diff options
| author | Takashi Iwai <tiwai@suse.de> | 2014-10-06 14:01:11 +0200 |
|---|---|---|
| committer | Takashi Iwai <tiwai@suse.de> | 2014-10-06 14:01:11 +0200 |
| commit | 8df22a4d6f5b81c9c1703579d4907b57002689ed (patch) | |
| tree | 064e9662d427a82076e1151fcd9aa78a1066f9f4 /sound | |
| parent | 0cae90a96c15f2fd3bd139ba5505755c9c9ef2eb (diff) | |
| parent | a5448c88b812390a3622e76d774e10c0da1fb970 (diff) | |
| download | linux-8df22a4d6f5b81c9c1703579d4907b57002689ed.tar.bz2 | |
Merge tag 'asoc-v3.18' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v3.18
- More componentisation work from Lars-Peter, this time mainly
cleaning up the suspend and bias level transition callbacks.
- Real system support for the Intel drivers and a bunch of fixes and
enhancements for the associated CODEC drivers, this is going to need
a lot quirks over time due to the lack of any firmware description of
the boards.
- Jack detect support for simple card from Dylan Reid.
- A bunch of small fixes and enhancements for the Freescale drivers.
- New drivers for Analog Devices SSM4567, Cirrus Logic CS35L32, Everest
Semiconductor ES8328 and Freescale cards using the ASRC in newer i.MX
processors.
Diffstat (limited to 'sound')
105 files changed, 5827 insertions, 1330 deletions
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index b03c7ae5f4e3..dfc28542a007 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1784,14 +1784,16 @@ static int snd_pcm_lib_ioctl_fifo_size(struct snd_pcm_substream *substream, { struct snd_pcm_hw_params *params = arg; snd_pcm_format_t format; - int channels, width; + int channels; + ssize_t frame_size; params->fifo_size = substream->runtime->hw.fifo_size; if (!(substream->runtime->hw.info & SNDRV_PCM_INFO_FIFO_IN_FRAMES)) { format = params_format(params); channels = params_channels(params); - width = snd_pcm_format_physical_width(format); - params->fifo_size /= width * channels; + frame_size = snd_pcm_format_size(format, channels); + if (frame_size > 0) + params->fifo_size /= (unsigned)frame_size; } return 0; } diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index d5b0582daaf0..d2eaf8bc10e1 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -777,6 +777,7 @@ static const struct hda_model_fixup cxt5066_fixup_models[] = { { .id = CXT_PINCFG_LENOVO_TP410, .name = "tp410" }, { .id = CXT_FIXUP_THINKPAD_ACPI, .name = "thinkpad" }, { .id = CXT_PINCFG_LEMOTE_A1004, .name = "lemote-a1004" }, + { .id = CXT_PINCFG_LEMOTE_A1205, .name = "lemote-a1205" }, { .id = CXT_FIXUP_OLPC_XO, .name = "olpc-xo" }, {} }; diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index 922006dd0583..4c3b0af39fd8 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -1337,8 +1337,6 @@ static int pm860x_probe(struct snd_soc_codec *codec) } } - pm860x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; out: @@ -1354,7 +1352,6 @@ static int pm860x_remove(struct snd_soc_codec *codec) for (i = 3; i >= 0; i--) free_irq(pm860x->irq[i], pm860x); - pm860x_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; } diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 8838838e25ed..a68d1731a8fd 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -43,6 +43,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_ALC5623 if I2C select SND_SOC_ALC5632 if I2C select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC + select SND_SOC_CS35L32 if I2C select SND_SOC_CS42L51_I2C if I2C select SND_SOC_CS42L52 if I2C && INPUT select SND_SOC_CS42L56 if I2C && INPUT @@ -56,7 +57,10 @@ config SND_SOC_ALL_CODECS select SND_SOC_DA7213 if I2C select SND_SOC_DA732X if I2C select SND_SOC_DA9055 if I2C + select SND_SOC_DMIC select SND_SOC_BT_SCO + select SND_SOC_ES8328_SPI if SPI_MASTER + select SND_SOC_ES8328_I2C if I2C select SND_SOC_ISABELLE if I2C select SND_SOC_JZ4740_CODEC select SND_SOC_LM4857 if I2C @@ -90,6 +94,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_SSM2518 if I2C select SND_SOC_SSM2602_SPI if SPI_MASTER select SND_SOC_SSM2602_I2C if I2C + select SND_SOC_SSM4567 if I2C select SND_SOC_STA32X if I2C select SND_SOC_STA350 if I2C select SND_SOC_STA529 if I2C @@ -323,6 +328,10 @@ config SND_SOC_ALC5632 config SND_SOC_CQ0093VC tristate +config SND_SOC_CS35L32 + tristate "Cirrus Logic CS35L32 CODEC" + depends on I2C + config SND_SOC_CS42L51 tristate @@ -405,6 +414,17 @@ config SND_SOC_DMIC config SND_SOC_HDMI_CODEC tristate "HDMI stub CODEC" +config SND_SOC_ES8328 + tristate "Everest Semi ES8328 CODEC" + +config SND_SOC_ES8328_I2C + tristate + select SND_SOC_ES8328 + +config SND_SOC_ES8328_SPI + tristate + select SND_SOC_ES8328 + config SND_SOC_ISABELLE tristate @@ -464,6 +484,7 @@ config SND_SOC_RL6231 config SND_SOC_RT286 tristate + depends on I2C config SND_SOC_RT5631 tristate @@ -520,12 +541,20 @@ config SND_SOC_SSM2602 tristate config SND_SOC_SSM2602_SPI + tristate "Analog Devices SSM2602 CODEC - SPI" + depends on SPI_MASTER select SND_SOC_SSM2602 - tristate + select REGMAP_SPI config SND_SOC_SSM2602_I2C + tristate "Analog Devices SSM2602 CODEC - I2C" + depends on I2C select SND_SOC_SSM2602 - tristate + select REGMAP_I2C + +config SND_SOC_SSM4567 + tristate "Analog Devices ssm4567 amplifier driver support" + depends on I2C config SND_SOC_STA32X tristate @@ -712,7 +741,8 @@ config SND_SOC_WM8974 tristate config SND_SOC_WM8978 - tristate + tristate "Wolfson Microelectronics WM8978 codec" + depends on I2C config SND_SOC_WM8983 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 20afe0f0c5be..5dce451661e4 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -32,6 +32,7 @@ snd-soc-ak4671-objs := ak4671.o snd-soc-ak5386-objs := ak5386.o snd-soc-arizona-objs := arizona.o snd-soc-cq93vc-objs := cq93vc.o +snd-soc-cs35l32-objs := cs35l32.o snd-soc-cs42l51-objs := cs42l51.o snd-soc-cs42l51-i2c-objs := cs42l51-i2c.o snd-soc-cs42l52-objs := cs42l52.o @@ -49,6 +50,9 @@ snd-soc-da732x-objs := da732x.o snd-soc-da9055-objs := da9055.o snd-soc-bt-sco-objs := bt-sco.o snd-soc-dmic-objs := dmic.o +snd-soc-es8328-objs := es8328.o +snd-soc-es8328-i2c-objs := es8328-i2c.o +snd-soc-es8328-spi-objs := es8328-spi.o snd-soc-isabelle-objs := isabelle.o snd-soc-jz4740-codec-objs := jz4740.o snd-soc-l3-objs := l3.o @@ -91,6 +95,7 @@ snd-soc-ssm2518-objs := ssm2518.o snd-soc-ssm2602-objs := ssm2602.o snd-soc-ssm2602-spi-objs := ssm2602-spi.o snd-soc-ssm2602-i2c-objs := ssm2602-i2c.o +snd-soc-ssm4567-objs := ssm4567.o snd-soc-sta32x-objs := sta32x.o snd-soc-sta350-objs := sta350.o snd-soc-sta529-objs := sta529.o @@ -203,6 +208,7 @@ obj-$(CONFIG_SND_SOC_ALC5623) += snd-soc-alc5623.o obj-$(CONFIG_SND_SOC_ALC5632) += snd-soc-alc5632.o obj-$(CONFIG_SND_SOC_ARIZONA) += snd-soc-arizona.o obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o +obj-$(CONFIG_SND_SOC_CS35L32) += snd-soc-cs35l32.o obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o obj-$(CONFIG_SND_SOC_CS42L51_I2C) += snd-soc-cs42l51-i2c.o obj-$(CONFIG_SND_SOC_CS42L52) += snd-soc-cs42l52.o @@ -220,6 +226,9 @@ obj-$(CONFIG_SND_SOC_DA732X) += snd-soc-da732x.o obj-$(CONFIG_SND_SOC_DA9055) += snd-soc-da9055.o obj-$(CONFIG_SND_SOC_BT_SCO) += snd-soc-bt-sco.o obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o +obj-$(CONFIG_SND_SOC_ES8328) += snd-soc-es8328.o +obj-$(CONFIG_SND_SOC_ES8328_I2C)+= snd-soc-es8328-i2c.o +obj-$(CONFIG_SND_SOC_ES8328_SPI)+= snd-soc-es8328-spi.o obj-$(CONFIG_SND_SOC_ISABELLE) += snd-soc-isabelle.o obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o @@ -258,6 +267,7 @@ obj-$(CONFIG_SND_SOC_SSM2518) += snd-soc-ssm2518.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o obj-$(CONFIG_SND_SOC_SSM2602_SPI) += snd-soc-ssm2602-spi.o obj-$(CONFIG_SND_SOC_SSM2602_I2C) += snd-soc-ssm2602-i2c.o +obj-$(CONFIG_SND_SOC_SSM4567) += snd-soc-ssm4567.o obj-$(CONFIG_SND_SOC_STA32X) += snd-soc-sta32x.o obj-$(CONFIG_SND_SOC_STA350) += snd-soc-sta350.o obj-$(CONFIG_SND_SOC_STA529) += snd-soc-sta529.o diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index 1fb4402bf72d..fd43827bb856 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -56,8 +56,7 @@ #define GPIO31_DIR_OUTPUT 0x40 /* Macrocell register definitions */ -#define AB8500_CTRL3_REG 0x0200 -#define AB8500_GPIO_DIR4_REG 0x1013 +#define AB8500_GPIO_DIR4_REG 0x13 /* Bank AB8500_MISC */ /* Nr of FIR/IIR-coeff banks in ANC-block */ #define AB8500_NR_OF_ANC_COEFF_BANKS 2 @@ -126,6 +125,8 @@ struct ab8500_codec_drvdata_dbg { /* Private data for AB8500 device-driver */ struct ab8500_codec_drvdata { + struct regmap *regmap; + /* Sidetone */ long *sid_fir_values; enum sid_state sid_status; @@ -166,49 +167,35 @@ static inline const char *amic_type_str(enum amic_type type) */ /* Read a register from the audio-bank of AB8500 */ -static unsigned int ab8500_codec_read_reg(struct snd_soc_codec *codec, - unsigned int reg) +static int ab8500_codec_read_reg(void *context, unsigned int reg, + unsigned int *value) { + struct device *dev = context; int status; - unsigned int value = 0; u8 value8; - status = abx500_get_register_interruptible(codec->dev, AB8500_AUDIO, - reg, &value8); - if (status < 0) { - dev_err(codec->dev, - "%s: ERROR: Register (0x%02x:0x%02x) read failed (%d).\n", - __func__, (u8)AB8500_AUDIO, (u8)reg, status); - } else { - dev_dbg(codec->dev, - "%s: Read 0x%02x from register 0x%02x:0x%02x\n", - __func__, value8, (u8)AB8500_AUDIO, (u8)reg); - value = (unsigned int)value8; - } + status = abx500_get_register_interruptible(dev, AB8500_AUDIO, + reg, &value8); + *value = (unsigned int)value8; - return value; + return status; } /* Write to a register in the audio-bank of AB8500 */ -static int ab8500_codec_write_reg(struct snd_soc_codec *codec, - unsigned int reg, unsigned int value) +static int ab8500_codec_write_reg(void *context, unsigned int reg, + unsigned int value) { - int status; - - status = abx500_set_register_interruptible(codec->dev, AB8500_AUDIO, - reg, value); - if (status < 0) - dev_err(codec->dev, - "%s: ERROR: Register (%02x:%02x) write failed (%d).\n", - __func__, (u8)AB8500_AUDIO, (u8)reg, status); - else - dev_dbg(codec->dev, - "%s: Wrote 0x%02x into register %02x:%02x\n", - __func__, (u8)value, (u8)AB8500_AUDIO, (u8)reg); + struct device *dev = context; - return status; + return abx500_set_register_interruptible(dev, AB8500_AUDIO, + reg, value); } +static const struct regmap_config ab8500_codec_regmap = { + .reg_read = ab8500_codec_read_reg, + .reg_write = ab8500_codec_write_reg, +}; + /* * Controls - DAPM */ @@ -1968,16 +1955,16 @@ static int ab8500_audio_setup_mics(struct snd_soc_codec *codec, dev_dbg(codec->dev, "%s: Enter.\n", __func__); /* Set DMic-clocks to outputs */ - status = abx500_get_register_interruptible(codec->dev, (u8)AB8500_MISC, - (u8)AB8500_GPIO_DIR4_REG, + status = abx500_get_register_interruptible(codec->dev, AB8500_MISC, + AB8500_GPIO_DIR4_REG, &value8); if (status < 0) return status; value = value8 | GPIO27_DIR_OUTPUT | GPIO29_DIR_OUTPUT | GPIO31_DIR_OUTPUT; status = abx500_set_register_interruptible(codec->dev, - (u8)AB8500_MISC, - (u8)AB8500_GPIO_DIR4_REG, + AB8500_MISC, + AB8500_GPIO_DIR4_REG, value); if (status < 0) return status; @@ -2565,9 +2552,6 @@ static int ab8500_codec_probe(struct snd_soc_codec *codec) static struct snd_soc_codec_driver ab8500_codec_driver = { .probe = ab8500_codec_probe, - .read = ab8500_codec_read_reg, - .write = ab8500_codec_write_reg, - .reg_word_size = sizeof(u8), .controls = ab8500_ctrls, .num_controls = ARRAY_SIZE(ab8500_ctrls), .dapm_widgets = ab8500_dapm_widgets, @@ -2592,6 +2576,15 @@ static int ab8500_codec_driver_probe(struct platform_device *pdev) drvdata->anc_status = ANC_UNCONFIGURED; dev_set_drvdata(&pdev->dev, drvdata); + drvdata->regmap = devm_regmap_init(&pdev->dev, NULL, &pdev->dev, + &ab8500_codec_regmap); + if (IS_ERR(drvdata->regmap)) { + status = PTR_ERR(drvdata->regmap); + dev_err(&pdev->dev, "%s: Failed to allocate regmap: %d\n", + __func__, status); + return status; + } + dev_dbg(&pdev->dev, "%s: Register codec.\n", __func__); status = snd_soc_register_codec(&pdev->dev, &ab8500_codec_driver, ab8500_codec_dai, diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index e889e1b84192..bd9b1839c8b0 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -69,19 +69,6 @@ static struct snd_soc_dai_driver ac97_dai = { .ops = &ac97_dai_ops, }; -static unsigned int ac97_read(struct snd_soc_codec *codec, - unsigned int reg) -{ - return soc_ac97_ops->read(codec->ac97, reg); -} - -static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int val) -{ - soc_ac97_ops->write(codec->ac97, reg, val); - return 0; -} - static int ac97_soc_probe(struct snd_soc_codec *codec) { struct snd_ac97_bus *ac97_bus; @@ -122,8 +109,6 @@ static int ac97_soc_resume(struct snd_soc_codec *codec) #endif static struct snd_soc_codec_driver soc_codec_dev_ac97 = { - .write = ac97_write, - .read = ac97_read, .probe = ac97_soc_probe, .suspend = ac97_soc_suspend, .resume = ac97_soc_resume, diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c index 1ff7d4d027e9..7c784ad3e8b2 100644 --- a/sound/soc/codecs/adau1373.c +++ b/sound/soc/codecs/adau1373.c @@ -1448,29 +1448,10 @@ static int adau1373_set_bias_level(struct snd_soc_codec *codec, return 0; } -static int adau1373_remove(struct snd_soc_codec *codec) -{ - adau1373_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - -static int adau1373_suspend(struct snd_soc_codec *codec) -{ - struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec); - int ret; - - ret = adau1373_set_bias_level(codec, SND_SOC_BIAS_OFF); - regcache_cache_only(adau1373->regmap, true); - - return ret; -} - static int adau1373_resume(struct snd_soc_codec *codec) { struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec); - regcache_cache_only(adau1373->regmap, false); - adau1373_set_bias_level(codec, SND_SOC_BIAS_STANDBY); regcache_sync(adau1373->regmap); return 0; @@ -1501,8 +1482,6 @@ static const struct regmap_config adau1373_regmap_config = { static struct snd_soc_codec_driver adau1373_codec_driver = { .probe = adau1373_probe, - .remove = adau1373_remove, - .suspend = adau1373_suspend, .resume = adau1373_resume, .set_bias_level = adau1373_set_bias_level, .idle_bias_off = true, diff --git a/sound/soc/codecs/adau1761.c b/sound/soc/codecs/adau1761.c index 848cab839553..5518ebd6947c 100644 --- a/sound/soc/codecs/adau1761.c +++ b/sound/soc/codecs/adau1761.c @@ -714,9 +714,9 @@ static int adau1761_codec_probe(struct snd_soc_codec *codec) static const struct snd_soc_codec_driver adau1761_codec_driver = { .probe = adau1761_codec_probe, - .suspend = adau17x1_suspend, .resume = adau17x1_resume, .set_bias_level = adau1761_set_bias_level, + .suspend_bias_off = true, .controls = adau1761_controls, .num_controls = ARRAY_SIZE(adau1761_controls), diff --git a/sound/soc/codecs/adau1781.c b/sound/soc/codecs/adau1781.c index 045a61413840..e9fc00fb13dd 100644 --- a/sound/soc/codecs/adau1781.c +++ b/sound/soc/codecs/adau1781.c @@ -446,9 +446,9 @@ static int adau1781_codec_probe(struct snd_soc_codec *codec) static const struct snd_soc_codec_driver adau1781_codec_driver = { .probe = adau1781_codec_probe, - .suspend = adau17x1_suspend, .resume = adau17x1_resume, .set_bias_level = adau1781_set_bias_level, + .suspend_bias_off = true, .controls = adau1781_controls, .num_controls = ARRAY_SIZE(adau1781_controls), diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c index 0b659704e60c..3e16c1c64115 100644 --- a/sound/soc/codecs/adau17x1.c +++ b/sound/soc/codecs/adau17x1.c @@ -815,13 +815,6 @@ int adau17x1_add_routes(struct snd_soc_codec *codec) } EXPORT_SYMBOL_GPL(adau17x1_add_routes); -int adau17x1_suspend(struct snd_soc_codec *codec) -{ - codec->driver->set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} -EXPORT_SYMBOL_GPL(adau17x1_suspend); - int adau17x1_resume(struct snd_soc_codec *codec) { struct adau *adau = snd_soc_codec_get_drvdata(codec); @@ -829,7 +822,6 @@ int adau17x1_resume(struct snd_soc_codec *codec) if (adau->switch_mode) adau->switch_mode(codec->dev); - codec->driver->set_bias_level(codec, SND_SOC_BIAS_STANDBY); regcache_sync(adau->regmap); return 0; diff --git a/sound/soc/codecs/adau17x1.h b/sound/soc/codecs/adau17x1.h index 3ffabaf4c7a8..e4a557fd7155 100644 --- a/sound/soc/codecs/adau17x1.h +++ b/sound/soc/codecs/adau17x1.h @@ -52,7 +52,6 @@ int adau17x1_set_micbias_voltage(struct snd_soc_codec *codec, enum adau17x1_micbias_voltage micbias); bool adau17x1_readable_register(struct device *dev, unsigned int reg); bool adau17x1_volatile_register(struct device *dev, unsigned int reg); -int adau17x1_suspend(struct snd_soc_codec *codec); int adau17x1_resume(struct snd_soc_codec *codec); extern const struct snd_soc_dai_ops adau17x1_dai_ops; diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c index c43b93fdf0df..ce3cdca9fc62 100644 --- a/sound/soc/codecs/adav80x.c +++ b/sound/soc/codecs/adav80x.c @@ -812,42 +812,23 @@ static int adav80x_probe(struct snd_soc_codec *codec) /* Disable DAC zero flag */ regmap_write(adav80x->regmap, ADAV80X_DAC_CTRL3, 0x6); - return adav80x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); -} - -static int adav80x_suspend(struct snd_soc_codec *codec) -{ - struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); - int ret; - - ret = adav80x_set_bias_level(codec, SND_SOC_BIAS_OFF); - regcache_cache_only(adav80x->regmap, true); - - return ret; + return 0; } static int adav80x_resume(struct snd_soc_codec *codec) { struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); - regcache_cache_only(adav80x->regmap, false); - adav80x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); regcache_sync(adav80x->regmap); return 0; } -static int adav80x_remove(struct snd_soc_codec *codec) -{ - return adav80x_set_bias_level(codec, SND_SOC_BIAS_OFF); -} - static struct snd_soc_codec_driver adav80x_codec_driver = { .probe = adav80x_probe, - .remove = adav80x_remove, - .suspend = adav80x_suspend, .resume = adav80x_resume, .set_bias_level = adav80x_set_bias_level, + .suspend_bias_off = true, .set_pll = adav80x_set_pll, .set_sysclk = adav80x_set_sysclk, diff --git a/sound/soc/codecs/cs35l32.c b/sound/soc/codecs/cs35l32.c new file mode 100644 index 000000000000..c125925da92e --- /dev/null +++ b/sound/soc/codecs/cs35l32.c @@ -0,0 +1,631 @@ +/* + * cs35l32.c -- CS35L32 ALSA SoC audio driver + * + * Copyright 2014 CirrusLogic, Inc. + * + * Author: Brian Austin <brian.austin@cirrus.com> + * + * 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/module.h> +#include <linux/moduleparam.h> +#include <linux/version.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/i2c.h> +#include <linux/gpio.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <linux/platform_device.h> +#include <linux/regulator/consumer.h> +#include <linux/gpio/consumer.h> +#include <linux/of_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <dt-bindings/sound/cs35l32.h> + +#include "cs35l32.h" + +#define CS35L32_NUM_SUPPLIES 2 +static const char *const cs35l32_supply_names[CS35L32_NUM_SUPPLIES] = { + "VA", + "VP", +}; + +struct cs35l32_private { + struct regmap *regmap; + struct snd_soc_codec *codec; + struct regulator_bulk_data supplies[CS35L32_NUM_SUPPLIES]; + struct cs35l32_platform_data pdata; + struct gpio_desc *reset_gpio; +}; + +static const struct reg_default cs35l32_reg_defaults[] = { + + { 0x06, 0x04 }, /* Power Ctl 1 */ + { 0x07, 0xE8 }, /* Power Ctl 2 */ + { 0x08, 0x40 }, /* Clock Ctl */ + { 0x09, 0x20 }, /* Low Battery Threshold */ + { 0x0A, 0x00 }, /* Voltage Monitor [RO] */ + { 0x0B, 0x40 }, /* Conv Peak Curr Protection CTL */ + { 0x0C, 0x07 }, /* IMON Scaling */ + { 0x0D, 0x03 }, /* Audio/LED Pwr Manager */ + { 0x0F, 0x20 }, /* Serial Port Control */ + { 0x10, 0x14 }, /* Class D Amp CTL */ + { 0x11, 0x00 }, /* Protection Release CTL */ + { 0x12, 0xFF }, /* Interrupt Mask 1 */ + { 0x13, 0xFF }, /* Interrupt Mask 2 */ + { 0x14, 0xFF }, /* Interrupt Mask 3 */ + { 0x19, 0x00 }, /* LED Flash Mode Current */ + { 0x1A, 0x00 }, /* LED Movie Mode Current */ + { 0x1B, 0x20 }, /* LED Flash Timer */ + { 0x1C, 0x00 }, /* LED Flash Inhibit Current */ +}; + +static bool cs35l32_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS35L32_DEVID_AB: + case CS35L32_DEVID_CD: + case CS35L32_DEVID_E: + case CS35L32_FAB_ID: + case CS35L32_REV_ID: + case CS35L32_PWRCTL1: + case CS35L32_PWRCTL2: + case CS35L32_CLK_CTL: + case CS35L32_BATT_THRESHOLD: + case CS35L32_VMON: + case CS35L32_BST_CPCP_CTL: + case CS35L32_IMON_SCALING: + case CS35L32_AUDIO_LED_MNGR: + case CS35L32_ADSP_CTL: + case CS35L32_CLASSD_CTL: + case CS35L32_PROTECT_CTL: + case CS35L32_INT_MASK_1: + case CS35L32_INT_MASK_2: + case CS35L32_INT_MASK_3: + case CS35L32_INT_STATUS_1: + case CS35L32_INT_STATUS_2: + case CS35L32_INT_STATUS_3: + case CS35L32_LED_STATUS: + case CS35L32_FLASH_MODE: + case CS35L32_MOVIE_MODE: + case CS35L32_FLASH_TIMER: + case CS35L32_FLASH_INHIBIT: + return true; + default: + return false; + } +} + +static bool cs35l32_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS35L32_DEVID_AB: + case CS35L32_DEVID_CD: + case CS35L32_DEVID_E: + case CS35L32_FAB_ID: + case CS35L32_REV_ID: + case CS35L32_INT_STATUS_1: + case CS35L32_INT_STATUS_2: + case CS35L32_INT_STATUS_3: + case CS35L32_LED_STATUS: + return true; + default: + return false; + } +} + +static bool cs35l32_precious_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS35L32_INT_STATUS_1: + case CS35L32_INT_STATUS_2: + case CS35L32_INT_STATUS_3: + case CS35L32_LED_STATUS: + return true; + default: + return false; + } +} + +static DECLARE_TLV_DB_SCALE(classd_ctl_tlv, 900, 300, 0); + +static const struct snd_kcontrol_new imon_ctl = + SOC_DAPM_SINGLE("Switch", CS35L32_PWRCTL2, 6, 1, 1); + +static const struct snd_kcontrol_new vmon_ctl = + SOC_DAPM_SINGLE("Switch", CS35L32_PWRCTL2, 7, 1, 1); + +static const struct snd_kcontrol_new vpmon_ctl = + SOC_DAPM_SINGLE("Switch", CS35L32_PWRCTL2, 5, 1, 1); + +static const struct snd_kcontrol_new cs35l32_snd_controls[] = { + SOC_SINGLE_TLV("Speaker Volume", CS35L32_CLASSD_CTL, + 3, 0x04, 1, classd_ctl_tlv), + SOC_SINGLE("Zero Cross Switch", CS35L32_CLASSD_CTL, 2, 1, 0), + SOC_SINGLE("Gain Manager Switch", CS35L32_AUDIO_LED_MNGR, 3, 1, 0), +}; + +static const struct snd_soc_dapm_widget cs35l32_dapm_widgets[] = { + + SND_SOC_DAPM_SUPPLY("BOOST", CS35L32_PWRCTL1, 2, 1, NULL, 0), + SND_SOC_DAPM_OUT_DRV("Speaker", CS35L32_PWRCTL1, 7, 1, NULL, 0), + + SND_SOC_DAPM_AIF_OUT("SDOUT", NULL, 0, CS35L32_PWRCTL2, 3, 1), + + SND_SOC_DAPM_INPUT("VP"), + SND_SOC_DAPM_INPUT("ISENSE"), + SND_SOC_DAPM_INPUT("VSENSE"), + + SND_SOC_DAPM_SWITCH("VMON ADC", CS35L32_PWRCTL2, 7, 1, &vmon_ctl), + SND_SOC_DAPM_SWITCH("IMON ADC", CS35L32_PWRCTL2, 6, 1, &imon_ctl), + SND_SOC_DAPM_SWITCH("VPMON ADC", CS35L32_PWRCTL2, 5, 1, &vpmon_ctl), +}; + +static const struct snd_soc_dapm_route cs35l32_audio_map[] = { + + {"Speaker", NULL, "BOOST"}, + + {"VMON ADC", NULL, "VSENSE"}, + {"IMON ADC", NULL, "ISENSE"}, + {"VPMON ADC", NULL, "VP"}, + + {"SDOUT", "Switch", "VMON ADC"}, + {"SDOUT", "Switch", "IMON ADC"}, + {"SDOUT", "Switch", "VPMON ADC"}, + + {"Capture", NULL, "SDOUT"}, +}; + +static int cs35l32_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + snd_soc_update_bits(codec, CS35L32_ADSP_CTL, + CS35L32_ADSP_MASTER_MASK, + CS35L32_ADSP_MASTER_MASK); + break; + case SND_SOC_DAIFMT_CBS_CFS: + snd_soc_update_bits(codec, CS35L32_ADSP_CTL, + CS35L32_ADSP_MASTER_MASK, 0); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int cs35l32_set_tristate(struct snd_soc_dai *dai, int tristate) +{ + struct snd_soc_codec *codec = dai->codec; + + return snd_soc_update_bits(codec, CS35L32_PWRCTL2, + CS35L32_SDOUT_3ST, tristate << 3); +} + +static const struct snd_soc_dai_ops cs35l32_ops = { + .set_fmt = cs35l32_set_dai_fmt, + .set_tristate = cs35l32_set_tristate, +}; + +static struct snd_soc_dai_driver cs35l32_dai[] = { + { + .name = "cs35l32-monitor", + .id = 0, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = CS35L32_RATES, + .formats = CS35L32_FORMATS, + }, + .ops = &cs35l32_ops, + .symmetric_rates = 1, + } +}; + +static int cs35l32_codec_set_sysclk(struct snd_soc_codec *codec, + int clk_id, int source, unsigned int freq, int dir) +{ + unsigned int val; + + switch (freq) { + case 6000000: + val = CS35L32_MCLK_RATIO; + break; + case 12000000: + val = CS35L32_MCLK_DIV2_MASK | CS35L32_MCLK_RATIO; + break; + case 6144000: + val = 0; + break; + case 12288000: + val = CS35L32_MCLK_DIV2_MASK; + break; + default: + return -EINVAL; + } + + return snd_soc_update_bits(codec, CS35L32_CLK_CTL, + CS35L32_MCLK_DIV2_MASK | CS35L32_MCLK_RATIO_MASK, val); +} + +static struct snd_soc_codec_driver soc_codec_dev_cs35l32 = { + .set_sysclk = cs35l32_codec_set_sysclk, + + .dapm_widgets = cs35l32_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs35l32_dapm_widgets), + .dapm_routes = cs35l32_audio_map, + .num_dapm_routes = ARRAY_SIZE(cs35l32_audio_map), + + .controls = cs35l32_snd_controls, + .num_controls = ARRAY_SIZE(cs35l32_snd_controls), +}; + +/* Current and threshold powerup sequence Pg37 in datasheet */ +static const struct reg_default cs35l32_monitor_patch[] = { + + { 0x00, 0x99 }, + { 0x48, 0x17 }, + { 0x49, 0x56 }, + { 0x43, 0x01 }, + { 0x3B, 0x62 }, + { 0x3C, 0x80 }, + { 0x00, 0x00 }, +}; + +static struct regmap_config cs35l32_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = CS35L32_MAX_REGISTER, + .reg_defaults = cs35l32_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(cs35l32_reg_defaults), + .volatile_reg = cs35l32_volatile_register, + .readable_reg = cs35l32_readable_register, + .precious_reg = cs35l32_precious_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int cs35l32_handle_of_data(struct i2c_client *i2c_client, + struct cs35l32_platform_data *pdata) +{ + struct device_node *np = i2c_client->dev.of_node; + unsigned int val; + + if (of_property_read_u32(np, "cirrus,sdout-share", &val) >= 0) + pdata->sdout_share = val; + + of_property_read_u32(np, "cirrus,boost-manager", &val); + switch (val) { + case CS35L32_BOOST_MGR_AUTO: + case CS35L32_BOOST_MGR_AUTO_AUDIO: + case CS35L32_BOOST_MGR_BYPASS: + case CS35L32_BOOST_MGR_FIXED: + pdata->boost_mng = val; + break; + default: + dev_err(&i2c_client->dev, + "Wrong cirrus,boost-manager DT value %d\n", val); + pdata->boost_mng = CS35L32_BOOST_MGR_BYPASS; + } + + of_property_read_u32(np, "cirrus,sdout-datacfg", &val); + switch (val) { + case CS35L32_DATA_CFG_LR_VP: + case CS35L32_DATA_CFG_LR_STAT: + case CS35L32_DATA_CFG_LR: + case CS35L32_DATA_CFG_LR_VPSTAT: + pdata->sdout_datacfg = val; + break; + default: + dev_err(&i2c_client->dev, + "Wrong cirrus,sdout-datacfg DT value %d\n", val); + pdata->sdout_datacfg = CS35L32_DATA_CFG_LR; + } + + of_property_read_u32(np, "cirrus,battery-threshold", &val); + switch (val) { + case CS35L32_BATT_THRESH_3_1V: + case CS35L32_BATT_THRESH_3_2V: + case CS35L32_BATT_THRESH_3_3V: + case CS35L32_BATT_THRESH_3_4V: + pdata->batt_thresh = val; + break; + default: + dev_err(&i2c_client->dev, + "Wrong cirrus,battery-threshold DT value %d\n", val); + pdata->batt_thresh = CS35L32_BATT_THRESH_3_3V; + } + + of_property_read_u32(np, "cirrus,battery-recovery", &val); + switch (val) { + case CS35L32_BATT_RECOV_3_1V: + case CS35L32_BATT_RECOV_3_2V: + case CS35L32_BATT_RECOV_3_3V: + case CS35L32_BATT_RECOV_3_4V: + case CS35L32_BATT_RECOV_3_5V: + case CS35L32_BATT_RECOV_3_6V: + pdata->batt_recov = val; + break; + default: + dev_err(&i2c_client->dev, + "Wrong cirrus,battery-recovery DT value %d\n", val); + pdata->batt_recov = CS35L32_BATT_RECOV_3_4V; + } + + return 0; +} + +static int cs35l32_i2c_probe(struct i2c_client *i2c_client, + const struct i2c_device_id *id) +{ + struct cs35l32_private *cs35l32; + struct cs35l32_platform_data *pdata = + dev_get_platdata(&i2c_client->dev); + int ret, i; + unsigned int devid = 0; + unsigned int reg; + + + cs35l32 = devm_kzalloc(&i2c_client->dev, sizeof(struct cs35l32_private), + GFP_KERNEL); + if (!cs35l32) { + dev_err(&i2c_client->dev, "could not allocate codec\n"); + return -ENOMEM; + } + + i2c_set_clientdata(i2c_client, cs35l32); + + cs35l32->regmap = devm_regmap_init_i2c(i2c_client, &cs35l32_regmap); + if (IS_ERR(cs35l32->regmap)) { + ret = PTR_ERR(cs35l32->regmap); + dev_err(&i2c_client->dev, "regmap_init() failed: %d\n", ret); + return ret; + } + + if (pdata) { + cs35l32->pdata = *pdata; + } else { + pdata = devm_kzalloc(&i2c_client->dev, + sizeof(struct cs35l32_platform_data), + GFP_KERNEL); + if (!pdata) { + dev_err(&i2c_client->dev, "could not allocate pdata\n"); + return -ENOMEM; + } + if (i2c_client->dev.of_node) { + ret = cs35l32_handle_of_data(i2c_client, + &cs35l32->pdata); + if (ret != 0) + return ret; + } + } + + for (i = 0; i < ARRAY_SIZE(cs35l32->supplies); i++) + cs35l32->supplies[i].supply = cs35l32_supply_names[i]; + + ret = devm_regulator_bulk_get(&i2c_client->dev, + ARRAY_SIZE(cs35l32->supplies), + cs35l32->supplies); + if (ret != 0) { + dev_err(&i2c_client->dev, + "Failed to request supplies: %d\n", ret); + return ret; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(cs35l32->supplies), + cs35l32->supplies); + if (ret != 0) { + dev_err(&i2c_client->dev, + "Failed to enable supplies: %d\n", ret); + return ret; + } + + /* 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; + gpiod_set_value_cansleep(cs35l32->reset_gpio, 1); + } + + /* initialize codec */ + ret = regmap_read(cs35l32->regmap, CS35L32_DEVID_AB, ®); + devid = (reg & 0xFF) << 12; + + ret = regmap_read(cs35l32->regmap, CS35L32_DEVID_CD, ®); + devid |= (reg & 0xFF) << 4; + + ret = regmap_read(cs35l32->regmap, CS35L32_DEVID_E, ®); + devid |= (reg & 0xF0) >> 4; + + if (devid != CS35L32_CHIP_ID) { + ret = -ENODEV; + dev_err(&i2c_client->dev, + "CS35L32 Device ID (%X). Expected %X\n", + devid, CS35L32_CHIP_ID); + return ret; + } + + ret = regmap_read(cs35l32->regmap, CS35L32_REV_ID, ®); + if (ret < 0) { + dev_err(&i2c_client->dev, "Get Revision ID failed\n"); + return ret; + } + + ret = regmap_register_patch(cs35l32->regmap, cs35l32_monitor_patch, + ARRAY_SIZE(cs35l32_monitor_patch)); + if (ret < 0) { + dev_err(&i2c_client->dev, "Failed to apply errata patch\n"); + return ret; + } + + dev_info(&i2c_client->dev, + "Cirrus Logic CS35L32, Revision: %02X\n", reg & 0xFF); + + /* Setup VBOOST Management */ + if (cs35l32->pdata.boost_mng) + regmap_update_bits(cs35l32->regmap, CS35L32_AUDIO_LED_MNGR, + CS35L32_BOOST_MASK, + cs35l32->pdata.boost_mng); + + /* Setup ADSP Format Config */ + if (cs35l32->pdata.sdout_share) + regmap_update_bits(cs35l32->regmap, CS35L32_ADSP_CTL, + CS35L32_ADSP_SHARE_MASK, + cs35l32->pdata.sdout_share << 3); + + /* Setup ADSP Data Configuration */ + if (cs35l32->pdata.sdout_datacfg) + regmap_update_bits(cs35l32->regmap, CS35L32_ADSP_CTL, + CS35L32_ADSP_DATACFG_MASK, + cs35l32->pdata.sdout_datacfg << 4); + + /* Setup Low Battery Recovery */ + if (cs35l32->pdata.batt_recov) + regmap_update_bits(cs35l32->regmap, CS35L32_BATT_THRESHOLD, + CS35L32_BATT_REC_MASK, + cs35l32->pdata.batt_recov << 1); + + /* Setup Low Battery Threshold */ + if (cs35l32->pdata.batt_thresh) + regmap_update_bits(cs35l32->regmap, CS35L32_BATT_THRESHOLD, + CS35L32_BATT_THRESH_MASK, + cs35l32->pdata.batt_thresh << 4); + + /* Power down the AMP */ + regmap_update_bits(cs35l32->regmap, CS35L32_PWRCTL1, CS35L32_PDN_AMP, + CS35L32_PDN_AMP); + + /* Clear MCLK Error Bit since we don't have the clock yet */ + ret = regmap_read(cs35l32->regmap, CS35L32_INT_STATUS_1, ®); + + ret = snd_soc_register_codec(&i2c_client->dev, + &soc_codec_dev_cs35l32, cs35l32_dai, + ARRAY_SIZE(cs35l32_dai)); + if (ret < 0) + goto err_disable; + + return 0; + +err_disable: + regulator_bulk_disable(ARRAY_SIZE(cs35l32->supplies), + cs35l32->supplies); + return ret; +} + +static int cs35l32_i2c_remove(struct i2c_client *i2c_client) +{ + struct cs35l32_private *cs35l32 = i2c_get_clientdata(i2c_client); + + snd_soc_unregister_codec(&i2c_client->dev); + + /* Hold down reset */ + if (cs35l32->reset_gpio) + gpiod_set_value_cansleep(cs35l32->reset_gpio, 0); + + return 0; +} + +#ifdef CONFIG_PM_RUNTIME +static int cs35l32_runtime_suspend(struct device *dev) +{ + struct cs35l32_private *cs35l32 = dev_get_drvdata(dev); + + regcache_cache_only(cs35l32->regmap, true); + regcache_mark_dirty(cs35l32->regmap); + + /* Hold down reset */ + if (cs35l32->reset_gpio) + gpiod_set_value_cansleep(cs35l32->reset_gpio, 0); + + /* remove power */ + regulator_bulk_disable(ARRAY_SIZE(cs35l32->supplies), + cs35l32->supplies); + + return 0; +} + +static int cs35l32_runtime_resume(struct device *dev) +{ + struct cs35l32_private *cs35l32 = dev_get_drvdata(dev); + int ret; + + /* Enable power */ + ret = regulator_bulk_enable(ARRAY_SIZE(cs35l32->supplies), + cs35l32->supplies); + if (ret != 0) { + dev_err(dev, "Failed to enable supplies: %d\n", + ret); + return ret; + } + + if (cs35l32->reset_gpio) + gpiod_set_value_cansleep(cs35l32->reset_gpio, 1); + + regcache_cache_only(cs35l32->regmap, false); + regcache_sync(cs35l32->regmap); + + return 0; +} +#endif + +static const struct dev_pm_ops cs35l32_runtime_pm = { + SET_RUNTIME_PM_OPS(cs35l32_runtime_suspend, cs35l32_runtime_resume, + NULL) +}; + +static const struct of_device_id cs35l32_of_match[] = { + { .compatible = "cirrus,cs35l32", }, + {}, +}; +MODULE_DEVICE_TABLE(of, cs35l32_of_match); + + +static const struct i2c_device_id cs35l32_id[] = { + {"cs35l32", 0}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, cs35l32_id); + +static struct i2c_driver cs35l32_i2c_driver = { + .driver = { + .name = "cs35l32", + .owner = THIS_MODULE, + .pm = &cs35l32_runtime_pm, + .of_match_table = cs35l32_of_match, + }, + .id_table = cs35l32_id, + .probe = cs35l32_i2c_probe, + .remove = cs35l32_i2c_remove, +}; + +module_i2c_driver(cs35l32_i2c_driver); + +MODULE_DESCRIPTION("ASoC CS35L32 driver"); +MODULE_AUTHOR("Brian Austin, Cirrus Logic Inc, <brian.austin@cirrus.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs35l32.h b/sound/soc/codecs/cs35l32.h new file mode 100644 index 000000000000..31ab804a22bc --- /dev/null +++ b/sound/soc/codecs/cs35l32.h @@ -0,0 +1,93 @@ +/* + * cs35l32.h -- CS35L32 ALSA SoC audio driver + * + * Copyright 2014 CirrusLogic, Inc. + * + * Author: Brian Austin <brian.austin@cirrus.com> + * + * 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. + * + */ + +#ifndef __CS35L32_H__ +#define __CS35L32_H__ + +struct cs35l32_platform_data { + /* Low Battery Threshold */ + unsigned int batt_thresh; + /* Low Battery Recovery */ + unsigned int batt_recov; + /* LED Current Management*/ + unsigned int led_mng; + /* Audio Gain w/ LED */ + unsigned int audiogain_mng; + /* Boost Management */ + unsigned int boost_mng; + /* Data CFG for DUAL device */ + unsigned int sdout_datacfg; + /* SDOUT Sharing */ + unsigned int sdout_share; +}; + +#define CS35L32_CHIP_ID 0x00035A32 +#define CS35L32_DEVID_AB 0x01 /* Device ID A & B [RO] */ +#define CS35L32_DEVID_CD 0x02 /* Device ID C & D [RO] */ +#define CS35L32_DEVID_E 0x03 /* Device ID E [RO] */ +#define CS35L32_FAB_ID 0x04 /* Fab ID [RO] */ +#define CS35L32_REV_ID 0x05 /* Revision ID [RO] */ +#define CS35L32_PWRCTL1 0x06 /* Power Ctl 1 */ +#define CS35L32_PWRCTL2 0x07 /* Power Ctl 2 */ +#define CS35L32_CLK_CTL 0x08 /* Clock Ctl */ +#define CS35L32_BATT_THRESHOLD 0x09 /* Low Battery Threshold */ +#define CS35L32_VMON 0x0A /* Voltage Monitor [RO] */ +#define CS35L32_BST_CPCP_CTL 0x0B /* Conv Peak Curr Protection CTL */ +#define CS35L32_IMON_SCALING 0x0C /* IMON Scaling */ +#define CS35L32_AUDIO_LED_MNGR 0x0D /* Audio/LED Pwr Manager */ +#define CS35L32_ADSP_CTL 0x0F /* Serial Port Control */ +#define CS35L32_CLASSD_CTL 0x10 /* Class D Amp CTL */ +#define CS35L32_PROTECT_CTL 0x11 /* Protection Release CTL */ +#define CS35L32_INT_MASK_1 0x12 /* Interrupt Mask 1 */ +#define CS35L32_INT_MASK_2 0x13 /* Interrupt Mask 2 */ +#define CS35L32_INT_MASK_3 0x14 /* Interrupt Mask 3 */ +#define CS35L32_INT_STATUS_1 0x15 /* Interrupt Status 1 [RO] */ +#define CS35L32_INT_STATUS_2 0x16 /* Interrupt Status 2 [RO] */ +#define CS35L32_INT_STATUS_3 0x17 /* Interrupt Status 3 [RO] */ +#define CS35L32_LED_STATUS 0x18 /* LED Lighting Status [RO] */ +#define CS35L32_FLASH_MODE 0x19 /* LED Flash Mode Current */ +#define CS35L32_MOVIE_MODE 0x1A /* LED Movie Mode Current */ +#define CS35L32_FLASH_TIMER 0x1B /* LED Flash Timer */ +#define CS35L32_FLASH_INHIBIT 0x1C /* LED Flash Inhibit Current */ +#define CS35L32_MAX_REGISTER 0x1C + +#define CS35L32_MCLK_DIV2 0x01 +#define CS35L32_MCLK_RATIO 0x01 +#define CS35L32_MCLKDIS 0x80 +#define CS35L32_PDN_ALL 0x01 +#define CS35L32_PDN_AMP 0x80 +#define CS35L32_PDN_BOOST 0x04 +#define CS35L32_PDN_IMON 0x40 +#define CS35L32_PDN_VMON 0x80 +#define CS35L32_PDN_VPMON 0x20 +#define CS35L32_PDN_ADSP 0x08 + +#define CS35L32_MCLK_DIV2_MASK 0x40 +#define CS35L32_MCLK_RATIO_MASK 0x01 +#define CS35L32_MCLK_MASK 0x41 +#define CS35L32_ADSP_MASTER_MASK 0x40 +#define CS35L32_BOOST_MASK 0x03 +#define CS35L32_GAIN_MGR_MASK 0x08 +#define CS35L32_ADSP_SHARE_MASK 0x08 +#define CS35L32_ADSP_DATACFG_MASK 0x30 +#define CS35L32_SDOUT_3ST 0x80 +#define CS35L32_BATT_REC_MASK 0x0E +#define CS35L32_BATT_THRESH_MASK 0x30 + +#define CS35L32_RATES (SNDRV_PCM_RATE_48000) +#define CS35L32_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + + +#endif diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c index 98523209f739..4fdd47d700e3 100644 --- a/sound/soc/codecs/cs4265.c +++ b/sound/soc/codecs/cs4265.c @@ -77,6 +77,7 @@ static bool cs4265_readable_register(struct device *dev, unsigned int reg) case CS4265_INT_MASK: case CS4265_STATUS_MODE_MSB: case CS4265_STATUS_MODE_LSB: + case CS4265_CHIP_ID: return true; default: return false; @@ -458,12 +459,12 @@ static int cs4265_pcm_hw_params(struct snd_pcm_substream *substream, if (params_width(params) == 16) { snd_soc_update_bits(codec, CS4265_DAC_CTL, CS4265_DAC_CTL_DIF, (1 << 5)); - snd_soc_update_bits(codec, CS4265_ADC_CTL, + snd_soc_update_bits(codec, CS4265_SPDIF_CTL2, CS4265_SPDIF_CTL2_DIF, (1 << 7)); } else { snd_soc_update_bits(codec, CS4265_DAC_CTL, CS4265_DAC_CTL_DIF, (3 << 5)); - snd_soc_update_bits(codec, CS4265_ADC_CTL, + snd_soc_update_bits(codec, CS4265_SPDIF_CTL2, CS4265_SPDIF_CTL2_DIF, (1 << 7)); } break; @@ -472,7 +473,7 @@ static int cs4265_pcm_hw_params(struct snd_pcm_substream *substream, CS4265_DAC_CTL_DIF, 0); snd_soc_update_bits(codec, CS4265_ADC_CTL, CS4265_ADC_DIF, 0); - snd_soc_update_bits(codec, CS4265_ADC_CTL, + snd_soc_update_bits(codec, CS4265_SPDIF_CTL2, CS4265_SPDIF_CTL2_DIF, (1 << 6)); break; diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 969167d8b71e..35fbef743fbe 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -176,9 +176,9 @@ static bool cs42l52_volatile_register(struct device *dev, unsigned int reg) case CS42L52_BATT_LEVEL: case CS42L52_SPK_STATUS: case CS42L52_CHARGE_PUMP: - return 1; + return true; default: - return 0; + return false; } } @@ -946,20 +946,6 @@ static struct snd_soc_dai_driver cs42l52_dai = { .ops = &cs42l52_ops, }; -static int cs42l52_suspend(struct snd_soc_codec *codec) -{ - cs42l52_set_bias_level(codec, SND_SOC_BIAS_OFF); - - return 0; -} - -static int cs42l52_resume(struct snd_soc_codec *codec) -{ - cs42l52_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - return 0; -} - static int beep_rates[] = { 261, 522, 585, 667, 706, 774, 889, 1000, 1043, 1200, 1333, 1412, 1600, 1714, 2000, 2182 @@ -1104,8 +1090,6 @@ static int cs42l52_probe(struct snd_soc_codec *codec) cs42l52_init_beep(codec); - cs42l52_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - cs42l52->sysclk = CS42L52_DEFAULT_CLK; cs42l52->config.format = CS42L52_DEFAULT_FORMAT; @@ -1115,7 +1099,6 @@ static int cs42l52_probe(struct snd_soc_codec *codec) static int cs42l52_remove(struct snd_soc_codec *codec) { cs42l52_free_beep(codec); - cs42l52_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; } @@ -1123,9 +1106,8 @@ static int cs42l52_remove(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_dev_cs42l52 = { .probe = cs42l52_probe, .remove = cs42l52_remove, - .suspend = cs42l52_suspend, - .resume = cs42l52_resume, .set_bias_level = cs42l52_set_bias_level, + .suspend_bias_off = true, .dapm_widgets = cs42l52_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(cs42l52_dapm_widgets), diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c index c766a5a9ce80..2ddc7ac10ad7 100644 --- a/sound/soc/codecs/cs42l56.c +++ b/sound/soc/codecs/cs42l56.c @@ -171,9 +171,9 @@ static bool cs42l56_volatile_register(struct device *dev, unsigned int reg) { switch (reg) { case CS42L56_INT_STATUS: - return 1; + return true; default: - return 0; + return false; } } @@ -1016,20 +1016,6 @@ static struct snd_soc_dai_driver cs42l56_dai = { .ops = &cs42l56_ops, }; -static int cs42l56_suspend(struct snd_soc_codec *codec) -{ - cs42l56_set_bias_level(codec, SND_SOC_BIAS_OFF); - - return 0; -} - -static int cs42l56_resume(struct snd_soc_codec *codec) -{ - cs42l56_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - return 0; -} - static int beep_freq[] = { 261, 522, 585, 667, 706, 774, 889, 1000, 1043, 1200, 1333, 1412, 1600, 1714, 2000, 2182 @@ -1168,18 +1154,12 @@ static int cs42l56_probe(struct snd_soc_codec *codec) { cs42l56_init_beep(codec); - cs42l56_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; } static int cs42l56_remove(struct snd_soc_codec *codec) { - struct cs42l56_private *cs42l56 = snd_soc_codec_get_drvdata(codec); - cs42l56_free_beep(codec); - cs42l56_set_bias_level(codec, SND_SOC_BIAS_OFF); - regulator_bulk_free(ARRAY_SIZE(cs42l56->supplies), cs42l56->supplies); return 0; } @@ -1187,9 +1167,8 @@ static int cs42l56_remove(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_dev_cs42l56 = { .probe = cs42l56_probe, .remove = cs42l56_remove, - .suspend = cs42l56_suspend, - .resume = cs42l56_resume, .set_bias_level = cs42l56_set_bias_level, + .suspend_bias_off = true, .dapm_widgets = cs42l56_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(cs42l56_dapm_widgets), diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 0e7b9eb2ba61..2f8b94683e83 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -1330,25 +1330,10 @@ static struct snd_soc_dai_driver cs42l73_dai[] = { } }; -static int cs42l73_suspend(struct snd_soc_codec *codec) -{ - cs42l73_set_bias_level(codec, SND_SOC_BIAS_OFF); - - return 0; -} - -static int cs42l73_resume(struct snd_soc_codec *codec) -{ - cs42l73_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; -} - static int cs42l73_probe(struct snd_soc_codec *codec) { struct cs42l73_private *cs42l73 = snd_soc_codec_get_drvdata(codec); - cs42l73_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - /* Set Charge Pump Frequency */ if (cs42l73->pdata.chgfreq) snd_soc_update_bits(codec, CS42L73_CPFCHC, @@ -1362,18 +1347,10 @@ static int cs42l73_probe(struct snd_soc_codec *codec) return 0; } -static int cs42l73_remove(struct snd_soc_codec *codec) -{ - cs42l73_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - static struct snd_soc_codec_driver soc_codec_dev_cs42l73 = { .probe = cs42l73_probe, - .remove = cs42l73_remove, - .suspend = cs42l73_suspend, - .resume = cs42l73_resume, .set_bias_level = cs42l73_set_bias_level, + .suspend_bias_off = true, .dapm_widgets = cs42l73_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(cs42l73_dapm_widgets), diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c index 2fae31cb0067..61b2f9a2eef1 100644 --- a/sound/soc/codecs/da732x.c +++ b/sound/soc/codecs/da732x.c @@ -35,7 +35,6 @@ struct da732x_priv { struct regmap *regmap; - struct snd_soc_codec *codec; unsigned int sysclk; bool pll_en; @@ -217,7 +216,7 @@ static void da732x_set_charge_pump(struct snd_soc_codec *codec, int state) snd_soc_write(codec, DA732X_REG_CP_CTRL1, DA723X_CP_DIS); break; default: - pr_err(KERN_ERR "Wrong charge pump state\n"); + pr_err("Wrong charge pump state\n"); break; } } @@ -1508,31 +1507,7 @@ static int da732x_set_bias_level(struct snd_soc_codec *codec, return 0; } -static int da732x_probe(struct snd_soc_codec *codec) -{ - struct da732x_priv *da732x = snd_soc_codec_get_drvdata(codec); - struct snd_soc_dapm_context *dapm = &codec->dapm; - - da732x->codec = codec; - - dapm->idle_bias_off = false; - - da732x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - return 0; -} - -static int da732x_remove(struct snd_soc_codec *codec) -{ - - da732x_set_bias_level(codec, SND_SOC_BIAS_OFF); - - return 0; -} - static struct snd_soc_codec_driver soc_codec_dev_da732x = { - .probe = da732x_probe, - .remove = da732x_remove, .set_bias_level = da732x_set_bias_level, .controls = da732x_snd_controls, .num_controls = ARRAY_SIZE(da732x_snd_controls), diff --git a/sound/soc/codecs/es8328-i2c.c b/sound/soc/codecs/es8328-i2c.c new file mode 100644 index 000000000000..aae410d122ee --- /dev/null +++ b/sound/soc/codecs/es8328-i2c.c @@ -0,0 +1,60 @@ +/* + * es8328-i2c.c -- ES8328 ALSA SoC I2C Audio driver + * + * Copyright 2014 Sutajio Ko-Usagi PTE LTD + * + * Author: Sean Cross <xobs@kosagi.com> + * + * 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/module.h> +#include <linux/i2c.h> +#include <linux/regmap.h> + +#include <sound/soc.h> + +#include "es8328.h" + +static const struct i2c_device_id es8328_id[] = { + { "everest,es8328", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, es8328_id); + +static const struct of_device_id es8328_of_match[] = { + { .compatible = "everest,es8328", }, + { } +}; +MODULE_DEVICE_TABLE(of, es8328_of_match); + +static int es8328_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + return es8328_probe(&i2c->dev, + devm_regmap_init_i2c(i2c, &es8328_regmap_config)); +} + +static int es8328_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_codec(&i2c->dev); + return 0; +} + +static struct i2c_driver es8328_i2c_driver = { + .driver = { + .name = "es8328", + .of_match_table = es8328_of_match, + }, + .probe = es8328_i2c_probe, + .remove = es8328_i2c_remove, + .id_table = es8328_id, +}; + +module_i2c_driver(es8328_i2c_driver); + +MODULE_DESCRIPTION("ASoC ES8328 audio CODEC I2C driver"); +MODULE_AUTHOR("Sean Cross <xobs@kosagi.