diff options
Diffstat (limited to 'sound')
-rw-r--r-- | sound/soc/codecs/Kconfig | 4 | ||||
-rw-r--r-- | sound/soc/codecs/Makefile | 2 | ||||
-rw-r--r-- | sound/soc/codecs/max98357a.c | 138 | ||||
-rw-r--r-- | sound/soc/codecs/rt286.c | 31 | ||||
-rw-r--r-- | sound/soc/codecs/rt286.h | 7 | ||||
-rw-r--r-- | sound/soc/codecs/rt5645.c | 81 | ||||
-rw-r--r-- | sound/soc/codecs/rt5645.h | 72 | ||||
-rw-r--r-- | sound/soc/codecs/rt5670.c | 1 | ||||
-rw-r--r-- | sound/soc/intel/Kconfig | 11 | ||||
-rw-r--r-- | sound/soc/intel/Makefile | 2 | ||||
-rw-r--r-- | sound/soc/intel/cht_bsw_rt5645.c | 326 | ||||
-rw-r--r-- | sound/soc/intel/sst-haswell-ipc.c | 168 | ||||
-rw-r--r-- | sound/soc/intel/sst-haswell-ipc.h | 31 | ||||
-rw-r--r-- | sound/soc/intel/sst-haswell-pcm.c | 70 | ||||
-rw-r--r-- | sound/soc/intel/sst/sst.h | 3 | ||||
-rw-r--r-- | sound/soc/intel/sst/sst_acpi.c | 6 | ||||
-rw-r--r-- | sound/soc/jz4740/jz4740-i2s.c | 21 | ||||
-rw-r--r-- | sound/soc/samsung/Kconfig | 11 | ||||
-rw-r--r-- | sound/soc/samsung/Makefile | 2 | ||||
-rw-r--r-- | sound/soc/samsung/goni_wm8994.c | 289 | ||||
-rw-r--r-- | sound/soc/soc-core.c | 2 |
21 files changed, 694 insertions, 584 deletions
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 3190eed43c38..064e6c18e109 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -69,6 +69,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX98088 if I2C select SND_SOC_MAX98090 if I2C select SND_SOC_MAX98095 if I2C + select SND_SOC_MAX98357A select SND_SOC_MAX9850 if I2C select SND_SOC_MAX9768 if I2C select SND_SOC_MAX9877 if I2C @@ -456,6 +457,9 @@ config SND_SOC_MAX98090 config SND_SOC_MAX98095 tristate +config SND_SOC_MAX98357A + tristate + config SND_SOC_MAX9850 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index bbdfd1e1c182..69b8666d187a 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -64,6 +64,7 @@ snd-soc-max9768-objs := max9768.o snd-soc-max98088-objs := max98088.o snd-soc-max98090-objs := max98090.o snd-soc-max98095-objs := max98095.o +snd-soc-max98357a-objs := max98357a.o snd-soc-max9850-objs := max9850.o snd-soc-mc13783-objs := mc13783.o snd-soc-ml26124-objs := ml26124.o @@ -245,6 +246,7 @@ obj-$(CONFIG_SND_SOC_MAX9768) += snd-soc-max9768.o obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o obj-$(CONFIG_SND_SOC_MAX98090) += snd-soc-max98090.o obj-$(CONFIG_SND_SOC_MAX98095) += snd-soc-max98095.o +obj-$(CONFIG_SND_SOC_MAX98357A) += snd-soc-max98357a.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c new file mode 100644 index 000000000000..1806333ea29e --- /dev/null +++ b/sound/soc/codecs/max98357a.c @@ -0,0 +1,138 @@ +/* Copyright (c) 2010-2011,2013-2015 The Linux Foundation. All rights reserved. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 and + * only version 2 as published by the Free Software Foundation. + * + * 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. + * + * max98357a.c -- MAX98357A ALSA SoC Codec driver + */ + +#include <linux/module.h> +#include <linux/gpio.h> +#include <sound/soc.h> + +#define DRV_NAME "max98357a" + +static int max98357a_daiops_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + struct gpio_desc *sdmode = snd_soc_dai_get_drvdata(dai); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + gpiod_set_value(sdmode, 1); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + gpiod_set_value(sdmode, 0); + break; + } + + return 0; +} + +static const struct snd_soc_dapm_widget