diff options
Diffstat (limited to 'sound/soc')
27 files changed, 1033 insertions, 425 deletions
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 8349f982a586..6ecac1e4428e 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/rt5645.c b/sound/soc/codecs/rt5645.c index 068dfd5c7640..c9a4c5be083b 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -31,6 +31,7 @@ #include "rt5645.h" #define RT5645_DEVICE_ID 0x6308 +#define RT5650_DEVICE_ID 0x6419 #define RT5645_PR_RANGE_BASE (0xff + 1) #define RT5645_PR_SPACING 0x100 @@ -59,6 +60,10 @@ static const struct reg_default init_list[] = { }; #define RT5645_INIT_REG_LEN ARRAY_SIZE(init_list) +static const struct reg_default rt5650_init_list[] = { + {0xf6, 0x0100}, +}; + static const struct reg_default rt5645_reg[] = { { 0x00, 0x0000 }, { 0x01, 0xc8c8 }, @@ -86,6 +91,7 @@ static const struct reg_default rt5645_reg[] = { { 0x2a, 0x5656 }, { 0x2b, 0x5454 }, { 0x2c, 0xaaa0 }, + { 0x2d, 0x0000 }, { 0x2f, 0x1002 }, { 0x31, 0x5000 }, { 0x32, 0x0000 }, @@ -193,6 +199,8 @@ static const struct reg_default rt5645_reg[] = { { 0xdb, 0x0003 }, { 0xdc, 0x0049 }, { 0xdd, 0x001b }, + { 0xdf, 0x0008 }, + { 0xe0, 0x4000 }, { 0xe6, 0x8000 }, { 0xe7, 0x0200 }, { 0xec, 0xb300 }, @@ -242,6 +250,7 @@ static bool rt5645_volatile_register(struct device *dev, unsigned int reg) case RT5645_IRQ_CTRL3: case RT5645_INT_IRQ_ST: case RT5645_IL_CMD: + case RT5650_4BTN_IL_CMD1: case RT5645_VENDOR_ID: case RT5645_VENDOR_ID1: case RT5645_VENDOR_ID2: @@ -287,6 +296,7 @@ static bool rt5645_readable_register(struct device *dev, unsigned int reg) case RT5645_STO_DAC_MIXER: case RT5645_MONO_DAC_MIXER: case RT5645_DIG_MIXER: + case RT5650_A_DAC_SOUR: case RT5645_DIG_INF1_DATA: case RT5645_PDM_OUT_CTRL: case RT5645_REC_L1_MIXER: @@ -378,6 +388,8 @@ static bool rt5645_readable_register(struct device *dev, unsigned int reg) case RT5645_IL_CMD: case RT5645_IL_CMD2: case RT5645_IL_CMD3: + case RT5650_4BTN_IL_CMD1: + case RT5650_4BTN_IL_CMD2: case RT5645_DRC1_HL_CTRL1: case RT5645_DRC2_HL_CTRL1: case RT5645_ADC_MONO_HP_CTRL1: @@ -603,6 +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, @@ -1009,6 +1102,44 @@ static SOC_ENUM_SINGLE_DECL( static const struct snd_kcontrol_new rt5645_if1_adc_in_mux = SOC_DAPM_ENUM("IF1 ADC IN source", rt5645_if1_adc_in_enum); +/* MX-2d [3] [2] */ +static const char * const rt5650_a_dac1_src[] = { + "DAC1", "Stereo DAC Mixer" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5650_a_dac1_l_enum, RT5650_A_DAC_SOUR, + RT5650_A_DAC1_L_IN_SFT, rt5650_a_dac1_src); + +static const struct snd_kcontrol_new rt5650_a_dac1_l_mux = + SOC_DAPM_ENUM("A DAC1 L source", rt5650_a_dac1_l_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5650_a_dac1_r_enum, RT5650_A_DAC_SOUR, + RT5650_A_DAC1_R_IN_SFT, rt5650_a_dac1_src); + +static const struct snd_kcontrol_new rt5650_a_dac1_r_mux = + SOC_DAPM_ENUM("A DAC1 R source", rt5650_a_dac1_r_enum); + +/* MX-2d [1] [0] */ +static const char * const rt5650_a_dac2_src[] = { + "Stereo DAC Mixer", "Mono DAC Mixer" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5650_a_dac2_l_enum, RT5650_A_DAC_SOUR, + RT5650_A_DAC2_L_IN_SFT, rt5650_a_dac2_src); + +static const struct snd_kcontrol_new rt5650_a_dac2_l_mux = + SOC_DAPM_ENUM("A DAC2 L source", rt5650_a_dac2_l_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5650_a_dac2_r_enum, RT5650_A_DAC_SOUR, + RT5650_A_DAC2_R_IN_SFT, rt5650_a_dac2_src); + +static const struct snd_kcontrol_new rt5650_a_dac2_r_mux = + SOC_DAPM_ENUM("A DAC2 R