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/es8328-spi.c b/sound/soc/codecs/es8328-spi.c new file mode 100644 index 000000000000..8fbd935e1c76 --- /dev/null +++ b/sound/soc/codecs/es8328-spi.c @@ -0,0 +1,49 @@ +/* + * es8328.c -- ES8328 ALSA SoC SPI Audio driver + * + * Copyright 2014 Sutajio Ko-Usagi PTE LTD + * + * Author: Sean Cross <xobs@kosagi.com> + * + * 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/module.h> +#include <linux/regmap.h> +#include <linux/spi/spi.h> +#include <sound/soc.h> +#include "es8328.h" + +static const struct of_device_id es8328_of_match[] = { + { .compatible = "everest,es8328", }, + { } +}; +MODULE_DEVICE_TABLE(of, es8328_of_match); + +static int es8328_spi_probe(struct spi_device *spi) +{ + return es8328_probe(&spi->dev, + devm_regmap_init_spi(spi, &es8328_regmap_config)); +} + +static int es8328_spi_remove(struct spi_device *spi) +{ + snd_soc_unregister_codec(&spi->dev); + return 0; +} + +static struct spi_driver es8328_spi_driver = { + .driver = { + .name = "es8328", + .of_match_table = es8328_of_match, + }, + .probe = es8328_spi_probe, + .remove = es8328_spi_remove, +}; + +module_spi_driver(es8328_spi_driver); +MODULE_DESCRIPTION("ASoC ES8328 audio CODEC SPI driver"); +MODULE_AUTHOR("Sean Cross <xobs@kosagi.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c new file mode 100644 index 000000000000..f27325155ace --- /dev/null +++ b/sound/soc/codecs/es8328.c @@ -0,0 +1,756 @@ +/* + * es8328.c -- ES8328 ALSA SoC Audio driver + * + * Copyright 2014 Sutajio Ko-Usagi PTE LTD + * + * Author: Sean Cross <xobs@kosagi.com> + * + * 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/clk.h> +#include <linux/delay.h> +#include <linux/of_device.h> +#include <linux/module.h> +#include <linux/pm.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <linux/regulator/consumer.h> +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include "es8328.h" + +#define ES8328_SYSCLK_RATE_1X 11289600 +#define ES8328_SYSCLK_RATE_2X 22579200 + +/* Run the codec at 22.5792 or 11.2896 MHz to support these rates */ +static struct { + int rate; + u8 ratio; +} mclk_ratios[] = { + { 8000, 9 }, + {11025, 7 }, + {22050, 4 }, + {44100, 2 }, +}; + +/* regulator supplies for sgtl5000, VDDD is an optional external supply */ +enum sgtl5000_regulator_supplies { + DVDD, + AVDD, + PVDD, + HPVDD, + ES8328_SUPPLY_NUM +}; + +/* vddd is optional supply */ +static const char * const supply_names[ES8328_SUPPLY_NUM] = { + "DVDD", + "AVDD", + "PVDD", + "HPVDD", +}; + +#define ES8328_RATES (SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_22050 | \ + SNDRV_PCM_RATE_11025) +#define ES8328_FORMATS (SNDRV_PCM_FMTBIT_S16_LE) + +struct es8328_priv { + struct regmap *regmap; + struct clk *clk; + int playback_fs; + bool deemph; + struct regulator_bulk_data supplies[ES8328_SUPPLY_NUM]; +}; + +/* + * ES8328 Controls + */ + +static const char * const adcpol_txt[] = {"Normal", "L Invert", "R Invert", + "L + R Invert"}; +static SOC_ENUM_SINGLE_DECL(adcpol, + ES8328_ADCCONTROL6, 6, adcpol_txt); + +static const DECLARE_TLV_DB_SCALE(play_tlv, -3000, 100, 0); +static const DECLARE_TLV_DB_SCALE(dac_adc_tlv, -9600, 50, 0); +static const DECLARE_TLV_DB_SCALE(pga_tlv, 0, 300, 0); +static const DECLARE_TLV_DB_SCALE(bypass_tlv, -1500, 300, 0); +static const DECLARE_TLV_DB_SCALE(mic_tlv, 0, 300, 0); + +static const int deemph_settings[] = { 0, 32000, 44100, 48000 }; + +static int es8328_set_deemph(struct snd_soc_codec *codec) +{ + struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec); + int val, i, best; + + /* + * If we're using deemphasis select the nearest available sample + * rate. + */ + if (es8328->deemph) { + best = 1; + for (i = 2; i < ARRAY_SIZE(deemph_settings); i++) { + if (abs(deemph_settings[i] - es8328->playback_fs) < + abs(deemph_settings[best] - es8328->playback_fs)) + best = i; + } + + val = best << 1; + } else { + val = 0; + } + + dev_dbg(codec->dev, "Set deemphasis %d\n", val); + + return snd_soc_update_bits(codec, ES8328_DACCONTROL6, 0x6, val); +} + +static int es8328_get_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.enumerated.item[0] = es8328->deemph; + return 0; +} + +static int es8328_put_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec); + int deemph = ucontrol->value.enumerated.item[0]; + int ret; + + if (deemph > 1) + return -EINVAL; + + ret = es8328_set_deemph(codec); + if (ret < 0) + return ret; + + es8328->deemph = deemph; + + return 0; +} + + + +static const struct snd_kcontrol_new es8328_snd_controls[] = { + SOC_DOUBLE_R_TLV("Capture Digital Volume", + ES8328_ADCCONTROL8, ES8328_ADCCONTROL9, + 0, 0xc0, 1, dac_adc_tlv), + SOC_SINGLE("Capture ZC Switch", ES8328_ADCCONTROL7, 6, 1, 0), + + SOC_SINGLE_BOOL_EXT("DAC Deemphasis Switch", 0, + es8328_get_deemph, es8328_put_deemph), + + SOC_ENUM("Capture Polarity", adcpol), + + SOC_SINGLE_TLV("Left Mixer Left Bypass Volume", + ES8328_DACCONTROL17, 3, 7, 1, bypass_tlv), + SOC_SINGLE_TLV("Left Mixer Right Bypass Volume", + ES8328_DACCONTROL19, 3, 7, 1, bypass_tlv), + SOC_SINGLE_TLV("Right Mixer Left Bypass Volume", + ES8328_DACCONTROL18, 3, 7, 1, bypass_tlv), + SOC_SINGLE_TLV("Right Mixer Right Bypass Volume", + ES8328_DACCONTROL20, 3, 7, 1, bypass_tlv), + + SOC_DOUBLE_R_TLV("PCM Volume", + ES8328_LDACVOL, ES8328_RDACVOL, + 0, ES8328_DACVOL_MAX, 1, dac_adc_tlv), + + SOC_DOUBLE_R_TLV("Output 1 Playback Volume", + ES8328_LOUT1VOL, ES8328_ROUT1VOL, + 0, ES8328_OUT1VOL_MAX, 0, play_tlv), + + SOC_DOUBLE_R_TLV("Output 2 Playback Volume", + ES8328_LOUT2VOL, ES8328_ROUT2VOL, + 0, ES8328_OUT2VOL_MAX, 0, play_tlv), + + SOC_DOUBLE_TLV("Mic PGA Volume", ES8328_ADCCONTROL1, + 4, 0, 8, 0, mic_tlv), +}; + +/* + * DAPM Controls + */ + +static const char * const es8328_line_texts[] = { + "Line 1", "Line 2", "PGA", "Differential"}; + +static const struct soc_enum es8328_lline_enum = + SOC_ENUM_SINGLE(ES8328_DACCONTROL16, 3, + ARRAY_SIZE(es8328_line_texts), + es8328_line_texts); +static const struct snd_kcontrol_new es8328_left_line_controls = + SOC_DAPM_ENUM("Route", es8328_lline_enum); + +static const struct soc_enum es8328_rline_enum = + SOC_ENUM_SINGLE(ES8328_DACCONTROL16, 0, + ARRAY_SIZE(es8328_line_texts), + es8328_line_texts); +static const struct snd_kcontrol_new es8328_right_line_controls = + SOC_DAPM_ENUM("Route", es8328_lline_enum); + +/* Left Mixer */ +static const struct snd_kcontrol_new es8328_left_mixer_controls[] = { + SOC_DAPM_SINGLE("Playback Switch", ES8328_DACCONTROL17, 8, 1, 0), + SOC_DAPM_SINGLE("Left Bypass Switch", ES8328_DACCONTROL17, 7, 1, 0), + SOC_DAPM_SINGLE("Right Playback Switch", ES8328_DACCONTROL18, 8, 1, 0), + SOC_DAPM_SINGLE("Right Bypass Switch", ES8328_DACCONTROL18, 7, 1, 0), +}; + +/* Right Mixer */ +static const struct snd_kcontrol_new es8328_right_mixer_controls[] = { + SOC_DAPM_SINGLE("Left Playback Switch", ES8328_DACCONTROL19, 8, 1, 0), + SOC_DAPM_SINGLE("Left Bypass Switch", ES8328_DACCONTROL19, 7, 1, 0), + SOC_DAPM_SINGLE("Playback Switch", ES8328_DACCONTROL20, 8, 1, 0), + SOC_DAPM_SINGLE("Right Bypass Switch", ES8328_DACCONTROL20, 7, 1, 0), +}; + +static const char * const es8328_pga_sel[] = { + "Line 1", "Line 2", "Line 3", "Differential"}; + +/* Left PGA Mux */ +static const struct soc_enum es8328_lpga_enum = + SOC_ENUM_SINGLE(ES8328_ADCCONTROL2, 6, + ARRAY_SIZE(es8328_pga_sel), + es8328_pga_sel); +static const struct snd_kcontrol_new es8328_left_pga_controls = + SOC_DAPM_ENUM("Route", es8328_lpga_enum); + +/* Right PGA Mux */ +static const struct soc_enum es8328_rpga_enum = + SOC_ENUM_SINGLE(ES8328_ADCCONTROL2, 4, + ARRAY_SIZE(es8328_pga_sel), + es8328_pga_sel); +static const struct snd_kcontrol_new es8328_right_pga_controls = + SOC_DAPM_ENUM("Route", es8328_rpga_enum); + +/* Differential Mux */ +static const char * const es8328_diff_sel[] = {"Line 1", "Line 2"}; +static SOC_ENUM_SINGLE_DECL(diffmux, + ES8328_ADCCONTROL3, 7, es8328_diff_sel); +static const struct snd_kcontrol_new es8328_diffmux_controls = + SOC_DAPM_ENUM("Route", diffmux); + +/* Mono ADC Mux */ +static const char * const es8328_mono_mux[] = {"Stereo", "Mono (Left)", + "Mono (Right)", "Digital Mono"}; +static SOC_ENUM_SINGLE_DECL(monomux, + ES8328_ADCCONTROL3, 3, es8328_mono_mux); +static const struct snd_kcontrol_new es8328_monomux_controls = + SOC_DAPM_ENUM("Route", monomux); + +static const struct snd_soc_dapm_widget es8328_dapm_widgets[] = { + SND_SOC_DAPM_MUX("Differential Mux", SND_SOC_NOPM, 0, 0, + &es8328_diffmux_controls), + SND_SOC_DAPM_MUX("Left ADC Mux", SND_SOC_NOPM, 0, 0, + &es8328_monomux_controls), + SND_SOC_DAPM_MUX("Right ADC Mux", SND_SOC_NOPM, 0, 0, + &es8328_monomux_controls), + + SND_SOC_DAPM_MUX("Left PGA Mux", ES8328_ADCPOWER, + ES8328_ADCPOWER_AINL_OFF, 1, + &es8328_left_pga_controls), + SND_SOC_DAPM_MUX("Right PGA Mux", ES8328_ADCPOWER, + ES8328_ADCPOWER_AINR_OFF, 1, + &es8328_right_pga_controls), + + SND_SOC_DAPM_MUX("Left Line Mux", SND_SOC_NOPM, 0, 0, + &es8328_left_line_controls), + SND_SOC_DAPM_MUX("Right Line Mux", SND_SOC_NOPM, 0, 0, + &es8328_right_line_controls), + + SND_SOC_DAPM_ADC("Right ADC", "Right Capture", ES8328_ADCPOWER, + ES8328_ADCPOWER_ADCR_OFF, 1), + SND_SOC_DAPM_ADC("Left ADC", "Left Capture", ES8328_ADCPOWER, + ES8328_ADCPOWER_ADCL_OFF, 1), + + SND_SOC_DAPM_SUPPLY("Mic Bias", ES8328_ADCPOWER, + ES8328_ADCPOWER_MIC_BIAS_OFF, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Mic Bias Gen", ES8328_ADCPOWER, + ES8328_ADCPOWER_ADC_BIAS_GEN_OFF, 1, NULL, 0), + + SND_SOC_DAPM_SUPPLY("DAC STM", ES8328_CHIPPOWER, + ES8328_CHIPPOWER_DACSTM_RESET, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC STM", ES8328_CHIPPOWER, + ES8328_CHIPPOWER_ADCSTM_RESET, 1, NULL, 0), + + SND_SOC_DAPM_SUPPLY("DAC DIG", ES8328_CHIPPOWER, + ES8328_CHIPPOWER_DACDIG_OFF, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC DIG", ES8328_CHIPPOWER, + ES8328_CHIPPOWER_ADCDIG_OFF, 1, NULL, 0), + + SND_SOC_DAPM_SUPPLY("DAC DLL", ES8328_CHIPPOWER, + ES8328_CHIPPOWER_DACDLL_OFF, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC DLL", ES8328_CHIPPOWER, + ES8328_CHIPPOWER_ADCDLL_OFF, 1, NULL, 0), + + SND_SOC_DAPM_SUPPLY("ADC Vref", ES8328_CHIPPOWER, + ES8328_CHIPPOWER_ADCVREF_OFF, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC Vref", ES8328_CHIPPOWER, + ES8328_CHIPPOWER_DACVREF_OFF, 1, NULL, 0), + + SND_SOC_DAPM_DAC("Right DAC", "Right Playback", ES8328_DACPOWER, + ES8328_DACPOWER_RDAC_OFF, 1), + SND_SOC_DAPM_DAC("Left DAC", "Left Playback", ES8328_DACPOWER, + ES8328_DACPOWER_LDAC_OFF, 1), + + SND_SOC_DAPM_MIXER("Left Mixer", SND_SOC_NOPM, 0, 0, + &es8328_left_mixer_controls[0], + ARRAY_SIZE(es8328_left_mixer_controls)), + SND_SOC_DAPM_MIXER("Right Mixer", SND_SOC_NOPM, 0, 0, + &es8328_right_mixer_controls[0], + ARRAY_SIZE(es8328_right_mixer_controls)), + + SND_SOC_DAPM_PGA("Right Out 2", ES8328_DACPOWER, + ES8328_DACPOWER_ROUT2_ON, 0, NULL, 0), + SND_SOC_DAPM_PGA("Left Out 2", ES8328_DACPOWER, + ES8328_DACPOWER_LOUT2_ON, 0, NULL, 0), + SND_SOC_DAPM_PGA("Right Out 1", ES8328_DACPOWER, + ES8328_DACPOWER_ROUT1_ON, 0, NULL, 0), + SND_SOC_DAPM_PGA("Left Out 1", ES8328_DACPOWER, + ES8328_DACPOWER_LOUT1_ON, 0, NULL, 0), + + SND_SOC_DAPM_OUTPUT("LOUT1"), + SND_SOC_DAPM_OUTPUT("ROUT1"), + SND_SOC_DAPM_OUTPUT("LOUT2"), + SND_SOC_DAPM_OUTPUT("ROUT2"), + + SND_SOC_DAPM_INPUT("LINPUT1"), + SND_SOC_DAPM_INPUT("LINPUT2"), + SND_SOC_DAPM_INPUT("RINPUT1"), + SND_SOC_DAPM_INPUT("RINPUT2"), +}; + +static const struct snd_soc_dapm_route es8328_dapm_routes[] = { + + { "Left Line Mux", "Line 1", "LINPUT1" }, + { "Left Line Mux", "Line 2", "LINPUT2" }, + { "Left Line Mux", "PGA", "Left PGA Mux" }, + { "Left Line Mux", "Differential", "Differential Mux" }, + + { "Right Line Mux", "Line 1", "RINPUT1" }, + { "Right Line Mux", "Line 2", "RINPUT2" }, + { "Right Line Mux", "PGA", "Right PGA Mux" }, + { "Right Line Mux", "Differential", "Differential Mux" }, + + { "Left PGA Mux", "Line 1", "LINPUT1" }, + { "Left PGA Mux", "Line 2", "LINPUT2" }, + { "Left PGA Mux", "Differential", "Differential Mux" }, + + { "Right PGA Mux", "Line 1", "RINPUT1" }, + { "Right PGA Mux", "Line 2", "RINPUT2" }, + { "Right PGA Mux", "Differential", "Differential Mux" }, + + { "Differential Mux", "Line 1", "LINPUT1" }, + { "Differential Mux", "Line 1", "RINPUT1" }, + { "Differential Mux", "Line 2", "LINPUT2" }, + { "Differential Mux", "Line 2", "RINPUT2" }, + + { "Left ADC Mux", "Stereo", "Left PGA Mux" }, + { "Left ADC Mux", "Mono (Left)", "Left PGA Mux" }, + { "Left ADC Mux", "Digital Mono", "Left PGA Mux" }, + + { "Right ADC Mux", "Stereo", "Right PGA Mux" }, + { "Right ADC Mux", "Mono (Right)", "Right PGA Mux" }, + { "Right ADC Mux", "Digital Mono", "Right PGA Mux" }, + + { "Left ADC", NULL, "Left ADC Mux" }, + { "Right ADC", NULL, "Right ADC Mux" }, + + { "ADC DIG", NULL, "ADC STM" }, + { "ADC DIG", NULL, "ADC Vref" }, + { "ADC DIG", NULL, "ADC DLL" }, + + { "Left ADC", NULL, "ADC DIG" }, + { "Right ADC", NULL, "ADC DIG" }, + + { "Mic Bias", NULL, "Mic Bias Gen" }, + + { "Left Line Mux", "Line 1", "LINPUT1" }, + { "Left Line Mux", "Line 2", "LINPUT2" }, + { "Left Line Mux", "PGA", "Left PGA Mux" }, + { "Left Line Mux", "Differential", "Differential Mux" }, + + { "Right Line Mux", "Line 1", "RINPUT1" }, + { "Right Line Mux", "Line 2", "RINPUT2" }, + { "Right Line Mux", "PGA", "Right PGA Mux" }, + { "Right Line Mux", "Differential", "Differential Mux" }, + + { "Left Out 1", NULL, "Left DAC" }, + { "Right Out 1", NULL, "Right DAC" }, + { "Left Out 2", NULL, "Left DAC" }, + { "Right Out 2", NULL, "Right DAC" }, + + { "Left Mixer", "Playback Switch", "Left DAC" }, + { "Left Mixer", "Left Bypass Switch", "Left Line Mux" }, + { "Left Mixer", "Right Playback Switch", "Right DAC" }, + { "Left Mixer", "Right Bypass Switch", "Right Line Mux" }, + + { "Right Mixer", "Left Playback Switch", "Left DAC" }, + { "Right Mixer", "Left Bypass Switch", "Left Line Mux" }, + { "Right Mixer", "Playback Switch", "Right DAC" }, + { "Right Mixer", "Right Bypass Switch", "Right Line Mux" }, + + { "DAC DIG", NULL, "DAC STM" }, + { "DAC DIG", NULL, "DAC Vref" }, + { "DAC DIG", NULL, "DAC DLL" }, + + { "Left DAC", NULL, "DAC DIG" }, + { "Right DAC", NULL, "DAC DIG" }, + + { "Left Out 1", NULL, "Left Mixer" }, + { "LOUT1", NULL, "Left Out 1" }, + { "Right Out 1", NULL, "Right Mixer" }, + { "ROUT1", NULL, "Right Out 1" }, + + { "Left Out 2", NULL, "Left Mixer" }, + { "LOUT2", NULL, "Left Out 2" }, + { "Right Out 2", NULL, "Right Mixer" }, + { "ROUT2", NULL, "Right Out 2" }, +}; + +static int es8328_mute(struct snd_soc_dai *dai, int mute) +{ + return snd_soc_update_bits(dai->codec, ES8328_DACCONTROL3, + ES8328_DACCONTROL3_DACMUTE, + mute ? ES8328_DACCONTROL3_DACMUTE : 0); +} + +static int es8328_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec); + int clk_rate; + int i; + int reg; + u8 ratio; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + reg = ES8328_DACCONTROL2; + else + reg = ES8328_ADCCONTROL5; + + clk_rate = clk_get_rate(es8328->clk); + + if ((clk_rate != ES8328_SYSCLK_RATE_1X) && + (clk_rate != ES8328_SYSCLK_RATE_2X)) { + dev_err(codec->dev, + "%s: clock is running at %d Hz, not %d or %d Hz\n", + __func__, clk_rate, + ES8328_SYSCLK_RATE_1X, ES8328_SYSCLK_RATE_2X); + return -EINVAL; + } + + /* find master mode MCLK to sampling frequency ratio */ + ratio = mclk_ratios[0].rate; + for (i = 1; i < ARRAY_SIZE(mclk_ratios); i++) + if (params_rate(params) <= mclk_ratios[i].rate) + ratio = mclk_ratios[i].ratio; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + es8328->playback_fs = params_rate(params); + es8328_set_deemph(codec); + } + + return snd_soc_update_bits(codec, reg, ES8328_RATEMASK, ratio); +} + +static int es8328_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec); + int clk_rate; + u8 mode = ES8328_DACCONTROL1_DACWL_16; + + /* set master/slave audio interface */ + if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBM_CFM) + return -EINVAL; + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + mode |= ES8328_DACCONTROL1_DACFORMAT_I2S; + break; + case SND_SOC_DAIFMT_RIGHT_J: + mode |= ES8328_DACCONTROL1_DACFORMAT_RJUST; + break; + case SND_SOC_DAIFMT_LEFT_J: + mode |= ES8328_DACCONTROL1_DACFORMAT_LJUST; + break; + default: + return -EINVAL; + } + + /* clock inversion */ + if ((fmt & SND_SOC_DAIFMT_INV_MASK) != SND_SOC_DAIFMT_NB_NF) + return -EINVAL; + + snd_soc_write(codec, ES8328_DACCONTROL1, mode); + snd_soc_write(codec, ES8328_ADCCONTROL4, mode); + + /* Master serial port mode, with BCLK generated automatically */ + clk_rate = clk_get_rate(es8328->clk); + if (clk_rate == ES8328_SYSCLK_RATE_1X) + snd_soc_write(codec, ES8328_MASTERMODE, + ES8328_MASTERMODE_MSC); + else + snd_soc_write(codec, ES8328_MASTERMODE, + ES8328_MASTERMODE_MCLKDIV2 | + ES8328_MASTERMODE_MSC); + + return 0; +} + +static int es8328_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + /* VREF, VMID=2x50k, digital enabled */ + snd_soc_write(codec, ES8328_CHIPPOWER, 0); + snd_soc_update_bits(codec, ES8328_CONTROL1, + ES8328_CONTROL1_VMIDSEL_MASK | + ES8328_CONTROL1_ENREF, + ES8328_CONTROL1_VMIDSEL_50k | + ES8328_CONTROL1_ENREF); + break; + + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + snd_soc_update_bits(codec, ES8328_CONTROL1, + ES8328_CONTROL1_VMIDSEL_MASK | + ES8328_CONTROL1_ENREF, + ES8328_CONTROL1_VMIDSEL_5k | + ES8328_CONTROL1_ENREF); + + /* Charge caps */ + msleep(100); + } + + snd_soc_write(codec, ES8328_CONTROL2, + ES8328_CONTROL2_OVERCURRENT_ON | + ES8328_CONTROL2_THERMAL_SHUTDOWN_ON); + + /* VREF, VMID=2*500k, digital stopped */ + snd_soc_update_bits(codec, ES8328_CONTROL1, + ES8328_CONTROL1_VMIDSEL_MASK | + ES8328_CONTROL1_ENREF, + ES8328_CONTROL1_VMIDSEL_500k | + ES8328_CONTROL1_ENREF); + break; + + case SND_SOC_BIAS_OFF: + snd_soc_update_bits(codec, ES8328_CONTROL1, + ES8328_CONTROL1_VMIDSEL_MASK | + ES8328_CONTROL1_ENREF, + 0); + break; + } + codec->dapm.bias_level = level; + return 0; +} + +static const struct snd_soc_dai_ops es8328_dai_ops = { + .hw_params = es8328_hw_params, + .digital_mute = es8328_mute, + .set_fmt = es8328_set_dai_fmt, +}; + +static struct snd_soc_dai_driver es8328_dai = { + .name = "es8328-hifi-analog", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = ES8328_RATES, + .formats = ES8328_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = ES8328_RATES, + .formats = ES8328_FORMATS, + }, + .ops = &es8328_dai_ops, +}; + +static int es8328_suspend(struct snd_soc_codec *codec) +{ + struct es8328_priv *es8328; + int ret; + + es8328 = snd_soc_codec_get_drvdata(codec); + + clk_disable_unprepare(es8328->clk); + + ret = regulator_bulk_disable(ARRAY_SIZE(es8328->supplies), + es8328->supplies); + if (ret) { + dev_err(codec->dev, "unable to disable regulators\n"); + return ret; + } + return 0; +} + +static int es8328_resume(struct snd_soc_codec *codec) +{ + struct regmap *regmap = dev_get_regmap(codec->dev, NULL); + struct es8328_priv *es8328; + int ret; + + es8328 = snd_soc_codec_get_drvdata(codec); + + ret = clk_prepare_enable(es8328->clk); + if (ret) { + dev_err(codec->dev, "unable to enable clock\n"); + return ret; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(es8328->supplies), + es8328->supplies); + if (ret) { + dev_err(codec->dev, "unable to enable regulators\n"); + return ret; + } + + regcache_mark_dirty(regmap); + ret = regcache_sync(regmap); + if (ret) { + dev_err(codec->dev, "unable to sync regcache\n"); + return ret; + } + + return 0; +} + +static int es8328_codec_probe(struct snd_soc_codec *codec) +{ + struct es8328_priv *es8328; + int ret; + + es8328 = snd_soc_codec_get_drvdata(codec); + + ret = regulator_bulk_enable(ARRAY_SIZE(es8328->supplies), + es8328->supplies); + if (ret) { + dev_err(codec->dev, "unable to enable regulators\n"); + return ret; + } + + /* Setup clocks */ + es8328->clk = devm_clk_get(codec->dev, NULL); + if (IS_ERR(es8328->clk)) { + dev_err(codec->dev, "codec clock missing or invalid\n"); + ret = PTR_ERR(es8328->clk); + goto clk_fail; + } + + ret = clk_prepare_enable(es8328->clk); + if (ret) { + dev_err(codec->dev, "unable to prepare codec clk\n"); + goto clk_fail; + } + + return 0; + +clk_fail: + regulator_bulk_disable(ARRAY_SIZE(es8328->supplies), + es8328->supplies); + return ret; +} + +static int es8328_remove(struct snd_soc_codec *codec) +{ + struct es8328_priv *es8328; + + es8328 = snd_soc_codec_get_drvdata(codec); + + if (es8328->clk) + clk_disable_unprepare(es8328->clk); + + regulator_bulk_disable(ARRAY_SIZE(es8328->supplies), + es8328->supplies); + + return 0; +} + +const struct regmap_config es8328_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + .max_register = ES8328_REG_MAX, + .cache_type = REGCACHE_RBTREE, +}; +EXPORT_SYMBOL_GPL(es8328_regmap_config); + +static struct snd_soc_codec_driver es8328_codec_driver = { + .probe = es8328_codec_probe, + .suspend = es8328_suspend, + .resume = es8328_resume, + .remove = es8328_remove, + .set_bias_level = es8328_set_bias_level, + .suspend_bias_off = true, + + .controls = es8328_snd_controls, + .num_controls = ARRAY_SIZE(es8328_snd_controls), + .dapm_widgets = es8328_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(es8328_dapm_widgets), + .dapm_routes = es8328_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(es8328_dapm_routes), +}; + +int es8328_probe(struct device *dev, struct regmap *regmap) +{ + struct es8328_priv *es8328; + int ret; + int i; + + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + + es8328 = devm_kzalloc(dev, sizeof(*es8328), GFP_KERNEL); + if (es8328 == NULL) + return -ENOMEM; + + es8328->regmap = regmap; + + for (i = 0; i < ARRAY_SIZE(es8328->supplies); i++) + es8328->supplies[i].supply = supply_names[i]; + + ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(es8328->supplies), + es8328->supplies); + if (ret) { + dev_err(dev, "unable to get regulators\n"); + return ret; + } + + dev_set_drvdata(dev, es8328); + + return snd_soc_register_codec(dev, + &es8328_codec_driver, &es8328_dai, 1); +} +EXPORT_SYMBOL_GPL(es8328_probe); + +MODULE_DESCRIPTION("ASoC ES8328 driver"); +MODULE_AUTHOR("Sean Cross <xobs@kosagi.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/es8328.h b/sound/soc/codecs/es8328.h new file mode 100644 index 000000000000..cb36afe10c0e --- /dev/null +++ b/sound/soc/codecs/es8328.h @@ -0,0 +1,314 @@ +/* + * es8328.h -- ES8328 ALSA SoC Audio driver + */ + +#ifndef _ES8328_H +#define _ES8328_H + +#include <linux/regmap.h> + +struct device; + +extern const struct regmap_config es8328_regmap_config; +int es8328_probe(struct device *dev, struct regmap *regmap); + +#define ES8328_DACLVOL 46 +#define ES8328_DACRVOL 47 +#define ES8328_DACCTL 28 +#define ES8328_RATEMASK (0x1f << 0) + +#define ES8328_CONTROL1 0x00 +#define ES8328_CONTROL1_VMIDSEL_OFF (0 << 0) +#define ES8328_CONTROL1_VMIDSEL_50k (1 << 0) +#define ES8328_CONTROL1_VMIDSEL_500k (2 << 0) +#define ES8328_CONTROL1_VMIDSEL_5k (3 << 0) +#define ES8328_CONTROL1_VMIDSEL_MASK (7 << 0) +#define ES8328_CONTROL1_ENREF (1 << 2) +#define ES8328_CONTROL1_SEQEN (1 << 3) +#define ES8328_CONTROL1_SAMEFS (1 << 4) +#define ES8328_CONTROL1_DACMCLK_ADC (0 << 5) +#define ES8328_CONTROL1_DACMCLK_DAC (1 << 5) +#define ES8328_CONTROL1_LRCM (1 << 6) +#define ES8328_CONTROL1_SCP_RESET (1 << 7) + +#define ES8328_CONTROL2 0x01 +#define ES8328_CONTROL2_VREF_BUF_OFF (1 << 0) +#define ES8328_CONTROL2_VREF_LOWPOWER (1 << 1) +#define ES8328_CONTROL2_IBIASGEN_OFF (1 << 2) +#define ES8328_CONTROL2_ANALOG_OFF (1 << 3) +#define ES8328_CONTROL2_VREF_BUF_LOWPOWER (1 << 4) +#define ES8328_CONTROL2_VCM_MOD_LOWPOWER (1 << 5) +#define ES8328_CONTROL2_OVERCURRENT_ON (1 << 6) +#define ES8328_CONTROL2_THERMAL_SHUTDOWN_ON (1 << 7) + +#define ES8328_CHIPPOWER 0x02 +#define ES8328_CHIPPOWER_DACVREF_OFF 0 +#define ES8328_CHIPPOWER_ADCVREF_OFF 1 +#define ES8328_CHIPPOWER_DACDLL_OFF 2 +#define ES8328_CHIPPOWER_ADCDLL_OFF 3 +#define ES8328_CHIPPOWER_DACSTM_RESET 4 +#define ES8328_CHIPPOWER_ADCSTM_RESET 5 +#define ES8328_CHIPPOWER_DACDIG_OFF 6 +#define ES8328_CHIPPOWER_ADCDIG_OFF 7 + +#define ES8328_ADCPOWER 0x03 +#define ES8328_ADCPOWER_INT1_LOWPOWER 0 +#define ES8328_ADCPOWER_FLASH_ADC_LOWPOWER 1 +#define ES8328_ADCPOWER_ADC_BIAS_GEN_OFF 2 +#define ES8328_ADCPOWER_MIC_BIAS_OFF 3 +#define ES8328_ADCPOWER_ADCR_OFF 4 +#define ES8328_ADCPOWER_ADCL_OFF 5 +#define ES8328_ADCPOWER_AINR_OFF 6 +#define ES8328_ADCPOWER_AINL_OFF 7 + +#define ES8328_DACPOWER 0x04 +#define ES8328_DACPOWER_OUT3_ON 0 +#define ES8328_DACPOWER_MONO_ON 1 +#define ES8328_DACPOWER_ROUT2_ON 2 +#define ES8328_DACPOWER_LOUT2_ON 3 +#define ES8328_DACPOWER_ROUT1_ON 4 +#define ES8328_DACPOWER_LOUT1_ON 5 +#define ES8328_DACPOWER_RDAC_OFF 6 +#define ES8328_DACPOWER_LDAC_OFF 7 + +#define ES8328_CHIPLOPOW1 0x05 +#define ES8328_CHIPLOPOW2 0x06 +#define ES8328_ANAVOLMANAG 0x07 + +#define ES8328_MASTERMODE 0x08 +#define ES8328_MASTERMODE_BCLKDIV (0 << 0) +#define ES8328_MASTERMODE_BCLK_INV (1 << 5) +#define ES8328_MASTERMODE_MCLKDIV2 (1 << 6) +#define ES8328_MASTERMODE_MSC (1 << 7) + +#define ES8328_ADCCONTROL1 0x09 +#define ES8328_ADCCONTROL2 0x0a +#define ES8328_ADCCONTROL3 0x0b +#define ES8328_ADCCONTROL4 0x0c +#define ES8328_ADCCONTROL5 0x0d +#define ES8328_ADCCONTROL5_RATEMASK (0x1f << 0) + +#define ES8328_ADCCONTROL6 0x0e + +#define ES8328_ADCCONTROL7 0x0f +#define ES8328_ADCCONTROL7_ADC_MUTE (1 << 2) +#define ES8328_ADCCONTROL7_ADC_LER (1 << 3) +#define ES8328_ADCCONTROL7_ADC_ZERO_CROSS (1 << 4) +#define ES8328_ADCCONTROL7_ADC_SOFT_RAMP (1 << 5) +#define ES8328_ADCCONTROL7_ADC_RAMP_RATE_4 (0 << 6) +#define ES8328_ADCCONTROL7_ADC_RAMP_RATE_8 (1 << 6) +#define ES8328_ADCCONTROL7_ADC_RAMP_RATE_16 (2 << 6) +#define ES8328_ADCCONTROL7_ADC_RAMP_RATE_32 (3 << 6) + +#define ES8328_ADCCONTROL8 0x10 +#define ES8328_ADCCONTROL9 0x11 +#define ES8328_ADCCONTROL10 0x12 +#define ES8328_ADCCONTROL11 0x13 +#define ES8328_ADCCONTROL12 0x14 +#define ES8328_ADCCONTROL13 0x15 +#define ES8328_ADCCONTROL14 0x16 + +#define ES8328_DACCONTROL1 0x17 +#define ES8328_DACCONTROL1_DACFORMAT_I2S (0 << 1) +#define ES8328_DACCONTROL1_DACFORMAT_LJUST (1 << 1) +#define ES8328_DACCONTROL1_DACFORMAT_RJUST (2 << 1) +#define ES8328_DACCONTROL1_DACFORMAT_PCM (3 << 1) +#define ES8328_DACCONTROL1_DACWL_24 (0 << 3) +#define ES8328_DACCONTROL1_DACWL_20 (1 << 3) +#define ES8328_DACCONTROL1_DACWL_18 (2 << 3) +#define ES8328_DACCONTROL1_DACWL_16 (3 << 3) +#define ES8328_DACCONTROL1_DACWL_32 (4 << 3) +#define ES8328_DACCONTROL1_DACLRP_I2S_POL_NORMAL (0 << 6) +#define ES8328_DACCONTROL1_DACLRP_I2S_POL_INV (1 << 6) +#define ES8328_DACCONTROL1_DACLRP_PCM_MSB_CLK2 (0 << 6) +#define ES8328_DACCONTROL1_DACLRP_PCM_MSB_CLK1 (1 << 6) +#define ES8328_DACCONTROL1_LRSWAP (1 << 7) + +#define ES8328_DACCONTROL2 0x18 +#define ES8328_DACCONTROL2_RATEMASK (0x1f << 0) +#define ES8328_DACCONTROL2_DOUBLESPEED (1 << 5) + +#define ES8328_DACCONTROL3 0x19 +#define ES8328_DACCONTROL3_AUTOMUTE (1 << 2) +#define ES8328_DACCONTROL3_DACMUTE (1 << 2) +#define ES8328_DACCONTROL3_LEFTGAINVOL (1 << 3) +#define ES8328_DACCONTROL3_DACZEROCROSS (1 << 4) +#define ES8328_DACCONTROL3_DACSOFTRAMP (1 << 5) +#define ES8328_DACCONTROL3_DACRAMPRATE (3 << 6) + +#define ES8328_LDACVOL 0x1a +#define ES8328_LDACVOL_MASK (0 << 0) +#define ES8328_LDACVOL_MAX (0xc0) + +#define ES8328_RDACVOL 0x1b +#define ES8328_RDACVOL_MASK (0 << 0) +#define ES8328_RDACVOL_MAX (0xc0) + +#define ES8328_DACVOL_MAX (0xc0) + +#define ES8328_DACCONTROL4 0x1a +#define ES8328_DACCONTROL5 0x1b + +#define ES8328_DACCONTROL6 0x1c +#define ES8328_DACCONTROL6_CLICKFREE (1 << 3) +#define ES8328_DACCONTROL6_DAC_INVR (1 << 4) +#define ES8328_DACCONTROL6_DAC_INVL (1 << 5) +#define ES8328_DACCONTROL6_DEEMPH_OFF (0 << 6) +#define ES8328_DACCONTROL6_DEEMPH_32k (1 << 6) +#define ES8328_DACCONTROL6_DEEMPH_44_1k (2 << 6) +#define ES8328_DACCONTROL6_DEEMPH_48k (3 << 6) + +#define ES8328_DACCONTROL7 0x1d +#define ES8328_DACCONTROL7_VPP_SCALE_3p5 (0 << 0) +#define ES8328_DACCONTROL7_VPP_SCALE_4p0 (1 << 0) +#define ES8328_DACCONTROL7_VPP_SCALE_3p0 (2 << 0) +#define ES8328_DACCONTROL7_VPP_SCALE_2p5 (3 << 0) +#define ES8328_DACCONTROL7_SHELVING_STRENGTH (1 << 2) /* In eights */ +#define ES8328_DACCONTROL7_MONO (1 << 5) +#define ES8328_DACCONTROL7_ZEROR (1 << 6) +#define ES8328_DACCONTROL7_ZEROL (1 << 7) + +/* Shelving filter */ +#define ES8328_DACCONTROL8 0x1e +#define ES8328_DACCONTROL9 0x1f +#define ES8328_DACCONTROL10 0x20 +#define ES8328_DACCONTROL11 0x21 +#define ES8328_DACCONTROL12 0x22 +#define ES8328_DACCONTROL13 0x23 +#define ES8328_DACCONTROL14 0x24 +#define ES8328_DACCONTROL15 0x25 + +#define ES8328_DACCONTROL16 0x26 +#define ES8328_DACCONTROL16_RMIXSEL_RIN1 (0 << 0) +#define ES8328_DACCONTROL16_RMIXSEL_RIN2 (1 << 0) +#define ES8328_DACCONTROL16_RMIXSEL_RIN3 (2 << 0) +#define ES8328_DACCONTROL16_RMIXSEL_RADC (3 << 0) +#define ES8328_DACCONTROL16_LMIXSEL_LIN1 (0 << 3) +#define ES8328_DACCONTROL16_LMIXSEL_LIN2 (1 << 3) +#define ES8328_DACCONTROL16_LMIXSEL_LIN3 (2 << 3) +#define ES8328_DACCONTROL16_LMIXSEL_LADC (3 << 3) + +#define ES8328_DACCONTROL17 0x27 +#define ES8328_DACCONTROL17_LI2LOVOL (7 << 3) +#define ES8328_DACCONTROL17_LI2LO (1 << 6) +#define ES8328_DACCONTROL17_LD2LO (1 << 7) + +#define ES8328_DACCONTROL18 0x28 +#define ES8328_DACCONTROL18_RI2LOVOL (7 << 3) +#define ES8328_DACCONTROL18_RI2LO (1 << 6) +#define ES8328_DACCONTROL18_RD2LO (1 << 7) + +#define ES8328_DACCONTROL19 0x29 +#define ES8328_DACCONTROL19_LI2ROVOL (7 << 3) +#define ES8328_DACCONTROL19_LI2RO (1 << 6) +#define ES8328_DACCONTROL19_LD2RO (1 << 7) + +#define ES8328_DACCONTROL20 0x2a +#define ES8328_DACCONTROL20_RI2ROVOL (7 << 3) +#define ES8328_DACCONTROL20_RI2RO (1 << 6) +#define ES8328_DACCONTROL20_RD2RO (1 << 7) + +#define ES8328_DACCONTROL21 0x2b +#define ES8328_DACCONTROL21_LI2MOVOL (7 << 3) +#define ES8328_DACCONTROL21_LI2MO (1 << 6) +#define ES8328_DACCONTROL21_LD2MO (1 << 7) + +#define ES8328_DACCONTROL22 0x2c +#define ES8328_DACCONTROL22_RI2MOVOL (7 << 3) +#define ES8328_DACCONTROL22_RI2MO (1 << 6) +#define ES8328_DACCONTROL22_RD2MO (1 << 7) + +#define ES8328_DACCONTROL23 0x2d +#define ES8328_DACCONTROL23_MOUTINV (1 << 1) +#define ES8328_DACCONTROL23_HPSWPOL (1 << 2) +#define ES8328_DACCONTROL23_HPSWEN (1 << 3) +#define ES8328_DACCONTROL23_VROI_1p5k (0 << 4) +#define ES8328_DACCONTROL23_VROI_40k (1 << 4) +#define ES8328_DACCONTROL23_OUT3_VREF (0 << 5) +#define ES8328_DACCONTROL23_OUT3_ROUT1 (1 << 5) +#define ES8328_DACCONTROL23_OUT3_MONOOUT (2 << 5) +#define ES8328_DACCONTROL23_OUT3_RIGHT_MIXER (3 << 5) +#define ES8328_DACCONTROL23_ROUT2INV (1 << 7) + +/* LOUT1 Amplifier */ +#define ES8328_LOUT1VOL 0x2e +#define ES8328_LOUT1VOL_MASK (0 << 5) +#define ES8328_LOUT1VOL_MAX (0x24) + +/* ROUT1 Amplifier */ +#define ES8328_ROUT1VOL 0x2f +#define ES8328_ROUT1VOL_MASK (0 << 5) +#define ES8328_ROUT1VOL_MAX (0x24) + +#define ES8328_OUT1VOL_MAX (0x24) + +/* LOUT2 Amplifier */ +#define ES8328_LOUT2VOL 0x30 +#define ES8328_LOUT2VOL_MASK (0 << 5) +#define ES8328_LOUT2VOL_MAX (0x24) + +/* ROUT2 Amplifier */ +#define ES8328_ROUT2VOL 0x31 +#define ES8328_ROUT2VOL_MASK (0 << 5) +#define ES8328_ROUT2VOL_MAX (0x24) + +#define ES8328_OUT2VOL_MAX (0x24) + +/* Mono Out Amplifier */ +#define ES8328_MONOOUTVOL 0x32 +#define ES8328_MONOOUTVOL_MASK (0 << 5) +#define ES8328_MONOOUTVOL_MAX (0x24) + +#define ES8328_DACCONTROL29 0x33 +#define ES8328_DACCONTROL30 0x34 + +#define ES8328_SYSCLK 0 + +#define ES8328_REG_MAX 0x35 + +#define ES8328_PLL1 0 +#define ES8328_PLL2 1 + +/* clock inputs */ +#define ES8328_MCLK 0 +#define ES8328_PCMCLK 1 + +/* clock divider id's */ +#define ES8328_PCMDIV 0 +#define ES8328_BCLKDIV 1 +#define ES8328_VXCLKDIV 2 + +/* PCM clock dividers */ +#define ES8328_PCM_DIV_1 (0 << 6) +#define ES8328_PCM_DIV_3 (2 << 6) +#define ES8328_PCM_DIV_5_5 (3 << 6) +#define ES8328_PCM_DIV_2 (4 << 6) +#define ES8328_PCM_DIV_4 (5 << 6) +#define ES8328_PCM_DIV_6 (6 << 6) +#define ES8328_PCM_DIV_8 (7 << 6) + +/* BCLK clock dividers */ +#define ES8328_BCLK_DIV_1 (0 << 7) +#define ES8328_BCLK_DIV_2 (1 << 7) +#define ES8328_BCLK_DIV_4 (2 << 7) +#define ES8328_BCLK_DIV_8 (3 << 7) + +/* VXCLK clock dividers */ +#define ES8328_VXCLK_DIV_1 (0 << 6) +#define ES8328_VXCLK_DIV_2 (1 << 6) +#define ES8328_VXCLK_DIV_4 (2 << 6) +#define ES8328_VXCLK_DIV_8 (3 << 6) +#define ES8328_VXCLK_DIV_16 (4 << 6) + +#define ES8328_DAI_HIFI 0 +#define ES8328_DAI_VOICE 1 + +#define ES8328_1536FS 1536 +#define ES8328_1024FS 1024 +#define ES8328_768FS 768 +#define ES8328_512FS 512 +#define ES8328_384FS 384 +#define ES8328_256FS 256 +#define ES8328_128FS 128 + +#endif diff --git a/sound/soc/codecs/jz4740.c b/sound/soc/codecs/jz4740.c index bcebd1a9ce31..df7c01cf7072 100644 --- a/sound/soc/codecs/jz4740.c +++ b/sound/soc/codecs/jz4740.c @@ -293,41 +293,13 @@ static int jz4740_codec_dev_probe(struct snd_soc_codec *codec) regmap_update_bits(jz4740_codec->regmap, JZ4740_REG_CODEC_1, JZ4740_CODEC_1_SW2_ENABLE, JZ4740_CODEC_1_SW2_ENABLE); - jz4740_codec_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; } -static int jz4740_codec_dev_remove(struct snd_soc_codec *codec) -{ - jz4740_codec_set_bias_level(codec, SND_SOC_BIAS_OFF); - - return 0; -} - -#ifdef CONFIG_PM_SLEEP - -static int jz4740_codec_suspend(struct snd_soc_codec *codec) -{ - return jz4740_codec_set_bias_level(codec, SND_SOC_BIAS_OFF); -} - -static int jz4740_codec_resume(struct snd_soc_codec *codec) -{ - return jz4740_codec_set_bias_level(codec, SND_SOC_BIAS_STANDBY); -} - -#else -#define jz4740_codec_suspend NULL -#define jz4740_codec_resume NULL -#endif - static struct snd_soc_codec_driver soc_codec_dev_jz4740_codec = { .probe = jz4740_codec_dev_probe, - .remove = jz4740_codec_dev_remove, - .suspend = jz4740_codec_suspend, - .resume = jz4740_codec_resume, .set_bias_level = jz4740_codec_set_bias_level, + .suspend_bias_off = true, .controls = jz4740_codec_controls, .num_controls = ARRAY_SIZE(jz4740_codec_controls), diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index 275b3f72f3f4..c1ae5764983f 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -1395,18 +1395,6 @@ static struct snd_soc_dai_driver lm49453_dai[] = { }, }; -static int lm49453_suspend(struct snd_soc_codec *codec) -{ - lm49453_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - -static int lm49453_resume(struct snd_soc_codec *codec) -{ - lm49453_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; -} - /* power down chip */ static int lm49453_remove(struct snd_soc_codec *codec) { @@ -1416,8 +1404,6 @@ static int lm49453_remove(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_dev_lm49453 = { .remove = lm49453_remove, - .suspend = lm49453_suspend, - .resume = lm49453_resume, .set_bias_level = lm49453_set_bias_level, .controls = lm49453_snd_controls, .num_controls = ARRAY_SIZE(lm49453_snd_controls), diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 4a063fa88526..d519294f57c7 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -1311,8 +1311,6 @@ static const struct snd_soc_dapm_route max98090_dapm_routes[] = { {"MIC1 Input", NULL, "MIC1"}, {"MIC2 Input", NULL, "MIC2"}, - {"DMICL", NULL, "DMICL_ENA"}, - {"DMICR", NULL, "DMICR_ENA"}, {"DMICL", NULL, "AHPF"}, {"DMICR", NULL, "AHPF"}, @@ -1370,6 +1368,8 @@ static const struct snd_soc_dapm_route max98090_dapm_routes[] = { {"DMIC Mux", "ADC", "ADCR"}, {"DMIC Mux", "DMIC", "DMICL"}, {"DMIC Mux", "DMIC", "DMICR"}, + {"DMIC Mux", "DMIC", "DMICL_ENA"}, + {"DMIC Mux", "DMIC", "DMICR_ENA"}, {"LBENL Mux", "Normal", "DMIC Mux"}, {"LBENL Mux", "Loopback", "LTENL Mux"}, @@ -1972,6 +1972,102 @@ static int max98090_dai_digital_mute(struct snd_soc_dai *codec_dai, int mute) return 0; } +static int max98090_dai_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (!max98090->master && dai->active == 1) + queue_delayed_work(system_power_efficient_wq, + &max98090->pll_det_enable_work, + msecs_to_jiffies(10)); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (!max98090->master && dai->active == 1) + schedule_work(&max98090->pll_det_disable_work); + break; + default: + break; + } + + return 0; +} + +static void max98090_pll_det_enable_work(struct work_struct *work) +{ + struct max98090_priv *max98090 = + container_of(work, struct max98090_priv, + pll_det_enable_work.work); + struct snd_soc_codec *codec = max98090->codec; + unsigned int status, mask; + + /* + * Clear status register in order to clear possibly already occurred + * PLL unlock. If PLL hasn't still locked, the status will be set + * again and PLL unlock interrupt will occur. + * Note this will clear all status bits + */ + regmap_read(max98090->regmap, M98090_REG_DEVICE_STATUS, &status); + + /* + * Queue jack work in case jack state has just changed but handler + * hasn't run yet + */ + regmap_read(max98090->regmap, M98090_REG_INTERRUPT_S, &mask); + status &= mask; + if (status & M98090_JDET_MASK) + queue_delayed_work(system_power_efficient_wq, + &max98090->jack_work, + msecs_to_jiffies(100)); + + /* Enable PLL unlock interrupt */ + snd_soc_update_bits(codec, M98090_REG_INTERRUPT_S, + M98090_IULK_MASK, + 1 << M98090_IULK_SHIFT); +} + +static void max98090_pll_det_disable_work(struct work_struct *work) +{ + struct max98090_priv *max98090 = + container_of(work, struct max98090_priv, pll_det_disable_work); + struct snd_soc_codec *codec = max98090->codec; + + cancel_delayed_work_sync(&max98090->pll_det_enable_work); + + /* Disable PLL unlock interrupt */ + snd_soc_update_bits(codec, M98090_REG_INTERRUPT_S, + M98090_IULK_MASK, 0); +} + +static void max98090_pll_work(struct work_struct *work) +{ + struct max98090_priv *max98090 = + container_of(work, struct max98090_priv, pll_work); + struct snd_soc_codec *codec = max98090->codec; + + if (!snd_soc_codec_is_active(codec)) + return; + + dev_info(codec->dev, "PLL unlocked\n"); + + /* Toggle shutdown OFF then ON */ + snd_soc_update_bits(codec, M98090_REG_DEVICE_SHUTDOWN, + M98090_SHDNN_MASK, 0); + msleep(10); + snd_soc_update_bits(codec, M98090_REG_DEVICE_SHUTDOWN, + M98090_SHDNN_MASK, M98090_SHDNN_MASK); + + /* Give PLL time to lock */ + msleep(10); +} + static void max98090_jack_work(struct work_struct *work) { struct max98090_priv *max98090 = container_of(work, @@ -2063,12 +2159,16 @@ static void max98090_jack_work(struct work_struct *work) static irqreturn_t max98090_interrupt(int irq, void *data) { - struct snd_soc_codec *codec = data; - struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + struct max98090_priv *max98090 = data; + struct snd_soc_codec *codec = max98090->codec; int ret; unsigned int mask; unsigned int active; + /* Treat interrupt before codec is initialized as spurious */ + if (codec == NULL) + return IRQ_NONE; + dev_dbg(codec->dev, "***** max98090_interrupt *****\n"); ret = regmap_read(max98090->regmap, M98090_REG_INTERRUPT_S, &mask); @@ -2103,8 +2203,10 @@ static irqreturn_t max98090_interrupt(int irq, void *data) if (active & M98090_SLD_MASK) dev_dbg(codec->dev, "M98090_SLD_MASK\n"); - if (active & M98090_ULK_MASK) - dev_err(codec->dev, "M98090_ULK_MASK\n"); + if (active & M98090_ULK_MASK) { + dev_dbg(codec->dev, "M98090_ULK_MASK\n"); + schedule_work(&max98090->pll_work); + } if (active & M98090_JDET_MASK) { dev_dbg(codec->dev, "M98090_JDET_MASK\n"); @@ -2177,6 +2279,7 @@ static struct snd_soc_dai_ops max98090_dai_ops = { .set_tdm_slot = max98090_set_tdm_slot, .hw_params = max98090_dai_hw_params, .digital_mute = max98090_dai_digital_mute, + .trigger = max98090_dai_trigger, }; static struct snd_soc_dai_driver max98090_dai[] = { @@ -2230,7 +2333,6 @@ static int max98090_probe(struct snd_soc_codec *codec) max98090->lin_state = 0; max98090->pa1en = 0; max98090->pa2en = 0; - max98090->extmic_mux = 0; ret = snd_soc_read(codec, M98090_REG_REVISION_ID); if (ret < 0) { @@ -2258,22 +2360,16 @@ static int max98090_probe(struct snd_soc_codec *codec) max98090->jack_state = M98090_JACK_STATE_NO_HEADSET; INIT_DELAYED_WORK(&max98090->jack_work, max98090_jack_work); + INIT_DELAYED_WORK(&max98090->pll_det_enable_work, + max98090_pll_det_enable_work); + INIT_WORK(&max98090->pll_det_disable_work, + max98090_pll_det_disable_work); + INIT_WORK(&max98090->pll_work, max98090_pll_work); /* Enable jack detection */ snd_soc_write(codec, M98090_REG_JACK_DETECT, M98090_JDETEN_MASK | M98090_JDEB_25MS); - /* Register for interrupts */ - dev_dbg(codec->dev, "irq = %d\n", max98090->irq); - - ret = devm_request_threaded_irq(codec->dev, max98090->irq, NULL, - max98090_interrupt, IRQF_TRIGGER_FALLING | IRQF_ONESHOT, - "max98090_interrupt", codec); - if (ret < 0) { - dev_err(codec->dev, "request_irq failed: %d\n", - ret); - } - /* * Clear any old interrupts. * An old interrupt ocurring prior to installing the ISR @@ -2310,6 +2406,10 @@ static int max98090_remove(struct snd_soc_codec *codec) struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); cancel_delayed_work_sync(&max98090->jack_work); + cancel_delayed_work_sync(&max98090->pll_det_enable_work); + cancel_work_sync(&max98090->pll_det_disable_work); + cancel_work_sync(&max98090->pll_work); + max98090->codec = NULL; return 0; } @@ -2362,7 +2462,6 @@ static int max98090_i2c_probe(struct i2c_client *i2c, max98090->devtype = driver_data; i2c_set_clientdata(i2c, max98090); max98090->pdata = i2c->dev.platform_data; - max98090->irq = i2c->irq; max98090->regmap = devm_regmap_init_i2c(i2c, &max98090_regmap); if (IS_ERR(max98090->regmap)) { @@ -2371,6 +2470,15 @@ static int max98090_i2c_probe(struct i2c_client *i2c, goto err_enable; } + ret = devm_request_threaded_irq(&i2c->dev, i2c->irq, NULL, + max98090_interrupt, IRQF_TRIGGER_FALLING | IRQF_ONESHOT, + "max98090_interrupt", max98090); + if (ret < 0) { + dev_err(&i2c->dev, "request_irq failed: %d\n", + ret); + return ret; + } + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_max98090, max98090_dai, ARRAY_SIZE(max98090_dai)); diff --git a/sound/soc/codecs/max98090.h b/sound/soc/codecs/max98090.h index cf1b6062ba8c..a5f6bada06da 100644 --- a/sound/soc/codecs/max98090.h +++ b/sound/soc/codecs/max98090.h @@ -11,11 +11,6 @@ #ifndef _MAX98090_H #define _MAX98090_H -#include <linux/version.h> - -/* One can override the Linux version here with an explicit version number */ -#define M98090_LINUX_VERSION LINUX_VERSION_CODE - /* * MAX98090 Register Definitions */ @@ -1502,9 +1497,6 @@ #define M98090_REVID_WIDTH 8 #define M98090_REVID_NUM (1<<M98090_REVID_WIDTH) -#define M98090_BYTE1(w) ((w >> 8) & 0xff) -#define M98090_BYTE0(w) (w & 0xff) - /* Silicon revision number */ #define M98090_REVA 0x40 #define M98091_REVA 0x50 @@ -1529,9 +1521,11 @@ struct max98090_priv { unsigned int bclk; unsigned int lrclk; struct max98090_cdata dai[1]; - int irq; int jack_state; struct delayed_work jack_work; + struct delayed_work pll_det_enable_work; + struct work_struct pll_det_disable_work; + struct work_struct pll_work; struct snd_soc_jack *jack; unsigned int dai_fmt; int tdm_slots; @@ -1539,7 +1533,6 @@ struct max98090_priv { u8 lin_state; unsigned int pa1en; unsigned int pa2en; - unsigned int extmic_mux; unsigned int sidetone; bool master; }; diff --git a/sound/soc/codecs/ml26124.c b/sound/soc/codecs/ml26124.c index e661e8420e3d..711f55039522 100644 --- a/sound/soc/codecs/ml26124.c +++ b/sound/soc/codecs/ml26124.c @@ -565,41 +565,19 @@ static struct snd_soc_dai_driver ml26124_dai = { .symmetric_rates = 1, }; -#ifdef CONFIG_PM -static int ml26124_suspend(struct snd_soc_codec *codec) -{ - ml26124_set_bias_level(codec, SND_SOC_BIAS_OFF); - - return 0; -} - -static int ml26124_resume(struct snd_soc_codec *codec) -{ - ml26124_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - return 0; -} -#else -#define ml26124_suspend NULL -#define ml26124_resume NULL -#endif - static int ml26124_probe(struct snd_soc_codec *codec) { /* Software Reset */ snd_soc_update_bits(codec, ML26124_SW_RST, 0x01, 1); snd_soc_update_bits(codec, ML26124_SW_RST, 0x01, 0); - ml26124_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; } static struct snd_soc_codec_driver soc_codec_dev_ml26124 = { .probe = ml26124_probe, - .suspend = ml26124_suspend, - .resume = ml26124_resume, .set_bias_level = ml26124_set_bias_level, + .suspend_bias_off = true, .dapm_widgets = ml26124_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(ml26124_dapm_widgets), .dapm_routes = ml26124_intercon, diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index e4f6102efc1a..4aa555cbcca8 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -51,7 +51,7 @@ static struct reg_default rt286_index_def[] = { { 0x04, 0xaf01 }, { 0x08, 0x000d }, { 0x09, 0xd810 }, - { 0x0a, 0x0060 }, + { 0x0a, 0x0120 }, { 0x0b, 0x0000 }, { 0x0d, 0x2800 }, { 0x0f, 0x0000 }, @@ -60,7 +60,7 @@ static struct reg_default rt286_index_def[] = { { 0x33, 0x0208 }, { 0x49, 0x0004 }, { 0x4f, 0x50e9 }, - { 0x50, 0x2c00 }, + { 0x50, 0x2000 }, { 0x63, 0x2902 }, { 0x67, 0x1111 }, { 0x68, 0x1016 }, @@ -104,7 +104,6 @@ static const struct reg_default rt286_reg[] = { { 0x02170700, 0x00000000 }, { 0x02270100, 0x00000000 }, { 0x02370100, 0x00000000 }, - { 0x02040000, 0x00004002 }, { 0x01870700, 0x00000020 }, { 0x00830000, 0x000000c3 }, { 0x00930000, 0x000000c3 }, @@ -192,7 +191,6 @@ static int rt286_hw_write(void *context, unsigned int reg, unsigned int value) /*handle index registers*/ if (reg <= 0xff) { rt286_hw_write(client, RT286_COEF_INDEX, reg); - reg = RT286_PROC_COEF; for (i = 0; i < INDEX_CACHE_SIZE; i++) { if (reg == rt286->index_cache[i].reg) { rt286->index_cache[i].def = value; @@ -200,6 +198,7 @@ static int rt286_hw_write(void *context, unsigned int reg, unsigned int value) } } + reg = RT286_PROC_COEF; } data[0] = (reg >> 24) & 0xff; @@ -270,6 +269,7 @@ static int rt286_hw_read(void *context, unsigned int reg, unsigned int *value) return 0; } +#ifdef CONFIG_PM static void rt286_index_sync(struct snd_soc_codec *codec) { struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); @@ -280,6 +280,7 @@ static void rt286_index_sync(struct snd_soc_codec *codec) rt286->index_cache[i].def); } } +#endif static int rt286_support_power_controls[] = { RT286_DAC_OUT1, diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index f1ec6e6bd08a..c3f2decd643c 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1906,6 +1906,32 @@ static int rt5640_set_bias_level(struct snd_soc_codec *codec, return 0; } +int rt5640_dmic_enable(struct snd_soc_codec *codec, + bool dmic1_data_pin, bool dmic2_data_pin) +{ + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + + regmap_update_bits(rt5640->regmap, RT5640_GPIO_CTRL1, + RT5640_GP2_PIN_MASK, RT5640_GP2_PIN_DMIC1_SCL); + + if (dmic1_data_pin) { + regmap_update_bits(rt5640->regmap, RT5640_DMIC, + RT5640_DMIC_1_DP_MASK, RT5640_DMIC_1_DP_GPIO3); + regmap_update_bits(rt5640->regmap, RT5640_GPIO_CTRL1, + RT5640_GP3_PIN_MASK, RT5640_GP3_PIN_DMIC1_SDA); + } + + if (dmic2_data_pin) { + regmap_update_bits(rt5640->regmap, RT5640_DMIC, + RT5640_DMIC_2_DP_MASK, RT5640_DMIC_2_DP_GPIO4); + regmap_update_bits(rt5640->regmap, RT5640_GPIO_CTRL1, + RT5640_GP4_PIN_MASK, RT5640_GP4_PIN_DMIC2_SDA); + } + + return 0; +} +EXPORT_SYMBOL_GPL(rt5640_dmic_enable); + static int rt5640_probe(struct snd_soc_codec *codec) { struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); @@ -1945,6 +1971,10 @@ static int rt5640_probe(struct snd_soc_codec *codec) return -ENODEV; } + if (rt5640->pdata.dmic_en) + rt5640_dmic_enable(codec, rt5640->pdata.dmic1_data_pin, + rt5640->pdata.dmic2_data_pin); + return 0; } @@ -2195,25 +2225,6 @@ static int rt5640_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt5640->regmap, RT5640_IN3_IN4, RT5640_IN_DF2, RT5640_IN_DF2); - if (rt5640->pdata.dmic_en) { - regmap_update_bits(rt5640->regmap, RT5640_GPIO_CTRL1, - RT5640_GP2_PIN_MASK, RT5640_GP2_PIN_DMIC1_SCL); - - if (rt5640->pdata.dmic1_data_pin) { - regmap_update_bits(rt5640->regmap, RT5640_DMIC, - RT5640_DMIC_1_DP_MASK, RT5640_DMIC_1_DP_GPIO3); - regmap_update_bits(rt5640->regmap, RT5640_GPIO_CTRL1, - RT5640_GP3_PIN_MASK, RT5640_GP3_PIN_DMIC1_SDA); - } - - if (rt5640->pdata.dmic2_data_pin) { - regmap_update_bits(rt5640->regmap, RT5640_DMIC, - RT5640_DMIC_2_DP_MASK, RT5640_DMIC_2_DP_GPIO4); - regmap_update_bits(rt5640->regmap, RT5640_GPIO_CTRL1, - RT5640_GP4_PIN_MASK, RT5640_GP4_PIN_DMIC2_SDA); - } - } - rt5640->hp_mute = 1; return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5640, diff --git a/sound/soc/codecs/rt5640.h b/sound/soc/codecs/rt5640.h index 58ebe96b86da..3deb8babeabb 100644 --- a/sound/soc/codecs/rt5640.h +++ b/sound/soc/codecs/rt5640.h @@ -2097,4 +2097,7 @@ struct rt5640_priv { bool hp_mute; }; +int rt5640_dmic_enable(struct snd_soc_codec *codec, + bool dmic1_data_pin, bool dmic2_data_pin); + #endif diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index a7762d0a623e..3fb83bf09768 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -17,6 +17,7 @@ #include <linux/i2c.h> #include <linux/platform_device.h> #include <linux/spi/spi.h> +#include <linux/gpio.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -2103,6 +2104,77 @@ static int rt5645_set_bias_level(struct snd_soc_codec *codec, return 0; } +static int rt5645_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack) +{ + struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); + int gpio_state, jack_type = 0; + unsigned int val; + + gpio_state = gpio_get_value(rt5645->pdata.hp_det_gpio); + + dev_dbg(codec->dev, "gpio = %d(%d)\n", rt5645->pdata.hp_det_gpio, + gpio_state); + + if ((rt5645->pdata.gpio_hp_det_active_high && gpio_state) || + (!rt5645->pdata.gpio_hp_det_active_high && !gpio_state)) { + snd_soc_dapm_force_enable_pin(&codec->dapm, "micbias1"); + snd_soc_dapm_force_enable_pin(&codec->dapm, "micbias2"); + snd_soc_dapm_force_enable_pin(&codec->dapm, "LDO2"); + snd_soc_dapm_force_enable_pin(&codec->dapm, "Mic Det Power"); + snd_soc_dapm_sync(&codec->dapm); + + snd_soc_write(codec, RT5645_IN1_CTRL1, 0x0006); + snd_soc_write(codec, RT5645_JD_CTRL3, 0x00b0); + + snd_soc_update_bits(codec, RT5645_IN1_CTRL2, + RT5645_CBJ_MN_JD, 0); + snd_soc_update_bits(codec, RT5645_IN1_CTRL2, + RT5645_CBJ_MN_JD, RT5645_CBJ_MN_JD); + + msleep(400); + val = snd_soc_read(codec, RT5645_IN1_CTRL3) & 0x7; + dev_dbg(codec->dev, "val = %d\n", val); + + if (val == 1 || val == 2) + jack_type = SND_JACK_HEADSET; + else + jack_type = SND_JACK_HEADPHONE; + + snd_soc_dapm_disable_pin(&codec->dapm, "micbias1"); + snd_soc_dapm_disable_pin(&codec->dapm, "micbias2"); + snd_soc_dapm_disable_pin(&codec->dapm, "LDO2"); + snd_soc_dapm_disable_pin(&codec->dapm, "Mic Det Power"); + snd_soc_dapm_sync(&codec->dapm); + } + + snd_soc_jack_report(rt5645->jack, jack_type, SND_JACK_HEADSET); + + return 0; +} + +int rt5645_set_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack) +{ + struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); + + rt5645->jack = jack; + + rt5645_jack_detect(codec, rt5645->jack); + + return 0; +} +EXPORT_SYMBOL_GPL(rt5645_set_jack_detect); + +static irqreturn_t rt5645_irq(int irq, void *data) +{ + struct rt5645_priv *rt5645 = data; + + rt5645_jack_detect(rt5645->codec, rt5645->jack); + + return IRQ_HANDLED; +} + static int rt5645_probe(struct snd_soc_codec *codec) { struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); @@ -2250,6 +2322,7 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, if (rt5645 == NULL) return -ENOMEM; + rt5645->i2c = i2c; i2c_set_clientdata(i2c, rt5645); if (pdata) @@ -2345,12 +2418,38 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, } + if (rt5645->i2c->irq) { + ret = request_threaded_irq(rt5645->i2c->irq, NULL, rt5645_irq, + IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING + | IRQF_ONESHOT, "rt5645", rt5645); + if (ret) + dev_err(&i2c->dev, "Failed to reguest IRQ: %d\n", ret); + } + + if (gpio_is_valid(rt5645->pdata.hp_det_gpio)) { + ret = gpio_request(rt5645->pdata.hp_det_gpio, "rt5645"); + if (ret) + dev_err(&i2c->dev, "Fail gpio_request hp_det_gpio\n"); + + ret = gpio_direction_input(rt5645->pdata.hp_det_gpio); + if (ret) + dev_err(&i2c->dev, "Fail gpio_direction hp_det_gpio\n"); + } + return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5645, rt5645_dai, ARRAY_SIZE(rt5645_dai)); } static int rt5645_i2c_remove(struct i2c_client *i2c) { + struct rt5645_priv *rt5645 = i2c_get_clientdata(i2c); + + if (i2c->irq) + free_irq(i2c->irq, rt5645); + + if (gpio_is_valid(rt5645->pdata.hp_det_gpio)) + gpio_free(rt5645->pdata.hp_det_gpio); + snd_soc_unregister_codec(&i2c->dev); return 0; diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h index 355b7e9eefab..50c62c5668ea 100644 --- a/sound/soc/codecs/rt5645.h +++ b/sound/soc/codecs/rt5645.h @@ -2166,6 +2166,8 @@ struct rt5645_priv { struct snd_soc_codec *codec; struct rt5645_platform_data pdata; struct regmap *regmap; + struct i2c_client *i2c; + struct snd_soc_jack *jack; int sysclk; int sysclk_src; @@ -2178,4 +2180,7 @@ struct rt5645_priv { int pll_out; }; +int rt5645_set_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack); + #endif /* __RT5645_H__ */ diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 5337c448b5e3..16aa4d99a713 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -15,10 +15,12 @@ #include <linux/init.h> #include <linux/delay.h> #include <linux/pm.h> +#include <linux/of_gpio.h> #include <linux/regmap.h> #include <linux/i2c.h> #include <linux/platform_device.h> #include <linux/spi/spi.h> +#include <linux/gpio.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -540,6 +542,7 @@ static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0); static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0); static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0); static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0); +static const DECLARE_TLV_DB_SCALE(st_vol_tlv, -4650, 150, 0); /* {0, +20, +24, +30, +35, +40, +44, +50, +52} dB */ static unsigned int bst_tlv[] = { @@ -604,6 +607,10 @@ static const struct snd_kcontrol_new rt5677_snd_controls[] = { RT5677_MONO_ADC_L_VOL_SFT, RT5677_MONO_ADC_R_VOL_SFT, 127, 0, adc_vol_tlv), + /* Sidetone Control */ + SOC_SINGLE_TLV("Sidetone Volume", RT5677_SIDETONE_CTRL, + RT5677_ST_VOL_SFT, 31, 0, st_vol_tlv), + /* ADC Boost Volume Control */ SOC_DOUBLE_TLV("STO1 ADC Boost Volume", RT5677_STO1_2_ADC_BST, RT5677_STO1_ADC_L_BST_SFT, RT5677_STO1_ADC_R_BST_SFT, 3, 0, @@ -1700,14 +1707,19 @@ static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = { SND_SOC_DAPM_INPUT("Haptic Generator"), - SND_SOC_DAPM_PGA("DMIC1", RT5677_DMIC_CTRL1, RT5677_DMIC_1_EN_SFT, 0, - NULL, 0), - SND_SOC_DAPM_PGA("DMIC2", RT5677_DMIC_CTRL1, RT5677_DMIC_2_EN_SFT, 0, - NULL, 0), - SND_SOC_DAPM_PGA("DMIC3", RT5677_DMIC_CTRL1, RT5677_DMIC_3_EN_SFT, 0, - NULL, 0), - SND_SOC_DAPM_PGA("DMIC4", RT5677_DMIC_CTRL2, RT5677_DMIC_4_EN_SFT, 0, - NULL, 0), + SND_SOC_DAPM_PGA("DMIC1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DMIC2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DMIC3", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DMIC4", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("DMIC1 power", RT5677_DMIC_CTRL1, + RT5677_DMIC_1_EN_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DMIC2 power", RT5677_DMIC_CTRL1, + RT5677_DMIC_2_EN_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DMIC3 power", RT5677_DMIC_CTRL1, + RT5677_DMIC_3_EN_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DMIC4 power", RT5677_DMIC_CTRL2, + RT5677_DMIC_4_EN_SFT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("DMIC CLK", SND_SOC_NOPM, 0, 0, set_dmic_clk, SND_SOC_DAPM_PRE_PMU), @@ -1987,6 +1999,9 @@ static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = { /* Sidetone Mux */ SND_SOC_DAPM_MUX("Sidetone Mux", SND_SOC_NOPM, 0, 0, &rt5677_sidetone_mux), + SND_SOC_DAPM_SUPPLY("Sidetone Power", RT5677_SIDETONE_CTRL, + RT5677_ST_EN_SFT, 0, NULL, 0), + /* VAD Mux*/ SND_SOC_DAPM_MUX("VAD ADC Mux", SND_SOC_NOPM, 0, 0, &rt5677_vad_src_mux), @@ -2130,6 +2145,13 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = { { "DMIC L4", NULL, "DMIC CLK" }, { "DMIC R4", NULL, "DMIC CLK" }, + { "DMIC L1", NULL, "DMIC1 power" }, + { "DMIC R1", NULL, "DMIC1 power" }, + { "DMIC L3", NULL, "DMIC3 power" }, + { "DMIC R3", NULL, "DMIC3 power" }, + { "DMIC L4", NULL, "DMIC4 power" }, + { "DMIC R4", NULL, "DMIC4 power" }, + { "BST1", NULL, "IN1P" }, { "BST1", NULL, "IN1N" }, { "BST2", NULL, "IN2P" }, @@ -2691,6 +2713,7 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = { { "Sidetone Mux", "DMIC4 L", "DMIC L4" }, { "Sidetone Mux", "ADC1", "ADC 1" }, { "Sidetone Mux", "ADC2", "ADC 2" }, + { "Sidetone Mux", NULL, "Sidetone Power" }, { "Stereo DAC MIXL", "ST L Switch", "Sidetone Mux" }, { "Stereo DAC MIXL", "DAC1 L Switch", "DAC1 MIXL" }, @@ -2793,6 +2816,16 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = { { "PDM2R", NULL, "PDM2 R Mux" }, }; +static const struct snd_soc_dapm_route rt5677_dmic2_clk_1[] = { + { "DMIC L2", NULL, "DMIC1 power" }, + { "DMIC R2", NULL, "DMIC1 power" }, +}; + +static const struct snd_soc_dapm_route rt5677_dmic2_clk_2[] = { + { "DMIC L2", NULL, "DMIC2 power" }, + { "DMIC R2", NULL, "DMIC2 power" }, +}; + static int rt5677_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { @@ -3084,6 +3117,59 @@ static int rt5677_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, return 0; } +static int rt5677_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, int slot_width) +{ + struct snd_soc_codec *codec = dai->codec; + unsigned int val = 0; + + if (rx_mask || tx_mask) + val |= (1 << 12); + + switch (slots) { + case 4: + val |= (1 << 10); + break; + case 6: + val |= (2 << 10); + break; + case 8: + val |= (3 << 10); + break; + case 2: + default: + break; + } + + switch (slot_width) { + case 20: + val |= (1 << 8); + break; + case 24: + val |= (2 << 8); + break; + case 32: + val |= (3 << 8); + break; + case 16: + default: + break; + } + + switch (dai->id) { + case RT5677_AIF1: + snd_soc_update_bits(codec, RT5677_TDM1_CTRL1, 0x1f00, val); + break; + case RT5677_AIF2: + snd_soc_update_bits(codec, RT5677_TDM2_CTRL1, 0x1f00, val); + break; + default: + break; + } + + return 0; +} + static int rt5677_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { @@ -3138,12 +3224,148 @@ static int rt5677_set_bias_level(struct snd_soc_codec *codec, return 0; } +#ifdef CONFIG_GPIOLIB +static inline struct rt5677_priv *gpio_to_rt5677(struct gpio_chip *chip) +{ + return container_of(chip, struct rt5677_priv, gpio_chip); +} + +static void rt5677_gpio_set(struct gpio_chip *chip, unsigned offset, int value) +{ + struct rt5677_priv *rt5677 = gpio_to_rt5677(chip); + + switch (offset) { + case RT5677_GPIO1 ... RT5677_GPIO5: + regmap_update_bits(rt5677->regmap, RT5677_GPIO_CTRL2, + 0x1 << (offset * 3 + 1), !!value << (offset * 3 + 1)); + break; + + case RT5677_GPIO6: + regmap_update_bits(rt5677->regmap, RT5677_GPIO_CTRL3, + RT5677_GPIO6_OUT_MASK, !!value << RT5677_GPIO6_OUT_SFT); + break; + + default: + break; + } +} + +static int rt5677_gpio_direction_out(struct gpio_chip *chip, + unsigned offset, int value) +{ + struct rt5677_priv *rt5677 = gpio_to_rt5677(chip); + + switch (offset) { + case RT5677_GPIO1 ... RT5677_GPIO5: + regmap_update_bits(rt5677->regmap, RT5677_GPIO_CTRL2, + 0x3 << (offset * 3 + 1), + (0x2 | !!value) << (offset * 3 + 1)); + break; + + case RT5677_GPIO6: + regmap_update_bits(rt5677->regmap, RT5677_GPIO_CTRL3, + RT5677_GPIO6_DIR_MASK | RT5677_GPIO6_OUT_MASK, + RT5677_GPIO6_DIR_OUT | !!value << RT5677_GPIO6_OUT_SFT); + break; + + default: + break; + } + + return 0; +} + +static int rt5677_gpio_get(struct gpio_chip *chip, unsigned offset) +{ + struct rt5677_priv *rt5677 = gpio_to_rt5677(chip); + int value, ret; + + ret = regmap_read(rt5677->regmap, RT5677_GPIO_ST, &value); + if (ret < 0) + return ret; + + return (value & (0x1 << offset)) >> offset; +} + +static int rt5677_gpio_direction_in(struct gpio_chip *chip, unsigned offset) +{ + struct rt5677_priv *rt5677 = gpio_to_rt5677(chip); + + switch (offset) { + case RT5677_GPIO1 ... RT5677_GPIO5: + regmap_update_bits(rt5677->regmap, RT5677_GPIO_CTRL2, + 0x1 << (offset * 3 + 2), 0x0); + break; + + case RT5677_GPIO6: + regmap_update_bits(rt5677->regmap, RT5677_GPIO_CTRL3, + RT5677_GPIO6_DIR_MASK, RT5677_GPIO6_DIR_IN); + break; + + default: + break; + } + + return 0; +} + +static struct gpio_chip rt5677_template_chip = { + .label = "rt5677", + .owner = THIS_MODULE, + .direction_output = rt5677_gpio_direction_out, + .set = rt5677_gpio_set, + .direction_input = rt5677_gpio_direction_in, + .get = rt5677_gpio_get, + .can_sleep = 1, +}; + +static void rt5677_init_gpio(struct i2c_client *i2c) +{ + struct rt5677_priv *rt5677 = i2c_get_clientdata(i2c); + int ret; + + rt5677->gpio_chip = rt5677_template_chip; + rt5677->gpio_chip.ngpio = RT5677_GPIO_NUM; + rt5677->gpio_chip.dev = &i2c->dev; + rt5677->gpio_chip.base = -1; + + ret = gpiochip_add(&rt5677->gpio_chip); + if (ret != 0) + dev_err(&i2c->dev, "Failed to add GPIOs: %d\n", ret); +} + +static void rt5677_free_gpio(struct i2c_client *i2c) +{ + struct rt5677_priv *rt5677 = i2c_get_clientdata(i2c); + + gpiochip_remove(&rt5677->gpio_chip); +} +#else +static void rt5677_init_gpio(struct i2c_client *i2c) +{ +} + +static void rt5677_free_gpio(struct i2c_client *i2c) +{ +} +#endif + static int rt5677_probe(struct snd_soc_codec *codec) { struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); rt5677->codec = codec; + if (rt5677->pdata.dmic2_clk_pin == RT5677_DMIC_CLK2) { + snd_soc_dapm_add_routes(&codec->dapm, + rt5677_dmic2_clk_2, + ARRAY_SIZE(rt5677_dmic2_clk_2)); + } else { /*use dmic1 clock by default*/ + snd_soc_dapm_add_routes(&codec->dapm, + rt5677_dmic2_clk_1, + ARRAY_SIZE(rt5677_dmic2_clk_1)); + } + rt5677_set_bias_level(codec, SND_SOC_BIAS_OFF); regmap_write(rt5677->regmap, RT5677_DIG_MISC, 0x0020); @@ -3157,6 +3379,8 @@ static int rt5677_remove(struct snd_soc_codec *codec) struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); regmap_write(rt5677->regmap, RT5677_RESET, 0x10ec); + if (gpio_is_valid(rt5677->pow_ldo2)) + gpio_set_value_cansleep(rt5677->pow_ldo2, 0); return 0; } @@ -3168,6 +3392,8 @@ static int rt5677_suspend(struct snd_soc_codec *codec) regcache_cache_only(rt5677->regmap, true); regcache_mark_dirty(rt5677->regmap); + if (gpio_is_valid(rt5677->pow_ldo2)) + gpio_set_value_cansleep(rt5677->pow_ldo2, 0); return 0; } @@ -3176,6 +3402,10 @@ static int rt5677_resume(struct snd_soc_codec *codec) { struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); + if (gpio_is_valid(rt5677->pow_ldo2)) { + gpio_set_value_cansleep(rt5677->pow_ldo2, 1); + msleep(10); + } regcache_cache_only(rt5677->regmap, false); regcache_sync(rt5677->regmap); @@ -3195,6 +3425,7 @@ static struct snd_soc_dai_ops rt5677_aif_dai_ops = { .set_fmt = rt5677_set_dai_fmt, .set_sysclk = rt5677_set_dai_sysclk, .set_pll = rt5677_set_dai_pll, + .set_tdm_slot = rt5677_set_tdm_slot, }; static struct snd_soc_dai_driver rt5677_dai[] = { @@ -3333,6 +3564,35 @@ static const struct i2c_device_id rt5677_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, rt5677_i2c_id); +static int rt5677_parse_dt(struct rt5677_priv *rt5677, struct device_node *np) +{ + rt5677->pdata.in1_diff = of_property_read_bool(np, + "realtek,in1-differential"); + rt5677->pdata.in2_diff = of_property_read_bool(np, + "realtek,in2-differential"); + rt5677->pdata.lout1_diff = of_property_read_bool(np, + "realtek,lout1-differential"); + rt5677->pdata.lout2_diff = of_property_read_bool(np, + "realtek,lout2-differential"); + rt5677->pdata.lout3_diff = of_property_read_bool(np, + "realtek,lout3-differential"); + + rt5677->pow_ldo2 = of_get_named_gpio(np, + "realtek,pow-ldo2-gpio", 0); + + /* + * POW_LDO2 is optional (it may be statically tied on the board). + * -ENOENT means that the property doesn't exist, i.e. there is no + * GPIO, so is not an error. Any other error code means the property + * exists, but could not be parsed. + */ + if (!gpio_is_valid(rt5677->pow_ldo2) && + (rt5677->pow_ldo2 != -ENOENT)) + return rt5677->pow_ldo2; + + return 0; +} + static int rt5677_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -3351,6 +3611,33 @@ static int rt5677_i2c_probe(struct i2c_client *i2c, if (pdata) rt5677->pdata = *pdata; + if (i2c->dev.of_node) { + ret = rt5677_parse_dt(rt5677, i2c->dev.of_node); + if (ret) { + dev_err(&i2c->dev, "Failed to parse device tree: %d\n", + ret); + return ret; + } + } else { + rt5677->pow_ldo2 = -EINVAL; + } + + if (gpio_is_valid(rt5677->pow_ldo2)) { + ret = devm_gpio_request_one(&i2c->dev, rt5677->pow_ldo2, + GPIOF_OUT_INIT_HIGH, + "RT5677 POW_LDO2"); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to request POW_LDO2 %d: %d\n", + rt5677->pow_ldo2, ret); + return ret; + } + /* Wait a while until I2C bus becomes available. The datasheet + * does not specify the exact we should wait but startup + * sequence mentiones at least a few milliseconds. + */ + msleep(10); + } + rt5677->regmap = devm_regmap_init_i2c(i2c, &rt5677_regmap); if (IS_ERR(rt5677->regmap)) { ret = PTR_ERR(rt5677->regmap); @@ -3381,6 +3668,29 @@ static int rt5677_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt5677->regmap, RT5677_IN1, RT5677_IN_DF2, RT5677_IN_DF2); + if (rt5677->pdata.lout1_diff) + regmap_update_bits(rt5677->regmap, RT5677_LOUT1, + RT5677_LOUT1_L_DF, RT5677_LOUT1_L_DF); + + if (rt5677->pdata.lout2_diff) + regmap_update_bits(rt5677->regmap, RT5677_LOUT1, + RT5677_LOUT2_L_DF, RT5677_LOUT2_L_DF); + + if (rt5677->pdata.lout3_diff) + regmap_update_bits(rt5677->regmap, RT5677_LOUT1, + RT5677_LOUT3_L_DF, RT5677_LOUT3_L_DF); + + if (rt5677->pdata.dmic2_clk_pin == RT5677_DMIC_CLK2) { + regmap_update_bits(rt5677->regmap, RT5677_GEN_CTRL2, + RT5677_GPIO5_FUNC_MASK, + RT5677_GPIO5_FUNC_DMIC); + regmap_update_bits(rt5677->regmap, RT5677_GPIO_CTRL2, + RT5677_GPIO5_DIR_MASK, + RT5677_GPIO5_DIR_OUT); + } + + rt5677_init_gpio(i2c); + return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5677, rt5677_dai, ARRAY_SIZE(rt5677_dai)); } @@ -3388,6 +3698,7 @@ static int rt5677_i2c_probe(struct i2c_client *i2c, static int rt5677_i2c_remove(struct i2c_client *i2c) { snd_soc_unregister_codec(&i2c->dev); + rt5677_free_gpio(i2c); return 0; } diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h index 863393e62096..d4eb6d5e6746 100644 --- a/sound/soc/codecs/rt5677.h +++ b/sound/soc/codecs/rt5677.h @@ -382,6 +382,10 @@ #define RT5677_ST_SEL_SFT 9 #define RT5677_ST_EN (0x1 << 6) #define RT5677_ST_EN_SFT 6 +#define RT5677_ST_GAIN (0x1 << 5) +#define RT5677_ST_GAIN_SFT 5 +#define RT5677_ST_VOL_MASK (0x1f << 0) +#define RT5677_ST_VOL_SFT 0 /* Analog DAC1/2/3 Source Control (0x15) */ #define RT5677_ANA_DAC3_SRC_SEL_MASK (0x3 << 4) @@ -1287,16 +1291,16 @@ #define RT5677_PLL1_PD_SFT 8 #define RT5677_PLL1_PD_1 (0x0 << 8) #define RT5677_PLL1_PD_2 (0x1 << 8) -#define RT5671_DAC_OSR_MASK (0x3 << 6) -#define RT5671_DAC_OSR_SFT 6 -#define RT5671_DAC_OSR_128 (0x0 << 6) -#define RT5671_DAC_OSR_64 (0x1 << 6) -#define RT5671_DAC_OSR_32 (0x2 << 6) -#define RT5671_ADC_OSR_MASK (0x3 << 4) -#define RT5671_ADC_OSR_SFT 4 -#define RT5671_ADC_OSR_128 (0x0 << 4) -#define RT5671_ADC_OSR_64 (0x1 << 4) -#define RT5671_ADC_OSR_32 (0x2 << 4) +#define RT5677_DAC_OSR_MASK (0x3 << 6) +#define RT5677_DAC_OSR_SFT 6 +#define RT5677_DAC_OSR_128 (0x0 << 6) +#define RT5677_DAC_OSR_64 (0x1 << 6) +#define RT5677_DAC_OSR_32 (0x2 << 6) +#define RT5677_ADC_OSR_MASK (0x3 << 4) +#define RT5677_ADC_OSR_SFT 4 +#define RT5677_ADC_OSR_128 (0x0 << 4) +#define RT5677_ADC_OSR_64 (0x1 << 4) +#define RT5677_ADC_OSR_32 (0x2 << 4) /* Global Clock Control 2 (0x81) */ #define RT5677_PLL2_PR_SRC_MASK (0x1 << 15) @@ -1312,18 +1316,18 @@ #define RT5677_PLL2_SRC_BCLK4 (0x4 << 12) #define RT5677_PLL2_SRC_RCCLK (0x5 << 12) #define RT5677_PLL2_SRC_SLIM (0x6 << 12) -#define RT5671_DSP_ASRC_O_SRC (0x3 << 10) -#define RT5671_DSP_ASRC_O_SRC_SFT 10 -#define RT5671_DSP_ASRC_O_MCLK (0x0 << 10) -#define RT5671_DSP_ASRC_O_PLL1 (0x1 << 10) -#define RT5671_DSP_ASRC_O_SLIM (0x2 << 10) -#define RT5671_DSP_ASRC_O_RCCLK (0x3 << 10) -#define RT5671_DSP_ASRC_I_SRC (0x3 << 8) -#define RT5671_DSP_ASRC_I_SRC_SFT 8 -#define RT5671_DSP_ASRC_I_MCLK (0x0 << 8) -#define RT5671_DSP_ASRC_I_PLL1 (0x1 << 8) -#define RT5671_DSP_ASRC_I_SLIM (0x2 << 8) -#define RT5671_DSP_ASRC_I_RCCLK (0x3 << 8) +#define RT5677_DSP_ASRC_O_SRC (0x3 << 10) +#define RT5677_DSP_ASRC_O_SRC_SFT 10 +#define RT5677_DSP_ASRC_O_MCLK (0x0 << 10) +#define RT5677_DSP_ASRC_O_PLL1 (0x1 << 10) +#define RT5677_DSP_ASRC_O_SLIM (0x2 << 10) +#define RT5677_DSP_ASRC_O_RCCLK (0x3 << 10) +#define RT5677_DSP_ASRC_I_SRC (0x3 << 8) +#define RT5677_DSP_ASRC_I_SRC_SFT 8 +#define RT5677_DSP_ASRC_I_MCLK (0x0 << 8) +#define RT5677_DSP_ASRC_I_PLL1 (0x1 << 8) +#define RT5677_DSP_ASRC_I_SLIM (0x2 << 8) +#define RT5677_DSP_ASRC_I_RCCLK (0x3 << 8) #define RT5677_DSP_CLK_SRC_MASK (0x1 << 7) #define RT5677_DSP_CLK_SRC_SFT 7 #define RT5677_DSP_CLK_SRC_PLL2 (0x0 << 7) @@ -1363,6 +1367,110 @@ #define RT5677_SEL_SRC_IB01 (0x1 << 0) #define RT5677_SEL_SRC_IB01_SFT 0 +/* GPIO status (0xbf) */ +#define RT5677_GPIO6_STATUS_MASK (0x1 << 5) +#define RT5677_GPIO6_STATUS_SFT 5 +#define RT5677_GPIO5_STATUS_MASK (0x1 << 4) +#define RT5677_GPIO5_STATUS_SFT 4 +#define RT5677_GPIO4_STATUS_MASK (0x1 << 3) +#define RT5677_GPIO4_STATUS_SFT 3 +#define RT5677_GPIO3_STATUS_MASK (0x1 << 2) +#define RT5677_GPIO3_STATUS_SFT 2 +#define RT5677_GPIO2_STATUS_MASK (0x1 << 1) +#define RT5677_GPIO2_STATUS_SFT 1 +#define RT5677_GPIO1_STATUS_MASK (0x1 << 0) +#define RT5677_GPIO1_STATUS_SFT 0 + +/* GPIO Control 1 (0xc0) */ +#define RT5677_GPIO1_PIN_MASK (0x1 << 15) +#define RT5677_GPIO1_PIN_SFT 15 +#define RT5677_GPIO1_PIN_GPIO1 (0x0 << 15) +#define RT5677_GPIO1_PIN_IRQ (0x1 << 15) +#define RT5677_IPTV_MODE_MASK (0x1 << 14) +#define RT5677_IPTV_MODE_SFT 14 +#define RT5677_IPTV_MODE_GPIO (0x0 << 14) +#define RT5677_IPTV_MODE_IPTV (0x1 << 14) +#define RT5677_FUNC_MODE_MASK (0x1 << 13) +#define RT5677_FUNC_MODE_SFT 13 +#define RT5677_FUNC_MODE_DMIC_GPIO (0x0 << 13) +#define RT5677_FUNC_MODE_JTAG (0x1 << 13) + +/* GPIO Control 2 (0xc1) */ +#define RT5677_GPIO5_DIR_MASK (0x1 << 14) +#define RT5677_GPIO5_DIR_SFT 14 +#define RT5677_GPIO5_DIR_IN (0x0 << 14) +#define RT5677_GPIO5_DIR_OUT (0x1 << 14) +#define RT5677_GPIO5_OUT_MASK (0x1 << 13) +#define RT5677_GPIO5_OUT_SFT 13 +#define RT5677_GPIO5_OUT_LO (0x0 << 13) +#define RT5677_GPIO5_OUT_HI (0x1 << 13) +#define RT5677_GPIO5_P_MASK (0x1 << 12) +#define RT5677_GPIO5_P_SFT 12 +#define RT5677_GPIO5_P_NOR (0x0 << 12) +#define RT5677_GPIO5_P_INV (0x1 << 12) +#define RT5677_GPIO4_DIR_MASK (0x1 << 11) +#define RT5677_GPIO4_DIR_SFT 11 +#define RT5677_GPIO4_DIR_IN (0x0 << 11) +#define RT5677_GPIO4_DIR_OUT (0x1 << 11) +#define RT5677_GPIO4_OUT_MASK (0x1 << 10) +#define RT5677_GPIO4_OUT_SFT 10 +#define RT5677_GPIO4_OUT_LO (0x0 << 10) +#define RT5677_GPIO4_OUT_HI (0x1 << 10) +#define RT5677_GPIO4_P_MASK (0x1 << 9) +#define RT5677_GPIO4_P_SFT 9 +#define RT5677_GPIO4_P_NOR (0x0 << 9) +#define RT5677_GPIO4_P_INV (0x1 << 9) +#define RT5677_GPIO3_DIR_MASK (0x1 << 8) +#define RT5677_GPIO3_DIR_SFT 8 +#define RT5677_GPIO3_DIR_IN (0x0 << 8) +#define RT5677_GPIO3_DIR_OUT (0x1 << 8) +#define RT5677_GPIO3_OUT_MASK (0x1 << 7) +#define RT5677_GPIO3_OUT_SFT 7 +#define RT5677_GPIO3_OUT_LO (0x0 << 7) +#define RT5677_GPIO3_OUT_HI (0x1 << 7) +#define RT5677_GPIO3_P_MASK (0x1 << 6) +#define RT5677_GPIO3_P_SFT 6 +#define RT5677_GPIO3_P_NOR (0x0 << 6) +#define RT5677_GPIO3_P_INV (0x1 << 6) +#define RT5677_GPIO2_DIR_MASK (0x1 << 5) +#define RT5677_GPIO2_DIR_SFT 5 +#define RT5677_GPIO2_DIR_IN (0x0 << 5) +#define RT5677_GPIO2_DIR_OUT (0x1 << 5) +#define RT5677_GPIO2_OUT_MASK (0x1 << 4) +#define RT5677_GPIO2_OUT_SFT 4 +#define RT5677_GPIO2_OUT_LO (0x0 << 4) +#define RT5677_GPIO2_OUT_HI (0x1 << 4) +#define RT5677_GPIO2_P_MASK (0x1 << 3) +#define RT5677_GPIO2_P_SFT 3 +#define RT5677_GPIO2_P_NOR (0x0 << 3) +#define RT5677_GPIO2_P_INV (0x1 << 3) +#define RT5677_GPIO1_DIR_MASK (0x1 << 2) +#define RT5677_GPIO1_DIR_SFT 2 +#define RT5677_GPIO1_DIR_IN (0x0 << 2) +#define RT5677_GPIO1_DIR_OUT (0x1 << 2) +#define RT5677_GPIO1_OUT_MASK (0x1 << 1) +#define RT5677_GPIO1_OUT_SFT 1 +#define RT5677_GPIO1_OUT_LO (0x0 << 1) +#define RT5677_GPIO1_OUT_HI (0x1 << 1) +#define RT5677_GPIO1_P_MASK (0x1 << 0) +#define RT5677_GPIO1_P_SFT 0 +#define RT5677_GPIO1_P_NOR (0x0 << 0) +#define RT5677_GPIO1_P_INV (0x1 << 0) + +/* GPIO Control 3 (0xc2) */ +#define RT5677_GPIO6_DIR_MASK (0x1 << 2) +#define RT5677_GPIO6_DIR_SFT 2 +#define RT5677_GPIO6_DIR_IN (0x0 << 2) +#define RT5677_GPIO6_DIR_OUT (0x1 << 2) +#define RT5677_GPIO6_OUT_MASK (0x1 << 1) +#define RT5677_GPIO6_OUT_SFT 1 +#define RT5677_GPIO6_OUT_LO (0x0 << 1) +#define RT5677_GPIO6_OUT_HI (0x1 << 1) +#define RT5677_GPIO6_P_MASK (0x1 << 0) +#define RT5677_GPIO6_P_SFT 0 +#define RT5677_GPIO6_P_NOR (0x0 << 0) +#define RT5677_GPIO6_P_INV (0x1 << 0) + /* Virtual DSP Mixer Control (0xf7 0xf8 0xf9) */ #define RT5677_DSP_IB_01_H (0x1 << 15) #define RT5677_DSP_IB_01_H_SFT 15 @@ -1393,6 +1501,11 @@ #define RT5677_DSP_IB_9_L (0x1 << 1) #define RT5677_DSP_IB_9_L_SFT 1 +/* General Control2 (0xfc)*/ +#define RT5677_GPIO5_FUNC_MASK (0x1 << 9) +#define RT5677_GPIO5_FUNC_GPIO (0x0 << 9) +#define RT5677_GPIO5_FUNC_DMIC (0x1 << 9) + /* System Clock Source */ enum { RT5677_SCLK_S_MCLK, @@ -1418,6 +1531,16 @@ enum { RT5677_AIFS, }; +enum { + RT5677_GPIO1, + RT5677_GPIO2, + RT5677_GPIO3, + RT5677_GPIO4, + RT5677_GPIO5, + RT5677_GPIO6, + RT5677_GPIO_NUM, +}; + struct rt5677_priv { struct snd_soc_codec *codec; struct rt5677_platform_data pdata; @@ -1431,6 +1554,10 @@ struct rt5677_priv { int pll_src; int pll_in; int pll_out; + int pow_ldo2; /* POW_LDO2 pin */ +#ifdef CONFIG_GPIOLIB + struct gpio_chip gpio_chip; +#endif }; #endif /* __RT5677_H__ */ diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index e997d271728d..6bb77d76561b 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -626,6 +626,9 @@ static int sgtl5000_set_clock(struct snd_soc_codec *codec, int frame_rate) } else { dev_err(codec->dev, "PLL not supported in slave mode\n"); + dev_err(codec->dev, "%d ratio is not supported. " + "SYS_MCLK needs to be 256, 384 or 512 * fs\n", + sgtl5000->sysclk / sys_fs); return -EINVAL; } } @@ -1073,26 +1076,6 @@ static bool sgtl5000_readable(struct device *dev, unsigned int reg) } } -#ifdef CONFIG_SUSPEND -static int sgtl5000_suspend(struct snd_soc_codec *codec) -{ - sgtl5000_set_bias_level(codec, SND_SOC_BIAS_OFF); - - return 0; -} - -static int sgtl5000_resume(struct snd_soc_codec *codec) -{ - /* Bring the codec back up to standby to enable regulators */ - sgtl5000_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - return 0; -} -#else -#define sgtl5000_suspend NULL -#define sgtl5000_resume NULL -#endif /* CONFIG_SUSPEND */ - /* * sgtl5000 has 3 internal power supplies: * 1. VAG, normally set to vdda/2 @@ -1352,11 +1335,6 @@ static int sgtl5000_probe(struct snd_soc_codec *codec) */ snd_soc_write(codec, SGTL5000_DAP_CTRL, 0); - /* leading to standby state */ - ret = sgtl5000_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - if (ret) - goto err; - return 0; err: @@ -1373,8 +1351,6 @@ static int sgtl5000_remove(struct snd_soc_codec *codec) { struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); - sgtl5000_set_bias_level(codec, SND_SOC_BIAS_OFF); - regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), sgtl5000->supplies); regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies), @@ -1387,9 +1363,8 @@ static int sgtl5000_remove(struct snd_soc_codec *codec) static struct snd_soc_codec_driver sgtl5000_driver = { .probe = sgtl5000_probe, .remove = sgtl5000_remove, - .suspend = sgtl5000_suspend, - .resume = sgtl5000_resume, .set_bias_level = sgtl5000_set_bias_level, + .suspend_bias_off = true, .controls = sgtl5000_snd_controls, .num_controls = ARRAY_SIZE(sgtl5000_snd_controls), .dapm_widgets = sgtl5000_dapm_widgets, @@ -1442,6 +1417,7 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, { struct sgtl5000_priv *sgtl5000; int ret, reg, rev; + unsigned int mclk; sgtl5000 = devm_kzalloc(&client->dev, sizeof(struct sgtl5000_priv), GFP_KERNEL); @@ -1465,6 +1441,14 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, return ret; } + /* SGTL5000 SYS_MCLK should be between 8 and 27 MHz */ + mclk = clk_get_rate(sgtl5000->mclk); + if (mclk < 8000000 || mclk > 27000000) { + dev_err(&client->dev, "Invalid SYS_CLK frequency: %u.