max98357a_dapm_widgets[] = { + SND_SOC_DAPM_DAC("SDMode", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_OUTPUT("Speaker"), +}; + +static const struct snd_soc_dapm_route max98357a_dapm_routes[] = { + {"Speaker", NULL, "SDMode"}, +}; + +static int max98357a_codec_probe(struct snd_soc_codec *codec) +{ + struct gpio_desc *sdmode; + + sdmode = devm_gpiod_get(codec->dev, "sdmode"); + if (IS_ERR(sdmode)) { + dev_err(codec->dev, "%s() unable to get sdmode GPIO: %ld\n", + __func__, PTR_ERR(sdmode)); + return PTR_ERR(sdmode); + } + gpiod_direction_output(sdmode, 0); + snd_soc_codec_set_drvdata(codec, sdmode); + + return 0; +} + +static struct snd_soc_codec_driver max98357a_codec_driver = { + .probe = max98357a_codec_probe, + .dapm_widgets = max98357a_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(max98357a_dapm_widgets), + .dapm_routes = max98357a_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(max98357a_dapm_routes), +}; + +static struct snd_soc_dai_ops max98357a_dai_ops = { + .trigger = max98357a_daiops_trigger, +}; + +static struct snd_soc_dai_driver max98357a_dai_driver = { + .name = DRV_NAME, + .playback = { + .stream_name = DRV_NAME "-playback", + .formats = SNDRV_PCM_FMTBIT_S16 | + SNDRV_PCM_FMTBIT_S24 | + SNDRV_PCM_FMTBIT_S32, + .rates = SNDRV_PCM_RATE_8000 | + SNDRV_PCM_RATE_16000 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_96000, + .rate_min = 8000, + .rate_max = 96000, + .channels_min = 1, + .channels_max = 2, + }, + .ops = &max98357a_dai_ops, +}; + +static int max98357a_platform_probe(struct platform_device *pdev) +{ + int ret; + + ret = snd_soc_register_codec(&pdev->dev, &max98357a_codec_driver, + &max98357a_dai_driver, 1); + if (ret) + dev_err(&pdev->dev, "%s() error registering codec driver: %d\n", + __func__, ret); + + return ret; +} + +static int max98357a_platform_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + + return 0; +} + +#ifdef CONFIG_OF +static const struct of_device_id max98357a_device_id[] = { + { .compatible = "maxim," DRV_NAME, }, + {} +}; +MODULE_DEVICE_TABLE(of, max98357a_device_id); +#endif + +static struct platform_driver max98357a_platform_driver = { + .driver = { + .name = DRV_NAME, + .of_match_table = of_match_ptr(max98357a_device_id), + }, + .probe = max98357a_platform_probe, + .remove = max98357a_platform_remove, +}; +module_platform_driver(max98357a_platform_driver); + +MODULE_DESCRIPTION("Maxim MAX98357A Codec Driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:" DRV_NAME); diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 8104d2285602..f374840a5a7c 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -34,6 +34,7 @@ #include "rt286.h" #define RT286_VENDOR_ID 0x10ec0286 +#define RT288_VENDOR_ID 0x10ec0288 struct rt286_priv { struct regmap *regmap; @@ -305,6 +306,8 @@ static int rt286_jack_detect(struct rt286_priv *rt286, bool *hp, bool *mic) *hp = false; *mic = false; + if (!rt286->codec) + return -EINVAL; if (rt286->pdata.cbj_en) { regmap_read(rt286->regmap, RT286_GET_HP_SENSE, &buf); *hp = buf & 0x80000000; @@ -1169,6 +1172,7 @@ static const struct regmap_config rt286_regmap = { static const struct i2c_device_id rt286_i2c_id[] = { {"rt286", 0}, + {"rt288", 0}, {} }; MODULE_DEVICE_TABLE(i2c, rt286_i2c_id); @@ -1189,6 +1193,17 @@ static struct dmi_system_id force_combo_jack_table[] = { { } }; +static struct dmi_system_id dmi_dell_dino[] = { + { + .ident = "Dell Dino", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc."), + DMI_MATCH(DMI_BOARD_NAME, "0144P8") + } + }, + { } +}; + static