source", rt5650_a_dac2_r_enum); + /* MX-2F [13:12] */ static const char * const rt5645_if2_adc_in_src[] = { "IF_ADC1", "IF_ADC2", "VAD_ADC" @@ -1153,11 +1284,16 @@ static int rt5645_hp_event(struct snd_soc_dapm_widget *w, case SND_SOC_DAPM_POST_PMU: hp_amp_power(codec, 1); /* headphone unmute sequence */ - snd_soc_update_bits(codec, RT5645_DEPOP_M3, RT5645_CP_FQ1_MASK | - RT5645_CP_FQ2_MASK | RT5645_CP_FQ3_MASK, - (RT5645_CP_FQ_192_KHZ << RT5645_CP_FQ1_SFT) | - (RT5645_CP_FQ_12_KHZ << RT5645_CP_FQ2_SFT) | - (RT5645_CP_FQ_192_KHZ << RT5645_CP_FQ3_SFT)); + if (rt5645->codec_type == CODEC_TYPE_RT5650) { + snd_soc_write(codec, RT5645_DEPOP_M3, 0x0737); + } else { + snd_soc_update_bits(codec, RT5645_DEPOP_M3, + RT5645_CP_FQ1_MASK | RT5645_CP_FQ2_MASK | + RT5645_CP_FQ3_MASK, + (RT5645_CP_FQ_192_KHZ << RT5645_CP_FQ1_SFT) | + (RT5645_CP_FQ_12_KHZ << RT5645_CP_FQ2_SFT) | + (RT5645_CP_FQ_192_KHZ << RT5645_CP_FQ3_SFT)); + } regmap_write(rt5645->regmap, RT5645_PR_BASE + RT5645_MAMP_INT_REG2, 0xfc00); snd_soc_update_bits(codec, RT5645_DEPOP_M1, @@ -1177,12 +1313,16 @@ static int rt5645_hp_event(struct snd_soc_dapm_widget *w, case SND_SOC_DAPM_PRE_PMD: /* headphone mute sequence */ - snd_soc_update_bits(codec, RT5645_DEPOP_M3, - RT5645_CP_FQ1_MASK | RT5645_CP_FQ2_MASK | - RT5645_CP_FQ3_MASK, - (RT5645_CP_FQ_96_KHZ << RT5645_CP_FQ1_SFT) | - (RT5645_CP_FQ_12_KHZ << RT5645_CP_FQ2_SFT) | - (RT5645_CP_FQ_96_KHZ << RT5645_CP_FQ3_SFT)); + if (rt5645->codec_type == CODEC_TYPE_RT5650) { + snd_soc_write(codec, RT5645_DEPOP_M3, 0x0737); + } else { + snd_soc_update_bits(codec, RT5645_DEPOP_M3, + RT5645_CP_FQ1_MASK | RT5645_CP_FQ2_MASK | + RT5645_CP_FQ3_MASK, + (RT5645_CP_FQ_96_KHZ << RT5645_CP_FQ1_SFT) | + (RT5645_CP_FQ_12_KHZ << RT5645_CP_FQ2_SFT) | + (RT5645_CP_FQ_96_KHZ << RT5645_CP_FQ3_SFT)); + } regmap_write(rt5645->regmap, RT5645_PR_BASE + RT5645_MAMP_INT_REG2, 0xfc00); snd_soc_update_bits(codec, RT5645_DEPOP_M1, @@ -1576,6 +1716,17 @@ static const struct snd_soc_dapm_widget rt5645_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("SPOR"), }; +static const struct snd_soc_dapm_widget rt5650_specific_dapm_widgets[] = { + SND_SOC_DAPM_MUX("A DAC1 L Mux", SND_SOC_NOPM, + 0, 0, &rt5650_a_dac1_l_mux), + SND_SOC_DAPM_MUX("A DAC1 R Mux", SND_SOC_NOPM, + 0, 0, &rt5650_a_dac1_r_mux), + SND_SOC_DAPM_MUX("A DAC2 L Mux", SND_SOC_NOPM, + 0, 0, &rt5650_a_dac2_l_mux), + SND_SOC_DAPM_MUX("A DAC2 R Mux", SND_SOC_NOPM, + 0, 0, &rt5650_a_dac2_r_mux), +}; + static const struct snd_soc_dapm_route rt5645_dapm_routes[] = { { "adc stereo1 filter", NULL, "ADC STO1 ASRC", is_using_asrc }, { "adc stereo2 filter", NULL, "ADC STO2 ASRC", is_using_asrc }, @@ -1781,13 +1932,9 @@ static const struct snd_soc_dapm_route rt5645_dapm_routes[] = { { "DAC MIXR", "DAC R2 Switch", "DAC R2 Volume" }, { "DAC MIXR", "DAC L2 Switch", "DAC L2 Volume" }, - { "DAC L1", NULL, "Stereo DAC MIXL" }, { "DAC L1", NULL, "PLL1", is_sys_clk_from_pll }, - { "DAC R1", NULL, "Stereo DAC MIXR" }, { "DAC R1", NULL, "PLL1", is_sys_clk_from_pll }, - { "DAC L2", NULL, "Mono DAC MIXL" }, { "DAC L2", NULL, "PLL1", is_sys_clk_from_pll }, - { "DAC R2", NULL, "Mono DAC MIXR" }, { "DAC R2", NULL, "PLL1", is_sys_clk_from_pll }, { "SPK MIXL", "BST1 Switch", "BST1" }, @@ -1876,6 +2023,30 @@ static const struct snd_soc_dapm_route rt5645_dapm_routes[] = { { "SPOR", NULL, "SPK amp" }, }; +static const struct snd_soc_dapm_route rt5650_specific_dapm_routes[] = { + { "A DAC1 L Mux", "DAC1", "DAC1 MIXL"}, + { "A DAC1 L Mux", "Stereo DAC Mixer", "Stereo DAC