%03uMHz\n", + mclk / 1000000, mclk / 1000 % 1000); + return -EINVAL; + } + ret = clk_prepare_enable(sgtl5000->mclk); if (ret) return ret; diff --git a/sound/soc/codecs/ssm2518.c b/sound/soc/codecs/ssm2518.c index e8680bea5f86..67ea55adb307 100644 --- a/sound/soc/codecs/ssm2518.c +++ b/sound/soc/codecs/ssm2518.c @@ -646,17 +646,6 @@ static struct snd_soc_dai_driver ssm2518_dai = { .ops = &ssm2518_dai_ops, }; -static int ssm2518_probe(struct snd_soc_codec *codec) -{ - return ssm2518_set_bias_level(codec, SND_SOC_BIAS_OFF); -} - -static int ssm2518_remove(struct snd_soc_codec *codec) -{ - ssm2518_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - static int ssm2518_set_sysclk(struct snd_soc_codec *codec, int clk_id, int source, unsigned int freq, int dir) { @@ -727,8 +716,6 @@ static int ssm2518_set_sysclk(struct snd_soc_codec *codec, int clk_id, } static struct snd_soc_codec_driver ssm2518_codec_driver = { - .probe = ssm2518_probe, - .remove = ssm2518_remove, .set_bias_level = ssm2518_set_bias_level, .set_sysclk = ssm2518_set_sysclk, .idle_bias_off = true, diff --git a/sound/soc/codecs/ssm2602-i2c.c b/sound/soc/codecs/ssm2602-i2c.c index abd63d537173..0d9779d6bfda 100644 --- a/sound/soc/codecs/ssm2602-i2c.c +++ b/sound/soc/codecs/ssm2602-i2c.c @@ -41,10 +41,19 @@ static const struct i2c_device_id ssm2602_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, ssm2602_i2c_id); +static const struct of_device_id ssm2602_of_match[] = { + { .compatible = "adi,ssm2602", }, + { .compatible = "adi,ssm2603", }, + { .compatible = "adi,ssm2604", }, + { } +}; +MODULE_DEVICE_TABLE(of, ssm2602_of_match); + static struct i2c_driver ssm2602_i2c_driver = { .driver = { .name = "ssm2602", .owner = THIS_MODULE, + .of_match_table = ssm2602_of_match, }, .probe = ssm2602_i2c_probe, .remove = ssm2602_i2c_remove, diff --git a/sound/soc/codecs/ssm2602-spi.c b/sound/soc/codecs/ssm2602-spi.c index 2bf55e24a7bb..b5df14fbe3ad 100644 --- a/sound/soc/codecs/ssm2602-spi.c +++ b/sound/soc/codecs/ssm2602-spi.c @@ -26,10 +26,17 @@ static int ssm2602_spi_remove(struct spi_device *spi) return 0; } +static const struct of_device_id ssm2602_of_match[] = { + { .compatible = "adi,ssm2602", }, + { } +}; +MODULE_DEVICE_TABLE(of, ssm2602_of_match); + static struct spi_driver ssm2602_spi_driver = { .driver = { .name = "ssm2602", .owner = THIS_MODULE, + .of_match_table = ssm2602_of_match, }, .probe = ssm2602_spi_probe, .remove = ssm2602_spi_remove, diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 484b3bbe8624..314eaece1b7d 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -192,7 +192,7 @@ static const struct snd_pcm_hw_constraint_list ssm2602_constraints_12288000 = { }; static const unsigned int ssm2602_rates_11289600[] = { - 8000, 44100, 88200, + 8000, 11025, 22050, 44100, 88200, }; static const struct snd_pcm_hw_constraint_list ssm2602_constraints_11289600 = { @@ -237,6 +237,16 @@ static const struct ssm2602_coeff ssm2602_coeff_table[] = { {18432000, 96000, SSM2602_COEFF_SRATE(0x7, 0x1, 0x0)}, {12000000, 96000, SSM2602_COEFF_SRATE(0x7, 0x0, 0x1)}, + /* 11.025k */ + {11289600, 11025, SSM2602_COEFF_SRATE(0xc, 0x0, 0x0)}, + {16934400, 11025, SSM2602_COEFF_SRATE(0xc, 0x1, 0x0)}, + {12000000, 11025, SSM2602_COEFF_SRATE(0xc, 0x1, 0x1)}, + + /* 22.05k */ + {11289600, 22050, SSM2602_COEFF_SRATE(0xd, 0x0, 0x0)}, + {16934400, 22050, SSM2602_COEFF_SRATE(0xd, 0x1, 0x0)}, + {12000000, 22050, SSM2602_COEFF_SRATE(0xd, 0x1, 0x1)}, + /* 44.1k */ {11289600, 44100, SSM2602_COEFF_SRATE(0x8, 0x0, 0x0)}, {16934400, 44100, SSM2602_COEFF_SRATE(0x8, 0x1, 0x0)}, @@ -467,7 +477,8 @@ static int ssm2602_set_bias_level(struct snd_soc_codec *codec, return 0; } -#define SSM2602_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\ +#define SSM2602_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\ SNDRV_PCM_RATE_96000) @@ -502,18 +513,11 @@ static struct snd_soc_dai_driver ssm2602_dai = { .symmetric_samplebits = 1, }; -static int ssm2602_suspend(struct snd_soc_codec *codec) -{ - ssm2602_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - static int ssm2602_resume(struct snd_soc_codec *codec) { struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); regcache_sync(ssm2602->regmap); - ssm2602_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; } @@ -586,27 +590,14 @@ static int ssm260x_codec_probe(struct snd_soc_codec *codec) break; } - if (ret) - return ret; - - ssm2602_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - return 0; -} - -/* remove everything here */ -static int ssm2602_remove(struct snd_soc_codec *codec) -{ - ssm2602_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; + return ret; } static struct snd_soc_codec_driver soc_codec_dev_ssm2602 = { .probe = ssm260x_codec_probe, - .remove = ssm2602_remove, - .suspend = ssm2602_suspend, .resume = ssm2602_resume, .set_bias_level = ssm2602_set_bias_level, + .suspend_bias_off = true, .controls = ssm260x_snd_controls, .num_controls = ARRAY_SIZE(ssm260x_snd_controls), @@ -647,7 +638,7 @@ int ssm2602_probe(struct device *dev, enum ssm2602_type type, return -ENOMEM; dev_set_drvdata(dev, ssm2602); - ssm2602->type = SSM2602; + ssm2602->type = type; ssm2602->regmap = regmap; return snd_soc_register_codec(dev, &soc_codec_dev_ssm2602, diff --git a/sound/soc/codecs/ssm4567.c b/sound/soc/codecs/ssm4567.c new file mode 100644 index 000000000000..4b5c17f8507e --- /dev/null +++ b/sound/soc/codecs/ssm4567.c @@ -0,0 +1,343 @@ +/* + * SSM4567 amplifier audio driver + * + * Copyright 2014 Google Chromium project. + * Author: Anatol Pomozov <anatol@chromium.org> + * + * Based on code copyright/by: + * Copyright 2013 Analog Devices Inc. + * + * Licensed under the GPL-2. + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#define SSM4567_REG_POWER_CTRL 0x00 +#define SSM4567_REG_AMP_SNS_CTRL 0x01 +#define SSM4567_REG_DAC_CTRL 0x02 +#define SSM4567_REG_DAC_VOLUME 0x03 +#define SSM4567_REG_SAI_CTRL_1 0x04 +#define SSM4567_REG_SAI_CTRL_2 0x05 +#define SSM4567_REG_SAI_PLACEMENT_1 0x06 +#define SSM4567_REG_SAI_PLACEMENT_2 0x07 +#define SSM4567_REG_SAI_PLACEMENT_3 0x08 +#define SSM4567_REG_SAI_PLACEMENT_4 0x09 +#define SSM4567_REG_SAI_PLACEMENT_5 0x0a +#define SSM4567_REG_SAI_PLACEMENT_6 0x0b +#define SSM4567_REG_BATTERY_V_OUT 0x0c +#define SSM4567_REG_LIMITER_CTRL_1 0x0d +#define SSM4567_REG_LIMITER_CTRL_2 0x0e +#define SSM4567_REG_LIMITER_CTRL_3 0x0f +#define SSM4567_REG_STATUS_1 0x10 +#define SSM4567_REG_STATUS_2 0x11 +#define SSM4567_REG_FAULT_CTRL 0x12 +#define SSM4567_REG_PDM_CTRL 0x13 +#define SSM4567_REG_MCLK_RATIO 0x14 +#define SSM4567_REG_BOOST_CTRL_1 0x15 +#define SSM4567_REG_BOOST_CTRL_2 0x16 +#define SSM4567_REG_SOFT_RESET 0xff + +/* POWER_CTRL */ +#define SSM4567_POWER_APWDN_EN BIT(7) +#define SSM4567_POWER_BSNS_PWDN BIT(6) +#define SSM4567_POWER_VSNS_PWDN BIT(5) +#define SSM4567_POWER_ISNS_PWDN BIT(4) +#define SSM4567_POWER_BOOST_PWDN BIT(3) +#define SSM4567_POWER_AMP_PWDN BIT(2) +#define SSM4567_POWER_VBAT_ONLY BIT(1) +#define SSM4567_POWER_SPWDN BIT(0) + +/* DAC_CTRL */ +#define SSM4567_DAC_HV BIT(7) +#define SSM4567_DAC_MUTE BIT(6) +#define SSM4567_DAC_HPF BIT(5) +#define SSM4567_DAC_LPM BIT(4) +#define SSM4567_DAC_FS_MASK 0x7 +#define SSM4567_DAC_FS_8000_12000 0x0 +#define SSM4567_DAC_FS_16000_24000 0x1 +#define SSM4567_DAC_FS_32000_48000 0x2 +#define SSM4567_DAC_FS_64000_96000 0x3 +#define SSM4567_DAC_FS_128000_192000 0x4 + +struct ssm4567 { + struct regmap *regmap; +}; + +static const struct reg_default ssm4567_reg_defaults[] = { + { SSM4567_REG_POWER_CTRL, 0x81 }, + { SSM4567_REG_AMP_SNS_CTRL, 0x09 }, + { SSM4567_REG_DAC_CTRL, 0x32 }, + { SSM4567_REG_DAC_VOLUME, 0x40 }, + { SSM4567_REG_SAI_CTRL_1, 0x00 }, + { SSM4567_REG_SAI_CTRL_2, 0x08 }, + { SSM4567_REG_SAI_PLACEMENT_1, 0x01 }, + { SSM4567_REG_SAI_PLACEMENT_2, 0x20 }, + { SSM4567_REG_SAI_PLACEMENT_3, 0x32 }, + { SSM4567_REG_SAI_PLACEMENT_4, 0x07 }, + { SSM4567_REG_SAI_PLACEMENT_5, 0x07 }, + { SSM4567_REG_SAI_PLACEMENT_6, 0x07 }, + { SSM4567_REG_BATTERY_V_OUT, 0x00 }, + { SSM4567_REG_LIMITER_CTRL_1, 0xa4 }, + { SSM4567_REG_LIMITER_CTRL_2, 0x73 }, + { SSM4567_REG_LIMITER_CTRL_3, 0x00 }, + { SSM4567_REG_STATUS_1, 0x00 }, + { SSM4567_REG_STATUS_2, 0x00 }, + { SSM4567_REG_FAULT_CTRL, 0x30 }, + { SSM4567_REG_PDM_CTRL, 0x40 }, + { SSM4567_REG_MCLK_RATIO, 0x11 }, + { SSM4567_REG_BOOST_CTRL_1, 0x03 }, + { SSM4567_REG_BOOST_CTRL_2, 0x00 }, + { SSM4567_REG_SOFT_RESET, 0x00 }, +}; + + +static bool ssm4567_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case SSM4567_REG_POWER_CTRL ... SSM4567_REG_BOOST_CTRL_2: + return true; + default: + return false; + } + +} + +static bool ssm4567_writeable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case SSM4567_REG_POWER_CTRL ... SSM4567_REG_SAI_PLACEMENT_6: + case SSM4567_REG_LIMITER_CTRL_1 ... SSM4567_REG_LIMITER_CTRL_3: + case SSM4567_REG_FAULT_CTRL ... SSM4567_REG_BOOST_CTRL_2: + /* The datasheet states that soft reset register is read-only, + * but logically it is write-only. */ + case SSM4567_REG_SOFT_RESET: + return true; + default: + return false; + } +} + +static bool ssm4567_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case SSM4567_REG_BATTERY_V_OUT: + case SSM4567_REG_STATUS_1 ... SSM4567_REG_STATUS_2: + case SSM4567_REG_SOFT_RESET: + return true; + default: + return false; + } +} + +static const DECLARE_TLV_DB_MINMAX_MUTE(ssm4567_vol_tlv, -7125, 2400); + +static const struct snd_kcontrol_new ssm4567_snd_controls[] = { + SOC_SINGLE_TLV("Master Playback Volume", SSM4567_REG_DAC_VOLUME, 0, + 0xff, 1, ssm4567_vol_tlv), + SOC_SINGLE("DAC Low Power Mode Switch", SSM4567_REG_DAC_CTRL, 4, 1, 0), +}; + +static const struct snd_soc_dapm_widget ssm4567_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC", "HiFi Playback", SSM4567_REG_POWER_CTRL, 2, 1), + + SND_SOC_DAPM_OUTPUT("OUT"), +}; + +static const struct snd_soc_dapm_route ssm4567_routes[] = { + { "OUT", NULL, "DAC" }, +}; + +static int ssm4567_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct ssm4567 *ssm4567 = snd_soc_codec_get_drvdata(codec); + unsigned int rate = params_rate(params); + unsigned int dacfs; + + if (rate >= 8000 && rate <= 12000) + dacfs = SSM4567_DAC_FS_8000_12000; + else if (rate >= 16000 && rate <= 24000) + dacfs = SSM4567_DAC_FS_16000_24000; + else if (rate >= 32000 && rate <= 48000) + dacfs = SSM4567_DAC_FS_32000_48000; + else if (rate >= 64000 && rate <= 96000) + dacfs = SSM4567_DAC_FS_64000_96000; + else if (rate >= 128000 && rate <= 192000) + dacfs = SSM4567_DAC_FS_128000_192000; + else + return -EINVAL; + + return regmap_update_bits(ssm4567->regmap, SSM4567_REG_DAC_CTRL, + SSM4567_DAC_FS_MASK, dacfs); +} + +static int ssm4567_mute(struct snd_soc_dai *dai, int mute) +{ + struct ssm4567 *ssm4567 = snd_soc_codec_get_drvdata(dai->codec); + unsigned int val; + + val = mute ? SSM4567_DAC_MUTE : 0; + return regmap_update_bits(ssm4567->regmap, SSM4567_REG_DAC_CTRL, + SSM4567_DAC_MUTE, val); +} + +static int ssm4567_set_power(struct ssm4567 *ssm4567, bool enable) +{ + int ret = 0; + + if (!enable) { + ret = regmap_update_bits(ssm4567->regmap, + SSM4567_REG_POWER_CTRL, + SSM4567_POWER_SPWDN, SSM4567_POWER_SPWDN); + regcache_mark_dirty(ssm4567->regmap); + } + + regcache_cache_only(ssm4567->regmap, !enable); + + if (enable) { + ret = regmap_update_bits(ssm4567->regmap, + SSM4567_REG_POWER_CTRL, + SSM4567_POWER_SPWDN, 0x00); + regcache_sync(ssm4567->regmap); + } + + return ret; +} + +static int ssm4567_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct ssm4567 *ssm4567 = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) + ret = ssm4567_set_power(ssm4567, true); + break; + case SND_SOC_BIAS_OFF: + ret = ssm4567_set_power(ssm4567, false); + break; + } + + if (ret) + return ret; + + codec->dapm.bias_level = level; + + return 0; +} + +static const struct snd_soc_dai_ops ssm4567_dai_ops = { + .hw_params = ssm4567_hw_params, + .digital_mute = ssm4567_mute, +}; + +static struct snd_soc_dai_driver ssm4567_dai = { + .name = "ssm4567-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32, + }, + .ops = &ssm4567_dai_ops, +}; + +static struct snd_soc_codec_driver ssm4567_codec_driver = { + .set_bias_level = ssm4567_set_bias_level, + .idle_bias_off = true, + + .controls = ssm4567_snd_controls, + .num_controls = ARRAY_SIZE(ssm4567_snd_controls), + .dapm_widgets = ssm4567_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ssm4567_dapm_widgets), + .dapm_routes = ssm4567_routes, + .num_dapm_routes = ARRAY_SIZE(ssm4567_routes), +}; + +static const struct regmap_config ssm4567_regmap_config = { + .val_bits = 8, + .reg_bits = 8, + + .max_register = SSM4567_REG_SOFT_RESET, + .readable_reg = ssm4567_readable_reg, + .writeable_reg = ssm4567_writeable_reg, + .volatile_reg = ssm4567_volatile_reg, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = ssm4567_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(ssm4567_reg_defaults), +}; + +static int ssm4567_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct ssm4567 *ssm4567; + int ret; + + ssm4567 = devm_kzalloc(&i2c->dev, sizeof(*ssm4567), GFP_KERNEL); + if (ssm4567 == NULL) + return -ENOMEM; + + i2c_set_clientdata(i2c, ssm4567); + + ssm4567->regmap = devm_regmap_init_i2c(i2c, &ssm4567_regmap_config); + if (IS_ERR(ssm4567->regmap)) + return PTR_ERR(ssm4567->regmap); + + ret = regmap_write(ssm4567->regmap, SSM4567_REG_SOFT_RESET, 0x00); + if (ret) + return ret; + + ret = ssm4567_set_power(ssm4567, false); + if (ret) + return ret; + + return snd_soc_register_codec(&i2c->dev, &ssm4567_codec_driver, + &ssm4567_dai, 1); +} + +static int ssm4567_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id ssm4567_i2c_ids[] = { + { "ssm4567", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ssm4567_i2c_ids); + +static struct i2c_driver ssm4567_driver = { + .driver = { + .name = "ssm4567", + .owner = THIS_MODULE, + }, + .probe = ssm4567_i2c_probe, + .remove = ssm4567_i2c_remove, + .id_table = ssm4567_i2c_ids, +}; +module_i2c_driver(ssm4567_driver); + +MODULE_DESCRIPTION("ASoC SSM4567 driver"); +MODULE_AUTHOR("Anatol Pomozov <anatol@chromium.org>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/sta529.c b/sound/soc/codecs/sta529.c index 9aa1323fb2ab..89c748dd3d6e 100644 --- a/sound/soc/codecs/sta529.c +++ b/sound/soc/codecs/sta529.c @@ -4,7 +4,7 @@ * sound/soc/codecs/sta529.c -- spear ALSA Soc codec driver * * Copyright (C) 2012 ST Microelectronics - * Rajeev Kumar <rajeev-dlh.kumar@st.com> + * Rajeev Kumar <rajeevkumar.linux@gmail.com> * * This file is licensed under the terms of the GNU General Public * License version 2. This program is licensed "as is" without any @@ -426,5 +426,5 @@ static struct i2c_driver sta529_i2c_driver = { module_i2c_driver(sta529_i2c_driver); MODULE_DESCRIPTION("ASoC STA529 codec driver"); -MODULE_AUTHOR("Rajeev Kumar <rajeev-dlh.kumar@st.com>"); +MODULE_AUTHOR("Rajeev Kumar <rajeevkumar.linux@gmail.com>"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c index 23b32960ff1d..f039dc825971 100644 --- a/sound/soc/codecs/tas2552.c +++ b/sound/soc/codecs/tas2552.c @@ -78,6 +78,44 @@ struct tas2552_data { unsigned int mclk; }; +/* Input mux controls */ +static const char *tas2552_input_texts[] = { + "Digital", "Analog" +}; + +static SOC_ENUM_SINGLE_DECL(tas2552_input_mux_enum, TAS2552_CFG_3, 7, + tas2552_input_texts); + +static const struct snd_kcontrol_new tas2552_input_mux_control[] = { + SOC_DAPM_ENUM("Input selection", tas2552_input_mux_enum) +}; + +static const struct snd_soc_dapm_widget tas2552_dapm_widgets[] = +{ + SND_SOC_DAPM_INPUT("IN"), + + /* MUX Controls */ + SND_SOC_DAPM_MUX("Input selection", SND_SOC_NOPM, 0, 0, + tas2552_input_mux_control), + + SND_SOC_DAPM_AIF_IN("DAC IN", "DAC Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DAC", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_OUT_DRV("ClassD", TAS2552_CFG_2, 7, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL", TAS2552_CFG_2, 3, 0, NULL, 0), + + SND_SOC_DAPM_OUTPUT("OUT") +}; + +static const struct snd_soc_dapm_route tas2552_audio_map[] = { + {"DAC", NULL, "DAC IN"}, + {"Input selection", "Digital", "DAC"}, + {"Input selection", "Analog", "IN"}, + {"ClassD", NULL, "Input selection"}, + {"OUT", NULL, "ClassD"}, + {"ClassD", NULL, "PLL"}, +}; + +#ifdef CONFIG_PM_RUNTIME static void tas2552_sw_shutdown(struct tas2552_data *tas_data, int sw_shutdown) { u8 cfg1_reg; @@ -90,6 +128,7 @@ static void tas2552_sw_shutdown(struct tas2552_data *tas_data, int sw_shutdown) snd_soc_update_bits(tas_data->codec, TAS2552_CFG_1, TAS2552_SWS_MASK, cfg1_reg); } +#endif static int tas2552_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, @@ -101,10 +140,6 @@ static int tas2552_hw_params(struct snd_pcm_substream *substream, int d; u8 p, j; - /* Turn on Class D amplifier */ - snd_soc_update_bits(codec, TAS2552_CFG_2, TAS2552_CLASSD_EN_MASK, - TAS2552_CLASSD_EN); - if (!tas2552->mclk) return -EINVAL; @@ -147,9 +182,6 @@ static int tas2552_hw_params(struct snd_pcm_substream *substream, } - snd_soc_update_bits(codec, TAS2552_CFG_2, TAS2552_PLL_ENABLE, - TAS2552_PLL_ENABLE); - return 0; } @@ -269,19 +301,10 @@ static const struct dev_pm_ops tas2552_pm = { NULL) }; -static void tas2552_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_codec *codec = dai->codec; - - snd_soc_update_bits(codec, TAS2552_CFG_2, TAS2552_PLL_ENABLE, 0); -} - static struct snd_soc_dai_ops tas2552_speaker_dai_ops = { .hw_params = tas2552_hw_params, .set_sysclk = tas2552_set_dai_sysclk, .set_fmt = tas2552_set_dai_fmt, - .shutdown = tas2552_shutdown, .digital_mute = tas2552_mute, }; @@ -294,7 +317,7 @@ static struct snd_soc_dai_driver tas2552_dai[] = { { .name = "tas2552-amplifier", .playback = { - .stream_name = "Speaker", + .stream_name = "Playback", .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_192000, @@ -312,6 +335,7 @@ static DECLARE_TLV_DB_SCALE(dac_tlv, -7, 100, 24); static const struct snd_kcontrol_new tas2552_snd_controls[] = { SOC_SINGLE_TLV("Speaker Driver Playback Volume", TAS2552_PGA_GAIN, 0, 0x1f, 1, dac_tlv), + SOC_DAPM_SINGLE("Playback AMP", SND_SOC_NOPM, 0, 1, 0), }; static const struct reg_default tas2552_init_regs[] = { @@ -321,6 +345,7 @@ static const struct reg_default tas2552_init_regs[] = { static int tas2552_codec_probe(struct snd_soc_codec *codec) { struct tas2552_data *tas2552 = snd_soc_codec_get_drvdata(codec); + struct snd_soc_dapm_context *dapm = &codec->dapm; int ret; tas2552->codec = codec; @@ -362,9 +387,14 @@ static int tas2552_codec_probe(struct snd_soc_codec *codec) goto patch_fail; } - snd_soc_write(codec, TAS2552_CFG_2, TAS2552_CLASSD_EN | - TAS2552_BOOST_EN | TAS2552_APT_EN | - TAS2552_LIM_EN); + snd_soc_write(codec, TAS2552_CFG_2, TAS2552_BOOST_EN | + TAS2552_APT_EN | TAS2552_LIM_EN); + + snd_soc_dapm_new_controls(dapm, tas2552_dapm_widgets, + ARRAY_SIZE(tas2552_dapm_widgets)); + snd_soc_dapm_add_routes(dapm, tas2552_audio_map, + ARRAY_SIZE(tas2552_audio_map)); + return 0; patch_fail: diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index 0f64c7890eed..145fe5b253d4 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -167,13 +167,13 @@ struct aic31xx_priv { struct regulator_bulk_data supplies[AIC31XX_NUM_SUPPLIES]; struct aic31xx_disable_nb disable_nb[AIC31XX_NUM_SUPPLIES]; unsigned int sysclk; + u8 p_div; int rate_div_line; }; struct aic31xx_rate_divs { - u32 mclk; + u32 mclk_p; u32 rate; - u8 p_val; u8 pll_j; u16 pll_d; u16 dosr; @@ -186,51 +186,51 @@ struct aic31xx_rate_divs { /* ADC dividers can be disabled by cofiguring them to 0 */ static const struct aic31xx_rate_divs aic31xx_divs[] = { - /* mclk rate pll: p j d dosr ndac mdac aors nadc madc */ + /* mclk/p rate pll: j d dosr ndac mdac aors nadc madc */ /* 8k rate */ - {12000000, 8000, 1, 8, 1920, 128, 48, 2, 128, 48, 2}, - {24000000, 8000, 2, 8, 1920, 128, 48, 2, 128, 48, 2}, - {25000000, 8000, 2, 7, 8643, 128, 48, 2, 128, 48, 2}, + {12000000, 8000, 8, 1920, 128, 48, 2, 128, 48, 2}, + {12000000, 8000, 8, 1920, 128, 32, 3, 128, 32, 3}, + {12500000, 8000, 7, 8643, 128, 48, 2, 128, 48, 2}, /* 11.025k rate */ - {12000000, 11025, 1, 7, 5264, 128, 32, 2, 128, 32, 2}, - {24000000, 11025, 2, 7, 5264, 128, 32, 2, 128, 32, 2}, - {25000000, 11025, 2, 7, 2253, 128, 32, 2, 128, 32, 2}, + {12000000, 11025, 7, 5264, 128, 32, 2, 128, 32, 2}, + {12000000, 11025, 8, 4672, 128, 24, 3, 128, 24, 3}, + {12500000, 11025, 7, 2253, 128, 32, 2, 128, 32, 2}, /* 16k rate */ - {12000000, 16000, 1, 8, 1920, 128, 24, 2, 128, 24, 2}, - {24000000, 16000, 2, 8, 1920, 128, 24, 2, 128, 24, 2}, - {25000000, 16000, 2, 7, 8643, 128, 24, 2, 128, 24, 2}, + {12000000, 16000, 8, 1920, 128, 24, 2, 128, 24, 2}, + {12000000, 16000, 8, 1920, 128, 16, 3, 128, 16, 3}, + {12500000, 16000, 7, 8643, 128, 24, 2, 128, 24, 2}, /* 22.05k rate */ - {12000000, 22050, 1, 7, 5264, 128, 16, 2, 128, 16, 2}, - {24000000, 22050, 2, 7, 5264, 128, 16, 2, 128, 16, 2}, - {25000000, 22050, 2, 7, 2253, 128, 16, 2, 128, 16, 2}, + {12000000, 22050, 7, 5264, 128, 16, 2, 128, 16, 2}, + {12000000, 22050, 8, 4672, 128, 12, 3, 128, 12, 3}, + {12500000, 22050, 7, 2253, 128, 16, 2, 128, 16, 2}, /* 32k rate */ - {12000000, 32000, 1, 8, 1920, 128, 12, 2, 128, 12, 2}, - {24000000, 32000, 2, 8, 1920, 128, 12, 2, 128, 12, 2}, - {25000000, 32000, 2, 7, 8643, 128, 12, 2, 128, 12, 2}, + {12000000, 32000, 8, 1920, 128, 12, 2, 128, 12, 2}, + {12000000, 32000, 8, 1920, 128, 8, 3, 128, 8, 3}, + {12500000, 32000, 7, 8643, 128, 12, 2, 128, 12, 2}, /* 44.1k rate */ - {12000000, 44100, 1, 7, 5264, 128, 8, 2, 128, 8, 2}, - {24000000, 44100, 2, 7, 5264, 128, 8, 2, 128, 8, 2}, - {25000000, 44100, 2, 7, 2253, 128, 8, 2, 128, 8, 2}, + {12000000, 44100, 7, 5264, 128, 8, 2, 128, 8, 2}, + {12000000, 44100, 8, 4672, 128, 6, 3, 128, 6, 3}, + {12500000, 44100, 7, 2253, 128, 8, 2, 128, 8, 2}, /* 48k rate */ - {12000000, 48000, 1, 8, 1920, 128, 8, 2, 128, 8, 2}, - {24000000, 48000, 2, 8, 1920, 128, 8, 2, 128, 8, 2}, - {25000000, 48000, 2, 7, 8643, 128, 8, 2, 128, 8, 2}, + {12000000, 48000, 8, 1920, 128, 8, 2, 128, 8, 2}, + {12000000, 48000, 7, 6800, 96, 5, 4, 96, 5, 4}, + {12500000, 48000, 7, 8643, 128, 8, 2, 128, 8, 2}, /* 88.2k rate */ - {12000000, 88200, 1, 7, 5264, 64, 8, 2, 64, 8, 2}, - {24000000, 88200, 2, 7, 5264, 64, 8, 2, 64, 8, 2}, - {25000000, 88200, 2, 7, 2253, 64, 8, 2, 64, 8, 2}, + {12000000, 88200, 7, 5264, 64, 8, 2, 64, 8, 2}, + {12000000, 88200, 8, 4672, 64, 6, 3, 64, 6, 3}, + {12500000, 88200, 7, 2253, 64, 8, 2, 64, 8, 2}, /* 96k rate */ - {12000000, 96000, 1, 8, 1920, 64, 8, 2, 64, 8, 2}, - {24000000, 96000, 2, 8, 1920, 64, 8, 2, 64, 8, 2}, - {25000000, 96000, 2, 7, 8643, 64, 8, 2, 64, 8, 2}, + {12000000, 96000, 8, 1920, 64, 8, 2, 64, 8, 2}, + {12000000, 96000, 7, 6800, 48, 5, 4, 48, 5, 4}, + {12500000, 96000, 7, 8643, 64, 8, 2, 64, 8, 2}, /* 176.4k rate */ - {12000000, 176400, 1, 7, 5264, 32, 8, 2, 32, 8, 2}, - {24000000, 176400, 2, 7, 5264, 32, 8, 2, 32, 8, 2}, - {25000000, 176400, 2, 7, 2253, 32, 8, 2, 32, 8, 2}, + {12000000, 176400, 7, 5264, 32, 8, 2, 32, 8, 2}, + {12000000, 176400, 8, 4672, 32, 6, 3, 32, 6, 3}, + {12500000, 176400, 7, 2253, 32, 8, 2, 32, 8, 2}, /* 192k rate */ - {12000000, 192000, 1, 8, 1920, 32, 8, 2, 32, 8, 2}, - {24000000, 192000, 2, 8, 1920, 32, 8, 2, 32, 8, 2}, - {25000000, 192000, 2, 7, 8643, 32, 8, 2, 32, 8, 2}, + {12000000, 192000, 8, 1920, 32, 8, 2, 32, 8, 2}, + {12000000, 192000, 7, 6800, 24, 5, 4, 24, 5, 4}, + {12500000, 192000, 7, 8643, 32, 8, 2, 32, 8, 2}, }; static const char * const ldac_in_text[] = { @@ -680,7 +680,10 @@ static int aic31xx_setup_pll(struct snd_soc_codec *codec, struct snd_pcm_hw_params *params) { struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + int bclk_score = snd_soc_params_to_frame_size(params); + int mclk_p = aic31xx->sysclk / aic31xx->p_div; int bclk_n = 0; + int match = -1; int i; /* Use PLL as CODEC_CLKIN and DAC_CLK as BDIV_CLKIN */ @@ -691,19 +694,41 @@ static int aic31xx_setup_pll(struct snd_soc_codec *codec, for (i = 0; i < ARRAY_SIZE(aic31xx_divs); i++) { if (aic31xx_divs[i].rate == params_rate(params) && - aic31xx_divs[i].mclk == aic31xx->sysclk) - break; + aic31xx_divs[i].mclk_p == mclk_p) { + int s = (aic31xx_divs[i].dosr * aic31xx_divs[i].mdac) % + snd_soc_params_to_frame_size(params); + int bn = (aic31xx_divs[i].dosr * aic31xx_divs[i].mdac) / + snd_soc_params_to_frame_size(params); + if (s < bclk_score && bn > 0) { + match = i; + bclk_n = bn; + bclk_score = s; + } + } } - if (i == ARRAY_SIZE(aic31xx_divs)) { - dev_err(codec->dev, "%s: Sampling rate %u not supported\n", + if (match == -1) { + dev_err(codec->dev, + "%s: Sample rate (%u) and format not supported\n", __func__, params_rate(params)); + /* See bellow for details how fix this. */ return -EINVAL; } + if (bclk_score != 0) { + dev_warn(codec->dev, "Can not produce exact bitclock"); + /* This is fine if using dsp format, but if using i2s + there may be trouble. To fix the issue edit the + aic31xx_divs table for your mclk and sample + rate. Details can be found from: + http://www.ti.com/lit/ds/symlink/tlv320aic3100.pdf + Section: 5.6 CLOCK Generation and PLL + */ + } + i = match; /* PLL configuration */ snd_soc_update_bits(codec, AIC31XX_PLLPR, AIC31XX_PLL_MASK, - (aic31xx_divs[i].p_val << 4) | 0x01); + (aic31xx->p_div << 4) | 0x01); snd_soc_write(codec, AIC31XX_PLLJ, aic31xx_divs[i].pll_j); snd_soc_write(codec, AIC31XX_PLLDMSB, @@ -729,14 +754,6 @@ static int aic31xx_setup_pll(struct snd_soc_codec *codec, snd_soc_write(codec, AIC31XX_AOSR, aic31xx_divs[i].aosr); /* Bit clock divider configuration. */ - bclk_n = (aic31xx_divs[i].dosr * aic31xx_divs[i].mdac) - / snd_soc_params_to_frame_size(params); - if (bclk_n == 0) { - dev_err(codec->dev, "%s: Not enough BLCK bandwidth\n", - __func__); - return -EINVAL; - } - snd_soc_update_bits(codec, AIC31XX_BCLKN, AIC31XX_PLL_MASK, bclk_n); @@ -745,7 +762,7 @@ static int aic31xx_setup_pll(struct snd_soc_codec *codec, dev_dbg(codec->dev, "pll %d.%04d/%d dosr %d n %d m %d aosr %d n %d m %d bclk_n %d\n", aic31xx_divs[i].pll_j, aic31xx_divs[i].pll_d, - aic31xx_divs[i].p_val, aic31xx_divs[i].dosr, + aic31xx->p_div, aic31xx_divs[i].dosr, aic31xx_divs[i].ndac, aic31xx_divs[i].mdac, aic31xx_divs[i].aosr, aic31xx_divs[i].nadc, aic31xx_divs[i].madc, bclk_n); @@ -813,7 +830,7 @@ static int aic31xx_set_dai_fmt(struct snd_soc_dai *codec_dai, { struct snd_soc_codec *codec = codec_dai->codec; u8 iface_reg1 = 0; - u8 iface_reg3 = 0; + u8 iface_reg2 = 0; u8 dsp_a_val = 0; dev_dbg(codec->dev, "## %s: fmt = 0x%x\n", __func__, fmt); @@ -838,7 +855,7 @@ static int aic31xx_set_dai_fmt(struct snd_soc_dai *codec_dai, /* NOTE: BCLKINV bit value 1 equas NB and 0 equals IB */ switch (fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_NB_NF: - iface_reg3 |= AIC31XX_BCLKINV_MASK; + iface_reg2 |= AIC31XX_BCLKINV_MASK; break; case SND_SOC_DAIFMT_IB_NF: break; @@ -870,7 +887,7 @@ static int aic31xx_set_dai_fmt(struct snd_soc_dai *codec_dai, dsp_a_val); snd_soc_update_bits(codec, AIC31XX_IFACE2, AIC31XX_BCLKINV_MASK, - iface_reg3); + iface_reg2); return 0; } @@ -885,7 +902,16 @@ static int aic31xx_set_dai_sysclk(struct snd_soc_dai *codec_dai, dev_dbg(codec->dev, "## %s: clk_id = %d, freq = %d, dir = %d\n", __func__, clk_id, freq, dir); - for (i = 0; aic31xx_divs[i].mclk != freq; i++) { + for (i = 1; freq/i > 20000000 && i < 8; i++) + ; + if (freq/i > 20000000) { + dev_err(aic31xx->dev, "%s: Too high mclk frequency %u\n", + __func__, freq); + return -EINVAL; + } + aic31xx->p_div = i; + + for (i = 0; aic31xx_divs[i].mclk_p != freq/aic31xx->p_div; i++) { if (i == ARRAY_SIZE(aic31xx_divs)) { dev_err(aic31xx->dev, "%s: Unsupported frequency %d\n", __func__, freq); diff --git a/sound/soc/codecs/tlv320aic31xx.h b/sound/soc/codecs/tlv320aic31xx.h index 52ed57c69dfa..fe16c34607bb 100644 --- a/sound/soc/codecs/tlv320aic31xx.h +++ b/sound/soc/codecs/tlv320aic31xx.h @@ -18,7 +18,8 @@ #define AIC31XX_RATES SNDRV_PCM_RATE_8000_192000 #define AIC31XX_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE \ - | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) + | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_LE \ + | SNDRV_PCM_FMTBIT_S32_LE) #define AIC31XX_STEREO_CLASS_D_BIT 0x1 diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 64f179ee9834..f7c2a575a892 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1121,6 +1121,7 @@ static int aic3x_regulator_event(struct notifier_block *nb, static int aic3x_set_power(struct snd_soc_codec *codec, int power) { struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); + unsigned int pll_c, pll_d; int ret; if (power) { @@ -1138,6 +1139,18 @@ static int aic3x_set_power(struct snd_soc_codec *codec, int power) /* Sync reg_cache with the hardware */ regcache_cache_only(aic3x->regmap, false); regcache_sync(aic3x->regmap); + + /* Rewrite paired PLL D registers in case cached sync skipped + * writing one of them and thus caused other one also not + * being written + */ + pll_c = snd_soc_read(codec, AIC3X_PLL_PROGC_REG); + pll_d = snd_soc_read(codec, AIC3X_PLL_PROGD_REG); + if (pll_c == aic3x_reg[AIC3X_PLL_PROGC_REG].def || + pll_d == aic3x_reg[AIC3X_PLL_PROGD_REG].def) { + snd_soc_write(codec, AIC3X_PLL_PROGC_REG, pll_c); + snd_soc_write(codec, AIC3X_PLL_PROGD_REG, pll_d); + } } else { /* * Do soft reset to this codec instance in order to clear @@ -1222,20 +1235,6 @@ static struct snd_soc_dai_driver aic3x_dai = { .symmetric_rates = 1, }; -static int aic3x_suspend(struct snd_soc_codec *codec) -{ - aic3x_set_bias_level(codec, SND_SOC_BIAS_OFF); - - return 0; -} - -static int aic3x_resume(struct snd_soc_codec *codec) -{ - aic3x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - return 0; -} - static void aic3x_mono_init(struct snd_soc_codec *codec) { /* DAC to Mono Line Out default volume and route to Output mixer */ @@ -1429,8 +1428,6 @@ static struct snd_soc_codec_driver soc_codec_dev_aic3x = { .idle_bias_off = true, .probe = aic3x_probe, .remove = aic3x_remove, - .suspend = aic3x_suspend, - .resume = aic3x_resume, .controls = aic3x_snd_controls, .num_controls = ARRAY_SIZE(aic3x_snd_controls), .dapm_widgets = aic3x_dapm_widgets, diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 7bb0d36d4c54..a01ad629ed61 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -2319,11 +2319,8 @@ static void wm5100_init_gpio(struct i2c_client *i2c) static void wm5100_free_gpio(struct i2c_client *i2c) { struct wm5100_priv *wm5100 = i2c_get_clientdata(i2c); - int ret; - ret = gpiochip_remove(&wm5100->gpio_chip); - if (ret != 0) - dev_err(&i2c->dev, "Failed to remove GPIOs: %d\n", ret); + gpiochip_remove(&wm5100->gpio_chip); } #else static void wm5100_init_gpio(struct i2c_client *i2c) diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 3dfdcc4197fa..628ec774cf22 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -212,7 +212,7 @@ static void wm8350_pga_work(struct work_struct *work) { struct snd_soc_dapm_context *dapm = container_of(work, struct snd_soc_dapm_context, delayed_work.work); - struct snd_soc_codec *codec = dapm->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm); struct wm8350_data *wm8350_data = snd_soc_codec_get_drvdata(codec); struct wm8350_output *out1 = &wm8350_data->out1, *out2 = &wm8350_data->out2; diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index a237f1627f61..31bb4801a005 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -413,7 +413,6 @@ static int wm8741_resume(struct snd_soc_codec *codec) return 0; } #else -#define wm8741_suspend NULL #define wm8741_resume NULL #endif diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index e54e097f4fcb..21ca3a94fc96 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1433,7 +1433,7 @@ static void wm8753_work(struct work_struct *work) struct snd_soc_dapm_context *dapm = container_of(work, struct snd_soc_dapm_context, delayed_work.work); - struct snd_soc_codec *codec = dapm->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm); wm8753_set_bias_level(codec, dapm->bias_level); } diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index 0ea01dfcb6e1..3addc5fe5cb2 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -518,23 +518,6 @@ static int wm8804_set_bias_level(struct snd_soc_codec *codec, return 0; } -#ifdef CONFIG_PM -static int wm8804_suspend(struct snd_soc_codec *codec) -{ - wm8804_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - -static int wm8804_resume(struct snd_soc_codec *codec) -{ - wm8804_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; -} -#else -#define wm8804_suspend NULL -#define wm8804_resume NULL -#endif - static int wm8804_remove(struct snd_soc_codec *codec) { struct wm8804_priv *wm8804; @@ -671,8 +654,6 @@ static struct snd_soc_dai_driver wm8804_dai = { static struct snd_soc_codec_driver soc_codec_dev_wm8804 = { .probe = wm8804_probe, .remove = wm8804_remove, - .suspend = wm8804_suspend, - .resume = wm8804_resume, .set_bias_level = wm8804_set_bias_level, .idle_bias_off = true, diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index aa0984864e76..c038b3e04398 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1877,11 +1877,7 @@ static void wm8903_init_gpio(struct wm8903_priv *wm8903) static void wm8903_free_gpio(struct wm8903_priv *wm8903) { - int ret; - - ret = gpiochip_remove(&wm8903->gpio_chip); - if (ret != 0) - dev_err(wm8903->dev, "Failed to remove GPIOs: %d\n", ret); + gpiochip_remove(&wm8903->gpio_chip); } #else static void wm8903_init_gpio(struct wm8903_priv *wm8903) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 1098ae32f1f9..9077411e62ce 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3398,11 +3398,8 @@ static void wm8962_init_gpio(struct snd_soc_codec *codec) static void wm8962_free_gpio(struct snd_soc_codec *codec) { struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); - int ret; - ret = gpiochip_remove(&wm8962->gpio_chip); - if (ret != 0) - dev_err(codec->dev, "Failed to remove GPIOs: %d\n", ret); + gpiochip_remove(&wm8962->gpio_chip); } #else static void wm8962_init_gpio(struct snd_soc_codec *codec) diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index 0499cd4cfb71..39ddb9b8834c 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -615,7 +615,7 @@ static void wm8971_work(struct work_struct *work) struct snd_soc_dapm_context *dapm = container_of(work, struct snd_soc_dapm_context, delayed_work.work); - struct snd_soc_codec *codec = dapm->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm); wm8971_set_bias_level(codec, codec->dapm.bias_level); } diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 6cc0566dc29a..1fcb9f3f3097 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -4082,17 +4082,23 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) switch (control->type) { case WM8994: - if (wm8994->micdet_irq) { + if (wm8994->micdet_irq) ret = request_threaded_irq(wm8994->micdet_irq, NULL, wm8994_mic_irq, IRQF_TRIGGER_RISING, "Mic1 detect", wm8994); - if (ret != 0) - dev_warn(codec->dev, - "Failed to request Mic1 detect IRQ: %d\n", - ret); - } + else + ret = wm8994_request_irq(wm8994->wm8994, + WM8994_IRQ_MIC1_DET, + wm8994_mic_irq, "Mic 1 detect", + wm8994); + + if (ret != 0) + dev_warn(codec->dev, + "Failed to request Mic1 detect IRQ: %d\n", + ret); + ret = wm8994_request_irq(wm8994->wm8994, WM8994_IRQ_MIC1_SHRT, diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index cae4ac5a5730..1288edeb8c7d 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -1998,23 +1998,6 @@ static int wm8995_set_bias_level(struct snd_soc_codec *codec, return 0; } -#ifdef CONFIG_PM -static int wm8995_suspend(struct snd_soc_codec *codec) -{ - wm8995_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - -static int wm8995_resume(struct snd_soc_codec *codec) -{ - wm8995_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; -} -#else -#define wm8995_suspend NULL -#define wm8995_resume NULL -#endif - static int wm8995_remove(struct snd_soc_codec *codec) { struct wm8995_priv *wm8995; @@ -2220,8 +2203,6 @@ static struct snd_soc_dai_driver wm8995_dai[] = { static struct snd_soc_codec_driver soc_codec_dev_wm8995 = { .probe = wm8995_probe, .remove = wm8995_remove, - .suspend = wm8995_suspend, - .resume = wm8995_resume, .set_bias_level = wm8995_set_bias_level, .idle_bias_off = true, }; diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index f16ff4f56923..b1dcc11c1b23 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -2216,11 +2216,7 @@ static void wm8996_init_gpio(struct wm8996_priv *wm8996) static void wm8996_free_gpio(struct wm8996_priv *wm8996) { - int ret; - - ret = gpiochip_remove(&wm8996->gpio_chip); - if (ret != 0) - dev_err(wm8996->dev, "Failed to remove GPIOs: %d\n", ret); + gpiochip_remove(&wm8996->gpio_chip); } #else static void wm8996_init_gpio(struct wm8996_priv *wm8996) diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index d69510c53239..8e948c63f3d9 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -63,7 +63,8 @@ config SND_DM365_AIC3X_CODEC Say Y if you want to add support for AIC3101 audio codec config SND_DM365_VOICE_CODEC - bool "Voice Codec - CQ93VC" + tristate "Voice Codec - CQ93VC" + depends on SND_DAVINCI_SOC select MFD_DAVINCI_VOICECODEC select SND_DAVINCI_SOC_VCIF select SND_SOC_CQ0093VC diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 6a6b2ff7d7d7..0eed9b1b24e1 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -42,14 +42,26 @@ #define MCASP_MAX_AFIFO_DEPTH 64 +static u32 context_regs[] = { + DAVINCI_MCASP_TXFMCTL_REG, + DAVINCI_MCASP_RXFMCTL_REG, + DAVINCI_MCASP_TXFMT_REG, + DAVINCI_MCASP_RXFMT_REG, + DAVINCI_MCASP_ACLKXCTL_REG, + DAVINCI_MCASP_ACLKRCTL_REG, + DAVINCI_MCASP_AHCLKXCTL_REG, + DAVINCI_MCASP_AHCLKRCTL_REG, + DAVINCI_MCASP_PDIR_REG, + DAVINCI_MCASP_RXMASK_REG, + DAVINCI_MCASP_TXMASK_REG, + DAVINCI_MCASP_RXTDM_REG, + DAVINCI_MCASP_TXTDM_REG, +}; + struct davinci_mcasp_context { - u32 txfmtctl; - u32 rxfmtctl; - u32 txfmt; - u32 rxfmt; - u32 aclkxctl; - u32 aclkrctl; - u32 pdir; + u32 config_regs[ARRAY_SIZE(context_regs)]; + u32 afifo_regs[2]; /* for read/write fifo control registers */ + u32 *xrsr_regs; /* for serializer configuration */ }; struct davinci_mcasp { @@ -467,8 +479,17 @@ static int davinci_config_channel_size(struct davinci_mcasp *mcasp, { u32 fmt; u32 tx_rotate = (word_length / 4) & 0x7; - u32 rx_rotate = (32 - word_length) / 4; u32 mask = (1ULL << word_length) - 1; + /* + * For captured data we should not rotate, inversion and masking is + * enoguh to get the data to the right position: + * Format data from bus after reverse (XRBUF) + * S16_LE: |LSB|MSB|xxx|xxx| |xxx|xxx|MSB|LSB| + * S24_3LE: |LSB|DAT|MSB|xxx| |xxx|MSB|DAT|LSB| + * S24_LE: |LSB|DAT|MSB|xxx| |xxx|MSB|DAT|LSB| + * S32_LE: |LSB|DAT|DAT|MSB| |MSB|DAT|DAT|LSB| + */ + u32 rx_rotate = 0; /* * if s BCLK-to-LRCLK ratio has been configured via the set_clkdiv() @@ -865,14 +886,24 @@ static int davinci_mcasp_suspend(struct snd_soc_dai *dai) { struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai); struct davinci_mcasp_context *context = &mcasp->context; + u32 reg; + int i; + + for (i = 0; i < ARRAY_SIZE(context_regs); i++) + context->config_regs[i] = mcasp_get_reg(mcasp, context_regs[i]); + + if (mcasp->txnumevt) { + reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET; + context->afifo_regs[0] = mcasp_get_reg(mcasp, reg); + } + if (mcasp->rxnumevt) { + reg = mcasp->fifo_base + MCASP_RFIFOCTL_OFFSET; + context->afifo_regs[1] = mcasp_get_reg(mcasp, reg); + } - context->txfmtctl = mcasp_get_reg(mcasp, DAVINCI_MCASP_TXFMCTL_REG); - context->rxfmtctl = mcasp_get_reg(mcasp, DAVINCI_MCASP_RXFMCTL_REG); - context->txfmt = mcasp_get_reg(mcasp, DAVINCI_MCASP_TXFMT_REG); - context->rxfmt = mcasp_get_reg(mcasp, DAVINCI_MCASP_RXFMT_REG); - context->aclkxctl = mcasp_get_reg(mcasp, DAVINCI_MCASP_ACLKXCTL_REG); - context->aclkrctl = mcasp_get_reg(mcasp, DAVINCI_MCASP_ACLKRCTL_REG); - context->pdir = mcasp_get_reg(mcasp, DAVINCI_MCASP_PDIR_REG); + for (i = 0; i < mcasp->num_serializer; i++) + context->xrsr_regs[i] = mcasp_get_reg(mcasp, + DAVINCI_MCASP_XRSRCTL_REG(i)); return 0; } @@ -881,14 +912,24 @@ static int davinci_mcasp_resume(struct snd_soc_dai *dai) { struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai); struct davinci_mcasp_context *context = &mcasp->context; + u32 reg; + int i; - mcasp_set_reg(mcasp, DAVINCI_MCASP_TXFMCTL_REG, context->txfmtctl); - mcasp_set_reg(mcasp, DAVINCI_MCASP_RXFMCTL_REG, context->rxfmtctl); - mcasp_set_reg(mcasp, DAVINCI_MCASP_TXFMT_REG, context->txfmt); - mcasp_set_reg(mcasp, DAVINCI_MCASP_RXFMT_REG, context->rxfmt); - mcasp_set_reg(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, context->aclkxctl); - mcasp_set_reg(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, context->aclkrctl); - mcasp_set_reg(mcasp, DAVINCI_MCASP_PDIR_REG, context->pdir); + for (i = 0; i < ARRAY_SIZE(context_regs); i++) + mcasp_set_reg(mcasp, context_regs[i], context->config_regs[i]); + + if (mcasp->txnumevt) { + reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET; + mcasp_set_reg(mcasp, reg, context->afifo_regs[0]); + } + if (mcasp->rxnumevt) { + reg = mcasp->fifo_base + MCASP_RFIFOCTL_OFFSET; + mcasp_set_reg(mcasp, reg, context->afifo_regs[1]); + } + + for (i = 0; i < mcasp->num_serializer; i++) + mcasp_set_reg(mcasp, DAVINCI_MCASP_XRSRCTL_REG(i), + context->xrsr_regs[i]); return 0; } @@ -1207,6 +1248,11 @@ static int davinci_mcasp_probe(struct platform_device *pdev) mcasp->op_mode = pdata->op_mode; mcasp->tdm_slots = pdata->tdm_slots; mcasp->num_serializer = pdata->num_serializer; +#ifdef CONFIG_PM_SLEEP + mcasp->context.xrsr_regs = devm_kzalloc(&pdev->dev, + sizeof(u32) * mcasp->num_serializer, + GFP_KERNEL); +#endif mcasp->serial_dir = pdata->serial_dir; mcasp->version = pdata->version; mcasp->txnumevt = pdata->txnumevt; diff --git a/sound/soc/davinci/edma-pcm.c b/sound/soc/davinci/edma-pcm.c index 605e643133db..59e588abe54b 100644 --- a/sound/soc/davinci/edma-pcm.c +++ b/sound/soc/davinci/edma-pcm.c @@ -25,6 +25,8 @@ #include <sound/dmaengine_pcm.h> #include <linux/edma.h> +#include "edma-pcm.h" + static const struct snd_pcm_hardware edma_pcm_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index 25c31f1655f6..e961388e6e9c 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -4,7 +4,7 @@ * sound/soc/dwc/designware_i2s.c * * Copyright (C) 2010 ST Microelectronics - * Rajeev Kumar <rajeev-dlh.kumar@st.com> + * Rajeev Kumar <rajeevkumar.linux@gmail.com> * * This file is licensed under the terms of the GNU General Public * License version 2. This program is licensed "as is" without any @@ -455,7 +455,7 @@ static struct platform_driver dw_i2s_driver = { module_platform_driver(dw_i2s_driver); -MODULE_AUTHOR("Rajeev Kumar <rajeev-dlh.kumar@st.com>"); +MODULE_AUTHOR("Rajeev Kumar <rajeevkumar.linux@gmail.com>"); MODULE_DESCRIPTION("DESIGNWARE I2S SoC Interface"); MODULE_LICENSE("GPL"); MODULE_ALIAS("platform:designware_i2s"); diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index f3012b645b51..081e406b3713 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -240,6 +240,18 @@ config SND_SOC_IMX_WM8962 Say Y if you want to add support for SoC audio on an i.MX board with a wm8962 codec. +config SND_SOC_IMX_ES8328 + tristate "SoC Audio support for i.MX boards with the ES8328 codec" + depends on OF && (I2C || SPI) + select SND_SOC_ES8328_I2C if I2C + select SND_SOC_ES8328_SPI if SPI_MASTER + select SND_SOC_IMX_PCM_DMA + select SND_SOC_IMX_AUDMUX + select SND_SOC_FSL_SSI + help + Say Y if you want to add support for the ES8328 audio codec connected + via SSI/I2S over either SPI or I2C. + config SND_SOC_IMX_SGTL5000 tristate "SoC Audio support for i.MX boards with sgtl5000" depends on OF && I2C @@ -268,6 +280,20 @@ config SND_SOC_IMX_MC13783 select SND_SOC_MC13783 select SND_SOC_IMX_PCM_DMA +config SND_SOC_FSL_ASOC_CARD + tristate "Generic ASoC Sound Card with ASRC support" + depends on OF && I2C + select SND_SOC_IMX_AUDMUX + select SND_SOC_IMX_PCM_DMA + select SND_SOC_FSL_ESAI + select SND_SOC_FSL_SAI + select SND_SOC_FSL_SSI + help + ALSA SoC Audio support with ASRC feature for Freescale SoCs that have + ESAI/SAI/SSI and connect with external CODECs such as WM8962, CS42888 + and SGTL5000. + Say Y if you want to add support for Freescale Generic ASoC Sound Card. + endif # SND_IMX_SOC endmenu diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index 9ff59267eac9..d28dc25c9375 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -11,6 +11,7 @@ snd-soc-p1022-rdk-objs := p1022_rdk.o obj-$(CONFIG_SND_SOC_P1022_RDK) += snd-soc-p1022-rdk.o # Freescale SSI/DMA/SAI/SPDIF Support +snd-soc-fsl-asoc-card-objs := fsl-asoc-card.o snd-soc-fsl-asrc-objs := fsl_asrc.o fsl_asrc_dma.o snd-soc-fsl-sai-objs := fsl_sai.o snd-soc-fsl-ssi-y := fsl_ssi.o @@ -19,6 +20,7 @@ snd-soc-fsl-spdif-objs := fsl_spdif.o snd-soc-fsl-esai-objs := fsl_esai.o snd-soc-fsl-utils-objs := fsl_utils.o snd-soc-fsl-dma-objs := fsl_dma.o +obj-$(CONFIG_SND_SOC_FSL_ASOC_CARD) += snd-soc-fsl-asoc-card.o obj-$(CONFIG_SND_SOC_FSL_ASRC) += snd-soc-fsl-asrc.o obj-$(CONFIG_SND_SOC_FSL_SAI) += snd-soc-fsl-sai.o obj-$(CONFIG_SND_SOC_FSL_SSI) += snd-soc-fsl-ssi.o @@ -50,6 +52,7 @@ snd-soc-eukrea-tlv320-objs := eukrea-tlv320.o snd-soc-phycore-ac97-objs := phycore-ac97.o snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o snd-soc-wm1133-ev1-objs := wm1133-ev1.o +snd-soc-imx-es8328-objs := imx-es8328.o snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o snd-soc-imx-wm8962-objs := imx-wm8962.o snd-soc-imx-spdif-objs := imx-spdif.o @@ -59,6 +62,7 @@ obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o obj-$(CONFIG_SND_SOC_PHYCORE_AC97) += snd-soc-phycore-ac97.o obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o +obj-$(CONFIG_SND_SOC_IMX_ES8328) += snd-soc-imx-es8328.o obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o obj-$(CONFIG_SND_SOC_IMX_WM8962) += snd-soc-imx-wm8962.o obj-$(CONFIG_SND_SOC_IMX_SPDIF) += snd-soc-imx-spdif.o diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c new file mode 100644 index 000000000000..007c772f3cef --- /dev/null +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -0,0 +1,574 @@ +/* + * Freescale Generic ASoC Sound Card driver with ASRC + * + * Copyright (C) 2014 Freescale Semiconductor, Inc. + * + * Author: Nicolin Chen <nicoleotsuka@gmail.com> + * + * This file is licensed under the terms of the GNU General Public License + * version 2. This program is licensed "as is" without any warranty of any + * kind, whether express or implied. + */ + +#include <linux/clk.h> +#include <linux/i2c.h> +#include <linux/module.h> +#include <linux/of_platform.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include "fsl_esai.h" +#include "fsl_sai.h" +#include "imx-audmux.h" + +#include "../codecs/sgtl5000.h" +#include "../codecs/wm8962.h" + +#define RX 0 +#define TX 1 + +/* Default DAI format without Master and Slave flag */ +#define DAI_FMT_BASE (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF) + +/** + * CODEC private data + * + * @mclk_freq: Clock rate of MCLK + * @mclk_id: MCLK (or main clock) id for set_sysclk() + * @fll_id: FLL (or secordary clock) id for set_sysclk() + * @pll_id: PLL id for set_pll() + */ +struct codec_priv { + unsigned long mclk_freq; + u32 mclk_id; + u32 fll_id; + u32 pll_id; +}; + +/** + * CPU private data + * + * @sysclk_freq[2]: SYSCLK rates for set_sysclk() + * @sysclk_dir[2]: SYSCLK directions for set_sysclk() + * @sysclk_id[2]: SYSCLK ids for set_sysclk() + * + * Note: [1] for tx and [0] for rx + */ +struct cpu_priv { + unsigned long sysclk_freq[2]; + u32 sysclk_dir[2]; + u32 sysclk_id[2]; +}; + +/** + * Freescale Generic ASOC card private data + * + * @dai_link[3]: DAI link structure including normal one and DPCM link + * @pdev: platform device pointer + * @codec_priv: CODEC private data + * @cpu_priv: CPU private data + * @card: ASoC card structure + * @sample_rate: Current sample rate + * @sample_format: Current sample format + * @asrc_rate: ASRC sample rate used by Back-Ends + * @asrc_format: ASRC sample format used by Back-Ends + * @dai_fmt: DAI format between CPU and CODEC + * @name: Card name + */ + +struct fsl_asoc_card_priv { + struct snd_soc_dai_link dai_link[3]; + struct platform_device *pdev; + struct codec_priv codec_priv; + struct cpu_priv cpu_priv; + struct snd_soc_card card; + u32 sample_rate; + u32 sample_format; + u32 asrc_rate; + u32 asrc_format; + u32 dai_fmt; + char name[32]; +}; + +/** + * This dapm route map exsits for DPCM link only. + * The other routes shall go through Device Tree. + */ +static const struct snd_soc_dapm_route audio_map[] = { + {"CPU-Playback", NULL, "ASRC-Playback"}, + {"Playback", NULL, "CPU-Playback"}, + {"ASRC-Capture", NULL, "CPU-Capture"}, + {"CPU-Capture", NULL, "Capture"}, +}; + +/* Add all possible widgets into here without being redundant */ +static const struct snd_soc_dapm_widget fsl_asoc_card_dapm_widgets[] = { + SND_SOC_DAPM_LINE("Line Out Jack", NULL), + SND_SOC_DAPM_LINE("Line In Jack", NULL), + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_MIC("AMIC", NULL), + SND_SOC_DAPM_MIC("DMIC", NULL), +}; + +static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + struct cpu_priv *cpu_priv = &priv->cpu_priv; + struct device *dev = rtd->card->dev; + int ret; + + priv->sample_rate = params_rate(params); + priv->sample_format = params_format(params); + + if (priv->card.set_bias_level) + return 0; + + /* Specific configurations of DAIs starts from here */ + ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, cpu_priv->sysclk_id[tx], + cpu_priv->sysclk_freq[tx], + cpu_priv->sysclk_dir[tx]); + if (ret) { + dev_err(dev, "failed to set sysclk for cpu dai\n"); + return ret; + } + + return 0; +} + +static struct snd_soc_ops fsl_asoc_card_ops = { + .hw_params = fsl_asoc_card_hw_params, +}; + +static int be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card); + struct snd_interval *rate; + struct snd_mask *mask; + + rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + rate->max = rate->min = priv->asrc_rate; + + mask = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + snd_mask_none(mask); + snd_mask_set(mask, priv->asrc_format); + + return 0; +} + +static struct snd_soc_dai_link fsl_asoc_card_dai[] = { + /* Default ASoC DAI Link*/ + { + .name = "HiFi", + .stream_name = "HiFi", + .ops = &fsl_asoc_card_ops, + }, + /* DPCM Link between Front-End and Back-End (Optional) */ + { + .name = "HiFi-ASRC-FE", + .stream_name = "HiFi-ASRC-FE", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .dpcm_playback = 1, + .dpcm_capture = 1, + .dynamic = 1, + }, + { + .name = "HiFi-ASRC-BE", + .stream_name = "HiFi-ASRC-BE", + .platform_name = "snd-soc-dummy", + .be_hw_params_fixup = be_hw_params_fixup, + .ops = &fsl_asoc_card_ops, + .dpcm_playback = 1, + .dpcm_capture = 1, + .no_pcm = 1, + }, +}; + +static int fsl_asoc_card_set_bias_level(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level) +{ + struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card); + struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + struct codec_priv *codec_priv = &priv->codec_priv; + struct device *dev = card->dev; + unsigned int pll_out; + int ret; + + if (dapm->dev != codec_dai->dev) + return 0; + + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (dapm->bias_level != SND_SOC_BIAS_STANDBY) + break; + + if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE) + pll_out = priv->sample_rate * 384; + else + pll_out = priv->sample_rate * 256; + + ret = snd_soc_dai_set_pll(codec_dai, codec_priv->pll_id, + codec_priv->mclk_id, + codec_priv->mclk_freq, pll_out); + if (ret) { + dev_err(dev, "failed to start FLL: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->fll_id, + pll_out, SND_SOC_CLOCK_IN); + if (ret) { + dev_err(dev, "failed to set SYSCLK: %d\n", ret); + return ret; + } + break; + + case SND_SOC_BIAS_STANDBY: + if (dapm->bias_level != SND_SOC_BIAS_PREPARE) + break; + + ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id, + codec_priv->mclk_freq, + SND_SOC_CLOCK_IN); + if (ret) { + dev_err(dev, "failed to switch away from FLL: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_pll(codec_dai, codec_priv->pll_id, 0, 0, 0); + if (ret) { + dev_err(dev, "failed to stop FLL: %d\n", ret); + return ret; + } + break; + + default: + break; + } + + return 0; +} + +static int fsl_asoc_card_audmux_init(struct device_node *np, + struct fsl_asoc_card_priv *priv) +{ + struct device *dev = &priv->pdev->dev; + u32 int_ptcr = 0, ext_ptcr = 0; + int int_port, ext_port; + int ret; + + ret = of_property_read_u32(np, "mux-int-port", &int_port); + if (ret) { + dev_err(dev, "mux-int-port missing or invalid\n"); + return ret; + } + ret = of_property_read_u32(np, "mux-ext-port", &ext_port); + if (ret) { + dev_err(dev, "mux-ext-port missing or invalid\n"); + return ret; + } + + /* + * The port numbering in the hardware manual starts at 1, while + * the AUDMUX API expects it starts at 0. + */ + int_port--; + ext_port--; + + /* + * Use asynchronous mode (6 wires) for all cases. + * If only 4 wires are needed, just set SSI into + * synchronous mode and enable 4 PADs in IOMUX. + */ + switch (priv->dai_fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + int_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | ext_port) | + IMX_AUDMUX_V2_PTCR_RCSEL(8 | ext_port) | + IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_RFSDIR | + IMX_AUDMUX_V2_PTCR_RCLKDIR | + IMX_AUDMUX_V2_PTCR_TFSDIR | + IMX_AUDMUX_V2_PTCR_TCLKDIR; + break; + case SND_SOC_DAIFMT_CBM_CFS: + int_ptcr = IMX_AUDMUX_V2_PTCR_RCSEL(8 | ext_port) | + IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_RCLKDIR | + IMX_AUDMUX_V2_PTCR_TCLKDIR; + ext_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | int_port) | + IMX_AUDMUX_V2_PTCR_TFSEL(int_port) | + IMX_AUDMUX_V2_PTCR_RFSDIR | + IMX_AUDMUX_V2_PTCR_TFSDIR; + break; + case SND_SOC_DAIFMT_CBS_CFM: + int_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | ext_port) | + IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_RFSDIR | + IMX_AUDMUX_V2_PTCR_TFSDIR; + ext_ptcr = IMX_AUDMUX_V2_PTCR_RCSEL(8 | int_port) | + IMX_AUDMUX_V2_PTCR_TCSEL(int_port) | + IMX_AUDMUX_V2_PTCR_RCLKDIR | + IMX_AUDMUX_V2_PTCR_TCLKDIR; + break; + case SND_SOC_DAIFMT_CBS_CFS: + ext_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | int_port) | + IMX_AUDMUX_V2_PTCR_RCSEL(8 | int_port) | + IMX_AUDMUX_V2_PTCR_TFSEL(int_port) | + IMX_AUDMUX_V2_PTCR_TCSEL(int_port) | + IMX_AUDMUX_V2_PTCR_RFSDIR | + IMX_AUDMUX_V2_PTCR_RCLKDIR | + IMX_AUDMUX_V2_PTCR_TFSDIR | + IMX_AUDMUX_V2_PTCR_TCLKDIR; + break; + default: + return -EINVAL; + } + + /* Asynchronous mode can not be set along with RCLKDIR */ + ret = imx_audmux_v2_configure_port(int_port, 0, + IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port)); + if (ret) { + dev_err(dev, "audmux internal port setup failed\n"); + return ret; + } + + ret = imx_audmux_v2_configure_port(int_port, int_ptcr, + IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port)); + if (ret) { + dev_err(dev, "audmux internal port setup failed\n"); + return ret; + } + + ret = imx_audmux_v2_configure_port(ext_port, 0, + IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)); + if (ret) { + dev_err(dev, "audmux external port setup failed\n"); + return ret; + } + + ret = imx_audmux_v2_configure_port(ext_port, ext_ptcr, + IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)); + if (ret) { + dev_err(dev, "audmux external port setup failed\n"); + return ret; + } + + return 0; +} + +static int fsl_asoc_card_late_probe(struct snd_soc_card *card) +{ + struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card); + struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + struct codec_priv *codec_priv = &priv->codec_priv; + struct device *dev = card->dev; + int ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id, + codec_priv->mclk_freq, SND_SOC_CLOCK_IN); + if (ret) { + dev_err(dev, "failed to set sysclk in %s\n", __func__); + return ret; + } + + return 0; +} + +static int fsl_asoc_card_probe(struct platform_device *pdev) +{ + struct device_node *cpu_np, *codec_np, *asrc_np; + struct device_node *np = pdev->dev.of_node; + struct platform_device *asrc_pdev = NULL; + struct platform_device *cpu_pdev; + struct fsl_asoc_card_priv *priv; + struct i2c_client *codec_dev; + struct clk *codec_clk; + u32 width; + int ret; + + priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + cpu_np = of_parse_phandle(np, "audio-cpu", 0); + /* Give a chance to old DT binding */ + if (!cpu_np) + cpu_np = of_parse_phandle(np, "ssi-controller", 0); + codec_np = of_parse_phandle(np, "audio-codec", 0); + if (!cpu_np || !codec_np) { + dev_err(&pdev->dev, "phandle missing or invalid\n"); + ret = -EINVAL; + goto fail; + } + + cpu_pdev = of_find_device_by_node(cpu_np); + if (!cpu_pdev) { + dev_err(&pdev->dev, "failed to find CPU DAI device\n"); + ret = -EINVAL; + goto fail; + } + + codec_dev = of_find_i2c_device_by_node(codec_np); + if (!codec_dev) { + dev_err(&pdev->dev, "failed to find codec platform device\n"); + ret = -EINVAL; + goto fail; + } + + asrc_np = of_parse_phandle(np, "audio-asrc", 0); + if (asrc_np) + asrc_pdev = of_find_device_by_node(asrc_np); + + /* Get the MCLK rate only, and leave it controlled by CODEC drivers */ + codec_clk = clk_get(&codec_dev->dev, NULL); + if (!IS_ERR(codec_clk)) { + priv->codec_priv.mclk_freq = clk_get_rate(codec_clk); + clk_put(codec_clk); + } + + /* Default sample rate and format, will be updated in hw_params() */ + priv->sample_rate = 44100; + priv->sample_format = SNDRV_PCM_FORMAT_S16_LE; + + /* Assign a default DAI format, and allow each card to overwrite it */ + priv->dai_fmt = DAI_FMT_BASE; + + /* Diversify the card configurations */ + if (of_device_is_compatible(np, "fsl,imx-audio-cs42888")) { + priv->card.set_bias_level = NULL; + priv->cpu_priv.sysclk_freq[TX] = priv->codec_priv.mclk_freq; + priv->cpu_priv.sysclk_freq[RX] = priv->codec_priv.mclk_freq; + priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_OUT; + priv->cpu_priv.sysclk_dir[RX] = SND_SOC_CLOCK_OUT; + priv->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; + } else if (of_device_is_compatible(np, "fsl,imx-audio-sgtl5000")) { + priv->codec_priv.mclk_id = SGTL5000_SYSCLK; + priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; + } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8962")) { + priv->card.set_bias_level = fsl_asoc_card_set_bias_level; + priv->codec_priv.mclk_id = WM8962_SYSCLK_MCLK; + priv->codec_priv.fll_id = WM8962_SYSCLK_FLL; + priv->codec_priv.pll_id = WM8962_FLL; + priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; + } else { + dev_err(&pdev->dev, "unknown Device Tree compatible\n"); + return -EINVAL; + } + + /* Common settings for corresponding Freescale CPU DAI driver */ + if (strstr(cpu_np->name, "ssi")) { + /* Only SSI needs to configure AUDMUX */ + ret = fsl_asoc_card_audmux_init(np, priv); + if (ret) { + dev_err(&pdev->dev, "failed to init audmux\n"); + goto asrc_fail; + } + } else if (strstr(cpu_np->name, "esai")) { + priv->cpu_priv.sysclk_id[1] = ESAI_HCKT_EXTAL; + priv->cpu_priv.sysclk_id[0] = ESAI_HCKR_EXTAL; + } else if (strstr(cpu_np->name, "sai")) { + priv->cpu_priv.sysclk_id[1] = FSL_SAI_CLK_MAST1; + priv->cpu_priv.sysclk_id[0] = FSL_SAI_CLK_MAST1; + } + + sprintf(priv->name, "%s-audio", codec_dev->name); + + /* Initialize sound card */ + priv->pdev = pdev; + priv->card.dev = &pdev->dev; + priv->card.name = priv->name; + priv->card.dai_link = priv->dai_link; + priv->card.dapm_routes = audio_map; + priv->card.late_probe = fsl_asoc_card_late_probe; + priv->card.num_dapm_routes = ARRAY_SIZE(audio_map); + priv->card.dapm_widgets = fsl_asoc_card_dapm_widgets; + priv->card.num_dapm_widgets = ARRAY_SIZE(fsl_asoc_card_dapm_widgets); + + memcpy(priv->dai_link, fsl_asoc_card_dai, + sizeof(struct snd_soc_dai_link) * ARRAY_SIZE(priv->dai_link)); + + /* Normal DAI Link */ + priv->dai_link[0].cpu_of_node = cpu_np; + priv->dai_link[0].codec_of_node = codec_np; + priv->dai_link[0].codec_dai_name = codec_dev->name; + priv->dai_link[0].platform_of_node = cpu_np; + priv->dai_link[0].dai_fmt = priv->dai_fmt; + priv->card.num_links = 1; + + if (asrc_pdev) { + /* DPCM DAI Links only if ASRC exsits */ + priv->dai_link[1].cpu_of_node = asrc_np; + priv->dai_link[1].platform_of_node = asrc_np; + priv->dai_link[2].codec_dai_name = codec_dev->name; + priv->dai_link[2].codec_of_node = codec_np; + priv->dai_link[2].cpu_of_node = cpu_np; + priv->dai_link[2].dai_fmt = priv->dai_fmt; + priv->card.num_links = 3; + + ret = of_property_read_u32(asrc_np, "fsl,asrc-rate", + &priv->asrc_rate); + if (ret) { + dev_err(&pdev->dev, "failed to get output rate\n"); + ret = -EINVAL; + goto asrc_fail; + } + + ret = of_property_read_u32(asrc_np, "fsl,asrc-width", &width); + if (ret) { + dev_err(&pdev->dev, "failed to get output rate\n"); + ret = -EINVAL; + goto asrc_fail; + } + + if (width == 24) + priv->asrc_format = SNDRV_PCM_FORMAT_S24_LE; + else + priv->asrc_format = SNDRV_PCM_FORMAT_S16_LE; + } + + /* Finish card registering */ + platform_set_drvdata(pdev, priv); + snd_soc_card_set_drvdata(&priv->card, priv); + + ret = devm_snd_soc_register_card(&pdev->dev, &priv->card); + if (ret) + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + +asrc_fail: + of_node_put(asrc_np); +fail: + of_node_put(codec_np); + of_node_put(cpu_np); + + return ret; +} + +static const struct of_device_id fsl_asoc_card_dt_ids[] = { + { .compatible = "fsl,imx-audio-cs42888", }, + { .compatible = "fsl,imx-audio-sgtl5000", }, + { .compatible = "fsl,imx-audio-wm8962", }, + {} +}; + +static struct platform_driver fsl_asoc_card_driver = { + .probe = fsl_asoc_card_probe, + .driver = { + .name = "fsl-asoc-card", + .pm = &snd_soc_pm_ops, + .of_match_table = fsl_asoc_card_dt_ids, + }, +}; +module_platform_driver(fsl_asoc_card_driver); + +MODULE_DESCRIPTION("Freescale Generic ASoC Sound Card driver with ASRC"); +MODULE_AUTHOR("Nicolin Chen <nicoleotsuka@gmail.com>"); +MODULE_ALIAS("platform:fsl-asoc-card"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index 822110420b71..3b145313f93e 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -684,7 +684,7 @@ static bool fsl_asrc_writeable_reg(struct device *dev, unsigned int reg) } } -static struct regmap_config fsl_asrc_regmap_config = { +static const struct regmap_config fsl_asrc_regmap_config = { .reg_bits = 32, .reg_stride = 4, .val_bits = 32, @@ -802,10 +802,6 @@ static int fsl_asrc_probe(struct platform_device *pdev) asrc_priv->paddr = res->start; - /* Register regmap and let it prepare core clock */ - if (of_property_read_bool(np, "big-endian")) - fsl_asrc_regmap_config.val_format_endian = REGMAP_ENDIAN_BIG; - asrc_priv->regmap = devm_regmap_init_mmio_clk(&pdev->dev, "mem", regs, &fsl_asrc_regmap_config); if (IS_ERR(asrc_priv->regmap)) { diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index a3b29ed84963..8bcdfda09d7a 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -37,6 +37,7 @@ * @fsysclk: system clock source to derive HCK, SCK and FS * @fifo_depth: depth of tx/rx FIFO * @slot_width: width of each DAI slot + * @slots: number of slots * @hck_rate: clock rate of desired HCKx clock * @sck_rate: clock rate of desired SCKx clock * @hck_dir: the direction of HCKx pads @@ -55,6 +56,7 @@ struct fsl_esai { struct clk *fsysclk; u32 fifo_depth; u32 slot_width; + u32 slots; u32 hck_rate[2]; u32 sck_rate[2]; bool hck_dir[2]; @@ -362,6 +364,7 @@ static int fsl_esai_set_dai_tdm_slot(struct snd_soc_dai *dai, u32 tx_mask, ESAI_xSMB_xS_MASK, ESAI_xSMB_xS(rx_mask)); esai_priv->slot_width = slot_width; + esai_priv->slots = slots; return 0; } @@ -509,10 +512,11 @@ static int fsl_esai_hw_params(struct snd_pcm_substream *substream, bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; u32 width = snd_pcm_format_width(params_format(params)); u32 channels = params_channels(params); + u32 pins = DIV_ROUND_UP(channels, esai_priv->slots); u32 bclk, mask, val; int ret; - bclk = params_rate(params) * esai_priv->slot_width * 2; + bclk = params_rate(params) * esai_priv->slot_width * esai_priv->slots; ret = fsl_esai_set_bclk(dai, tx, bclk); if (ret) @@ -529,7 +533,7 @@ static int fsl_esai_hw_params(struct snd_pcm_substream *substream, mask = ESAI_xFCR_xFR_MASK | ESAI_xFCR_xWA_MASK | ESAI_xFCR_xFWM_MASK | (tx ? ESAI_xFCR_TE_MASK | ESAI_xFCR_TIEN : ESAI_xFCR_RE_MASK); val = ESAI_xFCR_xWA(width) | ESAI_xFCR_xFWM(esai_priv->fifo_depth) | - (tx ? ESAI_xFCR_TE(channels) | ESAI_xFCR_TIEN : ESAI_xFCR_RE(channels)); + (tx ? ESAI_xFCR_TE(pins) | ESAI_xFCR_TIEN : ESAI_xFCR_RE(pins)); regmap_update_bits(esai_priv->regmap, REG_ESAI_xFCR(tx), mask, val); @@ -564,6 +568,7 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; u8 i, channels = substream->runtime->channels; + u32 pins = DIV_ROUND_UP(channels, esai_priv->slots); switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -578,7 +583,7 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx), tx ? ESAI_xCR_TE_MASK : ESAI_xCR_RE_MASK, - tx ? ESAI_xCR_TE(channels) : ESAI_xCR_RE(channels)); + tx ? ESAI_xCR_TE(pins) : ESAI_xCR_RE(pins)); break; case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_STOP: @@ -705,7 +710,7 @@ static bool fsl_esai_writeable_reg(struct device *dev, unsigned int reg) } } -static struct regmap_config fsl_esai_regmap_config = { +static const struct regmap_config fsl_esai_regmap_config = { .reg_bits = 32, .reg_stride = 4, .val_bits = 32, @@ -731,9 +736,6 @@ static int fsl_esai_probe(struct platform_device *pdev) esai_priv->pdev = pdev; strcpy(esai_priv->name, np->name); - if (of_property_read_bool(np, "big-endian")) - fsl_esai_regmap_config.val_format_endian = REGMAP_ENDIAN_BIG; - /* Get the addresses and IRQ */ res = platform_get_resource(pdev, IORESOURCE_MEM, 0); regs = devm_ioremap_resource(&pdev->dev, res); @@ -781,6 +783,9 @@ static int fsl_esai_probe(struct platform_device *pdev) /* Set a default slot size */ esai_priv->slot_width = 32; + /* Set a default slot number */ + esai_priv->slots = 2; + /* Set a default master/slave state */ esai_priv->slave_mode = true; diff --git a/sound/soc/fsl/fsl_esai.h b/sound/soc/fsl/fsl_esai.h index 75e14033e8d8..91a550f4a10d 100644 --- a/sound/soc/fsl/fsl_esai.h +++ b/sound/soc/fsl/fsl_esai.h @@ -130,8 +130,8 @@ #define ESAI_xFCR_RE_WIDTH 4 #define ESAI_xFCR_TE_MASK (((1 << ESAI_xFCR_TE_WIDTH) - 1) << ESAI_xFCR_xE_SHIFT) #define ESAI_xFCR_RE_MASK (((1 << ESAI_xFCR_RE_WIDTH) - 1) << ESAI_xFCR_xE_SHIFT) -#define ESAI_xFCR_TE(x) ((ESAI_xFCR_TE_MASK >> (ESAI_xFCR_TE_WIDTH - ((x + 1) >> 1))) & ESAI_xFCR_TE_MASK) -#define ESAI_xFCR_RE(x) ((ESAI_xFCR_RE_MASK >> (ESAI_xFCR_RE_WIDTH - ((x + 1) >> 1))) & ESAI_xFCR_RE_MASK) +#define ESAI_xFCR_TE(x) ((ESAI_xFCR_TE_MASK >> (ESAI_xFCR_TE_WIDTH - x)) & ESAI_xFCR_TE_MASK) +#define ESAI_xFCR_RE(x) ((ESAI_xFCR_RE_MASK >> (ESAI_xFCR_RE_WIDTH - x)) & ESAI_xFCR_RE_MASK) #define ESAI_xFCR_xFR_SHIFT 1 #define ESAI_xFCR_xFR_MASK (1 << ESAI_xFCR_xFR_SHIFT) #define ESAI_xFCR_xFR (1 << ESAI_xFCR_xFR_SHIFT) @@ -272,8 +272,8 @@ #define ESAI_xCR_RE_WIDTH 4 #define ESAI_xCR_TE_MASK (((1 << ESAI_xCR_TE_WIDTH) - 1) << ESAI_xCR_xE_SHIFT) #define ESAI_xCR_RE_MASK (((1 << ESAI_xCR_RE_WIDTH) - 1) << ESAI_xCR_xE_SHIFT) -#define ESAI_xCR_TE(x) ((ESAI_xCR_TE_MASK >> (ESAI_xCR_TE_WIDTH - ((x + 1) >> 1))) & ESAI_xCR_TE_MASK) -#define ESAI_xCR_RE(x) ((ESAI_xCR_RE_MASK >> (ESAI_xCR_RE_WIDTH - ((x + 1) >> 1))) & ESAI_xCR_RE_MASK) +#define ESAI_xCR_TE(x) ((ESAI_xCR_TE_MASK >> (ESAI_xCR_TE_WIDTH - x)) & ESAI_xCR_TE_MASK) +#define ESAI_xCR_RE(x) ((ESAI_xCR_RE_MASK >> (ESAI_xCR_RE_WIDTH - x)) & ESAI_xCR_RE_MASK) /* * Transmit Clock Control Register -- REG_ESAI_TCCR 0xD8 diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index faa049797897..7eeb1dd8ce27 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -175,7 +175,7 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, bool tx = fsl_dir == FSL_FMT_TRANSMITTER; u32 val_cr2 = 0, val_cr4 = 0; - if (!sai->big_endian_data) + if (!sai->is_lsb_first) val_cr4 |= FSL_SAI_CR4_MF; /* DAI mode */ @@ -304,7 +304,7 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, val_cr5 |= FSL_SAI_CR5_WNW(word_width); val_cr5 |= FSL_SAI_CR5_W0W(word_width); - if (sai->big_endian_data) + if (sai->is_lsb_first) val_cr5 |= FSL_SAI_CR5_FBT(0); else val_cr5 |= FSL_SAI_CR5_FBT(word_width - 1); @@ -330,13 +330,13 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, u32 xcsr, count = 100; /* - * The transmitter bit clock and frame sync are to be - * used by both the transmitter and receiver. + * Asynchronous mode: Clear SYNC for both Tx and Rx. + * Rx sync with Tx clocks: Clear SYNC for Tx, set it for Rx. + * Tx sync with Rx clocks: Clear SYNC for Rx, set it for Tx. */ - regmap_update_bits(sai->regmap, FSL_SAI_TCR2, FSL_SAI_CR2_SYNC, - ~FSL_SAI_CR2_SYNC); + regmap_update_bits(sai->regmap, FSL_SAI_TCR2, FSL_SAI_CR2_SYNC, 0); regmap_update_bits(sai->regmap, FSL_SAI_RCR2, FSL_SAI_CR2_SYNC, - FSL_SAI_CR2_SYNC); + sai->synchronous[RX] ? FSL_SAI_CR2_SYNC : 0); /* * It is recommended that the transmitter is the last enabled @@ -437,8 +437,13 @@ static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai) { struct fsl_sai *sai = dev_get_drvdata(cpu_dai->dev); - regmap_update_bits(sai->regmap, FSL_SAI_TCSR, 0xffffffff, 0x0); - regmap_update_bits(sai->regmap, FSL_SAI_RCSR, 0xffffffff, 0x0); + /* Software Reset for both Tx and Rx */ + regmap_write(sai->regmap, FSL_SAI_TCSR, FSL_SAI_CSR_SR); + regmap_write(sai->regmap, FSL_SAI_RCSR, FSL_SAI_CSR_SR); + /* Clear SR bit to finish the reset */ + regmap_write(sai->regmap, FSL_SAI_TCSR, 0); + regmap_write(sai->regmap, FSL_SAI_RCSR, 0); + regmap_update_bits(sai->regmap, FSL_SAI_TCR1, FSL_SAI_CR1_RFW_MASK, FSL_SAI_MAXBURST_TX * 2); regmap_update_bits(sai->regmap, FSL_SAI_RCR1, FSL_SAI_CR1_RFW_MASK, @@ -539,7 +544,7 @@ static bool fsl_sai_writeable_reg(struct device *dev, unsigned int reg) } } -static struct regmap_config fsl_sai_regmap_config = { +static const struct regmap_config fsl_sai_regmap_config = { .reg_bits = 32, .reg_stride = 4, .val_bits = 32, @@ -568,11 +573,7 @@ static int fsl_sai_probe(struct platform_device *pdev) if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx6sx-sai")) sai->sai_on_imx = true; - sai->big_endian_regs = of_property_read_bool(np, "big-endian-regs"); - if (sai->big_endian_regs) - fsl_sai_regmap_config.val_format_endian = REGMAP_ENDIAN_BIG; - - sai->big_endian_data = of_property_read_bool(np, "big-endian-data"); + sai->is_lsb_first = of_property_read_bool(np, "lsb-first"); res = platform_get_resource(pdev, IORESOURCE_MEM, 0); base = devm_ioremap_resource(&pdev->dev, res); @@ -621,6 +622,33 @@ static int fsl_sai_probe(struct platform_device *pdev) return ret; } + /* Sync Tx with Rx as default by following old DT binding */ + sai->synchronous[RX] = true; + sai->synchronous[TX] = false; + fsl_sai_dai.symmetric_rates = 1; + fsl_sai_dai.symmetric_channels = 1; + fsl_sai_dai.symmetric_samplebits = 1; + + if (of_find_property(np, "fsl,sai-synchronous-rx", NULL) && + of_find_property(np, "fsl,sai-asynchronous", NULL)) { + /* error out if both synchronous and asynchronous are present */ + dev_err(&pdev->dev, "invalid binding for synchronous mode\n"); + return -EINVAL; + } + + if (of_find_property(np, "fsl,sai-synchronous-rx", NULL)) { + /* Sync Rx with Tx */ + sai->synchronous[RX] = false; + sai->synchronous[TX] = true; + } else if (of_find_property(np, "fsl,sai-asynchronous", NULL)) { + /* Discard all settings for asynchronous mode */ + sai->synchronous[RX] = false; + sai->synchronous[TX] = false; + fsl_sai_dai.symmetric_rates = 0; + fsl_sai_dai.symmetric_channels = 0; + fsl_sai_dai.symmetric_samplebits = 0; + } + sai->dma_params_rx.addr = res->start + FSL_SAI_RDR; sai->dma_params_tx.addr = res->start + FSL_SAI_TDR; sai->dma_params_rx.maxburst = FSL_SAI_MAXBURST_RX; diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 0e6c9f595d75..34667209b607 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -48,6 +48,7 @@ /* SAI Transmit/Recieve Control Register */ #define FSL_SAI_CSR_TERE BIT(31) #define FSL_SAI_CSR_FR BIT(25) +#define FSL_SAI_CSR_SR BIT(24) #define FSL_SAI_CSR_xF_SHIFT 16 #define FSL_SAI_CSR_xF_W_SHIFT 18 #define FSL_SAI_CSR_xF_MASK (0x1f << FSL_SAI_CSR_xF_SHIFT) @@ -131,13 +132,16 @@ struct fsl_sai { struct clk *bus_clk; struct clk *mclk_clk[FSL_SAI_MCLK_MAX]; - bool big_endian_regs; - bool big_endian_data; + bool is_lsb_first; bool is_dsp_mode; bool sai_on_imx; + bool synchronous[2]; struct snd_dmaengine_dai_dma_data dma_params_rx; struct snd_dmaengine_dai_dma_data dma_params_tx; }; +#define TX 1 +#define RX 0 + #endif /* __FSL_SAI_H */ diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 70acfe4a9bd5..9b791621294c 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -15,7 +15,6 @@ #include <linux/bitrev.h> #include <linux/clk.h> -#include <linux/clk-private.h> #include <linux/module.h> #include <linux/of_address.h> #include <linux/of_device.h> @@ -1040,7 +1039,7 @@ static bool fsl_spdif_writeable_reg(struct device *dev, unsigned int reg) } } -static struct regmap_config fsl_spdif_regmap_config = { +static const struct regmap_config fsl_spdif_regmap_config = { .reg_bits = 32, .reg_stride = 4, .val_bits = 32, @@ -1184,9 +1183,6 @@ static int fsl_spdif_probe(struct platform_device *pdev) memcpy(&spdif_priv->cpu_dai_drv, &fsl_spdif_dai, sizeof(fsl_spdif_dai)); spdif_priv->cpu_dai_drv.name = spdif_priv->name; - if (of_property_read_bool(np, "big-endian")) - fsl_spdif_regmap_config.val_format_endian = REGMAP_ENDIAN_BIG; - /* Get the addresses and IRQ */ res = platform_get_resource(pdev, IORESOURCE_MEM, 0); regs = devm_ioremap_resource(&pdev->dev, res); diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 87eb5776a39b..e6955170dc42 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -169,6 +169,7 @@ struct fsl_ssi_private { u8 i2s_mode; bool use_dma; bool use_dual_fifo; + bool has_ipg_clk_name; unsigned int fifo_depth; struct fsl_ssi_rxtx_reg_val rxtx_reg_val; @@ -259,6 +260,11 @@ static bool fsl_ssi_is_i2s_master(struct fsl_ssi_private *ssi_private) SND_SOC_DAIFMT_CBS_CFS; } +static bool fsl_ssi_is_i2s_cbm_cfs(struct fsl_ssi_private *ssi_private) +{ + return (ssi_private->dai_fmt & SND_SOC_DAIFMT_MASTER_MASK) == + SND_SOC_DAIFMT_CBM_CFS; +} /** * fsl_ssi_isr: SSI interrupt handler * @@ -525,6 +531,11 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(rtd->cpu_dai); + int ret; + + ret = clk_prepare_enable(ssi_private->clk); + if (ret) + return ret; /* When using dual fifo mode, it is safer to ensure an even period * size. If appearing to an odd number while DMA always starts its @@ -539,6 +550,21 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, } /** + * fsl_ssi_shutdown: shutdown the SSI + * + */ +static void fsl_ssi_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct fsl_ssi_private *ssi_private = + snd_soc_dai_get_drvdata(rtd->cpu_dai); + + clk_disable_unprepare(ssi_private->clk); + +} + +/** * fsl_ssi_set_bclk - configure Digital Audio Interface bit clock * * Note: This function can be only called when using SSI as DAI master @@ -705,6 +731,23 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, } } + if (!fsl_ssi_is_ac97(ssi_private)) { + u8 i2smode; + /* + * Switch to normal net mode in order to have a frame sync + * signal every 32 bits instead of 16 bits + */ + if (fsl_ssi_is_i2s_cbm_cfs(ssi_private) && sample_size == 16) + i2smode = CCSR_SSI_SCR_I2S_MODE_NORMAL | + CCSR_SSI_SCR_NET; + else + i2smode = ssi_private->i2s_mode; + + regmap_update_bits(regs, CCSR_SSI_SCR, + CCSR_SSI_SCR_NET | CCSR_SSI_SCR_I2S_MODE_MASK, + channels == 1 ? 0 : i2smode); + } + /* * FIXME: The documentation says that SxCCR[WL] should not be * modified while the SSI is enabled. The only time this can @@ -724,11 +767,6 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, regmap_update_bits(regs, CCSR_SSI_SRCCR, CCSR_SSI_SxCCR_WL_MASK, wl); - if (!fsl_ssi_is_ac97(ssi_private)) - regmap_update_bits(regs, CCSR_SSI_SCR, - CCSR_SSI_SCR_NET | CCSR_SSI_SCR_I2S_MODE_MASK, - channels == 1 ? 0 : ssi_private->i2s_mode); - return 0; } @@ -748,8 +786,9 @@ static int fsl_ssi_hw_free(struct snd_pcm_substream *substream, return 0; } -static int _fsl_ssi_set_dai_fmt(struct fsl_ssi_private *ssi_private, - unsigned int fmt) +static int _fsl_ssi_set_dai_fmt(struct device *dev, + struct fsl_ssi_private *ssi_private, + unsigned int fmt) { struct regmap *regs = ssi_private->regs; u32 strcr = 0, stcr, srcr, scr, mask; @@ -758,7 +797,7 @@ static int _fsl_ssi_set_dai_fmt(struct fsl_ssi_private *ssi_private, ssi_private->dai_fmt = fmt; if (fsl_ssi_is_i2s_master(ssi_private) && IS_ERR(ssi_private->baudclk)) { - dev_err(&ssi_private->pdev->dev, "baudclk is missing which is necessary for master mode\n"); + dev_err(dev, "baudclk is missing which is necessary for master mode\n"); return -EINVAL; } @@ -780,6 +819,7 @@ static int _fsl_ssi_set_dai_fmt(struct fsl_ssi_private *ssi_private, switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFS: case SND_SOC_DAIFMT_CBS_CFS: ssi_private->i2s_mode |= CCSR_SSI_SCR_I2S_MODE_MASTER; regmap_update_bits(regs, CCSR_SSI_STCCR, @@ -853,6 +893,11 @@ static int _fsl_ssi_set_dai_fmt(struct fsl_ssi_private *ssi_private, case SND_SOC_DAIFMT_CBM_CFM: scr &= ~CCSR_SSI_SCR_SYS_CLK_EN; break; + case SND_SOC_DAIFMT_CBM_CFS: + strcr &= ~CCSR_SSI_STCR_TXDIR; + strcr |= CCSR_SSI_STCR_TFDIR; + scr &= ~CCSR_SSI_SCR_SYS_CLK_EN; + break; default: return -EINVAL; } @@ -913,7 +958,7 @@ static int fsl_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai); - return _fsl_ssi_set_dai_fmt(ssi_private, fmt); + return _fsl_ssi_set_dai_fmt(cpu_dai->dev, ssi_private, fmt); } /** @@ -1020,6 +1065,7 @@ static int fsl_ssi_dai_probe(struct snd_soc_dai *dai) static const struct snd_soc_dai_ops fsl_ssi_dai_ops = { .startup = fsl_ssi_startup, + .shutdown = fsl_ssi_shutdown, .hw_params = fsl_ssi_hw_params, .hw_free = fsl_ssi_hw_free, .set_fmt = fsl_ssi_set_dai_fmt, @@ -1145,17 +1191,22 @@ static int fsl_ssi_imx_probe(struct platform_device *pdev, u32 dmas[4]; int ret; - ssi_private->clk = devm_clk_get(&pdev->dev, NULL); + if (ssi_private->has_ipg_clk_name) + ssi_private->clk = devm_clk_get(&pdev->dev, "ipg"); + else + ssi_private->clk = devm_clk_get(&pdev->dev, NULL); if (IS_ERR(ssi_private->clk)) { ret = PTR_ERR(ssi_private->clk); dev_err(&pdev->dev, "could not get clock: %d\n", ret); return ret; } - ret = clk_prepare_enable(ssi_private->clk); - if (ret) { - dev_err(&pdev->dev, "clk_prepare_enable failed: %d\n", ret); - return ret; + if (!ssi_private->has_ipg_clk_name) { + ret = clk_prepare_enable(ssi_private->clk); + if (ret) { + dev_err(&pdev->dev, "clk_prepare_enable failed: %d\n", ret); + return ret; + } } /* For those SLAVE implementations, we ingore non-baudclk cases @@ -1213,8 +1264,9 @@ static int fsl_ssi_imx_probe(struct platform_device *pdev, return 0; error_pcm: - clk_disable_unprepare(ssi_private->clk); + if (!ssi_private->has_ipg_clk_name) + clk_disable_unprepare(ssi_private->clk); return ret; } @@ -1223,7 +1275,8 @@ static void fsl_ssi_imx_clean(struct platform_device *pdev, { if (!ssi_private->use_dma) imx_pcm_fiq_exit(pdev); - clk_disable_unprepare(ssi_private->clk); + if (!ssi_private->has_ipg_clk_name) + clk_disable_unprepare(ssi_private->clk); } static int fsl_ssi_probe(struct platform_device *pdev) @@ -1262,9 +1315,6 @@ static int fsl_ssi_probe(struct platform_device *pdev) if (sprop) { if (!strcmp(sprop, "ac97-slave")) ssi_private->dai_fmt = SND_SOC_DAIFMT_AC97; - else if (!strcmp(sprop, "i2s-slave")) - ssi_private->dai_fmt = SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_CBM_CFM; } ssi_private->use_dma = !of_property_read_bool(np, @@ -1298,8 +1348,16 @@ static int fsl_ssi_probe(struct platform_device *pdev) return -ENOMEM; } - ssi_private->regs = devm_regmap_init_mmio(&pdev->dev, iomem, + ret = of_property_match_string(np, "clock-names", "ipg"); + if (ret < 0) { + ssi_private->has_ipg_clk_name = false; + ssi_private->regs = devm_regmap_init_mmio(&pdev->dev, iomem, &fsl_ssi_regconfig); + } else { + ssi_private->has_ipg_clk_name = true; + ssi_private->regs = devm_regmap_init_mmio_clk(&pdev->dev, + "ipg", iomem, &fsl_ssi_regconfig); + } if (IS_ERR(ssi_private->regs)) { dev_err(&pdev->dev, "Failed to init register map\n"); return PTR_ERR(ssi_private->regs); @@ -1387,7 +1445,8 @@ static int fsl_ssi_probe(struct platform_device *pdev) done: if (ssi_private->dai_fmt) - _fsl_ssi_set_dai_fmt(ssi_private, ssi_private->dai_fmt); + _fsl_ssi_set_dai_fmt(&pdev->dev, ssi_private, + ssi_private->dai_fmt); return 0; diff --git a/sound/soc/fsl/imx-es8328.c b/sound/soc/fsl/imx-es8328.c new file mode 100644 index 000000000000..653e66d150c8 --- /dev/null +++ b/sound/soc/fsl/imx-es8328.c @@ -0,0 +1,232 @@ +/* + * Copyright 2012 Freescale Semiconductor, Inc. + * Copyright 2012 Linaro Ltd. + * + * The code contained herein is licensed under the GNU General Public + * License. You may obtain a copy of the GNU General Public License + * Version 2 or later at the following locations: + * + * http://www.opensource.org/licenses/gpl-license.html + * http://www.gnu.org/copyleft/gpl.html + */ + +#include <linux/gpio.