int rt286_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -1211,7 +1226,7 @@ static int rt286_i2c_probe(struct i2c_client *i2c, regmap_read(rt286->regmap, RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID), &ret); - if (ret != RT286_VENDOR_ID) { + if (ret != RT286_VENDOR_ID && ret != RT288_VENDOR_ID) { dev_err(&i2c->dev, "Device with ID register %x is not rt286\n", ret); return -ENODEV; @@ -1224,7 +1239,8 @@ static int rt286_i2c_probe(struct i2c_client *i2c, if (pdata) rt286->pdata = *pdata; - if (dmi_check_system(force_combo_jack_table)) + if (dmi_check_system(force_combo_jack_table) || + dmi_check_system(dmi_dell_dino)) rt286->pdata.cbj_en = true; regmap_write(rt286->regmap, RT286_SET_AUDIO_POWER, AC_PWRST_D3); @@ -1263,6 +1279,17 @@ static int rt286_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL3, 0xf777, 0x4737); regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL4, 0x00ff, 0x003f); + if (dmi_check_system(dmi_dell_dino)) { + regmap_update_bits(rt286->regmap, + RT286_SET_GPIO_MASK, 0x40, 0x40); + regmap_update_bits(rt286->regmap, + RT286_SET_GPIO_DIRECTION, 0x40, 0x40); + regmap_update_bits(rt286->regmap, + RT286_SET_GPIO_DATA, 0x40, 0x40); + regmap_update_bits(rt286->regmap, + RT286_GPIO_CTRL, 0xc, 0x8); + } + if (rt286->i2c->irq) { ret = request_threaded_irq(rt286->i2c->irq, NULL, rt286_irq, IRQF_TRIGGER_HIGH | IRQF_ONESHOT, "rt286", rt286); diff --git a/sound/soc/codecs/rt286.h b/sound/soc/codecs/rt286.h index b539b7320a79..7130edb152ef 100644 --- a/sound/soc/codecs/rt286.h +++ b/sound/soc/codecs/rt286.h @@ -117,6 +117,12 @@ VERB_CMD(AC_VERB_SET_COEF_INDEX, RT286_VENDOR_REGISTERS, 0) #define RT286_PROC_COEF\ VERB_CMD(AC_VERB_SET_PROC_COEF, RT286_VENDOR_REGISTERS, 0) +#define RT286_SET_GPIO_MASK\ + VERB_CMD(AC_VERB_SET_GPIO_MASK, RT286_AUDIO_FUNCTION_GROUP, 0) +#define RT286_SET_GPIO_DIRECTION\ + VERB_CMD(AC_VERB_SET_GPIO_DIRECTION, RT286_AUDIO_FUNCTION_GROUP, 0) +#define RT286_SET_GPIO_DATA\ + VERB_CMD(AC_VERB_SET_GPIO_DATA, RT286_AUDIO_FUNCTION_GROUP, 0) /* Index registers */ #define RT286_A_BIAS_CTRL1 0x01 @@ -131,6 +137,7 @@ #define RT286_POWER_CTRL3 0x0f #define RT286_MIC1_DET_CTRL 0x19 #define RT286_MISC_CTRL1 0x20 +#define RT286_GPIO_CTRL 0x29 #define RT286_IRQ_CTRL 0x33 #define RT286_PLL_CTRL1 0x49 #define RT286_CBJ_CTRL1 0x4f diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 3e16a889a806..c9a4c5be083b 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -615,6 +615,87 @@ static int is_using_asrc(struct snd_soc_dapm_widget *source, } +/** + * rt5645_sel_asrc_clk_src - select ASRC clock source for a set of filters + * @codec: SoC audio codec device. + * @filter_mask: mask of filters. + * @clk_src: clock source + * + * The ASRC function is for asynchronous MCLK and LRCK. Also, since RT5645 can + * only support standard 32fs or 64fs i2s format, ASRC should be enabled to + * support special i2s clock format such as Intel's 100fs(100 * sampling rate). + * ASRC function will track i2s clock and generate a corresponding system clock + * for codec. This function provides an API to select the clock source for a + * set of filters specified by the mask. And the codec driver will turn on ASRC + * for these filters if ASRC is selected as their clock source. + */ +int rt5645_sel_asrc_clk_src(struct snd_soc_codec *codec, + unsigned int filter_mask, unsigned int clk_src) +{ + unsigned int asrc2_mask = 0; + unsigned int asrc2_value = 0; + unsigned int