MIXL"}, + { "A DAC1 R Mux", "DAC1", "DAC1 MIXR"}, + { "A DAC1 R Mux", "Stereo DAC Mixer", "Stereo DAC MIXR"}, + + { "A DAC2 L Mux", "Stereo DAC Mixer", "Stereo DAC MIXL"}, + { "A DAC2 L Mux", "Mono DAC Mixer", "Mono DAC MIXL"}, + { "A DAC2 R Mux", "Stereo DAC Mixer", "Stereo DAC MIXR"}, + { "A DAC2 R Mux", "Mono DAC Mixer", "Mono DAC MIXR"}, + + { "DAC L1", NULL, "A DAC1 L Mux" }, + { "DAC R1", NULL, "A DAC1 R Mux" }, + { "DAC L2", NULL, "A DAC2 L Mux" }, + { "DAC R2", NULL, "A DAC2 R Mux" }, +}; + +static const struct snd_soc_dapm_route rt5645_specific_dapm_routes[] = { + { "DAC L1", NULL, "Stereo DAC MIXL" }, + { "DAC R1", NULL, "Stereo DAC MIXR" }, + { "DAC L2", NULL, "Mono DAC MIXL" }, + { "DAC R2", NULL, "Mono DAC MIXR" }, +}; + static int rt5645_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { @@ -2295,6 +2466,22 @@ static int rt5645_probe(struct snd_soc_codec *codec) rt5645->codec = codec; + switch (rt5645->codec_type) { + case CODEC_TYPE_RT5645: + snd_soc_dapm_add_routes(&codec->dapm, + rt5645_specific_dapm_routes, + ARRAY_SIZE(rt5645_specific_dapm_routes)); + break; + case CODEC_TYPE_RT5650: + snd_soc_dapm_new_controls(&codec->dapm, + rt5650_specific_dapm_widgets, + ARRAY_SIZE(rt5650_specific_dapm_widgets)); + snd_soc_dapm_add_routes(&codec->dapm, + rt5650_specific_dapm_routes, + ARRAY_SIZE(rt5650_specific_dapm_routes)); + break; + } + rt5645_set_bias_level(codec, SND_SOC_BIAS_OFF); snd_soc_update_bits(codec, RT5645_CHARGE_PUMP, 0x0300, 0x0200); @@ -2426,6 +2613,7 @@ static const struct regmap_config rt5645_regmap = { static const struct i2c_device_id rt5645_i2c_id[] = { { "rt5645", 0 }, + { "rt5650", 0 }, { } }; MODULE_DEVICE_TABLE(i2c, rt5645_i2c_id); @@ -2458,9 +2646,18 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, } regmap_read(rt5645->regmap, RT5645_VENDOR_ID2, &val); - if (val != RT5645_DEVICE_ID) { + + switch (val) { + case RT5645_DEVICE_ID: + rt5645->codec_type = CODEC_TYPE_RT5645; + break; + case RT5650_DEVICE_ID: + rt5645->codec_type = CODEC_TYPE_RT5650; + break; + default: dev_err(&i2c->dev, - "Device with ID register %x is not rt5645\n", val); + "Device with ID register %x is not rt5645 or rt5650\n", + val); return -ENODEV; } @@ -2471,6 +2668,14 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, if (ret != 0) dev_warn(&i2c->dev, "Failed to apply regmap patch: %d\n", ret); + if (rt5645->codec_type == CODEC_TYPE_RT5650) { + ret = regmap_register_patch(rt5645->regmap, rt5650_init_list, + ARRAY_SIZE(rt5650_init_list)); + if (ret != 0) + dev_warn(&i2c->dev, "Apply rt5650 patch failed: %d\n", + ret); + } + if (rt5645->pdata.in2_diff) regmap_update_bits(rt5645->regmap, RT5645_IN2_CTRL, RT5645_IN_DF2, RT5645_IN_DF2); diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h index a815e36a2bdb..dbfd98c22f4d 100644 --- a/sound/soc/codecs/rt5645.h +++ b/sound/soc/codecs/rt5645.h @@ -47,6 +47,7 @@ #define RT5645_STO_DAC_MIXER 0x2a #define RT5645_MONO_DAC_MIXER 0x2b #define RT5645_DIG_MIXER 0x2c +#define RT5650_A_DAC_SOUR 0x2d #define RT5645_DIG_INF1_DATA 0x2f /* Mixer - PDM */ #define RT5645_PDM_OUT_CTRL 0x31 @@ -150,6 +151,8 @@ #define RT5645_IL_CMD 0xdb #define RT5645_IL_CMD2 0xdc #define RT5645_IL_CMD3 0xdd +#define RT5650_4BTN_IL_CMD1 0xdf +#define RT5650_4BTN_IL_CMD2 0xe0 #define RT5645_DRC1_HL_CTRL1 0xe7 #define RT5645_DRC2_HL_CTRL1 0xe9 #define RT5645_MUTI_DRC_CTRL1 0xea @@ -472,6 +475,12 @@ #define RT5645_DAC_L2_DAC_R_VOL_MASK (0x1 << 4) #define RT5645_DAC_L2_DAC_R_VOL_SFT 4 +/* Analog DAC1/2 Input Source Control (0x2d) */ +#define RT5650_A_DAC1_L_IN_SFT 3 +#define RT5650_A_DAC1_R_IN_SFT 2 +#define RT5650_A_DAC2_L_IN_SFT 1 +#define RT5650_A_DAC2_R_IN_SFT 0 + /* Digital Interface Data Control (0x2f) */ #define RT5645_IF1_ADC2_IN_SEL (0x1 << 15) #define RT5645_IF1_ADC2_IN_SFT 15 @@ -1111,50 +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) @@ -2175,6 +2161,24 @@ enum { RT5645_DMIC_DATA_GPIO11, }; +enum { + CODEC_TYPE_RT5645, + 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; @@ -2184,6 +2188,7 @@ struct rt5645_priv { struct snd_soc_jack *mic_jack; struct delayed_work jack_detect_work; + int codec_type; int sysclk; int sysclk_src; int lrck[RT5645_AIFS]; diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index f5b054de481e..7b3d6b5992f1 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -592,6 +592,89 @@ static int can_use_asrc(struct snd_soc_dapm_widget *source, return 0; } + +/** + * rt5670_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 RT5670 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 rt5670_sel_asrc_clk_src(struct snd_soc_codec *codec, + unsigned int filter_mask, unsigned int clk_src) +{ + unsigned int asrc2_mask = 0, asrc2_value = 0; + unsigned int asrc3_mask = 0, asrc3_value = 0; + + if (clk_src > RT5670_CLK_SEL_SYS3) + return -EINVAL; + + if (filter_mask & RT5670_DA_STEREO_FILTER) { + asrc2_mask |= RT5670_DA_STO_CLK_SEL_MASK; + asrc2_value = (asrc2_value & ~RT5670_DA_STO_CLK_SEL_MASK) + | (clk_src << RT5670_DA_STO_CLK_SEL_SFT); + } + + if (filter_mask & RT5670_DA_MONO_L_FILTER) { + asrc2_mask |= RT5670_DA_MONOL_CLK_SEL_MASK; + asrc2_value = (asrc2_value & ~RT5670_DA_MONOL_CLK_SEL_MASK) + | (clk_src << RT5670_DA_MONOL_CLK_SEL_SFT); + } + + if (filter_mask & RT5670_DA_MONO_R_FILTER) { + asrc2_mask |= RT5670_DA_MONOR_CLK_SEL_MASK; + asrc2_value = (asrc2_value & ~RT5670_DA_MONOR_CLK_SEL_MASK) + | (clk_src << RT5670_DA_MONOR_CLK_SEL_SFT); + } + + if (filter_mask & RT5670_AD_STEREO_FILTER) { + asrc2_mask |= RT5670_AD_STO1_CLK_SEL_MASK; + asrc2_value = (asrc2_value & ~RT5670_AD_STO1_CLK_SEL_MASK) + | (clk_src << RT5670_AD_STO1_CLK_SEL_SFT); + } + + if (filter_mask & RT5670_AD_MONO_L_FILTER) { + asrc3_mask |= RT5670_AD_MONOL_CLK_SEL_MASK; + asrc3_value = (asrc3_value & ~RT5670_AD_MONOL_CLK_SEL_MASK) + | (clk_src << RT5670_AD_MONOL_CLK_SEL_SFT); + } + + if (filter_mask & RT5670_AD_MONO_R_FILTER) { + asrc3_mask |= RT5670_AD_MONOR_CLK_SEL_MASK; + asrc3_value = (asrc3_value & ~RT5670_AD_MONOR_CLK_SEL_MASK) + | (clk_src << RT5670_AD_MONOR_CLK_SEL_SFT); + } + + if (filter_mask & RT5670_UP_RATE_FILTER) { + asrc3_mask |= RT5670_UP_CLK_SEL_MASK; + asrc3_value = (asrc3_value & ~RT5670_UP_CLK_SEL_MASK) + | (clk_src << RT5670_UP_CLK_SEL_SFT); + } + + if (filter_mask & RT5670_DOWN_RATE_FILTER) { + asrc3_mask |= RT5670_DOWN_CLK_SEL_MASK; + asrc3_value = (asrc3_value & ~RT5670_DOWN_CLK_SEL_MASK) + | (clk_src << RT5670_DOWN_CLK_SEL_SFT); + } + + if (asrc2_mask) + snd_soc_update_bits(codec, RT5670_ASRC_2, + asrc2_mask, asrc2_value); + + if (asrc3_mask) + snd_soc_update_bits(codec, RT5670_ASRC_3, + asrc3_mask, asrc3_value); + return 0; +} +EXPORT_SYMBOL_GPL(rt5670_sel_asrc_clk_src); + /* Digital Mixer */ static const struct snd_kcontrol_new rt5670_sto1_adc_l_mix[] = { SOC_DAPM_SINGLE("ADC1 Switch", RT5670_STO1_ADC_MIXER, diff --git a/sound/soc/codecs/rt5670.h b/sound/soc/codecs/rt5670.h index d11b9c207e26..21f8e18c13c4 100644 --- a/sound/soc/codecs/rt5670.h +++ b/sound/soc/codecs/rt5670.h @@ -1023,50 +1023,33 @@ #define RT5670_DMIC_2_M_NOR (0x0 << 8) #define