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/of_platform.h> +#include <linux/i2c.h> +#include <linux/of_gpio.h> +#include <sound/soc.h> +#include <sound/jack.h> + +#include "imx-audmux.h" + +#define DAI_NAME_SIZE 32 +#define MUX_PORT_MAX 7 + +struct imx_es8328_data { + struct device *dev; + struct snd_soc_dai_link dai; + struct snd_soc_card card; + char codec_dai_name[DAI_NAME_SIZE]; + char platform_name[DAI_NAME_SIZE]; + int jack_gpio; +}; + +static struct snd_soc_jack_gpio headset_jack_gpios[] = { + { + .gpio = -1, + .name = "headset-gpio", + .report = SND_JACK_HEADSET, + .invert = 0, + .debounce_time = 200, + }, +}; + +static struct snd_soc_jack headset_jack; + +static int imx_es8328_dai_init(struct snd_soc_pcm_runtime *rtd) +{ + struct imx_es8328_data *data = container_of(rtd->card, + struct imx_es8328_data, card); + int ret = 0; + + /* Headphone jack detection */ + if (gpio_is_valid(data->jack_gpio)) { + ret = snd_soc_jack_new(rtd->codec, "Headphone", + SND_JACK_HEADPHONE | SND_JACK_BTN_0, + &headset_jack); + if (ret) + return ret; + + headset_jack_gpios[0].gpio = data->jack_gpio; + ret = snd_soc_jack_add_gpios(&headset_jack, + ARRAY_SIZE(headset_jack_gpios), + headset_jack_gpios); + } + + return ret; +} + +static const struct snd_soc_dapm_widget imx_es8328_dapm_widgets[] = { + SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_REGULATOR_SUPPLY("audio-amp", 1, 0), +}; + +static int imx_es8328_probe(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + struct device_node *ssi_np, *codec_np; + struct platform_device *ssi_pdev; + struct imx_es8328_data *data; + u32 int_port, ext_port; + int ret; + struct device *dev = &pdev->dev; + + ret = of_property_read_u32(np, "mux-int-port", &int_port); + if (ret) { + dev_err(dev, "mux-int-port missing or invalid\n"); + goto fail; + } + if (int_port > MUX_PORT_MAX || int_port == 0) { + dev_err(dev, "mux-int-port: hardware only has %d mux ports\n", + MUX_PORT_MAX); + goto fail; + } + + ret = of_property_read_u32(np, "mux-ext-port", &ext_port); + if (ret) { + dev_err(dev, "mux-ext-port missing or invalid\n"); + goto fail; + } + if (ext_port > MUX_PORT_MAX || ext_port == 0) { + dev_err(dev, "mux-ext-port: hardware only has %d mux ports\n", + MUX_PORT_MAX); + goto fail; + } + + /* + * The port numbering in the hardware manual starts at 1, while + * the audmux API expects it starts at 0. + */ + int_port--; + ext_port--; + ret = imx_audmux_v2_configure_port(int_port, + IMX_AUDMUX_V2_PTCR_SYN | + IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_TFSDIR | + IMX_AUDMUX_V2_PTCR_TCLKDIR, + IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port)); + if (ret) { + dev_err(dev, "audmux internal port setup failed\n"); + return ret; + } + ret = imx_audmux_v2_configure_port(ext_port, + IMX_AUDMUX_V2_PTCR_SYN, + IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)); + if (ret) { + dev_err(dev, "audmux external port setup failed\n"); + return ret; + } + + ssi_np = of_parse_phandle(pdev->dev.of_node, "ssi-controller", 0); + codec_np = of_parse_phandle(pdev->dev.of_node, "audio-codec", 0); + if (!ssi_np || !codec_np) { + dev_err(dev, "phandle missing or invalid\n"); + ret = -EINVAL; + goto fail; + } + + ssi_pdev = of_find_device_by_node(ssi_np); + if (!ssi_pdev) { + dev_err(dev, "failed to find SSI platform device\n"); + ret = -EINVAL; + goto fail; + } + + data = devm_kzalloc(dev, sizeof(*data), GFP_KERNEL); + if (!data) { + ret = -ENOMEM; + goto fail; + } + + data->dev = dev; + + data->jack_gpio = of_get_named_gpio(pdev->dev.of_node, "jack-gpio", 0); + + data->dai.name = "hifi"; + data->dai.stream_name = "hifi"; + data->dai.codec_dai_name = "es8328-hifi-analog"; + data->dai.codec_of_node = codec_np; + data->dai.cpu_of_node = ssi_np; + data->dai.platform_of_node = ssi_np; + data->dai.init = &imx_es8328_dai_init; + data->dai.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM; + + data->card.dev = dev; + data->card.dapm_widgets = imx_es8328_dapm_widgets; + data->card.num_dapm_widgets = ARRAY_SIZE(imx_es8328_dapm_widgets); + ret = snd_soc_of_parse_card_name(&data->card, "model"); + if (ret) { + dev_err(dev, "Unable to parse card name\n"); + goto fail; + } + ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing"); + if (ret) { + dev_err(dev, "Unable to parse routing: %d\n", ret); + goto fail; + } + data->card.num_links = 1; + data->card.owner = THIS_MODULE; + data->card.dai_link = &data->dai; + + ret = snd_soc_register_card(&data->card); + if (ret) { + dev_err(dev, "Unable to register: %d\n", ret); + goto fail; + } + + platform_set_drvdata(pdev, data); +fail: + of_node_put(ssi_np); + of_node_put(codec_np); + + return ret; +} + +static int imx_es8328_remove(struct platform_device *pdev) +{ + struct imx_es8328_data *data = platform_get_drvdata(pdev); + + snd_soc_jack_free_gpios(&headset_jack, ARRAY_SIZE(headset_jack_gpios), + headset_jack_gpios); + + snd_soc_unregister_card(&data->card); + + return 0; +} + +static const struct of_device_id imx_es8328_dt_ids[] = { + { .compatible = "fsl,imx-audio-es8328", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, imx_es8328_dt_ids); + +static struct platform_driver imx_es8328_driver = { + .driver = { + .name = "imx-es8328", + .of_match_table = imx_es8328_dt_ids, + }, + .probe = imx_es8328_probe, + .remove = imx_es8328_remove, +}; +module_platform_driver(imx_es8328_driver); + +MODULE_AUTHOR("Sean Cross <xobs@kosagi.com>"); +MODULE_DESCRIPTION("Kosagi i.MX6 ES8328 ASoC machine driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:imx-audio-es8328"); diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index cef7776b712c..fcb431fe20b4 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -10,10 +10,13 @@ */ #include <linux/clk.h> #include <linux/device.h> +#include <linux/gpio.h> #include <linux/module.h> #include <linux/of.h> +#include <linux/of_gpio.h> #include <linux/platform_device.h> #include <linux/string.h> +#include <sound/jack.h> #include <sound/simple_card.h> #include <sound/soc-dai.h> #include <sound/soc.h> @@ -25,9 +28,15 @@ struct simple_card_data { struct asoc_simple_dai codec_dai; } *dai_props; unsigned int mclk_fs; + int gpio_hp_det; + int gpio_mic_det; struct snd_soc_dai_link dai_link[]; /* dynamically allocated */ }; +#define simple_priv_to_dev(priv) ((priv)->snd_card.dev) +#define simple_priv_to_link(priv, i) ((priv)->snd_card.dai_link + i) +#define simple_priv_to_props(priv, i) ((priv)->dai_props + i) + static int asoc_simple_card_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -50,6 +59,32 @@ static struct snd_soc_ops asoc_simple_card_ops = { .hw_params = asoc_simple_card_hw_params, }; +static struct snd_soc_jack simple_card_hp_jack; +static struct snd_soc_jack_pin simple_card_hp_jack_pins[] = { + { + .pin = "Headphones", + .mask = SND_JACK_HEADPHONE, + }, +}; +static struct snd_soc_jack_gpio simple_card_hp_jack_gpio = { + .name = "Headphone detection", + .report = SND_JACK_HEADPHONE, + .debounce_time = 150, +}; + +static struct snd_soc_jack simple_card_mic_jack; +static struct snd_soc_jack_pin simple_card_mic_jack_pins[] = { + { + .pin = "Mic Jack", + .mask = SND_JACK_MICROPHONE, + }, +}; +static struct snd_soc_jack_gpio simple_card_mic_jack_gpio = { + .name = "Mic detection", + .report = SND_JACK_MICROPHONE, + .debounce_time = 150, +}; + static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai, struct asoc_simple_dai *set) { @@ -105,42 +140,70 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) if (ret < 0) return ret; + if (gpio_is_valid(priv->gpio_hp_det)) { + snd_soc_jack_new(codec->codec, "Headphones", SND_JACK_HEADPHONE, + &simple_card_hp_jack); + snd_soc_jack_add_pins(&simple_card_hp_jack, + ARRAY_SIZE(simple_card_hp_jack_pins), + simple_card_hp_jack_pins); + + simple_card_hp_jack_gpio.gpio = priv->gpio_hp_det; + snd_soc_jack_add_gpios(&simple_card_hp_jack, 1, + &simple_card_hp_jack_gpio); + } + + if (gpio_is_valid(priv->gpio_mic_det)) { + snd_soc_jack_new(codec->codec, "Mic Jack", SND_JACK_MICROPHONE, + &simple_card_mic_jack); + snd_soc_jack_add_pins(&simple_card_mic_jack, + ARRAY_SIZE(simple_card_mic_jack_pins), + simple_card_mic_jack_pins); + simple_card_mic_jack_gpio.gpio = priv->gpio_mic_det; + snd_soc_jack_add_gpios(&simple_card_mic_jack, 1, + &simple_card_mic_jack_gpio); + } return 0; } static int asoc_simple_card_sub_parse_of(struct device_node *np, struct asoc_simple_dai *dai, - const struct device_node **p_node, - const char **name) + struct device_node **p_node, + const char **name, + int *args_count) { - struct device_node *node; + struct of_phandle_args args; struct clk *clk; u32 val; int ret; /* - * get node via "sound-dai = <&phandle port>" + * Get node via "sound-dai = <&phandle port>" * it will be used as xxx_of_node on soc_bind_dai_link() */ - node = of_parse_phandle(np, "sound-dai", 0); - if (!node) - return -ENODEV; - *p_node = node; + ret = of_parse_phandle_with_args(np, "sound-dai", + "#sound-dai-cells", 0, &args); + if (ret) + return ret; + + *p_node = args.np; - /* get dai->name */ + if (args_count) + *args_count = args.args_count; + + /* Get dai->name */ ret = snd_soc_of_get_dai_name(np, name); if (ret < 0) return ret; - /* parse TDM slot */ + /* Parse TDM slot */ ret = snd_soc_of_parse_tdm_slot(np, &dai->slots, &dai->slot_width); if (ret) return ret; /* - * dai->sysclk come from - * "clocks = <&xxx>" (if system has common clock) + * Parse dai->sysclk come from "clocks = <&xxx>" + * (if system has common clock) * or "system-clock-frequency = <xxx>" * or device's module clock. */ @@ -155,7 +218,7 @@ asoc_simple_card_sub_parse_of(struct device_node *np, } else if (!of_property_read_u32(np, "system-clock-frequency", &val)) { dai->sysclk = val; } else { - clk = of_clk_get(node, 0); + clk = of_clk_get(args.np, 0); if (!IS_ERR(clk)) dai->sysclk = clk_get_rate(clk); } @@ -163,12 +226,14 @@ asoc_simple_card_sub_parse_of(struct device_node *np, return 0; } -static int simple_card_dai_link_of(struct device_node *node, - struct device *dev, - struct snd_soc_dai_link *dai_link, - struct simple_dai_props *dai_props, - bool is_top_level_node) +static int asoc_simple_card_dai_link_of(struct device_node *node, + struct simple_card_data *priv, + int idx, + bool is_top_level_node) { + struct device *dev = simple_priv_to_dev(priv); + struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, idx); + struct simple_dai_props *dai_props = simple_priv_to_props(priv, idx); struct device_node *np = NULL; struct device_node *bitclkmaster = NULL; struct device_node *framemaster = NULL; @@ -176,8 +241,9 @@ static int simple_card_dai_link_of(struct device_node *node, char *name; char prop[128]; char *prefix = ""; - int ret; + int ret, cpu_args; + /* For single DAI link & old style of DT node */ if (is_top_level_node) prefix = "simple-audio-card,"; @@ -195,7 +261,8 @@ static int simple_card_dai_link_of(struct device_node *node, ret = asoc_simple_card_sub_parse_of(np, &dai_props->cpu_dai, &dai_link->cpu_of_node, - &dai_link->cpu_dai_name); + &dai_link->cpu_dai_name, + &cpu_args); if (ret < 0) goto dai_link_of_err; @@ -226,14 +293,16 @@ static int simple_card_dai_link_of(struct device_node *node, ret = asoc_simple_card_sub_parse_of(np, &dai_props->codec_dai, &dai_link->codec_of_node, - &dai_link->codec_dai_name); + &dai_link->codec_dai_name, NULL); if (ret < 0) goto dai_link_of_err; if (strlen(prefix) && !bitclkmaster && !framemaster) { - /* No dai-link level and master setting was not found from - sound node level, revert back to legacy DT parsing and - take the settings from codec node. */ + /* + * No DAI link level and master setting was found + * from sound node level, revert back to legacy DT + * parsing and take the settings from codec node. + */ dev_dbg(dev, "%s: Revert to legacy daifmt parsing\n", __func__); dai_props->cpu_dai.fmt = dai_props->codec_dai.fmt = @@ -262,10 +331,10 @@ static int simple_card_dai_link_of(struct device_node *node, goto dai_link_of_err; } - /* simple-card assumes platform == cpu */ + /* Simple Card assumes platform == cpu */ dai_link->platform_of_node = dai_link->cpu_of_node; - /* Link name is created from CPU/CODEC dai name */ + /* DAI link name is created from CPU/CODEC dai name */ name = devm_kzalloc(dev, strlen(dai_link->cpu_dai_name) + strlen(dai_link->codec_dai_name) + 2, @@ -274,6 +343,7 @@ static int simple_card_dai_link_of(struct device_node *node, dai_link->codec_dai_name); dai_link->name = dai_link->stream_name = name; dai_link->ops = &asoc_simple_card_ops; + dai_link->init = asoc_simple_card_dai_init; dev_dbg(dev, "\tname : %s\n", dai_link->stream_name); dev_dbg(dev, "\tcpu : %s / %04x / %d\n", @@ -285,6 +355,18 @@ static int simple_card_dai_link_of(struct device_node *node, dai_props->codec_dai.fmt, dai_props->codec_dai.sysclk); + /* + * In soc_bind_dai_link() will check cpu name after + * of_node matching if dai_link has cpu_dai_name. + * but, it will never match if name was created by + * fmt_single_name() remove cpu_dai_name if cpu_args + * was 0. See: + * fmt_single_name() + * fmt_multiple_name() + */ + if (!cpu_args) + dai_link->cpu_dai_name = NULL; + dai_link_of_err: if (np) of_node_put(np); @@ -296,19 +378,19 @@ dai_link_of_err: } static int asoc_simple_card_parse_of(struct device_node *node, - struct simple_card_data *priv, - struct device *dev, - int multi) + struct simple_card_data *priv) { - struct snd_soc_dai_link *dai_link = priv->snd_card.dai_link; - struct simple_dai_props *dai_props = priv->dai_props; + struct device *dev = simple_priv_to_dev(priv); u32 val; int ret; - /* parsing the card name from DT */ + if (!node) + return -EINVAL; + + /* Parse the card name from DT */ snd_soc_of_parse_card_name(&priv->snd_card, "simple-audio-card,name"); - /* off-codec widgets */ + /* The off-codec widgets */ if (of_property_read_bool(node, "simple-audio-card,widgets")) { ret = snd_soc_of_parse_audio_simple_widgets(&priv->snd_card, "simple-audio-card,widgets"); @@ -332,32 +414,45 @@ static int asoc_simple_card_parse_of(struct device_node *node, dev_dbg(dev, "New simple-card: %s\n", priv->snd_card.name ? priv->snd_card.name : ""); - if (multi) { + /* Single/Muti DAI link(s) & New style of DT node */ + if (of_get_child_by_name(node, "simple-audio-card,dai-link")) { struct device_node *np = NULL; - int i; - for (i = 0; (np = of_get_next_child(node, np)); i++) { + int i = 0; + + for_each_child_of_node(node, np) { dev_dbg(dev, "\tlink %d:\n", i); - ret = simple_card_dai_link_of(np, dev, dai_link + i, - dai_props + i, false); + ret = asoc_simple_card_dai_link_of(np, priv, + i, false); if (ret < 0) { of_node_put(np); return ret; } + i++; } } else { - ret = simple_card_dai_link_of(node, dev, dai_link, dai_props, - true); + /* For single DAI link & old style of DT node */ + ret = asoc_simple_card_dai_link_of(node, priv, 0, true); if (ret < 0) return ret; } + priv->gpio_hp_det = of_get_named_gpio(node, + "simple-audio-card,hp-det-gpio", 0); + if (priv->gpio_hp_det == -EPROBE_DEFER) + return -EPROBE_DEFER; + + priv->gpio_mic_det = of_get_named_gpio(node, + "simple-audio-card,mic-det-gpio", 0); + if (priv->gpio_mic_det == -EPROBE_DEFER) + return -EPROBE_DEFER; + if (!priv->snd_card.name) priv->snd_card.name = priv->snd_card.dai_link->name; return 0; } -/* update the reference count of the devices nodes at end of probe */ +/* Decrease the reference count of the device nodes */ static int asoc_simple_card_unref(struct platform_device *pdev) { struct snd_soc_card *card = platform_get_drvdata(pdev); @@ -384,34 +479,29 @@ static int asoc_simple_card_probe(struct platform_device *pdev) struct snd_soc_dai_link *dai_link; struct device_node *np = pdev->dev.of_node; struct device *dev = &pdev->dev; - int num_links, multi, ret; + int num_links, ret; - /* get the number of DAI links */ - if (np && of_get_child_by_name(np, "simple-audio-card,dai-link")) { + /* Get the number of DAI links */ + if (np && of_get_child_by_name(np, "simple-audio-card,dai-link")) num_links = of_get_child_count(np); - multi = 1; - } else { + else num_links = 1; - multi = 0; - } - /* allocate the private data and the DAI link array */ + /* Allocate the private data and the DAI link array */ priv = devm_kzalloc(dev, sizeof(*priv) + sizeof(*dai_link) * num_links, GFP_KERNEL); if (!priv) return -ENOMEM; - /* - * init snd_soc_card - */ + /* Init snd_soc_card */ priv->snd_card.owner = THIS_MODULE; priv->snd_card.dev = dev; dai_link = priv->dai_link; priv->snd_card.dai_link = dai_link; priv->snd_card.num_links = num_links; - /* get room for the other properties */ + /* Get room for the other properties */ priv->dai_props = devm_kzalloc(dev, sizeof(*priv->dai_props) * num_links, GFP_KERNEL); @@ -420,25 +510,13 @@ static int asoc_simple_card_probe(struct platform_device *pdev) if (np && of_device_is_available(np)) { - ret = asoc_simple_card_parse_of(np, priv, dev, multi); + ret = asoc_simple_card_parse_of(np, priv); if (ret < 0) { if (ret != -EPROBE_DEFER) dev_err(dev, "parse error %d\n", ret); goto err; } - /* - * soc_bind_dai_link() will check cpu name - * after of_node matching if dai_link has cpu_dai_name. - * but, it will never match if name was created by fmt_single_name() - * remove cpu_dai_name to escape name matching. - * see - * fmt_single_name() - * fmt_multiple_name() - */ - if (num_links == 1) - dai_link->cpu_dai_name = NULL; - } else { struct asoc_simple_card_info *cinfo; @@ -464,6 +542,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) dai_link->codec_name = cinfo->codec; dai_link->cpu_dai_name = cinfo->cpu_dai.name; dai_link->codec_dai_name = cinfo->codec_dai.name; + dai_link->init = asoc_simple_card_dai_init; memcpy(&priv->dai_props->cpu_dai, &cinfo->cpu_dai, sizeof(priv->dai_props->cpu_dai)); memcpy(&priv->dai_props->codec_dai, &cinfo->codec_dai, @@ -473,11 +552,6 @@ static int asoc_simple_card_probe(struct platform_device *pdev) priv->dai_props->codec_dai.fmt |= cinfo->daifmt; } - /* - * init snd_soc_dai_link - */ - dai_link->init = asoc_simple_card_dai_init; - snd_soc_card_set_drvdata(&priv->snd_card, priv); ret = devm_snd_soc_register_card(&pdev->dev, &priv->snd_card); @@ -491,6 +565,16 @@ err: static int asoc_simple_card_remove(struct platform_device *pdev) { + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct simple_card_data *priv = snd_soc_card_get_drvdata(card); + + if (gpio_is_valid(priv->gpio_hp_det)) + snd_soc_jack_free_gpios(&simple_card_hp_jack, 1, + &simple_card_hp_jack_gpio); + if (gpio_is_valid(priv->gpio_mic_det)) + snd_soc_jack_free_gpios(&simple_card_mic_jack, 1, + &simple_card_mic_jack_gpio); + return asoc_simple_card_unref(pdev); } diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile index 7acbfc43a0c6..f841786dad15 100644 --- a/sound/soc/intel/Makefile +++ b/sound/soc/intel/Makefile @@ -2,7 +2,8 @@ snd-soc-sst-dsp-objs := sst-dsp.o sst-firmware.o snd-soc-sst-acpi-objs := sst-acpi.o -snd-soc-sst-mfld-platform-objs := sst-mfld-platform-pcm.o sst-mfld-platform-compress.o +snd-soc-sst-mfld-platform-objs := sst-mfld-platform-pcm.o \ + sst-mfld-platform-compress.o sst-atom-controls.o snd-soc-mfld-machine-objs := mfld_machine.o obj-$(CONFIG_SND_SST_MFLD_PLATFORM) += snd-soc-sst-mfld-platform.o diff --git a/sound/soc/intel/byt-max98090.c b/sound/soc/intel/byt-max98090.c index b8b8af571ef1..d52681e7225e 100644 --- a/sound/soc/intel/byt-max98090.c +++ b/sound/soc/intel/byt-max98090.c @@ -139,6 +139,7 @@ static struct snd_soc_card byt_max98090_card = { .num_dapm_routes = ARRAY_SIZE(byt_max98090_audio_map), .controls = byt_max98090_controls, .num_controls = ARRAY_SIZE(byt_max98090_controls), + .fully_routed = true, }; static int byt_max98090_probe(struct platform_device *pdev) diff --git a/sound/soc/intel/byt-rt5640.c b/sound/soc/intel/byt-rt5640.c index 234a58de3c53..e03abdf21c1b 100644 --- a/sound/soc/intel/byt-rt5640.c +++ b/sound/soc/intel/byt-rt5640.c @@ -17,6 +17,7 @@ #include <linux/platform_device.h> #include <linux/acpi.h> #include <linux/device.h> +#include <linux/dmi.h> #include <linux/slab.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -36,8 +37,6 @@ static const struct snd_soc_dapm_widget byt_rt5640_widgets[] = { static const struct snd_soc_dapm_route byt_rt5640_audio_map[] = { {"Headset Mic", NULL, "MICBIAS1"}, {"IN2P", NULL, "Headset Mic"}, - {"IN2N", NULL, "Headset Mic"}, - {"DMIC1", NULL, "Internal Mic"}, {"Headphone", NULL, "HPOL"}, {"Headphone", NULL, "HPOR"}, {"Speaker", NULL, "SPOLP"}, @@ -46,6 +45,31 @@ static const struct snd_soc_dapm_route byt_rt5640_audio_map[] = { {"Speaker", NULL, "SPORN"}, }; +static const struct snd_soc_dapm_route byt_rt5640_intmic_dmic1_map[] = { + {"DMIC1", NULL, "Internal Mic"}, +}; + +static const struct snd_soc_dapm_route byt_rt5640_intmic_dmic2_map[] = { + {"DMIC2", NULL, "Internal Mic"}, +}; + +static const struct snd_soc_dapm_route byt_rt5640_intmic_in1_map[] = { + {"Internal Mic", NULL, "MICBIAS1"}, + {"IN1P", NULL, "Internal Mic"}, +}; + +enum { + BYT_RT5640_DMIC1_MAP, + BYT_RT5640_DMIC2_MAP, + BYT_RT5640_IN1_MAP, +}; + +#define BYT_RT5640_MAP(quirk) ((quirk) & 0xff) +#define BYT_RT5640_DMIC_EN BIT(16) + +static unsigned long byt_rt5640_quirk = BYT_RT5640_DMIC1_MAP | + BYT_RT5640_DMIC_EN; + static const struct snd_kcontrol_new byt_rt5640_controls[] = { SOC_DAPM_PIN_SWITCH("Headphone"), SOC_DAPM_PIN_SWITCH("Headset Mic"), @@ -77,12 +101,41 @@ static int byt_rt5640_hw_params(struct snd_pcm_substream *substream, return 0; } +static int byt_rt5640_quirk_cb(const struct dmi_system_id *id) +{ + byt_rt5640_quirk = (unsigned long)id->driver_data; + return 1; +} + +static const struct dmi_system_id byt_rt5640_quirk_table[] = { + { + .callback = byt_rt5640_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."), + DMI_MATCH(DMI_PRODUCT_NAME, "T100TA"), + }, + .driver_data = (unsigned long *)BYT_RT5640_IN1_MAP, + }, + { + .callback = byt_rt5640_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "DellInc."), + DMI_MATCH(DMI_PRODUCT_NAME, "Venue 8 Pro 5830"), + }, + .driver_data = (unsigned long *)(BYT_RT5640_DMIC2_MAP | + BYT_RT5640_DMIC_EN), + }, + {} +}; + static int byt_rt5640_init(struct snd_soc_pcm_runtime *runtime) { int ret; struct snd_soc_codec *codec = runtime->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; struct snd_soc_card *card = runtime->card; + const struct snd_soc_dapm_route *custom_map; + int num_routes; card->dapm.idle_bias_off = true; @@ -93,6 +146,31 @@ static int byt_rt5640_init(struct snd_soc_pcm_runtime *runtime) return ret; } + dmi_check_system(byt_rt5640_quirk_table); + switch (BYT_RT5640_MAP(byt_rt5640_quirk)) { + case BYT_RT5640_IN1_MAP: + custom_map = byt_rt5640_intmic_in1_map; + num_routes = ARRAY_SIZE(byt_rt5640_intmic_in1_map); + break; + case BYT_RT5640_DMIC2_MAP: + custom_map = byt_rt5640_intmic_dmic2_map; + num_routes = ARRAY_SIZE(byt_rt5640_intmic_dmic2_map); + break; + default: + custom_map = byt_rt5640_intmic_dmic1_map; + num_routes = ARRAY_SIZE(byt_rt5640_intmic_dmic1_map); + } + + ret = snd_soc_dapm_add_routes(dapm, custom_map, num_routes); + if (ret) + return ret; + + if (byt_rt5640_quirk & BYT_RT5640_DMIC_EN) { + ret = rt5640_dmic_enable(codec, 0, 0); + if (ret) + return ret; + } + snd_soc_dapm_ignore_suspend(dapm, "HPOL"); snd_soc_dapm_ignore_suspend(dapm, "HPOR"); @@ -131,6 +209,7 @@ static struct snd_soc_card byt_rt5640_card = { .num_dapm_widgets = ARRAY_SIZE(byt_rt5640_widgets), .dapm_routes = byt_rt5640_audio_map, .num_dapm_routes = ARRAY_SIZE(byt_rt5640_audio_map), + .fully_routed = true, }; static int byt_rt5640_probe(struct platform_device *pdev) diff --git a/sound/soc/intel/sst-atom-controls.c b/sound/soc/intel/sst-atom-controls.c new file mode 100644 index 000000000000..7104a34181a9 --- /dev/null +++ b/sound/soc/intel/sst-atom-controls.c @@ -0,0 +1,218 @@ +/* + * sst-atom-controls.c - Intel MID Platform driver DPCM ALSA controls for Mrfld + * + * Copyright (C) 2013-14 Intel Corp + * Author: Omair Mohammed Abdullah <omair.m.abdullah@intel.com> + * Vinod Koul <vinod.koul@intel.com> + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * 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; version 2 of the License. + * + * 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. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + */ +#define pr_fmt(fmt) KBUILD_MODNAME ": " fmt + +#include <linux/slab.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include "sst-mfld-platform.h" +#include "sst-atom-controls.h" + +static int sst_fill_byte_control(struct sst_data *drv, + u8 ipc_msg, u8 block, + u8 task_id, u8 pipe_id, + u16 len, void *cmd_data) +{ + struct snd_sst_bytes_v2 *byte_data = drv->byte_stream; + + byte_data->type = SST_CMD_BYTES_SET; + byte_data->ipc_msg = ipc_msg; + byte_data->block = block; + byte_data->task_id = task_id; + byte_data->pipe_id = pipe_id; + + if (len > SST_MAX_BIN_BYTES - sizeof(*byte_data)) { + dev_err(&drv->pdev->dev, "command length too big (%u)", len); + return -EINVAL; + } + byte_data->len = len; + memcpy(byte_data->bytes, cmd_data, len); + print_hex_dump_bytes("writing to lpe: ", DUMP_PREFIX_OFFSET, + byte_data, len + sizeof(*byte_data)); + return 0; +} + +static int sst_fill_and_send_cmd_unlocked(struct sst_data *drv, + u8 ipc_msg, u8 block, u8 task_id, u8 pipe_id, + void *cmd_data, u16 len) +{ + int ret = 0; + + ret = sst_fill_byte_control(drv, ipc_msg, + block, task_id, pipe_id, len, cmd_data); + if (ret < 0) + return ret; + return sst->ops->send_byte_stream(sst->dev, drv->byte_stream); +} + +/** + * sst_fill_and_send_cmd - generate the IPC message and send it to the FW + * @ipc_msg: type of IPC (CMD, SET_PARAMS, GET_PARAMS) + * @cmd_data: the IPC payload + */ +static int sst_fill_and_send_cmd(struct sst_data *drv, + u8 ipc_msg, u8 block, u8 task_id, u8 pipe_id, + void *cmd_data, u16 len) +{ + int ret; + + mutex_lock(&drv->lock); + ret = sst_fill_and_send_cmd_unlocked(drv, ipc_msg, block, + task_id, pipe_id, cmd_data, len); + mutex_unlock(&drv->lock); + + return ret; +} + +static int sst_send_algo_cmd(struct sst_data *drv, + struct sst_algo_control *bc) +{ + int len, ret = 0; + struct sst_cmd_set_params *cmd; + + /*bc->max includes sizeof algos + length field*/ + len = sizeof(cmd->dst) + sizeof(cmd->command_id) + bc->max; + + cmd = kzalloc(len, GFP_KERNEL); + if (cmd == NULL) + return -ENOMEM; + + SST_FILL_DESTINATION(2, cmd->dst, bc->pipe_id, bc->module_id); + cmd->command_id = bc->cmd_id; + memcpy(cmd->params, bc->params, bc->max); + + ret = sst_fill_and_send_cmd_unlocked(drv, SST_IPC_IA_SET_PARAMS, + SST_FLAG_BLOCKED, bc->task_id, 0, cmd, len); + kfree(cmd); + return ret; +} + +static int sst_algo_bytes_ctl_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct sst_algo_control *bc = (void *)kcontrol->private_value; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; + uinfo->count = bc->max; + + return 0; +} + +static int sst_algo_control_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct sst_algo_control *bc = (void *)kcontrol->private_value; + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + + switch (bc->type) { + case SST_ALGO_PARAMS: + memcpy(ucontrol->value.bytes.data, bc->params, bc->max); + break; + default: + dev_err(component->dev, "Invalid Input- algo type:%d\n", + bc->type); + return -EINVAL; + + } + return 0; +} + +static int sst_algo_control_set(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int ret = 0; + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct sst_data *drv = snd_soc_component_get_drvdata(cmpnt); + struct sst_algo_control *bc = (void *)kcontrol->private_value; + + dev_dbg(cmpnt->dev, "control_name=%s\n", kcontrol->id.name); + mutex_lock(&drv->lock); + switch (bc->type) { + case SST_ALGO_PARAMS: + memcpy(bc->params, ucontrol->value.bytes.data, bc->max); + break; + default: + mutex_unlock(&drv->lock); + dev_err(cmpnt->dev, "Invalid Input- algo type:%d\n", + bc->type); + return -EINVAL; + } + /*if pipe is enabled, need to send the algo params from here*/ + if (bc->w && bc->w->power) + ret = sst_send_algo_cmd(drv, bc); + mutex_unlock(&drv->lock); + + return ret; +} + +static const struct snd_kcontrol_new sst_algo_controls[] = { + SST_ALGO_KCONTROL_BYTES("media_loop1_out", "fir", 272, SST_MODULE_ID_FIR_24, + SST_PATH_INDEX_MEDIA_LOOP1_OUT, 0, SST_TASK_SBA, SBA_VB_SET_FIR), + SST_ALGO_KCONTROL_BYTES("media_loop1_out", "iir", 300, SST_MODULE_ID_IIR_24, + SST_PATH_INDEX_MEDIA_LOOP1_OUT, 0, SST_TASK_SBA, SBA_VB_SET_IIR), + SST_ALGO_KCONTROL_BYTES("media_loop1_out", "mdrp", 286, SST_MODULE_ID_MDRP, + SST_PATH_INDEX_MEDIA_LOOP1_OUT, 0, SST_TASK_SBA, SBA_SET_MDRP), + SST_ALGO_KCONTROL_BYTES("media_loop2_out", "fir", 272, SST_MODULE_ID_FIR_24, + SST_PATH_INDEX_MEDIA_LOOP2_OUT, 0, SST_TASK_SBA, SBA_VB_SET_FIR), + SST_ALGO_KCONTROL_BYTES("media_loop2_out", "iir", 300, SST_MODULE_ID_IIR_24, + SST_PATH_INDEX_MEDIA_LOOP2_OUT, 0, SST_TASK_SBA, SBA_VB_SET_IIR), + SST_ALGO_KCONTROL_BYTES("media_loop2_out", "mdrp", 286, SST_MODULE_ID_MDRP, + SST_PATH_INDEX_MEDIA_LOOP2_OUT, 0, SST_TASK_SBA, SBA_SET_MDRP), + SST_ALGO_KCONTROL_BYTES("sprot_loop_out", "lpro", 192, SST_MODULE_ID_SPROT, + SST_PATH_INDEX_SPROT_LOOP_OUT, 0, SST_TASK_SBA, SBA_VB_LPRO), + SST_ALGO_KCONTROL_BYTES("codec_in0", "dcr", 52, SST_MODULE_ID_FILT_DCR, + SST_PATH_INDEX_CODEC_IN0, 0, SST_TASK_SBA, SBA_VB_SET_IIR), + SST_ALGO_KCONTROL_BYTES("codec_in1", "dcr", 52, SST_MODULE_ID_FILT_DCR, + SST_PATH_INDEX_CODEC_IN1, 0, SST_TASK_SBA, SBA_VB_SET_IIR), + +}; + +static int sst_algo_control_init(struct device *dev) +{ + int i = 0; + struct sst_algo_control *bc; + /*allocate space to cache the algo parameters in the driver*/ + for (i = 0; i < ARRAY_SIZE(sst_algo_controls); i++) { + bc = (struct sst_algo_control *)sst_algo_controls[i].private_value; + bc->params = devm_kzalloc(dev, bc->max, GFP_KERNEL); + if (bc->params == NULL) + return -ENOMEM; + } + return 0; +} + +int sst_dsp_init_v2_dpcm(struct snd_soc_platform *platform) +{ + int ret = 0; + struct sst_data *drv = snd_soc_platform_get_drvdata(platform); + + drv->byte_stream = devm_kzalloc(platform->dev, + SST_MAX_BIN_BYTES, GFP_KERNEL); + if (!drv->byte_stream) + return -ENOMEM; + + /*Initialize algo control params*/ + ret = sst_algo_control_init(platform->dev); + if (ret) + return ret; + ret = snd_soc_add_platform_controls(platform, sst_algo_controls, + ARRAY_SIZE(sst_algo_controls)); + return ret; +} diff --git a/sound/soc/intel/sst-atom-controls.h b/sound/soc/intel/sst-atom-controls.h index 14063ab8c7c5..a73e894b175c 100644 --- a/sound/soc/intel/sst-atom-controls.h +++ b/sound/soc/intel/sst-atom-controls.h @@ -1,4 +1,6 @@ /* + * sst-atom-controls.h - Intel MID Platform driver header file + * * Copyright (C) 2013-14 Intel Corp * Author: Ramesh Babu <ramesh.babu.koul@intel.com> * Omair M Abdullah <omair.m.abdullah@intel.com> @@ -18,13 +20,423 @@ * */ -#ifndef __SST_CONTROLS_V2_H__ -#define __SST_CONTROLS_V2_H__ +#ifndef __SST_ATOM_CONTROLS_H__ +#define __SST_ATOM_CONTROLS_H__ enum { MERR_DPCM_AUDIO = 0, MERR_DPCM_COMPR, }; +/* define a bit for each mixer input */ +#define SST_MIX_IP(x) (x) + +#define SST_IP_CODEC0 SST_MIX_IP(2) +#define SST_IP_CODEC1 SST_MIX_IP(3) +#define SST_IP_LOOP0 SST_MIX_IP(4) +#define SST_IP_LOOP1 SST_MIX_IP(5) +#define SST_IP_LOOP2 SST_MIX_IP(6) +#define SST_IP_PROBE SST_MIX_IP(7) +#define SST_IP_VOIP SST_MIX_IP(12) +#define SST_IP_PCM0 SST_MIX_IP(13) +#define SST_IP_PCM1 SST_MIX_IP(14) +#define SST_IP_MEDIA0 SST_MIX_IP(17) +#define SST_IP_MEDIA1 SST_MIX_IP(18) +#define SST_IP_MEDIA2 SST_MIX_IP(19) +#define SST_IP_MEDIA3 SST_MIX_IP(20) + +#define SST_IP_LAST SST_IP_MEDIA3 + +#define SST_SWM_INPUT_COUNT (SST_IP_LAST + 1) +#define SST_CMD_SWM_MAX_INPUTS 6 + +#define SST_PATH_ID_SHIFT 8 +#define SST_DEFAULT_LOCATION_ID 0xFFFF +#define SST_DEFAULT_CELL_NBR 0xFF +#define SST_DEFAULT_MODULE_ID 0xFFFF + +/* + * Audio DSP Path Ids. Specified by the audio DSP FW + */ +enum sst_path_index { + SST_PATH_INDEX_CODEC_OUT0 = (0x02 << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_CODEC_OUT1 = (0x03 << SST_PATH_ID_SHIFT), + + SST_PATH_INDEX_SPROT_LOOP_OUT = (0x04 << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_MEDIA_LOOP1_OUT = (0x05 << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_MEDIA_LOOP2_OUT = (0x06 << SST_PATH_ID_SHIFT), + + SST_PATH_INDEX_VOIP_OUT = (0x0C << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_PCM0_OUT = (0x0D << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_PCM1_OUT = (0x0E << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_PCM2_OUT = (0x0F << SST_PATH_ID_SHIFT), + + SST_PATH_INDEX_MEDIA0_OUT = (0x12 << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_MEDIA1_OUT = (0x13 << SST_PATH_ID_SHIFT), + + + /* Start of input paths */ + SST_PATH_INDEX_CODEC_IN0 = (0x82 << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_CODEC_IN1 = (0x83 << SST_PATH_ID_SHIFT), + + SST_PATH_INDEX_SPROT_LOOP_IN = (0x84 << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_MEDIA_LOOP1_IN = (0x85 << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_MEDIA_LOOP2_IN = (0x86 << SST_PATH_ID_SHIFT), + + SST_PATH_INDEX_VOIP_IN = (0x8C << SST_PATH_ID_SHIFT), + + SST_PATH_INDEX_PCM0_IN = (0x8D << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_PCM1_IN = (0x8E << SST_PATH_ID_SHIFT), + + SST_PATH_INDEX_MEDIA0_IN = (0x8F << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_MEDIA1_IN = (0x90 << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_MEDIA2_IN = (0x91 << SST_PATH_ID_SHIFT), + + SST_PATH_INDEX_MEDIA3_IN = (0x9C << SST_PATH_ID_SHIFT), + + SST_PATH_INDEX_RESERVED = (0xFF << SST_PATH_ID_SHIFT), +}; + +/* + * path IDs + */ +enum sst_swm_inputs { + SST_SWM_IN_CODEC0 = (SST_PATH_INDEX_CODEC_IN0 | SST_DEFAULT_CELL_NBR), + SST_SWM_IN_CODEC1 = (SST_PATH_INDEX_CODEC_IN1 | SST_DEFAULT_CELL_NBR), + SST_SWM_IN_SPROT_LOOP = (SST_PATH_INDEX_SPROT_LOOP_IN | SST_DEFAULT_CELL_NBR), + SST_SWM_IN_MEDIA_LOOP1 = (SST_PATH_INDEX_MEDIA_LOOP1_IN | SST_DEFAULT_CELL_NBR), + SST_SWM_IN_MEDIA_LOOP2 = (SST_PATH_INDEX_MEDIA_LOOP2_IN | SST_DEFAULT_CELL_NBR), + SST_SWM_IN_VOIP = (SST_PATH_INDEX_VOIP_IN | SST_DEFAULT_CELL_NBR), + SST_SWM_IN_PCM0 = (SST_PATH_INDEX_PCM0_IN | SST_DEFAULT_CELL_NBR), + SST_SWM_IN_PCM1 = (SST_PATH_INDEX_PCM1_IN | SST_DEFAULT_CELL_NBR), + SST_SWM_IN_MEDIA0 = (SST_PATH_INDEX_MEDIA0_IN | SST_DEFAULT_CELL_NBR), /* Part of Media Mixer */ + SST_SWM_IN_MEDIA1 = (SST_PATH_INDEX_MEDIA1_IN | SST_DEFAULT_CELL_NBR), /* Part of Media Mixer */ + SST_SWM_IN_MEDIA2 = (SST_PATH_INDEX_MEDIA2_IN | SST_DEFAULT_CELL_NBR), /* Part of Media Mixer */ + SST_SWM_IN_MEDIA3 = (SST_PATH_INDEX_MEDIA3_IN | SST_DEFAULT_CELL_NBR), /* Part of Media Mixer */ + SST_SWM_IN_END = (SST_PATH_INDEX_RESERVED | SST_DEFAULT_CELL_NBR) +}; + +/* + * path IDs + */ +enum sst_swm_outputs { + SST_SWM_OUT_CODEC0 = (SST_PATH_INDEX_CODEC_OUT0 | SST_DEFAULT_CELL_NBR), + SST_SWM_OUT_CODEC1 = (SST_PATH_INDEX_CODEC_OUT1 | SST_DEFAULT_CELL_NBR), + SST_SWM_OUT_SPROT_LOOP = (SST_PATH_INDEX_SPROT_LOOP_OUT | SST_DEFAULT_CELL_NBR), + SST_SWM_OUT_MEDIA_LOOP1 = (SST_PATH_INDEX_MEDIA_LOOP1_OUT | SST_DEFAULT_CELL_NBR), + SST_SWM_OUT_MEDIA_LOOP2 = (SST_PATH_INDEX_MEDIA_LOOP2_OUT | SST_DEFAULT_CELL_NBR), + SST_SWM_OUT_VOIP = (SST_PATH_INDEX_VOIP_OUT | SST_DEFAULT_CELL_NBR), + SST_SWM_OUT_PCM0 = (SST_PATH_INDEX_PCM0_OUT | SST_DEFAULT_CELL_NBR), + SST_SWM_OUT_PCM1 = (SST_PATH_INDEX_PCM1_OUT | SST_DEFAULT_CELL_NBR), + SST_SWM_OUT_PCM2 = (SST_PATH_INDEX_PCM2_OUT | SST_DEFAULT_CELL_NBR), + SST_SWM_OUT_MEDIA0 = (SST_PATH_INDEX_MEDIA0_OUT | SST_DEFAULT_CELL_NBR), /* Part of Media Mixer */ + SST_SWM_OUT_MEDIA1 = (SST_PATH_INDEX_MEDIA1_OUT | SST_DEFAULT_CELL_NBR), /* Part of Media Mixer */ + SST_SWM_OUT_END = (SST_PATH_INDEX_RESERVED | SST_DEFAULT_CELL_NBR), +}; + +enum sst_ipc_msg { + SST_IPC_IA_CMD = 1, + SST_IPC_IA_SET_PARAMS, + SST_IPC_IA_GET_PARAMS, +}; + +enum sst_cmd_type { + SST_CMD_BYTES_SET = 1, + SST_CMD_BYTES_GET = 2, +}; + +enum sst_task { + SST_TASK_SBA = 1, + SST_TASK_MMX, +}; + +enum sst_type { + SST_TYPE_CMD = 1, + SST_TYPE_PARAMS, +}; + +enum sst_flag { + SST_FLAG_BLOCKED = 1, + SST_FLAG_NONBLOCK, +}; + +/* + * Enumeration for indexing the gain cells in VB_SET_GAIN DSP command + */ +enum sst_gain_index { + /* GAIN IDs for SB task start here */ + SST_GAIN_INDEX_CODEC_OUT0, + SST_GAIN_INDEX_CODEC_OUT1, + SST_GAIN_INDEX_CODEC_IN0, + SST_GAIN_INDEX_CODEC_IN1, + + SST_GAIN_INDEX_SPROT_LOOP_OUT, + SST_GAIN_INDEX_MEDIA_LOOP1_OUT, + SST_GAIN_INDEX_MEDIA_LOOP2_OUT, + + SST_GAIN_INDEX_PCM0_IN_LEFT, + SST_GAIN_INDEX_PCM0_IN_RIGHT, + + SST_GAIN_INDEX_PCM1_OUT_LEFT, + SST_GAIN_INDEX_PCM1_OUT_RIGHT, + SST_GAIN_INDEX_PCM1_IN_LEFT, + SST_GAIN_INDEX_PCM1_IN_RIGHT, + SST_GAIN_INDEX_PCM2_OUT_LEFT, + + SST_GAIN_INDEX_PCM2_OUT_RIGHT, + SST_GAIN_INDEX_VOIP_OUT, + SST_GAIN_INDEX_VOIP_IN, + + /* Gain IDs for MMX task start here */ + SST_GAIN_INDEX_MEDIA0_IN_LEFT, + SST_GAIN_INDEX_MEDIA0_IN_RIGHT, + SST_GAIN_INDEX_MEDIA1_IN_LEFT, + SST_GAIN_INDEX_MEDIA1_IN_RIGHT, + + SST_GAIN_INDEX_MEDIA2_IN_LEFT, + SST_GAIN_INDEX_MEDIA2_IN_RIGHT, + + SST_GAIN_INDEX_GAIN_END +}; + +/* + * Audio DSP module IDs specified by FW spec + * TODO: Update with all modules + */ +enum sst_module_id { + SST_MODULE_ID_PCM = 0x0001, + SST_MODULE_ID_MP3 = 0x0002, + SST_MODULE_ID_MP24 = 0x0003, + SST_MODULE_ID_AAC = 0x0004, + SST_MODULE_ID_AACP = 0x0005, + SST_MODULE_ID_EAACP = 0x0006, + SST_MODULE_ID_WMA9 = 0x0007, + SST_MODULE_ID_WMA10 = 0x0008, + SST_MODULE_ID_WMA10P = 0x0009, + SST_MODULE_ID_RA = 0x000A, + SST_MODULE_ID_DDAC3 = 0x000B, + SST_MODULE_ID_TRUE_HD = 0x000C, + SST_MODULE_ID_HD_PLUS = 0x000D, + + SST_MODULE_ID_SRC = 0x0064, + SST_MODULE_ID_DOWNMIX = 0x0066, + SST_MODULE_ID_GAIN_CELL = 0x0067, + SST_MODULE_ID_SPROT = 0x006D, + SST_MODULE_ID_BASS_BOOST = 0x006E, + SST_MODULE_ID_STEREO_WDNG = 0x006F, + SST_MODULE_ID_AV_REMOVAL = 0x0070, + SST_MODULE_ID_MIC_EQ = 0x0071, + SST_MODULE_ID_SPL = 0x0072, + SST_MODULE_ID_ALGO_VTSV = 0x0073, + SST_MODULE_ID_NR = 0x0076, + SST_MODULE_ID_BWX = 0x0077, + SST_MODULE_ID_DRP = 0x0078, + SST_MODULE_ID_MDRP = 0x0079, + + SST_MODULE_ID_ANA = 0x007A, + SST_MODULE_ID_AEC = 0x007B, + SST_MODULE_ID_NR_SNS = 0x007C, + SST_MODULE_ID_SER = 0x007D, + SST_MODULE_ID_AGC = 0x007E, + + SST_MODULE_ID_CNI = 0x007F, + SST_MODULE_ID_CONTEXT_ALGO_AWARE = 0x0080, + SST_MODULE_ID_FIR_24 = 0x0081, + SST_MODULE_ID_IIR_24 = 0x0082, + + SST_MODULE_ID_ASRC = 0x0083, + SST_MODULE_ID_TONE_GEN = 0x0084, + SST_MODULE_ID_BMF = 0x0086, + SST_MODULE_ID_EDL = 0x0087, + SST_MODULE_ID_GLC = 0x0088, + + SST_MODULE_ID_FIR_16 = 0x0089, + SST_MODULE_ID_IIR_16 = 0x008A, + SST_MODULE_ID_DNR = 0x008B, + + SST_MODULE_ID_VIRTUALIZER = 0x008C, + SST_MODULE_ID_VISUALIZATION = 0x008D, + SST_MODULE_ID_LOUDNESS_OPTIMIZER = 0x008E, + SST_MODULE_ID_REVERBERATION = 0x008F, + + SST_MODULE_ID_CNI_TX = 0x0090, + SST_MODULE_ID_REF_LINE = 0x0091, + SST_MODULE_ID_VOLUME = 0x0092, + SST_MODULE_ID_FILT_DCR = 0x0094, + SST_MODULE_ID_SLV = 0x009A, + SST_MODULE_ID_NLF = 0x009B, + SST_MODULE_ID_TNR = 0x009C, + SST_MODULE_ID_WNR = 0x009D, + + SST_MODULE_ID_LOG = 0xFF00, + + SST_MODULE_ID_TASK = 0xFFFF, +}; + +enum sst_cmd { + SBA_IDLE = 14, + SBA_VB_SET_SPEECH_PATH = 26, + MMX_SET_GAIN = 33, + SBA_VB_SET_GAIN = 33, + FBA_VB_RX_CNI = 35, + MMX_SET_GAIN_TIMECONST = 36, + SBA_VB_SET_TIMECONST = 36, + SBA_VB_START = 85, + SBA_SET_SWM = 114, + SBA_SET_MDRP = 116, + SBA_HW_SET_SSP = 117, + SBA_SET_MEDIA_LOOP_MAP = 118, + SBA_SET_MEDIA_PATH = 119, + MMX_SET_MEDIA_PATH = 119, + SBA_VB_LPRO = 126, + SBA_VB_SET_FIR = 128, + SBA_VB_SET_IIR = 129, + SBA_SET_SSP_SLOT_MAP = 130, +}; + +enum sst_dsp_switch { + SST_SWITCH_OFF = 0, + SST_SWITCH_ON = 3, +}; + +enum sst_path_switch { + SST_PATH_OFF = 0, + SST_PATH_ON = 1, +}; + +enum sst_swm_state { + SST_SWM_OFF = 0, + SST_SWM_ON = 3, +}; + +#define SST_FILL_LOCATION_IDS(dst, cell_idx, pipe_id) do { \ + dst.location_id.p.cell_nbr_idx = (cell_idx); \ + dst.location_id.p.path_id = (pipe_id); \ + } while (0) +#define SST_FILL_LOCATION_ID(dst, loc_id) (\ + dst.location_id.f = (loc_id)) +#define SST_FILL_MODULE_ID(dst, mod_id) (\ + dst.module_id = (mod_id)) + +#define SST_FILL_DESTINATION1(dst, id) do { \ + SST_FILL_LOCATION_ID(dst, (id) & 0xFFFF); \ + SST_FILL_MODULE_ID(dst, ((id) & 0xFFFF0000) >> 16); \ + } while (0) +#define SST_FILL_DESTINATION2(dst, loc_id, mod_id) do { \ + SST_FILL_LOCATION_ID(dst, loc_id); \ + SST_FILL_MODULE_ID(dst, mod_id); \ + } while (0) +#define SST_FILL_DESTINATION3(dst, cell_idx, path_id, mod_id) do { \ + SST_FILL_LOCATION_IDS(dst, cell_idx, path_id); \ + SST_FILL_MODULE_ID(dst, mod_id); \ + } while (0) + +#define SST_FILL_DESTINATION(level, dst, ...) \ + SST_FILL_DESTINATION##level(dst, __VA_ARGS__) +#define SST_FILL_DEFAULT_DESTINATION(dst) \ + SST_FILL_DESTINATION(2, dst, SST_DEFAULT_LOCATION_ID, SST_DEFAULT_MODULE_ID) + +struct sst_destination_id { + union sst_location_id { + struct { + u8 cell_nbr_idx; /* module index */ + u8 path_id; /* pipe_id */ + } __packed p; /* part */ + u16 f; /* full */ + } __packed location_id; + u16 module_id; +} __packed; +struct sst_dsp_header { + struct sst_destination_id dst; + u16 command_id; + u16 length; +} __packed; + +/* + * + * Common Commands + * + */ +struct sst_cmd_generic { + struct sst_dsp_header header; +} __packed; +struct sst_cmd_set_params { + struct sst_destination_id dst; + u16 command_id; + char params[0]; +} __packed; +#define SST_CONTROL_NAME(xpname, xmname, xinstance, xtype) \ + xpname " " xmname " " #xinstance " " xtype + +#define SST_COMBO_CONTROL_NAME(xpname, xmname, xinstance, xtype, xsubmodule) \ + xpname " " xmname " " #xinstance " " xtype " " xsubmodule +enum sst_algo_kcontrol_type { + SST_ALGO_PARAMS, + SST_ALGO_BYPASS, +}; + +struct sst_algo_control { + enum sst_algo_kcontrol_type type; + int max; + u16 module_id; + u16 pipe_id; + u16 task_id; + u16 cmd_id; + bool bypass; + unsigned char *params; + struct snd_soc_dapm_widget *w; +}; + +/* size of the control = size of params + size of length field */ +#define SST_ALGO_CTL_VALUE(xcount, xtype, xpipe, xmod, xtask, xcmd) \ + (struct sst_algo_control){ \ + .max = xcount + sizeof(u16), .type = xtype, .module_id = xmod, \ + .pipe_id = xpipe, .task_id = xtask, .cmd_id = xcmd, \ + } + +#define SST_ALGO_KCONTROL(xname, xcount, xmod, xpipe, \ + xtask, xcmd, xtype, xinfo, xget, xput) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = xname, \ + .info = xinfo, .get = xget, .put = xput, \ + .private_value = (unsigned long)& \ + SST_ALGO_CTL_VALUE(xcount, xtype, xpipe, \ + xmod, xtask, xcmd), \ +} + +#define SST_ALGO_KCONTROL_BYTES(xpname, xmname, xcount, xmod, \ + xpipe, xinstance, xtask, xcmd) \ + SST_ALGO_KCONTROL(SST_CONTROL_NAME(xpname, xmname, xinstance, "params"), \ + xcount, xmod, xpipe, xtask, xcmd, SST_ALGO_PARAMS, \ + sst_algo_bytes_ctl_info, \ + sst_algo_control_get, sst_algo_control_set) + +#define SST_ALGO_KCONTROL_BOOL(xpname, xmname, xmod, xpipe, xinstance, xtask) \ + SST_ALGO_KCONTROL(SST_CONTROL_NAME(xpname, xmname, xinstance, "bypass"), \ + 0, xmod, xpipe, xtask, 0, SST_ALGO_BYPASS, \ + snd_soc_info_bool_ext, \ + sst_algo_control_get, sst_algo_control_set) + +#define SST_ALGO_BYPASS_PARAMS(xpname, xmname, xcount, xmod, xpipe, \ + xinstance, xtask, xcmd) \ + SST_ALGO_KCONTROL_BOOL(xpname, xmname, xmod, xpipe, xinstance, xtask), \ + SST_ALGO_KCONTROL_BYTES(xpname, xmname, xcount, xmod, xpipe, xinstance, xtask, xcmd) + +#define SST_COMBO_ALGO_KCONTROL_BYTES(xpname, xmname, xsubmod, xcount, xmod, \ + xpipe, xinstance, xtask, xcmd) \ + SST_ALGO_KCONTROL(SST_COMBO_CONTROL_NAME(xpname, xmname, xinstance, "params", \ + xsubmod), \ + xcount, xmod, xpipe, xtask, xcmd, SST_ALGO_PARAMS, \ + sst_algo_bytes_ctl_info, \ + sst_algo_control_get, sst_algo_control_set) + + +struct sst_enum { + bool tx; + unsigned short reg; + unsigned int max; + const char * const *texts; + struct snd_soc_dapm_widget *w; +}; #endif diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c index 61bf6da4bb02..33fc5c3abf55 100644 --- a/sound/soc/intel/sst-haswell-pcm.c +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -138,11 +138,10 @@ static inline unsigned int hsw_ipc_to_mixer(u32 value) static int hsw_stream_volume_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_platform *platform = snd_soc_kcontrol_platform(kcontrol); + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct hsw_priv_data *pdata = snd_soc_component_get_drvdata(cmpnt); struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; - struct hsw_priv_data *pdata = - snd_soc_platform_get_drvdata(platform); struct hsw_pcm_data *pcm_data = &pdata->pcm[mc->reg]; struct sst_hsw *hsw = pdata->hsw; u32 volume; @@ -176,11 +175,10 @@ static int hsw_stream_volume_put(struct snd_kcontrol *kcontrol, static int hsw_stream_volume_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_platform *platform = snd_soc_kcontrol_platform(kcontrol); + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct hsw_priv_data *pdata = snd_soc_component_get_drvdata(cmpnt); struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; - struct hsw_priv_data *pdata = - snd_soc_platform_get_drvdata(platform); struct hsw_pcm_data *pcm_data = &pdata->pcm[mc->reg]; struct sst_hsw *hsw = pdata->hsw; u32 volume; @@ -208,8 +206,8 @@ static int hsw_stream_volume_get(struct snd_kcontrol *kcontrol, static int hsw_volume_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_platform *platform = snd_soc_kcontrol_platform(kcontrol); - struct hsw_priv_data *pdata = snd_soc_platform_get_drvdata(platform); + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct hsw_priv_data *pdata = snd_soc_component_get_drvdata(cmpnt); struct sst_hsw *hsw = pdata->hsw; u32 volume; @@ -233,8 +231,8 @@ static int hsw_volume_put(struct snd_kcontrol *kcontrol, static int hsw_volume_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_platform *platform = snd_soc_kcontrol_platform(kcontrol); - struct hsw_priv_data *pdata = snd_soc_platform_get_drvdata(platform); + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct hsw_priv_data *pdata = snd_soc_component_get_drvdata(cmpnt); struct sst_hsw *hsw = pdata->hsw; unsigned int volume = 0; @@ -778,20 +776,11 @@ static const struct snd_soc_dapm_route graph[] = { static int hsw_pcm_probe(struct snd_soc_platform *platform) { + struct hsw_priv_data *priv_data = snd_soc_platform_get_drvdata(platform); struct sst_pdata *pdata = dev_get_platdata(platform->dev); - struct hsw_priv_data *priv_data; - struct device *dma_dev; + struct device *dma_dev = pdata->dma_dev; int i, ret = 0; - if (!pdata) - return -ENODEV; - - dma_dev = pdata->dma_dev; - - priv_data = devm_kzalloc(platform->dev, sizeof(*priv_data), GFP_KERNEL); - priv_data->hsw = pdata->dsp; - snd_soc_platform_set_drvdata(platform, priv_data); - /* allocate DSP buffer page tables */ for (i = 0; i < ARRAY_SIZE(hsw_dais); i++) { @@ -848,27 +837,38 @@ static struct snd_soc_platform_driver hsw_soc_platform = { .ops = &hsw_pcm_ops, .pcm_new = hsw_pcm_new, .pcm_free = hsw_pcm_free, - .controls = hsw_volume_controls, - .num_controls = ARRAY_SIZE(hsw_volume_controls), - .dapm_widgets = widgets, - .num_dapm_widgets = ARRAY_SIZE(widgets), - .dapm_routes = graph, - .num_dapm_routes = ARRAY_SIZE(graph), }; static const struct snd_soc_component_driver hsw_dai_component = { - .name = "haswell-dai", + .name = "haswell-dai", + .controls = hsw_volume_controls, + .num_controls = ARRAY_SIZE(hsw_volume_controls), + .dapm_widgets = widgets, + .num_dapm_widgets = ARRAY_SIZE(widgets), + .dapm_routes = graph, + .num_dapm_routes = ARRAY_SIZE(graph), }; static int hsw_pcm_dev_probe(struct platform_device *pdev) { struct sst_pdata *sst_pdata = dev_get_platdata(&pdev->dev); + struct hsw_priv_data *priv_data; int ret; + if (!sst_pdata) + return -EINVAL; + + priv_data = devm_kzalloc(&pdev->dev, sizeof(*priv_data), GFP_KERNEL); + if (!priv_data) + return -ENOMEM; + ret = sst_hsw_dsp_init(&pdev->dev, sst_pdata); if (ret < 0) return -ENODEV; + priv_data->hsw = sst_pdata->dsp; + platform_set_drvdata(pdev, priv_data); + ret = snd_soc_register_platform(&pdev->dev, &hsw_soc_platform); if (ret < 0) goto err_plat; diff --git a/sound/soc/intel/sst-mfld-platform-compress.c b/sound/soc/intel/sst-mfld-platform-compress.c index 29c059ca19e8..59467775c9b8 100644 --- a/sound/soc/intel/sst-mfld-platform-compress.c +++ b/sound/soc/intel/sst-mfld-platform-compress.c @@ -86,7 +86,7 @@ static int sst_platform_compr_free(struct snd_compr_stream *cstream) /*need to check*/ str_id = stream->id; if (str_id) - ret_val = stream->compr_ops->close(str_id); + ret_val = stream->compr_ops->close(sst->dev, str_id); module_put(sst->dev->driver->owner); kfree(stream); pr_debug("%s: %d\n", __func__, ret_val); @@ -158,7 +158,7 @@ static int sst_platform_compr_set_params(struct snd_compr_stream *cstream, cb.drain_cb_param = cstream; cb.drain_notify = sst_drain_notify; - retval = stream->compr_ops->open(&str_params, &cb); + retval = stream->compr_ops->open(sst->dev, &str_params, &cb); if (retval < 0) { pr_err("stream allocation failed %d\n", retval); return retval; @@ -170,10 +170,30 @@ static int sst_platform_compr_set_params(struct snd_compr_stream *cstream, static int sst_platform_compr_trigger(struct snd_compr_stream *cstream, int cmd) { - struct sst_runtime_stream *stream = - cstream->runtime->private_data; - - return stream->compr_ops->control(cmd, stream->id); + struct sst_runtime_stream *stream = cstream->runtime->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + if (stream->compr_ops->stream_start) + return stream->compr_ops->stream_start(sst->dev, stream->id); + case SNDRV_PCM_TRIGGER_STOP: + if (stream->compr_ops->stream_drop) + return stream->compr_ops->stream_drop(sst->dev, stream->id); + case SND_COMPR_TRIGGER_DRAIN: + if (stream->compr_ops->stream_drain) + return stream->compr_ops->stream_drain(sst->dev, stream->id); + case SND_COMPR_TRIGGER_PARTIAL_DRAIN: + if (stream->compr_ops->stream_partial_drain) + return stream->compr_ops->stream_partial_drain(sst->dev, stream->id); + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (stream->compr_ops->stream_pause) + return stream->compr_ops->stream_pause(sst->dev, stream->id); + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (stream->compr_ops->stream_pause_release) + return stream->compr_ops->stream_pause_release(sst->dev, stream->id); + default: + return -EINVAL; + } } static int sst_platform_compr_pointer(struct snd_compr_stream *cstream, @@ -182,7 +202,7 @@ static int sst_platform_compr_pointer(struct snd_compr_stream *cstream, struct sst_runtime_stream *stream; stream = cstream->runtime->private_data; - stream->compr_ops->tstamp(stream->id, tstamp); + stream->compr_ops->tstamp(sst->dev, stream->id, tstamp); tstamp->byte_offset = tstamp->copied_total % (u32)cstream->runtime->buffer_size; pr_debug("calc bytes offset/copied bytes as %d\n", tstamp->byte_offset); @@ -195,7 +215,7 @@ static int sst_platform_compr_ack(struct snd_compr_stream *cstream, struct sst_runtime_stream *stream; stream = cstream->runtime->private_data; - stream->compr_ops->ack(stream->id, (unsigned long)bytes); + stream->compr_ops->ack(sst->dev, stream->id, (unsigned long)bytes); stream->bytes_written += bytes; return 0; @@ -225,7 +245,7 @@ static int sst_platform_compr_set_metadata(struct snd_compr_stream *cstream, struct sst_runtime_stream *stream = cstream->runtime->private_data; - return stream->compr_ops->set_metadata(stream->id, metadata); + return stream->compr_ops->set_metadata(sst->dev, stream->id, metadata); } struct snd_compr_ops sst_platform_compr_ops = { diff --git a/sound/soc/intel/sst-mfld-platform-pcm.c b/sound/soc/intel/sst-mfld-platform-pcm.c index 706212a6a68c..aa9b600dfc9b 100644 --- a/sound/soc/intel/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/sst-mfld-platform-pcm.c @@ -43,12 +43,12 @@ int sst_register_dsp(struct sst_device *dev) return -ENODEV; mutex_lock(&sst_lock); if (sst) { - pr_err("we already have a device %s\n", sst->name); + dev_err(dev->dev, "we already have a device %s\n", sst->name); module_put(dev->dev->driver->owner); mutex_unlock(&sst_lock); return -EEXIST; } - pr_debug("registering device %s\n", dev->name); + dev_dbg(dev->dev, "registering device %s\n", dev->name); sst = dev; mutex_unlock(&sst_lock); return 0; @@ -70,7 +70,7 @@ int sst_unregister_dsp(struct sst_device *dev) } module_put(sst->dev->driver->owner); - pr_debug("unreg %s\n", sst->name); + dev_dbg(dev->dev, "unreg %s\n", sst->name); sst = NULL; mutex_unlock(&sst_lock); return 0; @@ -252,7 +252,7 @@ int sst_fill_stream_params(void *substream, } static int sst_platform_alloc_stream(struct snd_pcm_substream *substream, - struct snd_soc_platform *platform) + struct snd_soc_dai *dai) { struct sst_runtime_stream *stream = substream->runtime->private_data; @@ -260,7 +260,7 @@ static int sst_platform_alloc_stream(struct snd_pcm_substream *substream, struct snd_sst_params str_params = {0}; struct snd_sst_alloc_params_ext alloc_params = {0}; int ret_val = 0; - struct sst_data *ctx = snd_soc_platform_get_drvdata(platform); + struct sst_data *ctx = snd_soc_dai_get_drvdata(dai); /* set codec params and inform SST driver the same */ sst_fill_pcm_params(substream, ¶m); @@ -277,7 +277,7 @@ static int sst_platform_alloc_stream(struct snd_pcm_substream *substream, stream->stream_info.str_id = str_params.stream_id; - ret_val = stream->ops->open(&str_params); + ret_val = stream->ops->open(sst->dev, &str_params); if (ret_val <= 0) return ret_val; @@ -306,22 +306,31 @@ static int sst_platform_init_stream(struct snd_pcm_substream *substream) { struct sst_runtime_stream *stream = substream->runtime->private_data; + struct snd_soc_pcm_runtime *rtd = substream->private_data; int ret_val; - pr_debug("setting buffer ptr param\n"); + dev_dbg(rtd->dev, "setting buffer ptr param\n"); sst_set_stream_status(stream, SST_PLATFORM_INIT); stream->stream_info.period_elapsed = sst_period_elapsed; stream->stream_info.arg = substream; stream->stream_info.buffer_ptr = 0; stream->stream_info.sfreq = substream->runtime->rate; - ret_val = stream->ops->device_control( - SST_SND_STREAM_INIT, &stream->stream_info); + ret_val = stream->ops->stream_init(sst->dev, &stream->stream_info); if (ret_val) - pr_err("control_set ret error %d\n", ret_val); + dev_err(rtd->dev, "control_set ret error %d\n", ret_val); return ret_val; } -/* end -- helper functions */ + +static int power_up_sst(struct sst_runtime_stream *stream) +{ + return stream->ops->power(sst->dev, true); +} + +static void power_down_sst(struct sst_runtime_stream *stream) +{ + stream->ops->power(sst->dev, false); +} static int sst_media_open(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) @@ -339,7 +348,7 @@ static int sst_media_open(struct snd_pcm_substream *substream, mutex_lock(&sst_lock); if (!sst || !try_module_get(sst->dev->driver->owner)) { - pr_err("no device available to run\n"); + dev_err(dai->dev, "no device available to run\n"); ret_val = -ENODEV; goto out_ops; } @@ -352,6 +361,10 @@ static int sst_media_open(struct snd_pcm_substream *substream, /* allocate memory for SST API set */ runtime->private_data = stream; + ret_val = power_up_sst(stream); + if (ret_val < 0) + return ret_val; + /* Make sure, that the period size is always even */ snd_pcm_hw_constraint_step(substream->runtime, 0, SNDRV_PCM_HW_PARAM_PERIODS, 2); @@ -371,26 +384,29 @@ static void sst_media_close(struct snd_pcm_substream *substream, int ret_val = 0, str_id; stream = substream->runtime->private_data; + power_down_sst(stream); + str_id = stream->stream_info.str_id; if (str_id) - ret_val = stream->ops->close(str_id); + ret_val = stream->ops->close(sst->dev, str_id); module_put(sst->dev->driver->owner); kfree(stream); } -static inline unsigned int get_current_pipe_id(struct snd_soc_platform *platform, +static inline unsigned int get_current_pipe_id(struct snd_soc_dai *dai, struct snd_pcm_substream *substream) { - struct sst_data *sst = snd_soc_platform_get_drvdata(platform); + struct sst_data *sst = snd_soc_dai_get_drvdata(dai); struct sst_dev_stream_map *map = sst->pdata->pdev_strm_map; struct sst_runtime_stream *stream = substream->runtime->private_data; u32 str_id = stream->stream_info.str_id; unsigned int pipe_id; + pipe_id = map[str_id].device_id; - pr_debug("%s: got pipe_id = %#x for str_id = %d\n", - __func__, pipe_id, str_id); + dev_dbg(dai->dev, "got pipe_id = %#x for str_id = %d\n", + pipe_id, str_id); return pipe_id; } @@ -403,12 +419,11 @@ static int sst_media_prepare(struct snd_pcm_substream *substream, stream = substream->runtime->private_data; str_id = stream->stream_info.str_id; if (stream->stream_info.str_id) { - ret_val = stream->ops->device_control( - SST_SND_DROP, &str_id); + ret_val = stream->ops->stream_drop(sst->dev, str_id); return ret_val; } - ret_val = sst_platform_alloc_stream(substream, dai->platform); + ret_val = sst_platform_alloc_stream(substream, dai); if (ret_val <= 0) return ret_val; snprintf(substream->pcm->id, sizeof(substream->pcm->id), @@ -461,37 +476,40 @@ static int sst_platform_pcm_trigger(struct snd_pcm_substream *substream, { int ret_val = 0, str_id; struct sst_runtime_stream *stream; - int str_cmd, status; + int status; + struct snd_soc_pcm_runtime *rtd = substream->private_data; - pr_debug("sst_platform_pcm_trigger called\n"); + dev_dbg(rtd->dev, "sst_platform_pcm_trigger called\n"); + if (substream->pcm->internal) + return 0; stream = substream->runtime->private_data; str_id = stream->stream_info.str_id; switch (cmd) { case SNDRV_PCM_TRIGGER_START: - pr_debug("sst: Trigger Start\n"); - str_cmd = SST_SND_START; + dev_dbg(rtd->dev, "sst: Trigger Start\n"); status = SST_PLATFORM_RUNNING; stream->stream_info.arg = substream; + ret_val = stream->ops->stream_start(sst->dev, str_id); break; case SNDRV_PCM_TRIGGER_STOP: - pr_debug("sst: in stop\n"); - str_cmd = SST_SND_DROP; + dev_dbg(rtd->dev, "sst: in stop\n"); status = SST_PLATFORM_DROPPED; + ret_val = stream->ops->stream_drop(sst->dev, str_id); break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - pr_debug("sst: in pause\n"); - str_cmd = SST_SND_PAUSE; + dev_dbg(rtd->dev, "sst: in pause\n"); status = SST_PLATFORM_PAUSED; + ret_val = stream->ops->stream_pause(sst->dev, str_id); break; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - pr_debug("sst: in pause release\n"); - str_cmd = SST_SND_RESUME; + dev_dbg(rtd->dev, "sst: in pause release\n"); status = SST_PLATFORM_RUNNING; + ret_val = stream->ops->stream_pause_release(sst->dev, str_id); break; default: return -EINVAL; } - ret_val = stream->ops->device_control(str_cmd, &str_id); + if (!ret_val) sst_set_stream_status(stream, status); @@ -505,16 +523,16 @@ static snd_pcm_uframes_t sst_platform_pcm_pointer struct sst_runtime_stream *stream; int ret_val, status; struct pcm_stream_info *str_info; + struct snd_soc_pcm_runtime *rtd = substream->private_data; stream = substream->runtime->private_data; status = sst_get_stream_status(stream); if (status == SST_PLATFORM_INIT) return 0; str_info = &stream->stream_info; - ret_val = stream->ops->device_control( - SST_SND_BUFFER_POINTER, str_info); + ret_val = stream->ops->stream_read_tstamp(sst->dev, str_info); if (ret_val) { - pr_err("sst: error code = %d\n", ret_val); + dev_err(rtd->dev, "sst: error code = %d\n", ret_val); return ret_val; } substream->runtime->delay = str_info->pcm_delay; @@ -530,7 +548,7 @@ static struct snd_pcm_ops sst_platform_ops = { static void sst_pcm_free(struct snd_pcm *pcm) { - pr_debug("sst_pcm_free called\n"); + dev_dbg(pcm->dev, "sst_pcm_free called\n"); snd_pcm_lib_preallocate_free_for_all(pcm); } @@ -547,14 +565,20 @@ static int sst_pcm_new(struct snd_soc_pcm_runtime *rtd) snd_dma_continuous_data(GFP_DMA), SST_MIN_BUFFER, SST_MAX_BUFFER); if (retval) { - pr_err("dma buffer allocationf fail\n"); + dev_err(rtd->dev, "dma buffer allocationf fail\n"); return retval; } } return retval; } -static struct snd_soc_platform_driver sst_soc_platform_drv = { +static int sst_soc_probe(struct snd_soc_platform *platform) +{ + return sst_dsp_init_v2_dpcm(platform); +} + +static struct snd_soc_platform_driver sst_soc_platform_drv = { + .probe = sst_soc_probe, .ops = &sst_platform_ops, .compr_ops = &sst_platform_compr_ops, .pcm_new = sst_pcm_new, @@ -574,13 +598,11 @@ static int sst_platform_probe(struct platform_device *pdev) drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_KERNEL); if (drv == NULL) { - pr_err("kzalloc failed\n"); return -ENOMEM; } pdata = devm_kzalloc(&pdev->dev, sizeof(*pdata), GFP_KERNEL); if (pdata == NULL) { - pr_err("kzalloc failed for pdata\n"); return -ENOMEM; } @@ -592,14 +614,14 @@ static int sst_platform_probe(struct platform_device *pdev) ret = snd_soc_register_platform(&pdev->dev, &sst_soc_platform_drv); if (ret) { - pr_err("registering soc platform failed\n"); + dev_err(&pdev->dev, "registering soc platform failed\n"); return ret; } ret = snd_soc_register_component(&pdev->dev, &sst_component, sst_platform_dai, ARRAY_SIZE(sst_platform_dai)); if (ret) { - pr_err("registering cpu dais failed\n"); + dev_err(&pdev->dev, "registering cpu dais failed\n"); snd_soc_unregister_platform(&pdev->dev); } return ret; @@ -610,7 +632,7 @@ static int sst_platform_remove(struct platform_device *pdev) snd_soc_unregister_component(&pdev->dev); snd_soc_unregister_platform(&pdev->dev); - pr_debug("sst_platform_remove success\n"); + dev_dbg(&pdev->dev, "sst_platform_remove success\n"); return 0; } diff --git a/sound/soc/intel/sst-mfld-platform.h b/sound/soc/intel/sst-mfld-platform.h index 6c6a42c08e24..19f83ec51613 100644 --- a/sound/soc/intel/sst-mfld-platform.h +++ b/sound/soc/intel/sst-mfld-platform.h @@ -54,20 +54,6 @@ enum sst_drv_status { SST_PLATFORM_DROPPED, }; -enum sst_controls { - SST_SND_ALLOC = 0x00, - SST_SND_PAUSE = 0x01, - SST_SND_RESUME = 0x02, - SST_SND_DROP = 0x03, - SST_SND_FREE = 0x04, - SST_SND_BUFFER_POINTER = 0x05, - SST_SND_STREAM_INIT = 0x06, - SST_SND_START = 0x07, - SST_SET_BYTE_STREAM = 0x100A, - SST_GET_BYTE_STREAM = 0x100B, - SST_MAX_CONTROLS = SST_GET_BYTE_STREAM, -}; - enum sst_stream_ops { STREAM_OPS_PLAYBACK = 0, STREAM_OPS_CAPTURE, @@ -113,24 +99,37 @@ struct sst_compress_cb { struct compress_sst_ops { const char *name; - int (*open) (struct snd_sst_params *str_params, - struct sst_compress_cb *cb); - int (*control) (unsigned int cmd, unsigned int str_id); - int (*tstamp) (unsigned int str_id, struct snd_compr_tstamp *tstamp); - int (*ack) (unsigned int str_id, unsigned long bytes); - int (*close) (unsigned int str_id); - int (*get_caps) (struct snd_compr_caps *caps); - int (*get_codec_caps) (struct snd_compr_codec_caps *codec); - int (*set_metadata) (unsigned int str_id, + int (*open)(struct device *dev, + struct snd_sst_params *str_params, struct sst_compress_cb *cb); + int (*stream_start)(struct device *dev, unsigned int str_id); + int (*stream_drop)(struct device *dev, unsigned int str_id); + int (*stream_drain)(struct device *dev, unsigned int str_id); + int (*stream_partial_drain)(struct device *dev, unsigned int str_id); + int (*stream_pause)(struct device *dev, unsigned int str_id); + int (*stream_pause_release)(struct device *dev, unsigned int str_id); + + int (*tstamp)(struct device *dev, unsigned int str_id, + struct snd_compr_tstamp *tstamp); + int (*ack)(struct device *dev, unsigned int str_id, + unsigned long bytes); + int (*close)(struct device *dev, unsigned int str_id); + int (*get_caps)(struct snd_compr_caps *caps); + int (*get_codec_caps)(struct snd_compr_codec_caps *codec); + int (*set_metadata)(struct device *dev, unsigned int str_id, struct snd_compr_metadata *mdata); - }; struct sst_ops { - int (*open) (struct snd_sst_params *str_param); - int (*device_control) (int cmd, void *arg); - int (*set_generic_params)(enum sst_controls cmd, void *arg); - int (*close) (unsigned int str_id); + int (*open)(struct device *dev, struct snd_sst_params *str_param); + int (*stream_init)(struct device *dev, struct pcm_stream_info *str_info); + int (*stream_start)(struct device *dev, int str_id); + int (*stream_drop)(struct device *dev, int str_id); + int (*stream_pause)(struct device *dev, int str_id); + int (*stream_pause_release)(struct device *dev, int str_id); + int (*stream_read_tstamp)(struct device *dev, struct pcm_stream_info *str_info); + int (*send_byte_stream)(struct device *dev, struct snd_sst_bytes_v2 *bytes); + int (*close)(struct device *dev, unsigned int str_id); + int (*power)(struct device *dev, bool state); }; struct sst_runtime_stream { @@ -152,6 +151,8 @@ struct sst_device { }; struct sst_data; + +int sst_dsp_init_v2_dpcm(struct snd_soc_platform *platform); void sst_set_stream_status(struct sst_runtime_stream *stream, int state); int sst_fill_stream_params(void *substream, const struct sst_data *ctx, struct snd_sst_params *str_params, bool is_compress); @@ -166,6 +167,7 @@ struct sst_algo_int_control_v2 { struct sst_data { struct platform_device *pdev; struct sst_platform_data *pdata; + struct snd_sst_bytes_v2 *byte_stream; struct mutex lock; }; int sst_register_dsp(struct sst_device *sst); diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index 943922c79f78..b10ae8074461 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -168,7 +168,7 @@ static int rx51_spk_event(struct snd_soc_dapm_widget *w, static int rx51_hp_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *k, int event) { - struct snd_soc_codec *codec = w->dapm->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); if (SND_SOC_DAPM_EVENT_ON(event)) tpa6130a2_stereo_enable(codec, 1); diff --git a/sound/soc/rockchip/Kconfig b/sound/soc/rockchip/Kconfig index c196a466eef6..78fc159559b0 100644 --- a/sound/soc/rockchip/Kconfig +++ b/sound/soc/rockchip/Kconfig @@ -2,11 +2,10 @@ config SND_SOC_ROCKCHIP tristate "ASoC support for Rockchip" depends on COMPILE_TEST || ARCH_ROCKCHIP select SND_SOC_GENERIC_DMAENGINE_PCM - select SND_ROCKCHIP_I2S help Say Y or M if you want to add support for codecs attached to the Rockchip SoCs' Audio interfaces. You will also need to select the audio interfaces to support below. -config SND_ROCKCHIP_I2S +config SND_SOC_ROCKCHIP_I2S tristate diff --git a/sound/soc/rockchip/Makefile b/sound/soc/rockchip/Makefile index 1006418e1394..b9219092b47f 100644 --- a/sound/soc/rockchip/Makefile +++ b/sound/soc/rockchip/Makefile @@ -1,4 +1,4 @@ # ROCKCHIP Platform Support snd-soc-i2s-objs := rockchip_i2s.o -obj-$(CONFIG_SND_ROCKCHIP_I2S) += snd-soc-i2s.o +obj-$(CONFIG_SND_SOC_ROCKCHIP_I2S) += snd-soc-i2s.o diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 8d8e4b59049f..033487c9a164 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -165,13 +165,14 @@ static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai, struct rk_i2s_dev *i2s = to_info(cpu_dai); unsigned int mask = 0, val = 0; - mask = I2S_CKR_MSS_SLAVE; + mask = I2S_CKR_MSS_MASK; switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: - val = I2S_CKR_MSS_SLAVE; + /* Set source clock in Master mode */ + val = I2S_CKR_MSS_MASTER; break; case SND_SOC_DAIFMT_CBM_CFM: - val = I2S_CKR_MSS_MASTER; + val = I2S_CKR_MSS_SLAVE; break; default: return -EINVAL; @@ -243,16 +244,6 @@ static int rockchip_i2s_hw_params(struct snd_pcm_substream *substream, regmap_update_bits(i2s->regmap, I2S_TXCR, I2S_TXCR_VDW_MASK, val); regmap_update_bits(i2s->regmap, I2S_RXCR, I2S_RXCR_VDW_MASK, val); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - dai->playback_dma_data = &i2s->playback_dma_data; - regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_TDL_MASK, - I2S_DMACR_TDL(1) | I2S_DMACR_TDE_ENABLE); - } else { - dai->capture_dma_data = &i2s->capture_dma_data; - regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_RDL_MASK, - I2S_DMACR_RDL(1) | I2S_DMACR_RDE_ENABLE); - } - return 0; } @@ -300,6 +291,16 @@ static int rockchip_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, return ret; } +static int rockchip_i2s_dai_probe(struct snd_soc_dai *dai) +{ + struct rk_i2s_dev *i2s = snd_soc_dai_get_drvdata(dai); + + dai->capture_dma_data = &i2s->capture_dma_data; + dai->playback_dma_data = &i2s->playback_dma_data; + + return 0; +} + static const struct snd_soc_dai_ops rockchip_i2s_dai_ops = { .hw_params = rockchip_i2s_hw_params, .set_sysclk = rockchip_i2s_set_sysclk, @@ -308,7 +309,9 @@ static const struct snd_soc_dai_ops rockchip_i2s_dai_ops = { }; static struct snd_soc_dai_driver rockchip_i2s_dai = { + .probe = rockchip_i2s_dai_probe, .playback = { + .stream_name = "Playback", .channels_min = 2, .channels_max = 8, .rates = SNDRV_PCM_RATE_8000_192000, @@ -318,6 +321,7 @@ static struct snd_soc_dai_driver rockchip_i2s_dai = { SNDRV_PCM_FMTBIT_S24_LE), }, .capture = { + .stream_name = "Capture", .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_192000, @@ -361,6 +365,8 @@ static bool rockchip_i2s_rd_reg(struct device *dev, unsigned int reg) case I2S_XFER: case I2S_CLR: case I2S_RXDR: + case I2S_FIFOLR: + case I2S_INTSR: return true; default: return false; @@ -370,8 +376,8 @@ static bool rockchip_i2s_rd_reg(struct device *dev, unsigned int reg) static bool rockchip_i2s_volatile_reg(struct device *dev, unsigned int reg) { switch (reg) { - case I2S_FIFOLR: case I2S_INTSR: + case I2S_CLR: return true; default: return false; @@ -381,8 +387,6 @@ static bool rockchip_i2s_volatile_reg(struct device *dev, unsigned int reg) static bool rockchip_i2s_precious_reg(struct device *dev, unsigned int reg) { switch (reg) { - case I2S_FIFOLR: - return true; default: return false; } @@ -419,6 +423,11 @@ static int rockchip_i2s_probe(struct platform_device *pdev) dev_err(&pdev->dev, "Can't retrieve i2s bus clock\n"); return PTR_ERR(i2s->hclk); } + ret = clk_prepare_enable(i2s->hclk); + if (ret) { + dev_err(i2s->dev, "hclock enable failed %d\n", ret); + return ret; + } i2s->mclk = devm_clk_get(&pdev->dev, "i2s_clk"); if (IS_ERR(i2s->mclk)) { diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 03eec22f0f46..9d513473b300 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -462,7 +462,7 @@ static int i2s_set_sysclk(struct snd_soc_dai *dai, if (dir == SND_SOC_CLOCK_IN) rfs = 0; - if ((rfs && other->rfs && (other->rfs != rfs)) || + if ((rfs && other && other->rfs && (other->rfs != rfs)) || (any_active(i2s) && (((dir == SND_SOC_CLOCK_IN) && !(mod & MOD_CDCLKCON)) || @@ -762,7 +762,8 @@ static void i2s_shutdown(struct snd_pcm_substream *substream, } else { u32 mod = readl(i2s->addr + I2SMOD); i2s->cdclk_out = !(mod & MOD_CDCLKCON); - other->cdclk_out = i2s->cdclk_out; + if (other) + other->cdclk_out = i2s->cdclk_out; } /* Reset any constraint on RFS and BFS */ i2s->rfs = 0; diff --git a/sound/soc/samsung/idma.c b/sound/soc/samsung/idma.c index db6cefa18017..0e8dd985fcb3 100644 --- a/sound/soc/samsung/idma.c +++ b/sound/soc/samsung/idma.c @@ -351,7 +351,7 @@ static void idma_free(struct snd_pcm *pcm) if (!buf->area) return; - iounmap(buf->area); + iounmap((void __iomem *)buf->area); buf->area = NULL; buf->addr = 0; @@ -369,7 +369,7 @@ static int preallocate_idma_buffer(struct snd_pcm *pcm, int stream) buf->dev.type = SNDRV_DMA_TYPE_CONTINUOUS; buf->addr = idma.lp_tx_addr; buf->bytes = idma_hardware.buffer_bytes_max; - buf->area = (unsigned char *)ioremap(buf->addr, buf->bytes); + buf->area = (unsigned char * __force)ioremap(buf->addr, buf->bytes); return 0; } diff --git a/sound/soc/samsung/odroidx2_max98090.c b/sound/soc/samsung/odroidx2_max98090.c index 278edf9e2a87..3c8f60423e82 100644 --- a/sound/soc/samsung/odroidx2_max98090.c +++ b/sound/soc/samsung/odroidx2_max98090.c @@ -66,12 +66,12 @@ static struct snd_soc_card odroidx2 = { .late_probe = odroidx2_late_probe, }; -struct odroidx2_drv_data odroidx2_drvdata = { +static const struct odroidx2_drv_data odroidx2_drvdata = { .dapm_widgets = odroidx2_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(odroidx2_dapm_widgets), }; -struct odroidx2_drv_data odroidu3_drvdata = { +static const struct odroidx2_drv_data odroidu3_drvdata = { .dapm_widgets = odroidu3_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(odroidu3_dapm_widgets), }; diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c index 9902efcb8ea1..a05482651aae 100644 --- a/sound/soc/samsung/speyside.c +++ b/sound/soc/samsung/speyside.c @@ -228,10 +228,12 @@ static struct snd_soc_dai_link speyside_dai[] = { }, }; -static int speyside_wm9081_init(struct snd_soc_dapm_context *dapm) +static int speyside_wm9081_init(struct snd_soc_component *component) { + struct snd_soc_codec *codec = snd_soc_component_to_codec(component); + /* At any time the WM9081 is active it will have this clock */ - return snd_soc_codec_set_sysclk(dapm->codec, WM9081_SYSCLK_MCLK, 0, + return snd_soc_codec_set_sysclk(codec, WM9081_SYSCLK_MCLK, 0, MCLK_AUDIO_RATE, 0); } diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index c76344350e44..66fddec9543d 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -1297,9 +1297,14 @@ static int fsi_dma_transfer(struct fsi_priv *fsi, struct fsi_stream *io) struct snd_pcm_substream *substream = io->substream; struct dma_async_tx_descriptor *desc; int is_play = fsi_stream_is_play(fsi, io); - enum dma_data_direction dir = is_play ? DMA_TO_DEVICE : DMA_FROM_DEVICE; + enum dma_transfer_direction dir; int ret = -EIO; + if (is_play) + dir = DMA_MEM_TO_DEV; + else + dir = DMA_DEV_TO_MEM; + desc = dmaengine_prep_dma_cyclic(io->chan, substream->runtime->dma_addr, snd_pcm_lib_buffer_bytes(substream), diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 19f78963e8b9..1922ec57d10a 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -798,10 +798,8 @@ if (name##_node) { \ mod_parse(src); mod_parse(dvc); - if (playback) - of_node_put(playback); - if (capture) - of_node_put(capture); + of_node_put(playback); + of_node_put(capture); } dai_i++; diff --git a/sound/soc/sh/siu_pcm.c b/sound/soc/sh/siu_pcm.c index 488f9becb44f..32eb6da2d2bd 100644 --- a/sound/soc/sh/siu_pcm.c +++ b/sound/soc/sh/siu_pcm.c @@ -139,7 +139,7 @@ static int siu_pcm_wr_set(struct siu_port *port_info, desc->callback = siu_dma_tx_complete; desc->callback_param = siu_stream; - cookie = desc->tx_submit(desc); + cookie = dmaengine_submit(desc); if (cookie < 0) { dev_err(dev, "Failed to submit a dma transfer\n"); return cookie; @@ -189,7 +189,7 @@ static int siu_pcm_rd_set(struct siu_port *port_info, desc->callback = siu_dma_tx_complete; desc->callback_param = siu_stream; - cookie = desc->tx_submit(desc); + cookie = dmaengine_submit(desc); if (cookie < 0) { dev_err(dev, "Failed to submit dma descriptor\n"); return cookie; diff --git a/sound/soc/sirf/sirf-usp.c b/sound/soc/sirf/sirf-usp.c index 3a730374e259..186dc7f33a55 100644 --- a/sound/soc/sirf/sirf-usp.c +++ b/sound/soc/sirf/sirf-usp.c @@ -100,6 +100,16 @@ static int sirf_usp_pcm_set_dai_fmt(struct snd_soc_dai *dai, return -EINVAL; } + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + usp->daifmt_format |= (fmt & SND_SOC_DAIFMT_INV_MASK); + break; + default: + return -EINVAL; + } + return 0; } @@ -177,7 +187,7 @@ static int sirf_usp_pcm_hw_params(struct snd_pcm_substream *substream, shifter_len = data_len; - switch (usp->daifmt_format) { + switch (usp->daifmt_format & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: regmap_update_bits(usp->regmap, USP_RX_FRAME_CTRL, USP_I2S_SYNC_CHG, USP_I2S_SYNC_CHG); @@ -193,6 +203,18 @@ static int sirf_usp_pcm_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } + switch (usp->daifmt_format & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + regmap_update_bits(usp->regmap, USP_MODE1, + USP_RXD_ACT_EDGE_FALLING | USP_TXD_ACT_EDGE_FALLING, + USP_RXD_ACT_EDGE_FALLING); + break; + default: + return -EINVAL; + } + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) regmap_update_bits(usp->regmap, USP_TX_FRAME_CTRL, USP_TXC_DATA_LEN_MASK | USP_TXC_FRAME_LEN_MASK diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 27c06acce205..cecfab3cc948 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -101,10 +101,12 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream) fe->dpcm[stream].runtime = fe_substream->runtime; - if (dpcm_path_get(fe, stream, &list) <= 0) { + ret = dpcm_path_get(fe, stream, &list); + if (ret < 0) + goto fe_err; + else if (ret == 0) dev_dbg(fe->dev, "ASoC: %s no valid %s route\n", fe->dai_link->name, stream ? "capture" : "playback"); - } /* calculate valid and active FE <-> BE dpcms */ dpcm_process_paths(fe, stream, &list, 1); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 889f4e3d35dc..3d8cff629a18 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -270,79 +270,54 @@ static const struct file_operations codec_reg_fops = { .llseek = default_llseek, }; -static struct dentry *soc_debugfs_create_dir(struct dentry *parent, - const char *fmt, ...) +static void soc_init_component_debugfs(struct snd_soc_component *component) { - struct dentry *de; - va_list ap; - char *s; + if (component->debugfs_prefix) { + char *name; - va_start(ap, fmt); - s = kvasprintf(GFP_KERNEL, fmt, ap); - va_end(ap); + name = kasprintf(GFP_KERNEL, "%s:%s", + component->debugfs_prefix, component->name); + if (name) { + component->debugfs_root = debugfs_create_dir(name, + component->card->debugfs_card_root); + kfree(name); + } + } else { + component->debugfs_root = debugfs_create_dir(component->name, + component->card->debugfs_card_root); + } - if (!s) - return NULL; + if (!component->debugfs_root) { + dev_warn(component->dev, + "ASoC: Failed to create component debugfs directory\n"); + return; + } - de = debugfs_create_dir(s, parent); - kfree(s); + snd_soc_dapm_debugfs_init(snd_soc_component_get_dapm(component), + component->debugfs_root); - return de; + if (component->init_debugfs) + component->init_debugfs(component); } -static void soc_init_codec_debugfs(struct snd_soc_codec *codec) +static void soc_cleanup_component_debugfs(struct snd_soc_component *component) { - struct dentry *debugfs_card_root = codec->component.card->debugfs_card_root; + debugfs_remove_recursive(component->debugfs_root); +} - codec->debugfs_codec_root = soc_debugfs_create_dir(debugfs_card_root, - "codec:%s", - codec->component.name); - if (!codec->debugfs_codec_root) { - dev_warn(codec->dev, - "ASoC: Failed to create codec debugfs directory\n"); - return; - } +static void soc_init_codec_debugfs(struct snd_soc_component *component) +{ + struct snd_soc_codec *codec = snd_soc_component_to_codec(component); - debugfs_create_bool("cache_sync", 0444, codec->debugfs_codec_root, + debugfs_create_bool("cache_sync", 0444, codec->component.debugfs_root, &codec->cache_sync); - debugfs_create_bool("cache_only", 0444, codec->debugfs_codec_root, - &codec->cache_only); codec->debugfs_reg = debugfs_create_file("codec_reg", 0644, - codec->debugfs_codec_root, + codec->component.debugfs_root, codec, &codec_reg_fops); if (!codec->debugfs_reg) dev_warn(codec->dev, "ASoC: Failed to create codec register debugfs file\n"); - - snd_soc_dapm_debugfs_init(&codec->dapm, codec->debugfs_codec_root); -} - -static void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec) -{ - debugfs_remove_recursive(codec->debugfs_codec_root); -} - -static void soc_init_platform_debugfs(struct snd_soc_platform *platform) -{ - struct dentry *debugfs_card_root = platform->component.card->debugfs_card_root; - - platform->debugfs_platform_root = soc_debugfs_create_dir(debugfs_card_root, - "platform:%s", - platform->component.name); - if (!platform->debugfs_platform_root) { - dev_warn(platform->dev, - "ASoC: Failed to create platform debugfs directory\n"); - return; - } - - snd_soc_dapm_debugfs_init(&platform->component.dapm, - platform->debugfs_platform_root); -} - -static void soc_cleanup_platform_debugfs(struct snd_soc_platform *platform) -{ - debugfs_remove_recursive(platform->debugfs_platform_root); } static ssize_t codec_list_read_file(struct file *file, char __user *user_buf, @@ -474,19 +449,15 @@ static void