asrc3_mask = 0; + unsigned int asrc3_value = 0; + + switch (clk_src) { + case RT5645_CLK_SEL_SYS: + case RT5645_CLK_SEL_I2S1_ASRC: + case RT5645_CLK_SEL_I2S2_ASRC: + case RT5645_CLK_SEL_SYS2: + break; + + default: + return -EINVAL; + } + + if (filter_mask & RT5645_DA_STEREO_FILTER) { + asrc2_mask |= RT5645_DA_STO_CLK_SEL_MASK; + asrc2_value = (asrc2_value & ~RT5645_DA_STO_CLK_SEL_MASK) + | (clk_src << RT5645_DA_STO_CLK_SEL_SFT); + } + + if (filter_mask & RT5645_DA_MONO_L_FILTER) { + asrc2_mask |= RT5645_DA_MONOL_CLK_SEL_MASK; + asrc2_value = (asrc2_value & ~RT5645_DA_MONOL_CLK_SEL_MASK) + | (clk_src << RT5645_DA_MONOL_CLK_SEL_SFT); + } + + if (filter_mask & RT5645_DA_MONO_R_FILTER) { + asrc2_mask |= RT5645_DA_MONOR_CLK_SEL_MASK; + asrc2_value = (asrc2_value & ~RT5645_DA_MONOR_CLK_SEL_MASK) + | (clk_src << RT5645_DA_MONOR_CLK_SEL_SFT); + } + + if (filter_mask & RT5645_AD_STEREO_FILTER) { + asrc2_mask |= RT5645_AD_STO1_CLK_SEL_MASK; + asrc2_value = (asrc2_value & ~RT5645_AD_STO1_CLK_SEL_MASK) + | (clk_src << RT5645_AD_STO1_CLK_SEL_SFT); + } + + if (filter_mask & RT5645_AD_MONO_L_FILTER) { + asrc3_mask |= RT5645_AD_MONOL_CLK_SEL_MASK; + asrc3_value = (asrc3_value & ~RT5645_AD_MONOL_CLK_SEL_MASK) + | (clk_src << RT5645_AD_MONOL_CLK_SEL_SFT); + } + + if (filter_mask & RT5645_AD_MONO_R_FILTER) { + asrc3_mask |= RT5645_AD_MONOR_CLK_SEL_MASK; + asrc3_value = (asrc3_value & ~RT5645_AD_MONOR_CLK_SEL_MASK) + | (clk_src << RT5645_AD_MONOR_CLK_SEL_SFT); + } + + if (asrc2_mask) + snd_soc_update_bits(codec, RT5645_ASRC_2, + asrc2_mask, asrc2_value); + + if (asrc3_mask) + snd_soc_update_bits(codec, RT5645_ASRC_3, + asrc3_mask, asrc3_value); + + return 0; +} +EXPORT_SYMBOL_GPL(rt5645_sel_asrc_clk_src); + /* Digital Mixer */ static const struct snd_kcontrol_new rt5645_sto1_adc_l_mix[] = { SOC_DAPM_SINGLE("ADC1 Switch", RT5645_STO1_ADC_MIXER, diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h index 74542310d3f0..dbfd98c22f4d 100644 --- a/sound/soc/codecs/rt5645.h +++ b/sound/soc/codecs/rt5645.h @@ -1120,50 +1120,27 @@ #define RT5645_DMIC_2_M_NOR (0x0 << 8) #define RT5645_DMIC_2_M_ASYN (0x1 << 8) +/* ASRC clock source selection (0x84, 0x85) */ +#define RT5645_CLK_SEL_SYS (0x0) +#define RT5645_CLK_SEL_I2S1_ASRC (0x1) +#define RT5645_CLK_SEL_I2S2_ASRC (0x2) +#define RT5645_CLK_SEL_SYS2 (0x5) + /* ASRC Control 2 (0x84) */ -#define RT5645_MDA_L_M_MASK (0x1 << 15) -#define RT5645_MDA_L_M_SFT 15 -#define RT5645_MDA_L_M_NOR (0x0 << 15) -#define RT5645_MDA_L_M_ASYN (0x1 << 15) -#define RT5645_MDA_R_M_MASK (0x1 << 14) -#define RT5645_MDA_R_M_SFT 14 -#define RT5645_MDA_R_M_NOR (0x0 << 14) -#define RT5645_MDA_R_M_ASYN (0x1 << 14) -#define RT5645_MAD_L_M_MASK (0x1 << 13) -#define RT5645_MAD_L_M_SFT 13 -#define RT5645_MAD_L_M_NOR (0x0 << 13) -#define RT5645_MAD_L_M_ASYN (0x1 << 13) -#define RT5645_MAD_R_M_MASK (0x1 << 12) -#define RT5645_MAD_R_M_SFT 12 -#define RT5645_MAD_R_M_NOR (0x0 << 12) -#define RT5645_MAD_R_M_ASYN (0x1 << 12) -#define RT5645_ADC_M_MASK (0x1 << 11) -#define RT5645_ADC_M_SFT 11 -#define RT5645_ADC_M_NOR (0x0 << 11) -#define RT5645_ADC_M_ASYN (0x1 << 11) -#define RT5645_STO_DAC_M_MASK (0x1 << 5) -#define RT5645_STO_DAC_M_SFT 5 -#define RT5645_STO_DAC_M_NOR (0x0 << 5) -#define RT5645_STO_DAC_M_ASYN (0x1 << 5) -#define RT5645_I2S1_R_D_MASK (0x1 << 4) -#define RT5645_I2S1_R_D_SFT 4 -#define RT5645_I2S1_R_D_DIS (0x0 << 4) -#define RT5645_I2S1_R_D_EN (0x1 << 4) -#define RT5645_I2S2_R_D_MASK (0x1 << 3) -#define