RT5670_DMIC_2_M_ASYN (0x1 << 8) +/* ASRC clock source selection (0x84, 0x85) */ +#define RT5670_CLK_SEL_SYS (0x0) +#define RT5670_CLK_SEL_I2S1_ASRC (0x1) +#define RT5670_CLK_SEL_I2S2_ASRC (0x2) +#define RT5670_CLK_SEL_I2S3_ASRC (0x3) +#define RT5670_CLK_SEL_SYS2 (0x5) +#define RT5670_CLK_SEL_SYS3 (0x6) + /* ASRC Control 2 (0x84) */ -#define RT5670_MDA_L_M_MASK (0x1 << 15) -#define RT5670_MDA_L_M_SFT 15 -#define RT5670_MDA_L_M_NOR (0x0 << 15) -#define RT5670_MDA_L_M_ASYN (0x1 << 15) -#define RT5670_MDA_R_M_MASK (0x1 << 14) -#define RT5670_MDA_R_M_SFT 14 -#define RT5670_MDA_R_M_NOR (0x0 << 14) -#define RT5670_MDA_R_M_ASYN (0x1 << 14) -#define RT5670_MAD_L_M_MASK (0x1 << 13) -#define RT5670_MAD_L_M_SFT 13 -#define RT5670_MAD_L_M_NOR (0x0 << 13) -#define RT5670_MAD_L_M_ASYN (0x1 << 13) -#define RT5670_MAD_R_M_MASK (0x1 << 12) -#define RT5670_MAD_R_M_SFT 12 -#define RT5670_MAD_R_M_NOR (0x0 << 12) -#define RT5670_MAD_R_M_ASYN (0x1 << 12) -#define RT5670_ADC_M_MASK (0x1 << 11) -#define RT5670_ADC_M_SFT 11 -#define RT5670_ADC_M_NOR (0x0 << 11) -#define RT5670_ADC_M_ASYN (0x1 << 11) -#define RT5670_STO_DAC_M_MASK (0x1 << 5) -#define RT5670_STO_DAC_M_SFT 5 -#define RT5670_STO_DAC_M_NOR (0x0 << 5) -#define RT5670_STO_DAC_M_ASYN (0x1 << 5) -#define RT5670_I2S1_R_D_MASK (0x1 << 4) -#define RT5670_I2S1_R_D_SFT 4 -#define RT5670_I2S1_R_D_DIS (0x0 << 4) -#define RT5670_I2S1_R_D_EN (0x1 << 4) -#define RT5670_I2S2_R_D_MASK (0x1 << 3) -#define RT5670_I2S2_R_D_SFT 3 -#define RT5670_I2S2_R_D_DIS (0x0 << 3) -#define RT5670_I2S2_R_D_EN (0x1 << 3) -#define RT5670_PRE_SCLK_MASK (0x3) -#define RT5670_PRE_SCLK_SFT 0 -#define RT5670_PRE_SCLK_512 (0x0) -#define RT5670_PRE_SCLK_1024 (0x1) -#define RT5670_PRE_SCLK_2048 (0x2) +#define RT5670_DA_STO_CLK_SEL_MASK (0xf << 12) +#define RT5670_DA_STO_CLK_SEL_SFT 12 +#define RT5670_DA_MONOL_CLK_SEL_MASK (0xf << 8) +#define RT5670_DA_MONOL_CLK_SEL_SFT 8 +#define RT5670_DA_MONOR_CLK_SEL_MASK (0xf << 4) +#define RT5670_DA_MONOR_CLK_SEL_SFT 4 +#define RT5670_AD_STO1_CLK_SEL_MASK (0xf << 0) +#define RT5670_AD_STO1_CLK_SEL_SFT 0 /* ASRC Control 3 (0x85) */ -#define RT5670_I2S1_RATE_MASK (0xf << 12) -#define RT5670_I2S1_RATE_SFT 12 -#define RT5670_I2S2_RATE_MASK (0xf << 8) -#define RT5670_I2S2_RATE_SFT 8 +#define RT5670_UP_CLK_SEL_MASK (0xf << 12) +#define RT5670_UP_CLK_SEL_SFT 12 +#define RT5670_DOWN_CLK_SEL_MASK (0xf << 8) +#define RT5670_DOWN_CLK_SEL_SFT 8 +#define RT5670_AD_MONOL_CLK_SEL_MASK (0xf << 4) +#define RT5670_AD_MONOL_CLK_SEL_SFT 4 +#define RT5670_AD_MONOR_CLK_SEL_MASK (0xf << 0) +#define RT5670_AD_MONOR_CLK_SEL_SFT 0 /* ASRC Control 4 (0x89) */ #define RT5670_I2S1_PD_MASK (0x7 << 12) @@ -1983,6 +1966,21 @@ enum { RT5670_DMIC_DATA_GPIO5, }; +/* filter mask */ +enum { + RT5670_DA_STEREO_FILTER = 0x1, + RT5670_DA_MONO_L_FILTER = (0x1 << 1), + RT5670_DA_MONO_R_FILTER = (0x1 << 2), + RT5670_AD_STEREO_FILTER = (0x1 << 3), + RT5670_AD_MONO_L_FILTER = (0x1 << 4), + RT5670_AD_MONO_R_FILTER = (0x1 << 5), + RT5670_UP_RATE_FILTER = (0x1 << 6), + RT5670_DOWN_RATE_FILTER = (0x1 << 7), +}; + +int rt5670_sel_asrc_clk_src(struct snd_soc_codec *codec, + unsigned int filter_mask, unsigned int clk_src); + struct rt5670_priv { struct snd_soc_codec *codec; struct rt5670_platform_data pdata; diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 735e2eec52f7..75870c0ea2c9 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -90,7 +90,6 @@ struct spdif_mixer_control { * @sysclk: system clock for rx clock rate measurement * @dma_params_tx: DMA