soc_cleanup_card_debugfs(struct snd_soc_card *card) #else -static inline void soc_init_codec_debugfs(struct snd_soc_codec *codec) -{ -} - -static inline void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec) -{ -} +#define soc_init_codec_debugfs NULL -static inline void soc_init_platform_debugfs(struct snd_soc_platform *platform) +static inline void soc_init_component_debugfs( + struct snd_soc_component *component) { } -static inline void soc_cleanup_platform_debugfs(struct snd_soc_platform *platform) +static inline void soc_cleanup_component_debugfs( + struct snd_soc_component *component) { } @@ -579,10 +550,8 @@ int snd_soc_suspend(struct device *dev) struct snd_soc_codec *codec; int i, j; - /* If the initialization of this soc device failed, there is no codec - * associated with it. Just bail out in this case. - */ - if (list_empty(&card->codec_dev_list)) + /* If the card is not initialized yet there is nothing to do */ + if (!card->instantiated) return 0; /* Due to the resume being scheduled into a workqueue we could @@ -668,7 +637,7 @@ int snd_soc_suspend(struct device *dev) list_for_each_entry(codec, &card->codec_dev_list, card_list) { /* If there are paths active then the CODEC will be held with * bias _ON and should not be suspended. */ - if (!codec->suspended && codec->driver->suspend) { + if (!codec->suspended) { switch (codec->dapm.bias_level) { case SND_SOC_BIAS_STANDBY: /* @@ -682,8 +651,10 @@ int snd_soc_suspend(struct device *dev) "ASoC: idle_bias_off CODEC on over suspend\n"); break; } + case SND_SOC_BIAS_OFF: - codec->driver->suspend(codec); + if (codec->driver->suspend) + codec->driver->suspend(codec); codec->suspended = 1; codec->cache_sync = 1; if (codec->component.regmap) @@ -757,11 +728,12 @@ static void soc_resume_deferred(struct work_struct *work) * left with bias OFF or STANDBY and suspended so we must now * resume. Otherwise the suspend was suppressed. */ - if (codec->driver->resume && codec->suspended) { + if (codec->suspended) { switch (codec->dapm.bias_level) { case SND_SOC_BIAS_STANDBY: case SND_SOC_BIAS_OFF: - codec->driver->resume(codec); + if (codec->driver->resume) + codec->driver->resume(codec); codec->suspended = 0; break; default: @@ -835,10 +807,8 @@ int snd_soc_resume(struct device *dev) struct snd_soc_card *card = dev_get_drvdata(dev); int i, ac97_control = 0; - /* If the initialization of this soc device failed, there is no codec - * associated with it. Just bail out in this case. - */ - if (list_empty(&card->codec_dev_list)) + /* If the card is not initialized yet there is nothing to do */ + if (!card->instantiated) return 0; /* activate pins from sleep state */ @@ -887,35 +857,40 @@ EXPORT_SYMBOL_GPL(snd_soc_resume); static const struct snd_soc_dai_ops null_dai_ops = { }; -static struct snd_soc_codec *soc_find_codec( - const struct device_node *codec_of_node, - const char *codec_name) +static struct snd_soc_component *soc_find_component( + const struct device_node *of_node, const char *name) { - struct snd_soc_codec *codec; + struct snd_soc_component *component; - list_for_each_entry(codec, &codec_list, list) { - if (codec_of_node) { - if (codec->dev->of_node != codec_of_node) - continue; - } else { - if (strcmp(codec->component.name, codec_name)) - continue; + list_for_each_entry(component, &component_list, list) { + if (of_node) { + if (component->dev->of_node == of_node) + return component; + } else if (strcmp(component->name, name) == 0) { + return component; } - - return codec; } return NULL; } -static struct snd_soc_dai *soc_find_codec_dai(struct snd_soc_codec *codec, - const char *codec_dai_name) +static struct snd_soc_dai *snd_soc_find_dai( + const struct snd_soc_dai_link_component *dlc) { - struct snd_soc_dai *codec_dai; + struct snd_soc_component *component; + struct snd_soc_dai *dai; + + /* Find CPU DAI from registered DAIs*/ + list_for_each_entry(component, &component_list, list) { + if (dlc->of_node && component->dev->of_node != dlc->of_node) + continue; + if (dlc->name && strcmp(dev_name(component->dev), dlc->name)) + continue; + list_for_each_entry(dai, &component->dai_list, list) { + if (dlc->dai_name && strcmp(dai->name, dlc->dai_name)) + continue; - list_for_each_entry(codec_dai, &codec->component.dai_list, list) { - if (!strcmp(codec_dai->name, codec_dai_name)) { - return codec_dai; + return dai; } } @@ -926,33 +901,19 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) { struct snd_soc_dai_link *dai_link = &card->dai_link[num]; struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; - struct snd_soc_component *component; struct snd_soc_dai_link_component *codecs = dai_link->codecs; + struct snd_soc_dai_link_component cpu_dai_component; struct snd_soc_dai **codec_dais = rtd->codec_dais; struct snd_soc_platform *platform; - struct snd_soc_dai *cpu_dai; const char *platform_name; int i; dev_dbg(card->dev, "ASoC: binding %s at idx %d\n", dai_link->name, num); - /* Find CPU DAI from registered DAIs*/ - list_for_each_entry(component, &component_list, list) { - if (dai_link->cpu_of_node && - component->dev->of_node != dai_link->cpu_of_node) - continue; - if (dai_link->cpu_name && - strcmp(dev_name(component->dev), dai_link->cpu_name)) - continue; - list_for_each_entry(cpu_dai, &component->dai_list, list) { - if (dai_link->cpu_dai_name && - strcmp(cpu_dai->name, dai_link->cpu_dai_name)) - continue; - - rtd->cpu_dai = cpu_dai; - } - } - + cpu_dai_component.name = dai_link->cpu_name; + cpu_dai_component.of_node = dai_link->cpu_of_node; + cpu_dai_component.dai_name = dai_link->cpu_dai_name; + rtd->cpu_dai = snd_soc_find_dai(&cpu_dai_component); if (!rtd->cpu_dai) { dev_err(card->dev, "ASoC: CPU DAI %s not registered\n", dai_link->cpu_dai_name); @@ -963,15 +924,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) /* Find CODEC from registered CODECs */ for (i = 0; i < rtd->num_codecs; i++) { - struct snd_soc_codec *codec; - codec = soc_find_codec(codecs[i].of_node, codecs[i].name); - if (!codec) { - dev_err(card->dev, "ASoC: CODEC %s not registered\n", - codecs[i].name); - return -EPROBE_DEFER; - } - - codec_dais[i] = soc_find_codec_dai(codec, codecs[i].dai_name); + codec_dais[i] = snd_soc_find_dai(&codecs[i]); if (!codec_dais[i]) { dev_err(card->dev, "ASoC: CODEC DAI %s not registered\n", codecs[i].dai_name); @@ -1012,68 +965,46 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) return 0; } -static int soc_remove_platform(struct snd_soc_platform *platform) +static void soc_remove_component(struct snd_soc_component *component) { - int ret; - - if (platform->driver->remove) { - ret = platform->driver->remove(platform); - if (ret < 0) - dev_err(platform->dev, "ASoC: failed to remove %d\n", - ret); - } - - /* Make sure all DAPM widgets are freed */ - snd_soc_dapm_free(&platform->component.dapm); - - soc_cleanup_platform_debugfs(platform); - platform->probed = 0; - module_put(platform->dev->driver->owner); - - return 0; -} + if (!component->probed) + return; -static void soc_remove_codec(struct snd_soc_codec *codec) -{ - int err; + /* This is a HACK and will be removed soon */ + if (component->codec) + list_del(&component->codec->card_list); - if (codec->driver->remove) { - err = codec->driver->remove(codec); - if (err < 0) - dev_err(codec->dev, "ASoC: failed to remove %d\n", err); - } + if (component->remove) + component->remove(component); - /* Make sure all DAPM widgets are freed */ - snd_soc_dapm_free(&codec->dapm); + snd_soc_dapm_free(snd_soc_component_get_dapm(component)); - soc_cleanup_codec_debugfs(codec); - codec->probed = 0; - list_del(&codec->card_list); - module_put(codec->dev->driver->owner); + soc_cleanup_component_debugfs(component); + component->probed = 0; + module_put(component->dev->driver->owner); } -static void soc_remove_codec_dai(struct snd_soc_dai *codec_dai, int order) +static void soc_remove_dai(struct snd_soc_dai *dai, int order) { int err; - if (codec_dai && codec_dai->probed && - codec_dai->driver->remove_order == order) { - if (codec_dai->driver->remove) { - err = codec_dai->driver->remove(codec_dai); + if (dai && dai->probed && + dai->driver->remove_order == order) { + if (dai->driver->remove) { + err = dai->driver->remove(dai); if (err < 0) - dev_err(codec_dai->dev, + dev_err(dai->dev, "ASoC: failed to remove %s: %d\n", - codec_dai->name, err); + dai->name, err); } - codec_dai->probed = 0; + dai->probed = 0; } } static void soc_remove_link_dais(struct snd_soc_card *card, int num, int order) { struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int i, err; + int i; /* unregister the rtd device */ if (rtd->dev_registered) { @@ -1085,22 +1016,9 @@ static void soc_remove_link_dais(struct snd_soc_card *card, int num, int order) /* remove the CODEC DAI */ for (i = 0; i < rtd->num_codecs; i++) - soc_remove_codec_dai(rtd->codec_dais[i], order); + soc_remove_dai(rtd->codec_dais[i], order); - /* remove the cpu_dai */ - if (cpu_dai && cpu_dai->probed && - cpu_dai->driver->remove_order == order) { - if (cpu_dai->driver->remove) { - err = cpu_dai->driver->remove(cpu_dai); - if (err < 0) - dev_err(cpu_dai->dev, - "ASoC: failed to remove %s: %d\n", - cpu_dai->name, err); - } - cpu_dai->probed = 0; - if (!cpu_dai->codec) - module_put(cpu_dai->dev->driver->owner); - } + soc_remove_dai(rtd->cpu_dai, order); } static void soc_remove_link_components(struct snd_soc_card *card, int num, @@ -1109,29 +1027,24 @@ static void soc_remove_link_components(struct snd_soc_card *card, int num, struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_platform *platform = rtd->platform; - struct snd_soc_codec *codec; + struct snd_soc_component *component; int i; /* remove the platform */ - if (platform && platform->probed && - platform->driver->remove_order == order) { - soc_remove_platform(platform); - } + if (platform && platform->component.driver->remove_order == order) + soc_remove_component(&platform->component); /* remove the CODEC-side CODEC */ for (i = 0; i < rtd->num_codecs; i++) { - codec = rtd->codec_dais[i]->codec; - if (codec && codec->probed && - codec->driver->remove_order == order) - soc_remove_codec(codec); + component = rtd->codec_dais[i]->component; + if (component->driver->remove_order == order) + soc_remove_component(component); } /* remove any CPU-side CODEC */ if (cpu_dai) { - codec = cpu_dai->codec; - if (codec && codec->probed && - codec->driver->remove_order == order) - soc_remove_codec(codec); + if (cpu_dai->component->driver->remove_order == order) + soc_remove_component(cpu_dai->component); } } @@ -1173,137 +1086,78 @@ static void soc_set_name_prefix(struct snd_soc_card *card, } } -static int soc_probe_codec(struct snd_soc_card *card, - struct snd_soc_codec *codec) +static int soc_probe_component(struct snd_soc_card *card, + struct snd_soc_component *component) { - int ret = 0; - const struct snd_soc_codec_driver *driver = codec->driver; + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); struct snd_soc_dai *dai; + int ret; + + if (component->probed) + return 0; - codec->component.card = card; - codec->dapm.card = card; - soc_set_name_prefix(card, &codec->component); + component->card = card; + dapm->card = card; + soc_set_name_prefix(card, component); - if (!try_module_get(codec->dev->driver->owner)) + if (!try_module_get(component->dev->driver->owner)) return -ENODEV; - soc_init_codec_debugfs(codec); + soc_init_component_debugfs(component); - if (driver->dapm_widgets) { - ret = snd_soc_dapm_new_controls(&codec->dapm, - driver->dapm_widgets, - driver->num_dapm_widgets); + if (component->dapm_widgets) { + ret = snd_soc_dapm_new_controls(dapm, component->dapm_widgets, + component->num_dapm_widgets); if (ret != 0) { - dev_err(codec->dev, + dev_err(component->dev, "Failed to create new controls %d\n", ret); goto err_probe; } } - /* Create DAPM widgets for each DAI stream */ - list_for_each_entry(dai, &codec->component.dai_list, list) { - ret = snd_soc_dapm_new_dai_widgets(&codec->dapm, dai); - + list_for_each_entry(dai, &component->dai_list, list) { + ret = snd_soc_dapm_new_dai_widgets(dapm, dai); if (ret != 0) { - dev_err(codec->dev, + dev_err(component->dev, "Failed to create DAI widgets %d\n", ret); goto err_probe; } } - codec->dapm.idle_bias_off = driver->idle_bias_off; - - if (driver->probe) { - ret = driver->probe(codec); + if (component->probe) { + ret = component->probe(component); if (ret < 0) { - dev_err(codec->dev, - "ASoC: failed to probe CODEC %d\n", ret); + dev_err(component->dev, + "ASoC: failed to probe component %d\n", ret); goto err_probe; } - WARN(codec->dapm.idle_bias_off && - codec->dapm.bias_level != SND_SOC_BIAS_OFF, - "codec %s can not start from non-off bias with idle_bias_off==1\n", - codec->component.name); - } - - if (driver->controls) - snd_soc_add_codec_controls(codec, driver->controls, - driver->num_controls); - if (driver->dapm_routes) - snd_soc_dapm_add_routes(&codec->dapm, driver->dapm_routes, - driver->num_dapm_routes); - - /* mark codec as probed and add to card codec list */ - codec->probed = 1; - list_add(&codec->card_list, &card->codec_dev_list); - list_add(&codec->dapm.list, &card->dapm_list); - return 0; - -err_probe: - soc_cleanup_codec_debugfs(codec); - module_put(codec->dev->driver->owner); - - return ret; -} - -static int soc_probe_platform(struct snd_soc_card *card, - struct snd_soc_platform *platform) -{ - int ret = 0; - const struct snd_soc_platform_driver *driver = platform->driver; - struct snd_soc_component *component; - struct snd_soc_dai *dai; - - platform->component.card = card; - platform->component.dapm.card = card; - - if (!try_module_get(platform->dev->driver->owner)) - return -ENODEV; - - soc_init_platform_debugfs(platform); - - if (driver->dapm_widgets) - snd_soc_dapm_new_controls(&platform->component.dapm, - driver->dapm_widgets, driver->num_dapm_widgets); - - /* Create DAPM widgets for each DAI stream */ - list_for_each_entry(component, &component_list, list) { - if (component->dev != platform->dev) - continue; - list_for_each_entry(dai, &component->dai_list, list) - snd_soc_dapm_new_dai_widgets(&platform->component.dapm, - dai); + WARN(dapm->idle_bias_off && + dapm->bias_level != SND_SOC_BIAS_OFF, + "codec %s can not start from non-off bias with idle_bias_off==1\n", + component->name); } - platform->component.dapm.idle_bias_off = 1; - - if (driver->probe) { - ret = driver->probe(platform); - if (ret < 0) { - dev_err(platform->dev, - "ASoC: failed to probe platform %d\n", ret); - goto err_probe; - } - } + if (component->controls) + snd_soc_add_component_controls(component, component->controls, + component->num_controls); + if (component->dapm_routes) + snd_soc_dapm_add_routes(dapm, component->dapm_routes, + component->num_dapm_routes); - if (driver->controls) - snd_soc_add_platform_controls(platform, driver->controls, - driver->num_controls); - if (driver->dapm_routes) - snd_soc_dapm_add_routes(&platform->component.dapm, - driver->dapm_routes, driver->num_dapm_routes); + component->probed = 1; + list_add(&dapm->list, &card->dapm_list); - /* mark platform as probed and add to card platform list */ - platform->probed = 1; - list_add(&platform->component.dapm.list, &card->dapm_list); + /* This is a HACK and will be removed soon */ + if (component->codec) + list_add(&component->codec->card_list, &card->codec_dev_list); return 0; err_probe: - soc_cleanup_platform_debugfs(platform); - module_put(platform->dev->driver->owner); + soc_cleanup_component_debugfs(component); + module_put(component->dev->driver->owner); return ret; } @@ -1342,17 +1196,21 @@ static int soc_post_component_init(struct snd_soc_pcm_runtime *rtd, } rtd->dev_registered = 1; - /* add DAPM sysfs entries for this codec */ - ret = snd_soc_dapm_sys_add(rtd->dev); - if (ret < 0) - dev_err(rtd->dev, - "ASoC: failed to add codec dapm sysfs entries: %d\n", ret); + if (rtd->codec) { + /* add DAPM sysfs entries for this codec */ + ret = snd_soc_dapm_sys_add(rtd->dev); + if (ret < 0) + dev_err(rtd->dev, + "ASoC: failed to add codec dapm sysfs entries: %d\n", + ret); - /* add codec sysfs entries */ - ret = device_create_file(rtd->dev, &dev_attr_codec_reg); - if (ret < 0) - dev_err(rtd->dev, - "ASoC: failed to add codec sysfs files: %d\n", ret); + /* add codec sysfs entries */ + ret = device_create_file(rtd->dev, &dev_attr_codec_reg); + if (ret < 0) + dev_err(rtd->dev, + "ASoC: failed to add codec sysfs files: %d\n", + ret); + } return 0; } @@ -1361,33 +1219,31 @@ static int soc_probe_link_components(struct snd_soc_card *card, int num, int order) { struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_component *component; int i, ret; /* probe the CPU-side component, if it is a CODEC */ - if (cpu_dai->codec && - !cpu_dai->codec->probed && - cpu_dai->codec->driver->probe_order == order) { - ret = soc_probe_codec(card, cpu_dai->codec); + component = rtd->cpu_dai->component; + if (component->driver->probe_order == order) { + ret = soc_probe_component(card, component); if (ret < 0) return ret; } /* probe the CODEC-side components */ for (i = 0; i < rtd->num_codecs; i++) { - if (!rtd->codec_dais[i]->codec->probed && - rtd->codec_dais[i]->codec->driver->probe_order == order) { - ret = soc_probe_codec(card, rtd->codec_dais[i]->codec); + component = rtd->codec_dais[i]->component; + if (component->driver->probe_order == order) { + ret = soc_probe_component(card, component); if (ret < 0) return ret; } } /* probe the platform */ - if (!platform->probed && - platform->driver->probe_order == order) { - ret = soc_probe_platform(card, platform); + if (platform->component.driver->probe_order == order) { + ret = soc_probe_component(card, &platform->component); if (ret < 0) return ret; } @@ -1482,18 +1338,12 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) /* probe the cpu_dai */ if (!cpu_dai->probed && cpu_dai->driver->probe_order == order) { - if (!cpu_dai->codec) { - if (!try_module_get(cpu_dai->dev->driver->owner)) - return -ENODEV; - } - if (cpu_dai->driver->probe) { ret = cpu_dai->driver->probe(cpu_dai); if (ret < 0) { dev_err(cpu_dai->dev, "ASoC: failed to probe CPU DAI %s: %d\n", cpu_dai->name, ret); - module_put(cpu_dai->dev->driver->owner); return ret; } } @@ -1654,17 +1504,24 @@ static int soc_bind_aux_dev(struct snd_soc_card *card, int num) { struct snd_soc_pcm_runtime *rtd = &card->rtd_aux[num]; struct snd_soc_aux_dev *aux_dev = &card->aux_dev[num]; - const char *codecname = aux_dev->codec_name; + const char *name = aux_dev->codec_name; - rtd->codec = soc_find_codec(aux_dev->codec_of_node, codecname); - if (!rtd->codec) { + rtd->component = soc_find_component(aux_dev->codec_of_node, name); + if (!rtd->component) { if (aux_dev->codec_of_node) - codecname = of_node_full_name(aux_dev->codec_of_node); + name = of_node_full_name(aux_dev->codec_of_node); - dev_err(card->dev, "ASoC: %s not registered\n", codecname); + dev_err(card->dev, "ASoC: %s not registered\n", name); return -EPROBE_DEFER; } + /* + * Some places still reference rtd->codec, so we have to keep that + * initialized if the component is a CODEC. Once all those references + * have been removed, this code can be removed as well. + */ + rtd->codec = rtd->component->codec; + return 0; } @@ -1674,18 +1531,13 @@ static int soc_probe_aux_dev(struct snd_soc_card *card, int num) struct snd_soc_aux_dev *aux_dev = &card->aux_dev[num]; int ret; - if (rtd->codec->probed) { - dev_err(rtd->codec->dev, "ASoC: codec already probed\n"); - return -EBUSY; - } - - ret = soc_probe_codec(card, rtd->codec); + ret = soc_probe_component(card, rtd->component); if (ret < 0) return ret; /* do machine specific initialization */ if (aux_dev->init) { - ret = aux_dev->init(&rtd->codec->dapm); + ret = aux_dev->init(rtd->component); if (ret < 0) { dev_err(card->dev, "ASoC: failed to init %s: %d\n", aux_dev->name, ret); @@ -1699,7 +1551,7 @@ static int soc_probe_aux_dev(struct snd_soc_card *card, int num) static void soc_remove_aux_dev(struct snd_soc_card *card, int num) { struct snd_soc_pcm_runtime *rtd = &card->rtd_aux[num]; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_component *component = rtd->component; /* unregister the rtd device */ if (rtd->dev_registered) { @@ -1708,8 +1560,8 @@ static void soc_remove_aux_dev(struct snd_soc_card *card, int num) rtd->dev_registered = 0; } - if (codec && codec->probed) - soc_remove_codec(codec); + if (component && component->probed) + soc_remove_component(component); } static int snd_soc_init_codec_cache(struct snd_soc_codec *codec) @@ -2107,19 +1959,14 @@ static struct platform_driver soc_driver = { int snd_soc_new_ac97_codec(struct snd_soc_codec *codec, struct snd_ac97_bus_ops *ops, int num) { - mutex_lock(&codec->mutex); - codec->ac97 = kzalloc(sizeof(struct snd_ac97), GFP_KERNEL); - if (codec->ac97 == NULL) { - mutex_unlock(&codec->mutex); + if (codec->ac97 == NULL) return -ENOMEM; - } codec->ac97->bus = kzalloc(sizeof(struct snd_ac97_bus), GFP_KERNEL); if (codec->ac97->bus == NULL) { kfree(codec->ac97); codec->ac97 = NULL; - mutex_unlock(&codec->mutex); return -ENOMEM; } @@ -2132,7 +1979,6 @@ int snd_soc_new_ac97_codec(struct snd_soc_codec *codec, */ codec->ac97_created = 1; - mutex_unlock(&codec->mutex); return 0; } EXPORT_SYMBOL_GPL(snd_soc_new_ac97_codec); @@ -2302,7 +2148,6 @@ EXPORT_SYMBOL_GPL(snd_soc_set_ac97_ops_of_reset); */ void snd_soc_free_ac97_codec(struct snd_soc_codec *codec) { - mutex_lock(&codec->mutex); #ifdef CONFIG_SND_SOC_AC97_BUS soc_unregister_ac97_codec(codec); #endif @@ -2310,7 +2155,6 @@ void snd_soc_free_ac97_codec(struct snd_soc_codec *codec) kfree(codec->ac97); codec->ac97 = NULL; codec->ac97_created = 0; - mutex_unlock(&codec->mutex); } EXPORT_SYMBOL_GPL(snd_soc_free_ac97_codec); @@ -3027,9 +2871,10 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, unsigned int val, val_mask; int ret; - val = ((ucontrol->value.integer.value[0] + min) & mask); if (invert) - val = max - val; + val = (max - ucontrol->value.integer.value[0]) & mask; + else + val = ((ucontrol->value.integer.value[0] + min) & mask); val_mask = mask << shift; val = val << shift; @@ -3038,9 +2883,10 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, return ret; if (snd_soc_volsw_is_stereo(mc)) { - val = ((ucontrol->value.integer.value[1] + min) & mask); if (invert) - val = max - val; + val = (max - ucontrol->value.integer.value[1]) & mask; + else + val = ((ucontrol->value.integer.value[1] + min) & mask); val_mask = mask << shift; val = val << shift; @@ -3085,8 +2931,9 @@ int snd_soc_get_volsw_range(struct snd_kcontrol *kcontrol, if (invert) ucontrol->value.integer.value[0] = max - ucontrol->value.integer.value[0]; - ucontrol->value.integer.value[0] = - ucontrol->value.integer.value[0] - min; + else + ucontrol->value.integer.value[0] = + ucontrol->value.integer.value[0] - min; if (snd_soc_volsw_is_stereo(mc)) { ret = snd_soc_component_read(component, rreg, &val); @@ -3097,8 +2944,9 @@ int snd_soc_get_volsw_range(struct snd_kcontrol *kcontrol, if (invert) ucontrol->value.integer.value[1] = max - ucontrol->value.integer.value[1]; - ucontrol->value.integer.value[1] = - ucontrol->value.integer.value[1] - min; + else + ucontrol->value.integer.value[1] = + ucontrol->value.integer.value[1] - min; } return 0; @@ -3203,7 +3051,7 @@ int snd_soc_bytes_put(struct snd_kcontrol *kcontrol, unsigned int val, mask; void *data; - if (!component->regmap) + if (!component->regmap || !params->num_regs) return -EINVAL; len = params->num_regs * component->val_bytes; @@ -3928,8 +3776,11 @@ EXPORT_SYMBOL_GPL(snd_soc_register_card); */ int snd_soc_unregister_card(struct snd_soc_card *card) { - if (card->instantiated) + if (card->instantiated) { + card->instantiated = false; + snd_soc_dapm_shutdown(card); soc_cleanup_card_resources(card); + } dev_dbg(card->dev, "ASoC: Unregistered card '%s'\n", card->name); return 0; @@ -4116,6 +3967,8 @@ static int snd_soc_component_initialize(struct snd_soc_component *component, component->dev = dev; component->driver = driver; + component->probe = component->driver->probe; + component->remove = component->driver->remove; if (!component->dapm_ptr) component->dapm_ptr = &component->dapm; @@ -4124,19 +3977,42 @@ static int snd_soc_component_initialize(struct snd_soc_component *component, dapm->dev = dev; dapm->component = component; dapm->bias_level = SND_SOC_BIAS_OFF; + dapm->idle_bias_off = true; if (driver->seq_notifier) dapm->seq_notifier = snd_soc_component_seq_notifier; if (driver->stream_event) dapm->stream_event = snd_soc_component_stream_event; + component->controls = driver->controls; + component->num_controls = driver->num_controls; + component->dapm_widgets = driver->dapm_widgets; + component->num_dapm_widgets = driver->num_dapm_widgets; + component->dapm_routes = driver->dapm_routes; + component->num_dapm_routes = driver->num_dapm_routes; + INIT_LIST_HEAD(&component->dai_list); mutex_init(&component->io_mutex); return 0; } +static void snd_soc_component_init_regmap(struct snd_soc_component *component) +{ + if (!component->regmap) + component->regmap = dev_get_regmap(component->dev, NULL); + if (component->regmap) { + int val_bytes = regmap_get_val_bytes(component->regmap); + /* Errors are legitimate for non-integer byte multiples */ + if (val_bytes > 0) + component->val_bytes = val_bytes; + } +} + static void snd_soc_component_add_unlocked(struct snd_soc_component *component) { + if (!component->write && !component->read) + snd_soc_component_init_regmap(component); + list_add(&component->list, &component_list); } @@ -4225,22 +4101,18 @@ found: } EXPORT_SYMBOL_GPL(snd_soc_unregister_component); -static int snd_soc_platform_drv_write(struct snd_soc_component *component, - unsigned int reg, unsigned int val) +static int snd_soc_platform_drv_probe(struct snd_soc_component *component) { struct snd_soc_platform *platform = snd_soc_component_to_platform(component); - return platform->driver->write(platform, reg, val); + return platform->driver->probe(platform); } -static int snd_soc_platform_drv_read(struct snd_soc_component *component, - unsigned int reg, unsigned int *val) +static void snd_soc_platform_drv_remove(struct snd_soc_component *component) { struct snd_soc_platform *platform = snd_soc_component_to_platform(component); - *val = platform->driver->read(platform, reg); - - return 0; + platform->driver->remove(platform); } /** @@ -4261,10 +4133,15 @@ int snd_soc_add_platform(struct device *dev, struct snd_soc_platform *platform, platform->dev = dev; platform->driver = platform_drv; - if (platform_drv->write) - platform->component.write = snd_soc_platform_drv_write; - if (platform_drv->read) - platform->component.read = snd_soc_platform_drv_read; + + if (platform_drv->probe) + platform->component.probe = snd_soc_platform_drv_probe; + if (platform_drv->remove) + platform->component.remove = snd_soc_platform_drv_remove; + +#ifdef CONFIG_DEBUG_FS + platform->component.debugfs_prefix = "platform"; +#endif mutex_lock(&client_mutex); snd_soc_component_add_unlocked(&platform->component); @@ -4386,6 +4263,20 @@ static void fixup_codec_formats(struct snd_soc_pcm_stream *stream) stream->formats |= codec_format_map[i]; } +static int snd_soc_codec_drv_probe(struct snd_soc_component *component) +{ + struct snd_soc_codec *codec = snd_soc_component_to_codec(component); + + return codec->driver->probe(codec); +} + +static void snd_soc_codec_drv_remove(struct snd_soc_component *component) +{ + struct snd_soc_codec *codec = snd_soc_component_to_codec(component); + + codec->driver->remove(codec); +} + static int snd_soc_codec_drv_write(struct snd_soc_component *component, unsigned int reg, unsigned int val) { @@ -4424,7 +4315,6 @@ int snd_soc_register_codec(struct device *dev, { struct snd_soc_codec *codec; struct snd_soc_dai *dai; - struct regmap *regmap; int ret, i; dev_dbg(dev, "codec register %s\n", dev_name(dev)); @@ -4434,18 +4324,37 @@ int snd_soc_register_codec(struct device *dev, return -ENOMEM; codec->component.dapm_ptr = &codec->dapm; + codec->component.codec = codec; ret = snd_soc_component_initialize(&codec->component, &codec_drv->component_driver, dev); if (ret) goto err_free; + if (codec_drv->controls) { + codec->component.controls = codec_drv->controls; + codec->component.num_controls = codec_drv->num_controls; + } + if (codec_drv->dapm_widgets) { + codec->component.dapm_widgets = codec_drv->dapm_widgets; + codec->component.num_dapm_widgets = codec_drv->num_dapm_widgets; + } + if (codec_drv->dapm_routes) { + codec->component.dapm_routes = codec_drv->dapm_routes; + codec->component.num_dapm_routes = codec_drv->num_dapm_routes; + } + + if (codec_drv->probe) + codec->component.probe = snd_soc_codec_drv_probe; + if (codec_drv->remove) + codec->component.remove = snd_soc_codec_drv_remove; if (codec_drv->write) codec->component.write = snd_soc_codec_drv_write; if (codec_drv->read) codec->component.read = snd_soc_codec_drv_read; codec->component.ignore_pmdown_time = codec_drv->ignore_pmdown_time; - codec->dapm.codec = codec; + codec->dapm.idle_bias_off = codec_drv->idle_bias_off; + codec->dapm.suspend_bias_off = codec_drv->suspend_bias_off; if (codec_drv->seq_notifier) codec->dapm.seq_notifier = codec_drv->seq_notifier; if (codec_drv->set_bias_level) @@ -4455,23 +4364,13 @@ int snd_soc_register_codec(struct device *dev, codec->component.val_bytes = codec_drv->reg_word_size; mutex_init(&codec->mutex); - if (!codec->component.write) { - if (codec_drv->get_regmap) - regmap = codec_drv->get_regmap(dev); - else - regmap = dev_get_regmap(dev, NULL); - - if (regmap) { - ret = snd_soc_component_init_io(&codec->component, - regmap); - if (ret) { - dev_err(codec->dev, - "Failed to set cache I/O:%d\n", - ret); - goto err_cleanup; - } - } - } +#ifdef CONFIG_DEBUG_FS + codec->component.init_debugfs = soc_init_codec_debugfs; + codec->component.debugfs_prefix = "codec"; +#endif + + if (codec_drv->get_regmap) + codec->component.regmap = codec_drv->get_regmap(dev); for (i = 0; i < num_dai; i++) { fixup_codec_formats(&dai_drv[i].playback); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 177bd8639ef9..2c456a376ade 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -326,12 +326,13 @@ static struct list_head *dapm_kcontrol_get_path_list( list_for_each_entry(path, dapm_kcontrol_get_path_list(kcontrol), \ list_kcontrol) -static unsigned int dapm_kcontrol_get_value(const struct snd_kcontrol *kcontrol) +unsigned int dapm_kcontrol_get_value(const struct snd_kcontrol *kcontrol) { struct dapm_kcontrol_data *data = snd_kcontrol_chip(kcontrol); return data->value; } +EXPORT_SYMBOL_GPL(dapm_kcontrol_get_value); static bool dapm_kcontrol_set_value(const struct snd_kcontrol *kcontrol, unsigned int value) @@ -1683,6 +1684,22 @@ static void dapm_power_one_widget(struct snd_soc_dapm_widget *w, } } +static bool dapm_idle_bias_off(struct snd_soc_dapm_context *dapm) +{ + if (dapm->idle_bias_off) + return true; + + switch (snd_power_get_state(dapm->card->snd_card)) { + case SNDRV_CTL_POWER_D3hot: + case SNDRV_CTL_POWER_D3cold: + return dapm->suspend_bias_off; + default: + break; + } + + return false; +} + /* * Scan each dapm widget for complete audio path. * A complete path is a route that has valid endpoints i.e.:- @@ -1706,7 +1723,7 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event) trace_snd_soc_dapm_start(card); list_for_each_entry(d, &card->dapm_list, list) { - if (d->idle_bias_off) + if (dapm_idle_bias_off(d)) d->target_bias_level = SND_SOC_BIAS_OFF; else d->target_bias_level = SND_SOC_BIAS_STANDBY; @@ -1772,7 +1789,7 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event) if (d->target_bias_level > bias) bias = d->target_bias_level; list_for_each_entry(d, &card->dapm_list, list) - if (!d->idle_bias_off) + if (!dapm_idle_bias_off(d)) d->target_bias_level = bias; trace_snd_soc_dapm_walk_done(card); @@ -3109,7 +3126,8 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, } w->dapm = dapm; - w->codec = dapm->codec; + if (dapm->component) + w->codec = dapm->component->codec; INIT_LIST_HEAD(&w->sources); INIT_LIST_HEAD(&w->sinks); INIT_LIST_HEAD(&w->list); diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 6307f85e871b..b329b84bc5af 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -336,10 +336,12 @@ static const struct snd_pcm_ops dmaengine_pcm_ops = { }; static const struct snd_soc_platform_driver dmaengine_pcm_platform = { + .component_driver = { + .probe_order = SND_SOC_COMP_ORDER_LATE, + }, .ops = &dmaengine_pcm_ops, .pcm_new = dmaengine_pcm_new, .pcm_free = dmaengine_pcm_free, - .probe_order = SND_SOC_COMP_ORDER_LATE, }; static const char * const dmaengine_pcm_dma_channel_names[] = { diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c index 7767fbd73eb7..9b3939049cef 100644 --- a/sound/soc/soc-io.c +++ b/sound/soc/soc-io.c @@ -271,31 +271,3 @@ int snd_soc_platform_write(struct snd_soc_platform *platform, return snd_soc_component_write(&platform->component, reg, val); } EXPORT_SYMBOL_GPL(snd_soc_platform_write); - -/** - * snd_soc_component_init_io() - Initialize regmap IO - * - * @component: component to initialize - * @regmap: regmap instance to use for IO operations - * - * Return: 0 on success, a negative error code otherwise - */ -int snd_soc_component_init_io(struct snd_soc_component *component, - struct regmap *regmap) -{ - int ret; - - if (!regmap) - return -EINVAL; - - ret = regmap_get_val_bytes(regmap); - /* Errors are legitimate for non-integer byte - * multiples */ - if (ret > 0) - component->val_bytes = ret; - - component->regmap = regmap; - - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_component_init_io); diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 731fdb5b5f9b..642c86240752 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2352,7 +2352,11 @@ static int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream) mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME); fe->dpcm[stream].runtime = fe_substream->runtime; - if (dpcm_path_get(fe, stream, &list) <= 0) { + ret = dpcm_path_get(fe, stream, &list); + if (ret < 0) { + mutex_unlock(&fe->card->mutex); + return ret; + } else if (ret == 0) { dev_dbg(fe->dev, "ASoC: %s no valid %s route\n", fe->dai_link->name, stream ? "capture" : "playback"); } diff --git a/sound/soc/spear/spear_pcm.c b/sound/soc/spear/spear_pcm.c index 0e5a8f35d0ad..a7dc3c56f44d 100644 --- a/sound/soc/spear/spear_pcm.c +++ b/sound/soc/spear/spear_pcm.c @@ -4,7 +4,7 @@ * sound/soc/spear/spear_pcm.c * * Copyright (C) 2012 ST Microelectronics - * Rajeev Kumar<rajeev-dlh.kumar@st.com> + * Rajeev Kumar<rajeevkumar.linux@gmail.com> * * This file is licensed under the terms of the GNU General Public * License version 2. This program is licensed "as is" without any @@ -50,6 +50,6 @@ int devm_spear_pcm_platform_register(struct device *dev, } EXPORT_SYMBOL_GPL(devm_spear_pcm_platform_register); -MODULE_AUTHOR("Rajeev Kumar <rajeev-dlh.kumar@st.com>"); +MODULE_AUTHOR("Rajeev Kumar <rajeevkumar.linux@gmail.com>"); MODULE_DESCRIPTION("SPEAr PCM DMA module"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/tegra/tegra_max98090.c b/sound/soc/tegra/tegra_max98090.c index b86cd9936ef1..01921d7e73fa 100644 --- a/sound/soc/tegra/tegra_max98090.c +++ b/sound/soc/tegra/tegra_max98090.c @@ -42,6 +42,7 @@ struct tegra_max98090 { struct tegra_asoc_utils_data util_data; int gpio_hp_det; + int gpio_mic_det; }; static int tegra_max98090_asoc_hw_params(struct snd_pcm_substream *substream, @@ -112,6 +113,22 @@ static struct snd_soc_jack_gpio tegra_max98090_hp_jack_gpio = { .invert = 1, }; +static struct snd_soc_jack tegra_max98090_mic_jack; + +static struct snd_soc_jack_pin tegra_max98090_mic_jack_pins[] = { + { + .pin = "Mic Jack", + .mask = SND_JACK_MICROPHONE, + }, +}; + +static struct snd_soc_jack_gpio tegra_max98090_mic_jack_gpio = { + .name = "Mic detection", + .report = SND_JACK_MICROPHONE, + .debounce_time = 150, + .invert = 1, +}; + static const struct snd_soc_dapm_widget tegra_max98090_dapm_widgets[] = { SND_SOC_DAPM_HP("Headphones", NULL), SND_SOC_DAPM_SPK("Speakers", NULL), @@ -141,6 +158,19 @@ static int tegra_max98090_asoc_init(struct snd_soc_pcm_runtime *rtd) &tegra_max98090_hp_jack_gpio); } + if (gpio_is_valid(machine->gpio_mic_det)) { + snd_soc_jack_new(codec, "Mic Jack", SND_JACK_MICROPHONE, + &tegra_max98090_mic_jack); + snd_soc_jack_add_pins(&tegra_max98090_mic_jack, + ARRAY_SIZE(tegra_max98090_mic_jack_pins), + tegra_max98090_mic_jack_pins); + + tegra_max98090_mic_jack_gpio.gpio = machine->gpio_mic_det; + snd_soc_jack_add_gpios(&tegra_max98090_mic_jack, + 1, + &tegra_max98090_mic_jack_gpio); + } + return 0; } @@ -153,6 +183,11 @@ static int tegra_max98090_card_remove(struct snd_soc_card *card) &tegra_max98090_hp_jack_gpio); } + if (gpio_is_valid(machine->gpio_mic_det)) { + snd_soc_jack_free_gpios(&tegra_max98090_mic_jack, 1, + &tegra_max98090_mic_jack_gpio); + } + return 0; } @@ -201,6 +236,11 @@ static int tegra_max98090_probe(struct platform_device *pdev) if (machine->gpio_hp_det == -EPROBE_DEFER) return -EPROBE_DEFER; + machine->gpio_mic_det = + of_get_named_gpio(np, "nvidia,mic-det-gpios", 0); + if (machine->gpio_mic_det == -EPROBE_DEFER) + return -EPROBE_DEFER; + ret = snd_soc_of_parse_card_name(card, "nvidia,model"); if (ret) goto err; diff --git a/sound/soc/txx9/txx9aclc.c b/sound/soc/txx9/txx9aclc.c index f0829de28708..cd71fd889d8b 100644 --- a/sound/soc/txx9/txx9aclc.c +++ b/sound/soc/txx9/txx9aclc.c @@ -16,6 +16,7 @@ #include <linux/platform_device.h> #include <linux/scatterlist.h> #include <linux/slab.h> +#include <linux/dmaengine.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -137,7 +138,7 @@ txx9aclc_dma_submit(struct txx9aclc_dmadata *dmadata, dma_addr_t buf_dma_addr) } desc->callback = txx9aclc_dma_complete; desc->callback_param = dmadata; - desc->tx_submit(desc); + dmaengine_submit(desc); return desc; } @@ -160,7 +161,7 @@ static void txx9aclc_dma_tasklet(unsigned long data) void __iomem *base = drvdata->base; spin_unlock_irqrestore(&dmadata->dma_lock, flags); - chan->device->device_control(chan, DMA_TERMINATE_ALL, 0); + dmaengine_terminate_all(chan); /* first time */ for (i = 0; i < NR_DMA_CHAIN; i++) { desc = txx9aclc_dma_submit(dmadata, @@ -169,7 +170,7 @@ static void txx9aclc_dma_tasklet(unsigned long data) return; } dmadata->dmacount = NR_DMA_CHAIN; - chan->device->device_issue_pending(chan); + dma_async_issue_pending(chan); spin_lock_irqsave(&dmadata->dma_lock, flags); __raw_writel(ctlbit, base + ACCTLEN); dmadata->frag_count = NR_DMA_CHAIN % dmadata->frags; @@ -188,7 +189,7 @@ static void txx9aclc_dma_tasklet(unsigned long data) dmadata->frag_count * dmadata->frag_bytes); if (!desc) return; - chan->device->device_issue_pending(chan); + dma_async_issue_pending(chan); spin_lock_irqsave(&dmadata->dma_lock, flags); dmadata->frag_count++; @@ -266,7 +267,7 @@ static int txx9aclc_pcm_close(struct snd_pcm_substream *substream) struct dma_chan *chan = dmadata->dma_chan; dmadata->frag_count = -1; - chan->device->device_control(chan, DMA_TERMINATE_ALL, 0); + dmaengine_terminate_all(chan); return 0; } @@ -398,8 +399,7 @@ static int txx9aclc_pcm_remove(struct snd_soc_platform *platform) struct dma_chan *chan = dmadata->dma_chan; if (chan) { dmadata->frag_count = -1; - chan->device->device_control(chan, - DMA_TERMINATE_ALL, 0); + dmaengine_terminate_all(chan); dma_release_channel(chan); } dev->dmadata[i].dma_chan = NULL; diff --git a/sound/usb/caiaq/control.c b/sound/usb/caiaq/control.c index f65fc0987cfb..b7a7c805d63f 100644 --- a/sound/usb/caiaq/control.c +++ b/sound/usb/caiaq/control.c @@ -100,15 +100,19 @@ static int control_put(struct snd_kcontrol *kcontrol, struct snd_usb_caiaqdev *cdev = caiaqdev(chip->card); int pos = kcontrol->private_value; int v = ucontrol->value.integer.value[0]; - unsigned char cmd = EP1_CMD_WRITE_IO; + unsigned char cmd; - if (cdev->chip.usb_id == - USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLX1)) - cmd = EP1_CMD_DIMM_LEDS; - - if (cdev->chip.usb_id == - USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_MASCHINECONTROLLER)) + switch (cdev->chip.usb_id) { + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_MASCHINECONTROLLER): + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLX1): + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_KORECONTROLLER2): + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_KORECONTROLLER): cmd = EP1_CMD_DIMM_LEDS; + break; + default: + cmd = EP1_CMD_WRITE_IO; + break; + } if (pos & CNT_INTVAL) { int i = pos & ~CNT_INTVAL; |