RT5645_I2S2_R_D_SFT 3 -#define RT5645_I2S2_R_D_DIS (0x0 << 3) -#define RT5645_I2S2_R_D_EN (0x1 << 3) -#define RT5645_PRE_SCLK_MASK (0x3) -#define RT5645_PRE_SCLK_SFT 0 -#define RT5645_PRE_SCLK_512 (0x0) -#define RT5645_PRE_SCLK_1024 (0x1) -#define RT5645_PRE_SCLK_2048 (0x2) +#define RT5645_DA_STO_CLK_SEL_MASK (0xf << 12) +#define RT5645_DA_STO_CLK_SEL_SFT 12 +#define RT5645_DA_MONOL_CLK_SEL_MASK (0xf << 8) +#define RT5645_DA_MONOL_CLK_SEL_SFT 8 +#define RT5645_DA_MONOR_CLK_SEL_MASK (0xf << 4) +#define RT5645_DA_MONOR_CLK_SEL_SFT 4 +#define RT5645_AD_STO1_CLK_SEL_MASK (0xf << 0) +#define RT5645_AD_STO1_CLK_SEL_SFT 0 /* ASRC Control 3 (0x85) */ -#define RT5645_I2S1_RATE_MASK (0xf << 12) -#define RT5645_I2S1_RATE_SFT 12 -#define RT5645_I2S2_RATE_MASK (0xf << 8) -#define RT5645_I2S2_RATE_SFT 8 +#define RT5645_AD_MONOL_CLK_SEL_MASK (0xf << 4) +#define RT5645_AD_MONOL_CLK_SEL_SFT 4 +#define RT5645_AD_MONOR_CLK_SEL_MASK (0xf << 0) +#define RT5645_AD_MONOR_CLK_SEL_SFT 0 /* ASRC Control 4 (0x89) */ #define RT5645_I2S1_PD_MASK (0x7 << 12) @@ -2189,6 +2166,19 @@ enum { CODEC_TYPE_RT5650, }; +/* filter mask */ +enum { + RT5645_DA_STEREO_FILTER = 0x1, + RT5645_DA_MONO_L_FILTER = (0x1 << 1), + RT5645_DA_MONO_R_FILTER = (0x1 << 2), + RT5645_AD_STEREO_FILTER = (0x1 << 3), + RT5645_AD_MONO_L_FILTER = (0x1 << 4), + RT5645_AD_MONO_R_FILTER = (0x1 << 5), +}; + +int rt5645_sel_asrc_clk_src(struct snd_soc_codec *codec, + unsigned int filter_mask, unsigned int clk_src); + struct rt5645_priv { struct snd_soc_codec *codec; struct rt5645_platform_data pdata; diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 7b3d6b5992f1..e1a4a45c57e2 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -2616,6 +2616,7 @@ static struct snd_soc_codec_driver soc_codec_dev_rt5670 = { static const struct regmap_config rt5670_regmap = { .reg_bits = 8, .val_bits = 16, + .use_single_rw = true, .max_register = RT5670_VENDOR_ID2 + 1 + (ARRAY_SIZE(rt5670_ranges) * RT5670_PR_SPACING), .volatile_reg = rt5670_volatile_register, diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index c0813f546d1f..ee03dbdda235 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -110,3 +110,14 @@ config SND_SOC_INTEL_CHT_BSW_RT5672_MACH platforms with RT5672 audio codec. Say Y if you have such a device If unsure select "N". + +config SND_SOC_INTEL_CHT_BSW_RT5645_MACH + tristate "ASoC Audio driver for Intel Cherrytrail & Braswell with RT5645 codec" + depends on X86_INTEL_LPSS + select SND_SOC_RT5645 + select SND_SST_MFLD_PLATFORM + select SND_SST_IPC_ACPI + help + This adds support for ASoC machine driver for Intel(R) Cherrytrail & Braswell + platforms with RT5645 audio codec. + If unsure select "N". diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile index e928ec385300..a8e53c45c6b6 100644 --- a/sound/soc/intel/Makefile +++ b/sound/soc/intel/Makefile @@ -28,6 +28,7 @@ snd-soc-sst-byt-max98090-mach-objs := byt-max98090.o snd-soc-sst-broadwell-objs := broadwell.o snd-soc-sst-bytcr-dpcm-rt5640-objs := bytcr_dpcm_rt5640.o snd-soc-sst-cht-bsw-rt5672-objs := cht_bsw_rt5672.o +snd-soc-sst-cht-bsw-rt5645-objs := cht_bsw_rt5645.o obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o @@ -35,6 +36,7 @@ obj-$(CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH) += snd-soc-sst-byt-max98090-mach.o obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-sst-broadwell.o obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5640_MACH) += snd-soc-sst-bytcr-dpcm-rt5640.o obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5672_MACH) += snd-soc-sst-cht-bsw-rt5672.o +obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5645_MACH) += snd-soc-sst-cht-bsw-rt5645.o # DSP driver obj-$(CONFIG_SND_SST_IPC) += sst/ diff --git a/sound/soc/intel/cht_bsw_rt5645.c b/sound/soc/intel/cht_bsw_rt5645.c new file mode 100644 index 000000000000..bd29617a9ab9 --- /dev/null +++ b/sound/soc/intel/cht_bsw_rt5645.c @@ -0,0 +1,326 @@ +/* + * cht-bsw-rt5645.c - ASoc Machine driver for Intel Cherryview-based platforms + * Cherrytrail and Braswell, with RT5645 codec. + * + * Copyright (C) 2015 Intel Corp + * Author: Fang, Yang A <yang.a.fang@intel.com> + * N,Harshapriya <harshapriya.n@intel.com> + * This file is modified from cht_bsw_rt5672.c + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * 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. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include "../codecs/rt5645.h" +#include "sst-atom-controls.h" + +#define CHT_PLAT_CLK_3_HZ 19200000 +#define CHT_CODEC_DAI "rt5645-aif1" + +struct cht_mc_private { + struct snd_soc_jack hp_jack; + struct snd_soc_jack mic_jack; +}; + +static inline struct snd_soc_dai *cht_get_codec_dai(struct snd_soc_card *card) +{ + int i; + + for (i = 0; i < card->num_rtd; i++) { + struct snd_soc_pcm_runtime *rtd; + + rtd = card->rtd + i; + if (!strncmp(rtd->codec_dai->name, CHT_CODEC_DAI, + strlen(CHT_CODEC_DAI))) + return rtd->codec_dai; + } + return NULL; +} + +static int platform_clock_control(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_soc_card *card = dapm->card; + struct snd_soc_dai *codec_dai; + int ret; + + codec_dai = cht_get_codec_dai(card); + if (!codec_dai) { + dev_err(card->dev, "Codec dai not found; Unable to set platform clock\n"); + return -EIO; + } + + if (!SND_SOC_DAPM_EVENT_OFF(event)) + return 0; + + /* Set codec sysclk source to its internal clock because codec PLL will + * be off when idle and MCLK will also be off by ACPI when codec is + * runtime suspended. Codec needs clock for jack detection and button + * press. + */ + ret = snd_soc_dai_set_sysclk(codec_dai, RT5645_SCLK_S_RCCLK, + 0, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(card->dev, "can't set codec sysclk: %d\n", ret); + return ret; + } + + return 0; +} + +static const struct snd_soc_dapm_widget cht_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Int Mic", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), + SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, + platform_clock_control, SND_SOC_DAPM_POST_PMD), +}; + +static const struct snd_soc_dapm_route cht_audio_map[] = { + {"IN1P", NULL, "Headset Mic"}, + {"IN1N", NULL, "Headset Mic"}, + {"DMIC L1", NULL, "Int Mic"}, + {"DMIC R1", NULL, "Int Mic"}, + {"Headphone", NULL, "HPOL"}, + {"Headphone", NULL, "HPOR"}, + {"Ext Spk", NULL, "SPOL"}, + {"Ext Spk", NULL, "SPOR"}, + {"AIF1 Playback", NULL, "ssp2 Tx"}, + {"ssp2 Tx", NULL, "codec_out0"}, + {"ssp2 Tx", NULL, "codec_out1"}, + {"codec_in0", NULL, "ssp2 Rx" }, + {"codec_in1", NULL, "ssp2 Rx" }, + {"ssp2 Rx", NULL, "AIF1 Capture"}, + {"Headphone", NULL, "Platform Clock"}, + {"Headset Mic", NULL, "Platform Clock"}, + {"Int Mic", NULL, "Platform Clock"}, + {"Ext Spk", NULL, "Platform Clock"}, +}; + +static const struct snd_kcontrol_new cht_mc_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Int Mic"), + SOC_DAPM_PIN_SWITCH("Ext Spk"), +}; + +static int cht_aif1_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + /* set codec PLL source to the 19.2MHz platform clock (MCLK) */ + ret = snd_soc_dai_set_pll(codec_dai, 0, RT5645_PLL1_S_MCLK, + CHT_PLAT_CLK_3_HZ, params_rate(params) * 512); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec pll: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, RT5645_SCLK_S_PLL1, + params_rate(params) * 512, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec sysclk: %d\n", ret); + return ret; + } + + return 0; +} + +static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) +{ + int ret; + struct snd_soc_codec *codec = runtime->codec; + struct snd_soc_dai *codec_dai = runtime->codec_dai; + struct cht_mc_private *ctx = snd_soc_card_get_drvdata(runtime->card); + + /* Select clk_i2s1_asrc as ASRC clock source */ + rt5645_sel_asrc_clk_src(codec, + RT5645_DA_STEREO_FILTER | + RT5645_DA_MONO_L_FILTER | + RT5645_DA_MONO_R_FILTER | + RT5645_AD_STEREO_FILTER, + RT5645_CLK_SEL_I2S1_ASRC); + + /* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */ + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xF, 0xF, 4, 24); + if (ret < 0) { + dev_err(runtime->dev, "can't set codec TDM slot %d\n", ret); + return ret; + } + + ret = snd_soc_jack_new(codec, "Headphone Jack", + SND_JACK_HEADPHONE, + &ctx->hp_jack); + if (ret) { + dev_err(runtime->dev, "HP jack creation failed %d\n", ret); + return ret; + } + + ret = snd_soc_jack_new(codec, "Mic Jack", + SND_JACK_MICROPHONE, + &ctx->mic_jack); + if (ret) { + dev_err(runtime->dev, "Mic jack creation failed %d\n", ret); + return ret; + } + + rt5645_set_jack_detect(codec, &ctx->hp_jack, &ctx->mic_jack); + + return ret; +} + +static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + + /* The DSP will covert the FE rate to 48k, stereo, 24bits */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + /* set SSP2 to 24-bit */ + snd_mask_set(¶ms->masks[SNDRV_PCM_HW_PARAM_FORMAT - + SNDRV_PCM_HW_PARAM_FIRST_MASK], + SNDRV_PCM_FORMAT_S24_LE); + return 0; +} + +static unsigned int rates_48000[] = { + 48000, +}; + +static struct snd_pcm_hw_constraint_list constraints_48000 = { + .count = ARRAY_SIZE(rates_48000), + .list = rates_48000, +}; + +static int cht_aif1_startup(struct snd_pcm_substream *substream) +{ + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_48000); +} + +static struct snd_soc_ops cht_aif1_ops = { + .startup = cht_aif1_startup, +}; + +static struct snd_soc_ops cht_be_ssp2_ops = { + .hw_params = cht_aif1_hw_params, +}; + +static struct snd_soc_dai_link cht_dailink[] = { + [MERR_DPCM_AUDIO] = { + .name = "Audio Port", + .stream_name = "Audio", + .cpu_dai_name = "media-cpu-dai", + .codec_dai_name = "snd-soc-dummy-dai", + .codec_name = "snd-soc-dummy", + .platform_name = "sst-mfld-platform", + .ignore_suspend = 1, + .dynamic = 1, + .dpcm_playback = 1, + .dpcm_capture = 1, + .ops = &cht_aif1_ops, + }, + [MERR_DPCM_COMPR] = { + .name = "Compressed Port", + .stream_name = "Compress", + .cpu_dai_name = "compress-cpu-dai", + .codec_dai_name = "snd-soc-dummy-dai", + .codec_name = "snd-soc-dummy", + .platform_name = "sst-mfld-platform", + }, + /* CODEC<->CODEC link */ + /* back ends */ + { + .name = "SSP2-Codec", + .be_id = 1, + .cpu_dai_name = "ssp2-port", + .platform_name = "sst-mfld-platform", + .no_pcm = 1, + .codec_dai_name = "rt5645-aif1", + .codec_name = "i2c-10EC5645:00", + .