diff options
author | Linus Torvalds <torvalds@linux-foundation.org> | 2015-04-15 15:41:41 -0700 |
---|---|---|
committer | Linus Torvalds <torvalds@linux-foundation.org> | 2015-04-15 15:41:41 -0700 |
commit | d0a3997c0c3f9351e24029349dee65dd1d9e8d84 (patch) | |
tree | 7a04fe282b0c7b329cd87cdb891f0f3879dc71a6 /sound/soc/codecs | |
parent | 6d50ff91d9780263160262daeb6adfdda8ddbc6c (diff) | |
parent | d6eb9e3ec78c98324097bab8eea266c3bb0d0ac7 (diff) |
Merge tag 'sound-4.1-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"There have been major modernization with the standard bus: in ALSA
sequencer core and HD-audio. Also, HD-audio receives the regmap
support replacing the in-house cache register cache code. These
changes shouldn't impact the existing behavior, but rather
refactoring.
In addition, HD-audio got the code split to a core library part and
the "legacy" driver parts. This is a preliminary work for adapting
the upcoming ASoC HD-audio driver, and the whole transition is still
work in progress, likely finished in 4.1.
Along with them, there are many updates in ASoC area as usual, too:
lots of cleanups, Intel code shuffling, etc.
Here are some highlights:
ALSA core:
- PCM: the audio timestamp / wallclock enhancement
- PCM: fixes in DPCM management
- Fixes / cleanups of user-space control element management
- Sequencer: modernization using the standard bus
HD-audio:
- Modernization using the standard bus
- Regmap support
- Use standard runtime PM for codec power saving
- Widget-path based power-saving for IDT, VIA and Realtek codecs
- Reorganized sysfs entries for each codec object
- More Dell headset support
ASoC:
- Move of jack registration to the card level
- Lots of ASoC cleanups, mainly moving things from the CODEC level to
the card level
- Support for DAPM routes specified by both the machine driver and DT
- Continuing improvements to rcar
- pcm512x enhacements
- Intel platforms updates
- rt5670 updates / fixes
- New platforms / devices: some non-DSP Qualcomm platforms, Google's
Storm platform, Maxmim MAX98925 CODECs and the Ingenic JZ4780 SoC
Misc:
- ice1724: Improved ESI W192M support
- emu10k1: Emu 1010 fixes/enhancement"
* tag 'sound-4.1-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (411 commits)
ALSA: hda - set GET bit when adding a vendor verb to the codec regmap
ALSA: hda/realtek - Enable the ALC292 dock fixup on the Thinkpad T450
ALSA: hda - Fix another race in runtime PM refcounting
ALSA: hda - Expose codec type sysfs
ALSA: ctl: fix to handle several elements added by one operation for userspace element
ASoC: Intel: fix array_size.cocci warnings
ASoC: n810: Automatically disconnect non-connected pins
ASoC: n810: Consistently pass the card DAPM context to n810_ext_control()
ASoC: davinci-evm: Use card DAPM context to access widgets
ASoC: mop500_ab8500: Use card DAPM context to access widgets
ASoC: wm1133-ev1: Use card DAPM context to access widgets
ASoC: atmel: Improve machine driver compile test coverage
ASoC: atmel: Add dependency to SND_SOC_I2C_AND_SPI where necessary
ALSA: control: Fix a typo of SNDRV_CTL_ELEM_ACCESS_TLV_* with SNDRV_CTL_TLV_OP_*
ALSA: usb-audio: Don't attempt to get Microsoft Lifecam Cinema sample rate
ASoC: rnsd: fix build regression without CONFIG_OF
ALSA: emu10k1: add toggles for E-mu 1010 optical ports
ALSA: ctl: fill identical information to return value when adding userspace elements
ALSA: ctl: fix a bug to return no identical information in info operation for userspace controls
ALSA: ctl: confirm to return all identical information in 'activate' event
...