parameters for transmit channel * @dma_params_rx: DMA parameters for receive channel - * @name: driver name */ struct fsl_spdif_priv { struct spdif_mixer_control fsl_spdif_control; @@ -109,12 +108,8 @@ struct fsl_spdif_priv { struct clk *sysclk; struct snd_dmaengine_dai_dma_data dma_params_tx; struct snd_dmaengine_dai_dma_data dma_params_rx; - - /* The name space will be allocated dynamically */ - char name[0]; }; - /* DPLL locked and lock loss interrupt handler */ static void spdif_irq_dpll_lock(struct fsl_spdif_priv *spdif_priv) { @@ -1169,19 +1164,15 @@ static int fsl_spdif_probe(struct platform_device *pdev) if (!np) return -ENODEV; - spdif_priv = devm_kzalloc(&pdev->dev, - sizeof(struct fsl_spdif_priv) + strlen(np->name) + 1, - GFP_KERNEL); + spdif_priv = devm_kzalloc(&pdev->dev, sizeof(*spdif_priv), GFP_KERNEL); if (!spdif_priv) return -ENOMEM; - strcpy(spdif_priv->name, np->name); - spdif_priv->pdev = pdev; /* Initialize this copy of the CPU DAI driver structure */ memcpy(&spdif_priv->cpu_dai_drv, &fsl_spdif_dai, sizeof(fsl_spdif_dai)); - spdif_priv->cpu_dai_drv.name = spdif_priv->name; + spdif_priv->cpu_dai_drv.name = dev_name(&pdev->dev); /* Get the addresses and IRQ */ res = platform_get_resource(pdev, IORESOURCE_MEM, 0); @@ -1203,7 +1194,7 @@ static int fsl_spdif_probe(struct platform_device *pdev) } ret = devm_request_irq(&pdev->dev, irq, spdif_isr, 0, - spdif_priv->name, spdif_priv); + dev_name(&pdev->dev), spdif_priv); if (ret) { dev_err(&pdev->dev, "could not claim irq %u\n", irq); return ret; diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 65400bef013c..46549de60e50 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -160,7 +160,7 @@ struct fsl_ssi_soc_data { */ struct fsl_ssi_private { struct regmap *regs; - unsigned int irq; + int irq; struct snd_soc_dai_driver cpu_dai_drv; unsigned int dai_fmt; diff --git a/sound/soc/fsl/imx-spdif.c b/sound/soc/fsl/imx-spdif.c index e94704f1b9ee..33da26a12457 100644 --- a/sound/soc/fsl/imx-spdif.c +++ b/sound/soc/fsl/imx-spdif.c @@ -60,6 +60,7 @@ static int imx_spdif_audio_probe(struct platform_device *pdev) data->card.dev = &pdev->dev; data->card.dai_link = &data->dai; data->card.num_links = 1; + data->card.owner = THIS_MODULE; ret = snd_soc_of_parse_card_name(&data->card, "model"); if (ret) 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/broadwell.c b/sound/soc/intel/broadwell.c index 7cf95d5d5d80..9cf7d01479ad 100644 --- a/sound/soc/intel/broadwell.c +++ b/sound/soc/intel/broadwell.c @@ -140,8 +140,6 @@ static struct snd_soc_ops broadwell_rt286_ops = { static int broadwell_rtd_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; struct sst_pdata *pdata = dev_get_platdata(rtd->platform->dev); struct sst_hsw *broadwell = pdata->dsp; int ret; @@ -155,14 +153,6 @@ static int broadwell_rtd_init(struct snd_soc_pcm_runtime *rtd) return ret; } - /* always connected - check HP for jack detect */ - snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); - snd_soc_dapm_enable_pin(dapm, "Speaker"); - snd_soc_dapm_enable_pin(dapm, "Mic Jack"); - snd_soc_dapm_enable_pin(dapm, "Line Jack"); - snd_soc_dapm_enable_pin(dapm, "DMIC1"); - snd_soc_dapm_enable_pin(dapm, "DMIC2"); - return 0; } diff --git a/sound/soc/intel/byt-rt5640.c b/sound/soc/intel/byt-rt5640.c index 0cba7830c5e9..354eaad886e1 100644 --- a/sound/soc/intel/byt-rt5640.c +++ b/sound/soc/intel/byt-rt5640.c @@ -132,7 +132,6 @@ 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; @@ -161,7 +160,7 @@ static int byt_rt5640_init(struct