0

[PATCH] Revert "ALSA: hda: cs35l41: Allow compilation test on non-ACPI configurations"
by Takashi Iwai 15 Jul '22

15 Jul '22

Since the recent change in CS35L41 codec requires the reference of acpi_dev handle, the current Kconfig may lead to a build breakage. Revert the Kconfig change and re-introduce the hard dependency on CONFIG_ACPI again as a temporary workaround. Fixes: eef375960210 ("ALSA: hda: cs35l41: Support reading subsystem id from ACPI") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index 44c33bc0740e..a8e8cf98befa 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -103,7 +103,7 @@ config SND_HDA_CS_DSP_CONTROLS config SND_HDA_SCODEC_CS35L41_I2C tristate "Build CS35L41 HD-audio side codec support for I2C Bus" depends on I2C - depends on ACPI || COMPILE_TEST + depends on ACPI depends on SND_SOC select SND_SOC_CS35L41_LIB select SND_HDA_SCODEC_CS35L41 @@ -118,7 +118,7 @@ comment "Set to Y if you want auto-loading the side codec driver" config SND_HDA_SCODEC_CS35L41_SPI tristate "Build CS35L41 HD-audio codec support for SPI Bus" depends on SPI_MASTER - depends on ACPI || COMPILE_TEST + depends on ACPI depends on SND_SOC select SND_SOC_CS35L41_LIB select SND_HDA_SCODEC_CS35L41 -- 2.35.3

1 0