Diffstat (limited to 'sound/soc/codecs')
43 files changed, 2845 insertions, 656 deletions
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index ea9f0e31f9d4..061c46587628 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -70,6 +70,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX98090 if I2C select SND_SOC_MAX98095 if I2C select SND_SOC_MAX98357A if GPIOLIB + select SND_SOC_MAX98925 if I2C select SND_SOC_MAX9850 if I2C select SND_SOC_MAX9768 if I2C select SND_SOC_MAX9877 if I2C @@ -141,7 +142,8 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM8770 if SPI_MASTER select SND_SOC_WM8776 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8782 - select SND_SOC_WM8804 if SND_SOC_I2C_AND_SPI + select SND_SOC_WM8804_I2C if I2C + select SND_SOC_WM8804_SPI if SPI_MASTER select SND_SOC_WM8900 if I2C select SND_SOC_WM8903 if I2C select SND_SOC_WM8904 if I2C @@ -460,6 +462,9 @@ config SND_SOC_MAX98095 config SND_SOC_MAX98357A tristate +config SND_SOC_MAX98925 + tristate + config SND_SOC_MAX9850 tristate @@ -744,8 +749,19 @@ config SND_SOC_WM8782 tristate config SND_SOC_WM8804 - tristate "Wolfson Microelectronics WM8804 S/PDIF transceiver" - depends on SND_SOC_I2C_AND_SPI + tristate + +config SND_SOC_WM8804_I2C + tristate "Wolfson Microelectronics WM8804 S/PDIF transceiver I2C" + depends on I2C + select SND_SOC_WM8804 + select REGMAP_I2C + +config SND_SOC_WM8804_SPI + tristate "Wolfson Microelectronics WM8804 S/PDIF transceiver SPI" + depends on SPI_MASTER + select SND_SOC_WM8804 + select REGMAP_SPI config SND_SOC_WM8900 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 69b8666d187a..abe2d7edf65c 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -65,6 +65,7 @@ snd-soc-max98088-objs := max98088.o snd-soc-max98090-objs := max98090.o snd-soc-max98095-objs := max98095.o snd-soc-max98357a-objs := max98357a.o +snd-soc-max98925-objs := max98925.o snd-soc-max9850-objs := max9850.o snd-soc-mc13783-objs := mc13783.o snd-soc-ml26124-objs := ml26124.o @@ -145,6 +146,8 @@ snd-soc-wm8770-objs := wm8770.o snd-soc-wm8776-objs := wm8776.o snd-soc-wm8782-objs := wm8782.o snd-soc-wm8804-objs := wm8804.o +snd-soc-wm8804-i2c-objs := wm8804-i2c.o +snd-soc-wm8804-spi-objs := wm8804-spi.o snd-soc-wm8900-objs := wm8900.o snd-soc-wm8903-objs := wm8903.o snd-soc-wm8904-objs := wm8904.o @@ -247,6 +250,7 @@ obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o obj-$(CONFIG_SND_SOC_MAX98090) += snd-soc-max98090.o obj-$(CONFIG_SND_SOC_MAX98095) += snd-soc-max98095.o obj-$(CONFIG_SND_SOC_MAX98357A) += snd-soc-max98357a.o +obj-$(CONFIG_SND_SOC_MAX98925) += snd-soc-max98925.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o @@ -323,6 +327,8 @@ obj-$(CONFIG_SND_SOC_WM8770) += snd-soc-wm8770.o obj-$(CONFIG_SND_SOC_WM8776) += snd-soc-wm8776.o obj-$(CONFIG_SND_SOC_WM8782) += snd-soc-wm8782.o obj-$(CONFIG_SND_SOC_WM8804) += snd-soc-wm8804.o +obj-$(CONFIG_SND_SOC_WM8804_I2C) += snd-soc-wm8804-i2c.o +obj-$(CONFIG_SND_SOC_WM8804_SPI) += snd-soc-wm8804-spi.o obj-$(CONFIG_SND_SOC_WM8900) += snd-soc-wm8900.o obj-$(CONFIG_SND_SOC_WM8903) += snd-soc-wm8903.o obj-$(CONFIG_SND_SOC_WM8904) += snd-soc-wm8904.o diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index 7895689588da..88ca9cb0ce79 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -2003,7 +2003,6 @@ static int ab8500_audio_setup_mics(struct