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF + | SND_SOC_DAIFMT_CBS_CFS, + .init = cht_codec_init, + .be_hw_params_fixup = cht_codec_fixup, + .ignore_suspend = 1, + .dpcm_playback = 1, + .dpcm_capture = 1, + .ops = &cht_be_ssp2_ops, + }, +}; + +/* SoC card */ +static struct snd_soc_card snd_soc_card_cht = { + .name = "chtrt5645", + .dai_link = cht_dailink, + .num_links = ARRAY_SIZE(cht_dailink), + .dapm_widgets = cht_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cht_dapm_widgets), + .dapm_routes = cht_audio_map, + .num_dapm_routes = ARRAY_SIZE(cht_audio_map), + .controls = cht_mc_controls, + .num_controls = ARRAY_SIZE(cht_mc_controls), +}; + +static int snd_cht_mc_probe(struct platform_device *pdev) +{ + int ret_val = 0; + struct cht_mc_private *drv; + + drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_ATOMIC); + if (!drv) + return -ENOMEM; + + snd_soc_card_cht.dev = &pdev->dev; + snd_soc_card_set_drvdata(&snd_soc_card_cht, drv); + ret_val = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_cht); + if (ret_val) { + dev_err(&pdev->dev, + "snd_soc_register_card failed %d\n", ret_val); + return ret_val; + } + platform_set_drvdata(pdev, &snd_soc_card_cht); + return ret_val; +} + +static struct platform_driver snd_cht_mc_driver = { + .driver = { + .name = "cht-bsw-rt5645", + .pm = &snd_soc_pm_ops, + }, + .probe = snd_cht_mc_probe, +}; + +module_platform_driver(snd_cht_mc_driver) + +MODULE_DESCRIPTION("ASoC Intel(R) Braswell Machine driver"); +MODULE_AUTHOR("Fang, Yang A,N,Harshapriya"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:cht-bsw-rt5645"); diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index a282179a3064..0ab1309ef274 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -338,12 +338,6 @@ static inline u32 msg_get_stage_type(u32 msg) return (msg & IPC_STG_TYPE_MASK) >> IPC_STG_TYPE_SHIFT; } -static inline u32 msg_set_stage_type(u32 msg, u32 type) -{ - return (msg & ~IPC_STG_TYPE_MASK) + - (type << IPC_STG_TYPE_SHIFT); -} - static inline u32 msg_get_stream_id(u32 msg) { return (msg & IPC_STR_ID_MASK) >> IPC_STR_ID_SHIFT; @@ -970,45 +964,6 @@ int sst_hsw_fw_get_version(struct sst_hsw *hsw, } /* Mixer Controls */ -int sst_hsw_stream_mute(struct sst_hsw *hsw, struct sst_hsw_stream *stream, - u32 stage_id, u32 channel) -{ - int ret; - - ret = sst_hsw_stream_get_volume(hsw, stream, stage_id, channel, - &stream->mute_volume[channel]); - if (ret < 0) - return ret; - - ret = sst_hsw_stream_set_volume(hsw, stream, stage_id, channel, 0); - if (ret < 0) { - dev_err(hsw->dev, "error: can't unmute stream %d channel %d\n", - stream->reply.stream_hw_id, channel); - return ret; - } - - stream->mute[channel] = 1; - return 0; -} - -int sst_hsw_stream_unmute(struct sst_hsw *hsw, struct sst_hsw_stream *stream, - u32 stage_id, u32 channel) - -{ - int ret; - - stream->mute[channel] = 0; - ret = sst_hsw_stream_set_volume(hsw, stream, stage_id, channel, - stream->mute_volume[channel]); - if (ret < 0) { - dev_err(hsw->dev, "error: can't unmute stream %d channel %d\n", - stream->reply.stream_hw_id, channel); - return ret; - } - - return 0; -} - int sst_hsw_stream_get_volume(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 stage_id, u32 channel, u32 *volume) { @@ -1022,17 +977,6 @@ int sst_hsw_stream_get_volume(struct sst_hsw *hsw, struct sst_hsw_stream *stream return 0; } -int sst_hsw_stream_set_volume_curve(struct sst_hsw *hsw, - struct sst_hsw_stream *stream, u64 curve_duration, - enum sst_hsw_volume_curve curve) -{ - /* curve duration in steps of 100ns */ - stream->vol_req.curve_duration = curve_duration; - stream->vol_req.curve_type = curve; |