snd_soc_codec *codec, return 0; } -EXPORT_SYMBOL_GPL(ab8500_audio_setup_mics); static int ab8500_audio_set_ear_cmv(struct snd_soc_codec *codec, enum ear_cm_voltage ear_cmv) @@ -2036,7 +2035,6 @@ static int ab8500_audio_set_ear_cmv(struct snd_soc_codec *codec, return 0; } -EXPORT_SYMBOL_GPL(ab8500_audio_set_ear_cmv); static int ab8500_audio_set_bit_delay(struct snd_soc_dai *dai, unsigned int delay) diff --git a/sound/soc/codecs/adau1977.c b/sound/soc/codecs/adau1977.c index 70ab35744aba..7ad8e156e2df 100644 --- a/sound/soc/codecs/adau1977.c +++ b/sound/soc/codecs/adau1977.c @@ -938,22 +938,15 @@ int adau1977_probe(struct device *dev, struct regmap *regmap, adau1977->dvdd_reg = NULL; } - adau1977->reset_gpio = devm_gpiod_get(dev, "reset"); - if (IS_ERR(adau1977->reset_gpio)) { - ret = PTR_ERR(adau1977->reset_gpio); - if (ret != -ENOENT && ret != -ENOSYS) - return PTR_ERR(adau1977->reset_gpio); - adau1977->reset_gpio = NULL; - } + adau1977->reset_gpio = devm_gpiod_get_optional(dev, "reset", + GPIOD_OUT_LOW); + if (IS_ERR(adau1977->reset_gpio)) + return PTR_ERR(adau1977->reset_gpio); dev_set_drvdata(dev, adau1977); - if (adau1977->reset_gpio) { - ret = gpiod_direction_output(adau1977->reset_gpio, 0); - if (ret) - return ret; + if (adau1977->reset_gpio) ndelay(100); - } ret = adau1977_power_enable(adau1977); if (ret) diff --git a/sound/soc/codecs/ak4554.c b/sound/soc/codecs/ak4554.c index 16ce9f9fefa1..298dedc05140 100644 --- a/sound/soc/codecs/ak4554.c +++ b/sound/soc/codecs/ak4554.c @@ -84,7 +84,7 @@ static int ak4554_soc_remove(struct platform_device *pdev) return 0; } -static struct of_device_id ak4554_of_match[] = { +static const struct of_device_id ak4554_of_match[] = { { .compatible = "asahi-kasei,ak4554" }, {}, }; diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index dde8b49c19ad..13585e88f597 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -97,6 +97,9 @@ #define PMMP (1 << 2) /* MPWR pin Power Management */ #define MGAIN0 (1 << 0) /* MIC amp gain*/ +/* SG_SL2 */ +#define LOPS (1 << 6) /* Stero Line-out Power Save Mode */ + /* TIMER */ #define ZTM(param) ((param & 0x3) << 4) /* ALC Zero Crossing TimeOut */ #define WTM(param) (((param & 0x4) << 4) | ((param & 0x3) << 2)) @@ -168,6 +171,29 @@ static const struct snd_kcontrol_new ak4642_lout_mixer_controls[] = { SOC_DAPM_SINGLE("DACL", SG_SL1, 4, 1, 0), }; +/* event handlers */ +static int ak4642_lout_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + + switch (event) { + case SND_SOC_DAPM_PRE_PMD: + case SND_SOC_DAPM_PRE_PMU: + /* Power save mode ON */ + snd_soc_update_bits(codec, SG_SL2, LOPS, LOPS); + break; + case SND_SOC_DAPM_POST_PMU: + case SND_SOC_DAPM_POST_PMD: + /* Power save mode OFF */ + mdelay(300); + snd_soc_update_bits(codec, SG_SL2, LOPS, 0); + break; + } + + return 0; +} + static const struct snd_soc_dapm_widget ak4642_dapm_widgets[] = { /* Outputs */ @@ -182,12 +208,15 @@ static const struct snd_soc_dapm_widget ak4642_dapm_widgets[] = { SND_SOC_DAPM_PGA("DACH", MD_CTL4, 0, 0, NULL, 0), - SND_SOC_DAPM_MIXER("LINEOUT Mixer", PW_MGMT1, 3, 0, + SND_SOC_DAPM_MIXER_E("LINEOUT Mixer", PW_MGMT1, 3, 0, &ak4642_lout_mixer_controls[0], - ARRAY_SIZE(ak4642_lout_mixer_controls)), + ARRAY_SIZE(ak4642_lout_mixer_controls), + ak4642_lout_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD), /* DAC */ - SND_SOC_DAPM_DAC("DAC", "HiFi Playback", PW_MGMT1, 2, 0), + SND_SOC_DAPM_DAC("DAC", NULL, PW_MGMT1, 2, 0), }; static const struct snd_soc_dapm_route ak4642_intercon[] = { @@ -205,6 +234,8 @@ static const struct snd_soc_dapm_route ak4642_intercon[] = { {"DACH", NULL, "DAC"}, {"LINEOUT Mixer", "DACL", "DAC"}, + + { "DAC", NULL, "Playback" }, }; /* @@ -468,13 +499,13 @@ static struct snd_soc_dai_driver ak4642_dai = { .name = "ak4642-hifi", .playback = { .stream_name = "Playback", - .channels_min = 1, + .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_48000, .formats = SNDRV_PCM_FMTBIT_S16_LE }, .capture = { .stream_name = "Capture", - .channels_min = 1, + .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_48000, .formats = SNDRV_PCM_FMTBIT_S16_LE }, diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 95d31d6291ac..57da0ceda03f 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1902,7 +1902,7 @@ static int arizona_is_enabled_fll(struct arizona_fll *fll) static int arizona_enable_fll(struct arizona_fll *fll) { struct arizona *arizona = fll->arizona; - int ret; + unsigned long time_left; bool use_sync = false; int already_enabled = arizona_is_enabled_fll(fll); struct arizona_fll_cfg cfg; @@ -1978,9 +1978,9 @@ static int arizona_enable_fll(struct arizona_fll *fll) regmap_update_bits_async(arizona->regmap, fll->base + 1, ARIZONA_FLL1_FREERUN, 0); - ret = wait_for_completion_timeout(&fll->ok, + time_left = wait_for_completion_timeout(&fll->ok, msecs_to_jiffies(250)); - if (ret == 0) + if (time_left == 0) arizona_fll_warn(fll, "Timed out waiting for lock\n"); return 0; diff --git a/sound/soc/codecs/cs35l32.c b/sound/soc/codecs/cs35l32.c index f2b8aad21274..60598b230341 100644 --- a/sound/soc/codecs/cs35l32.c +++ b/sound/soc/codecs/cs35l32.c @@ -437,20 +437,13 @@ static int cs35l32_i2c_probe(struct i2c_client *i2c_client, } /* Reset the Device */ - cs35l32->reset_gpio = devm_gpiod_get(&i2c_client->dev, - "reset-gpios"); - if (IS_ERR(cs35l32->reset_gpio)) { - ret = PTR_ERR(cs35l32->reset_gpio); - if (ret != -ENOENT && ret != -ENOSYS) - return ret; - - cs35l32->reset_gpio = NULL; - } else { - ret = gpiod_direction_output(cs35l32->reset_gpio, 0); - if (ret) - return ret; + cs35l32->reset_gpio = devm_gpiod_get_optional(&i2c_client->dev, + "reset", GPIOD_OUT_LOW); + if (IS_ERR(cs35l32->reset_gpio)) + return PTR_ERR(cs35l32->reset_gpio); + + if (cs35l32->reset_gpio) gpiod_set_value_cansleep(cs35l32->reset_gpio, 1); - } /* initialize codec */ ret = regmap_read(cs35l32->regmap, CS35L32_DEVID_AB, ®); diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c index ce6086835ebd..cac48ddf3ba6 100644 --- a/sound/soc/codecs/cs4265.c +++ b/sound/soc/codecs/cs4265.c @@ -605,21 +605,14 @@ static int cs4265_i2c_probe(struct i2c_client *i2c_client, return ret; } - cs4265->reset_gpio = devm_gpiod_get(&i2c_client->dev, - "reset-gpios"); - if (IS_ERR(cs4265->reset_gpio)) { - ret = PTR_ERR(cs4265->reset_gpio); - if (ret != -ENOENT && ret != -ENOSYS) - return ret; - - cs4265->reset_gpio = NULL; - } else { - ret = gpiod_direction_output(cs4265->reset_gpio, 0); - if (ret) - return ret; + cs4265->reset_gpio = devm_gpiod_get_optional(&i2c_client->dev, + "reset", GPIOD_OUT_LOW); + if (IS_ERR(cs4265->reset_gpio)) + return PTR_ERR(cs4265->reset_gpio); + + if (cs4265->reset_gpio) { mdelay(1); gpiod_set_value_cansleep(cs4265->reset_gpio, 1); - } i2c_set_clientdata(i2c_client, cs4265); diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 7d3a6accaf9a..e770ee6f36da 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -561,10 +561,10 @@ static int cs4271_codec_probe(struct snd_soc_codec *codec) if (gpio_is_valid(cs4271->gpio_nreset)) { /* Reset codec */ gpio_direction_output(cs4271->gpio_nreset, 0); - udelay(1); + mdelay(1); gpio_set_value(cs4271->gpio_nreset, 1); /* Give the codec time to wake up */ - udelay(1); + mdelay(1); } ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2, diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c index 0b10979513c4..0f334bc1b63c 100644 --- a/sound/soc/codecs/cx20442.c +++ b/sound/soc/codecs/cx20442.c @@ -420,7 +420,7 @@ static int cx20442_platform_probe(struct platform_device *pdev) &cx20442_codec_dev, &cx20442_dai, 1); } -static int __exit cx20442_platform_remove(struct platform_device *pdev) +static int cx20442_platform_remove(struct platform_device *pdev) { snd_soc_unregister_codec(&pdev->dev); return 0; @@ -431,7 +431,7 @@ static struct platform_driver cx20442_platform_driver = { .name = "cx20442-codec", }, .probe = cx20442_platform_probe, - .remove = __exit_p(cx20442_platform_remove), + .remove = cx20442_platform_remove, }; module_platform_driver(cx20442_platform_driver); diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index b112b1c2c394..3e33ef2acf3c 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2605,8 +2605,24 @@ err_enable: return ret; } +static void max98090_i2c_shutdown(struct i2c_client *i2c) +{ + struct max98090_priv *max98090 = dev_get_drvdata(&i2c->dev); + + /* + * Enable volume smoothing, disable zero cross. This will cause + * a quick 40ms ramp to mute on shutdown. + */ + regmap_write(max98090->regmap, + M98090_REG_LEVEL_CONTROL, M98090_VSENN_MASK); + regmap_write(max98090->regmap, + M98090_REG_DEVICE_SHUTDOWN, 0x00); + msleep(40); +} + static int max98090_i2c_remove(struct i2c_client *client) { + max98090_i2c_shutdown(client); snd_soc_unregister_codec(&client->dev); return 0; } @@ -2696,6 +2712,7 @@ static struct i2c_driver max98090_i2c_driver = { .acpi_match_table = ACPI_PTR(max98090_acpi_match), }, .probe = max98090_i2c_probe, + .shutdown = max98090_i2c_shutdown, .remove = max98090_i2c_remove, .id_table = max98090_i2c_id, }; diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c index e9e6efbc21dd..bf3e933ee895 100644 --- a/sound/soc/codecs/max98357a.c +++ b/sound/soc/codecs/max98357a.c @@ -26,8 +26,6 @@ #include <sound/soc-dai.h> #include <sound/soc-dapm.h> -#define DRV_NAME "max98357a" - static int max98357a_daiops_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { @@ -87,9 +85,9 @@ static struct snd_soc_dai_ops max98357a_dai_ops = { }; static struct snd_soc_dai_driver max98357a_dai_driver = { - .name = DRV_NAME, + .name = "HiFi", .playback = { - .stream_name = DRV_NAME "-playback", + .stream_name = "HiFi Playback", .formats = SNDRV_PCM_FMTBIT_S16 | SNDRV_PCM_FMTBIT_S24 | SNDRV_PCM_FMTBIT_S32, @@ -127,7 +125,7 @@ static int max98357a_platform_remove(struct platform_device *pdev) #ifdef CONFIG_OF static const struct of_device_id max98357a_device_id[] = { - { .compatible = "maxim," DRV_NAME, }, + { .compatible = "maxim,max98357a" }, {} }; MODULE_DEVICE_TABLE(of, max98357a_device_id); @@ -135,7 +133,7 @@ MODULE_DEVICE_TABLE(of, max98357a_device_id); static struct platform_driver max98357a_platform_driver = { .driver = { - .name = DRV_NAME, + .name = "max98357a", .of_match_table = of_match_ptr(max98357a_device_id), }, .probe = max98357a_platform_probe, @@ -145,4 +143,3 @@ module_platform_driver(max98357a_platform_driver); MODULE_DESCRIPTION("Maxim MAX98357A Codec Driver"); MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platform:" DRV_NAME); diff --git a/sound/soc/codecs/max98925.c b/sound/soc/codecs/max98925.c new file mode 100644 index 000000000000..9b5a17de4690 --- /dev/null +++ b/sound/soc/codecs/max98925.c @@ -0,0 +1,655 @@ +/* + * max98925.c -- ALSA SoC Stereo MAX98925 driver + * Copyright 2013-15 Maxim Integrated Products + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ +#include <linux/delay.h> +#include <linux/i2c.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <linux/cdev.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include "max98925.h" + +static const char *const dai_text[] = { + "Left", "Right", "LeftRight", "LeftRightDiv2", +}; + +static const char * const max98925_boost_voltage_text[] = { + "8.5V", "8.25V", "8.0V", "7.75V", "7.5V", "7.25V", "7.0V", "6.75V", + "6.5V", "6.5V", "6.5V", "6.5V", "6.5V", "6.5V", "6.5V", "6.5V" +}; + +static SOC_ENUM_SINGLE_DECL(max98925_boost_voltage, + MAX98925_CONFIGURATION, M98925_BST_VOUT_SHIFT, + max98925_boost_voltage_text); + +static const char *const hpf_text[] = { + "Disable", "DC Block", "100Hz", "200Hz", "400Hz", "800Hz", +}; + +static const struct reg_default max98925_reg[] = { + { 0x0B, 0x00 }, /* IRQ Enable0 */ + { 0x0C, 0x00 }, /* IRQ Enable1 */ + { 0x0D, 0x00 }, /* IRQ Enable2 */ + { 0x0E, 0x00 }, /* IRQ Clear0 */ + { 0x0F, 0x00 }, /* IRQ Clear1 */ + { 0x10, 0x00 }, /* IRQ Clear2 */ + { 0x11, 0xC0 }, /* Map0 */ + { 0x12, 0x00 }, /* Map1 */ + { 0x13, 0x00 }, /* Map2 */ + { 0x14, 0xF0 }, /* Map3 */ + { 0x15, 0x00 }, /* Map4 */ + { 0x16, 0xAB }, /* Map5 */ + { 0x17, 0x89 }, /* Map6 */ + { 0x18, 0x00 }, /* Map7 */ + { 0x19, 0x00 }, /* Map8 */ + { 0x1A, 0x06 }, /* DAI Clock Mode 1 */ + { 0x1B, 0xC0 }, /* DAI Clock Mode 2 */ + { 0x1C, 0x00 }, /* DAI Clock Divider Denominator MSBs */ + { 0x1D, 0x00 }, /* DAI Clock Divider Denominator LSBs */ + { 0x1E, 0xF0 }, /* DAI Clock Divider Numerator MSBs */ + { 0x1F, 0x00 }, /* DAI Clock Divider Numerator LSBs */ + { 0x20, 0x50 }, /* Format */ + { 0x21, 0x00 }, /* TDM Slot Select */ + { 0x22, 0x00 }, /* DOUT Configuration VMON */ + { 0x23, 0x00 }, /* DOUT Configuration IMON */ + { 0x24, 0x00 }, /* DOUT Configuration VBAT */ + { 0x25, 0x00 }, /* DOUT Configuration VBST */ + { 0x26, 0x00 }, /* DOUT Configuration FLAG */ + { 0x27, 0xFF }, /* DOUT HiZ Configuration 1 */ + { 0x28, 0xFF }, /* DOUT HiZ Configuration 2 */ + { 0x29, 0xFF }, /* DOUT HiZ Configuration 3 */ + { 0x2A, 0xFF }, /* DOUT HiZ Configuration 4 */ + { 0x2B, 0x02 }, /* DOUT Drive Strength */ + { 0x2C, 0x90 }, /* Filters */ + { 0x2D, 0x00 }, /* Gain */ + { 0x2E, 0x02 }, /* Gain Ramping */ + { 0x2F, 0x00 }, /* Speaker Amplifier */ + { 0x30, 0x0A }, /* Threshold */ + { 0x31, 0x00 }, /* ALC Attack */ + { 0x32, 0x80 }, /* ALC Atten and Release */ + { 0x33, 0x00 }, /* ALC Infinite Hold Release */ + { 0x34, 0x92 }, /* ALC Configuration */ + { 0x35, 0x01 }, /* Boost Converter */ + { 0x36, 0x00 }, /* Block Enable */ + { 0x37, 0x00 }, /* Configuration */ + { 0x38, 0x00 }, /* Global Enable */ + { 0x3A, 0x00 }, /* Boost Limiter */ +}; + +static const struct soc_enum max98925_dai_enum = + SOC_ENUM_SINGLE(MAX98925_GAIN, 5, ARRAY_SIZE(dai_text), dai_text); + +static const struct soc_enum max98925_hpf_enum = + SOC_ENUM_SINGLE(MAX98925_FILTERS, 0, ARRAY_SIZE(hpf_text), hpf_text); + +static const struct snd_kcontrol_new max98925_hpf_sel_mux = + SOC_DAPM_ENUM("Rc Filter MUX Mux", max98925_hpf_enum); + +static const struct snd_kcontrol_new max98925_dai_sel_mux = + SOC_DAPM_ENUM("DAI IN MUX Mux", max98925_dai_enum); + +static int max98925_dac_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct max98925_priv *max98925 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + regmap_update_bits(max98925->regmap, + MAX98925_BLOCK_ENABLE, + M98925_BST_EN_MASK | + M98925_ADC_IMON_EN_MASK | M98925_ADC_VMON_EN_MASK, + M98925_BST_EN_MASK | + M98925_ADC_IMON_EN_MASK | M98925_ADC_VMON_EN_MASK); + break; + case SND_SOC_DAPM_POST_PMD: + regmap_update_bits(max98925->regmap, + MAX98925_BLOCK_ENABLE, M98925_BST_EN_MASK | + M98925_ADC_IMON_EN_MASK | M98925_ADC_VMON_EN_MASK, 0); + break; + default: + return 0; + } + return 0; +} + +static const struct snd_soc_dapm_widget max98925_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("DAI_OUT", "HiFi Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_MUX("DAI IN MUX", SND_SOC_NOPM, 0, 0, + &max98925_dai_sel_mux), + SND_SOC_DAPM_MUX("Rc Filter MUX", SND_SOC_NOPM, 0, 0, + &max98925_hpf_sel_mux), + SND_SOC_DAPM_DAC_E("Amp Enable", NULL, MAX98925_BLOCK_ENABLE, + M98925_SPK_EN_SHIFT, 0, max98925_dac_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY("Global Enable", MAX98925_GLOBAL_ENABLE, + M98925_EN_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_OUTPUT("BE_OUT"), +}; + +static const struct snd_soc_dapm_route max98925_audio_map[] = { + {"DAI IN MUX", "Left", "DAI_OUT"}, + {"DAI IN MUX", "Right", "DAI_OUT"}, + {"DAI IN MUX", "LeftRight", "DAI_OUT"}, + {"DAI IN MUX", "LeftRightDiv2", "DAI_OUT"}, + {"Rc Filter MUX", "Disable", "DAI IN MUX"}, + {"Rc Filter MUX", "DC Block", "DAI IN MUX"}, + {"Rc Filter MUX", "100Hz", "DAI IN MUX"}, + {"Rc Filter MUX", "200Hz", "DAI IN MUX"}, + {"Rc Filter MUX", "400Hz", "DAI IN MUX"}, + {"Rc Filter MUX", "800Hz", "DAI IN MUX"}, + {"Amp Enable", NULL, "Rc Filter MUX"}, + {"BE_OUT", NULL, "Amp Enable"}, + {"BE_OUT", NULL, "Global Enable"}, +}; + +static bool max98925_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX98925_VBAT_DATA: + case MAX98925_VBST_DATA: + case MAX98925_LIVE_STATUS0: + case MAX98925_LIVE_STATUS1: + case MAX98925_LIVE_STATUS2: + case MAX98925_STATE0: + case MAX98925_STATE1: + case MAX98925_STATE2: + case MAX98925_FLAG0: + case MAX98925_FLAG1: + case MAX98925_FLAG2: + case MAX98925_REV_VERSION: + return true; + default: + return false; + } +} + +static bool max98925_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX98925_IRQ_CLEAR0: + case MAX98925_IRQ_CLEAR1: + case MAX98925_IRQ_CLEAR2: + case MAX98925_ALC_HOLD_RLS: + return false; + default: + return true; + } +} + +static DECLARE_TLV_DB_SCALE(max98925_spk_tlv, -600, 100, 0); + +static const struct snd_kcontrol_new max98925_snd_controls[] = { + SOC_SINGLE_TLV("Speaker Volume", MAX98925_GAIN, + M98925_SPK_GAIN_SHIFT, (1<<M98925_SPK_GAIN_WIDTH |