snd_soc_pcm_runtime *runtime) num_routes = ARRAY_SIZE(byt_rt5640_intmic_dmic1_map); } - ret = snd_soc_dapm_add_routes(dapm, custom_map, num_routes); + ret = snd_soc_dapm_add_routes(&card->dapm, custom_map, num_routes); if (ret) return ret; @@ -171,13 +170,8 @@ static int byt_rt5640_init(struct snd_soc_pcm_runtime *runtime) return ret; } - snd_soc_dapm_ignore_suspend(dapm, "HPOL"); - snd_soc_dapm_ignore_suspend(dapm, "HPOR"); - - snd_soc_dapm_ignore_suspend(dapm, "SPOLP"); - snd_soc_dapm_ignore_suspend(dapm, "SPOLN"); - snd_soc_dapm_ignore_suspend(dapm, "SPORP"); - snd_soc_dapm_ignore_suspend(dapm, "SPORN"); + snd_soc_dapm_ignore_suspend(&card->dapm, "Headphone"); + snd_soc_dapm_ignore_suspend(&card->dapm, "Speaker"); return ret; } 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/cht_bsw_rt5672.c b/sound/soc/intel/cht_bsw_rt5672.c index a406c6104897..ff016621583a 100644 --- a/sound/soc/intel/cht_bsw_rt5672.c +++ b/sound/soc/intel/cht_bsw_rt5672.c @@ -140,6 +140,7 @@ static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) { int ret; struct snd_soc_dai *codec_dai = runtime->codec_dai; + struct snd_soc_codec *codec = codec_dai->codec; /* 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); @@ -148,6 +149,19 @@ static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) return ret; } + /* Select codec ASRC clock source to track I2S1 clock, because codec + * is in slave mode and 100fs I2S format (BCLK = 100 * LRCLK) cannot + * be supported by RT5672. Otherwise, ASRC will be disabled and cause + * noise. + */ + rt5670_sel_asrc_clk_src(codec, + RT5670_DA_STEREO_FILTER + | RT5670_DA_MONO_L_FILTER + | RT5670_DA_MONO_R_FILTER + | RT5670_AD_STEREO_FILTER + | RT5670_AD_MONO_L_FILTER + | RT5670_AD_MONO_R_FILTER, + RT5670_CLK_SEL_I2S1_ASRC); return 0; } diff --git a/sound/soc/intel/sst-baytrail-pcm.c b/sound/soc/intel/sst-baytrail-pcm.c index 3bb6288d8b4d..224c49c9f135 100644 --- a/sound/soc/intel/sst-baytrail-pcm.c +++ b/sound/soc/intel/sst-baytrail-pcm.c @@ -320,11 +320,6 @@ static struct snd_pcm_ops sst_byt_pcm_ops = { .mmap = sst_byt_pcm_mmap, }; -static void sst_byt_pcm_free(struct snd_pcm *pcm) -{ - snd_pcm_lib_preallocate_free_for_all(pcm); -} - static int sst_byt_pcm_new(struct snd_soc_pcm_runtime *rtd) { struct snd_pcm *pcm = rtd->pcm; @@ -403,7 +398,6 @@ static struct snd_soc_platform_driver byt_soc_platform = { .remove = sst_byt_pcm_remove, .ops = &sst_byt_pcm_ops, .pcm_new = sst_byt_pcm_new, - .pcm_free = sst_byt_pcm_free, }; static const struct snd_soc_component_driver byt_dai_component = { diff --git a/sound/soc/intel/sst-dsp.c b/sound/soc/intel/sst-dsp.c index 86e410845670..64e94212d2d2 100644 --- a/sound/soc/intel/sst-dsp.c +++ b/sound/soc/intel/sst-dsp.c @@ -410,8 +410,7 @@ void sst_dsp_free(struct sst_dsp *sst) if (sst->ops->free) sst->ops->free(sst); - if (sst->dma) - sst_dma_free(sst->dma); + sst_dma_free(sst->dma); } EXPORT_SYMBOL_GPL(sst_dsp_free); diff --git a/sound/soc/intel/sst-firmware.c b/sound/soc/intel/sst-firmware.c index a2ae2c5f2e9f..5f71ef607a57 100644 --- a/sound/soc/intel/sst-firmware.c +++ b/sound/soc/intel/sst-firmware.c @@ -791,6 +791,7 @@ int sst_module_alloc_blocks(struct sst_module *module) struct sst_block_allocator ba; int ret; + memset(&ba, 0, sizeof(ba)); ba.size = module->size; ba.type = module->type; ba.offset = module->offset; @@ -864,6 +865,7 @@ int sst_module_runtime_alloc_blocks(struct sst_module_runtime *runtime, if (module->persistent_size == 0) return 0; + memset(&ba, 0, sizeof(ba)); ba.size = module->persistent_size; ba.type = SST_MEM_DRAM; diff --git a/sound/soc/intel/sst-haswell-dsp.c b/sound/soc/intel/sst-haswell-dsp.c index 57039b00efc2..c42ffae5fe9f 100644 --- a/sound/soc/intel/sst-haswell-dsp.c +++ b/sound/soc/intel/sst-haswell-dsp.c @@ -306,7 +306,7 @@ static void hsw_reset(struct sst_dsp *sst) static int hsw_set_dsp_D0(struct sst_dsp *sst) { int tries = 10; - u32 reg; + u32 reg, fw_dump_bit; /* Disable core clock gating (VDRTCTL2.DCLCGE = 0) */ reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2); @@ -368,7 +368,9 @@ finish: can't be accessed, please enable each block before accessing. */ reg = readl(sst->addr.pci_cfg + SST_VDRTCTL0); reg |= SST_VDRTCL0_DSRAMPGE_MASK | SST_VDRTCL0_ISRAMPGE_MASK; - writel(reg, sst->addr.pci_cfg + SST_VDRTCTL0); + /* for D0, always enable the block(DSRAM[0]) used for FW dump */ + fw_dump_bit = 1 << SST_VDRTCL0_DSRAMPGE_SHIFT; + writel(reg & ~fw_dump_bit, sst->addr.pci_cfg + SST_VDRTCTL0); /* disable DMA finish function for SSP0 & SSP1 */ @@ -491,6 +493,7 @@ static const struct sst_sram_shift sram_shift[] = { {SST_DEV_ID_LYNX_POINT, 6, 16}, /* lp */ {SST_DEV_ID_WILDCAT_POINT, 2, 12}, /* wpt */ }; + static u32 hsw_block_get_bit(struct sst_mem_block *block) { u32 bit = 0, shift = 0, index; @@ -587,7 +590,9 @@ static int hsw_block_disable(struct sst_mem_block *block) val = readl(sst->addr.pci_cfg + SST_VDRTCTL0); bit = hsw_block_get_bit(block); - writel(val | bit, sst->addr.pci_cfg + SST_VDRTCTL0); + /* don't disable DSRAM[0], keep it always enable for FW dump*/ + if (bit != (1 << SST_VDRTCL0_DSRAMPGE_SHIFT)) + writel(val | bit, sst->addr.pci_cfg + SST_VDRTCTL0); /* wait 18 DSP clock ticks */ udelay(10); @@ -612,7 +617,7 @@ static int hsw_init(struct sst_dsp *sst, struct sst_pdata *pdata) const struct sst_adsp_memregion *region; struct device *dev; int ret = -ENODEV, i, j, region_count; - u32 offset, size; + u32 offset, size, fw_dump_bit; dev = sst->dma_dev; @@ -669,9 +674,11 @@ static int hsw_init(struct sst_dsp *sst, struct sst_pdata *pdata) } } + /* always enable the block(DSRAM[0]) used for FW dump */ + fw_dump_bit = 1 << SST_VDRTCL0_DSRAMPGE_SHIFT; /* set default power gating control, enable power gating control for all blocks. that is, can't be accessed, please enable each block before accessing. */ - writel(0xffffffff, sst->addr.pci_cfg + SST_VDRTCTL0); + writel(0xffffffff & ~fw_dump_bit, sst->addr.pci_cfg + SST_VDRTCTL0); return 0; } diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index 8156cc1accb7..0ab1309ef274 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -94,6 +94,8 @@ /* Mailbox */ #define IPC_MAX_MAILBOX_BYTES 256 +#define INVALID_STREAM_HW_ID 0xffffffff + /* Global Message - Types and Replies */ enum ipc_glb_type { IPC_GLB_GET_FW_VERSION = 0, /* Retrieves firmware version */ @@ -275,7 +277,6 @@ struct sst_hsw { /* FW config */ struct sst_hsw_ipc_fw_ready fw_ready; struct sst_hsw_ipc_fw_version version; - struct sst_module *scratch; bool fw_done; struct sst_fw *sst_fw; @@ -337,12 +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; @@ -969,45 +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) { @@ -1021,17 +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; - - return 0; -} - /* stream volume */ int sst_hsw_stream_set_volume(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 stage_id, u32 channel, u32 volume) @@ -1083,42 +1028,6 @@ int sst_hsw_stream_set_volume(struct sst_hsw *hsw, return 0; } -int sst_hsw_mixer_mute(struct sst_hsw *hsw, u32 stage_id, u32 channel) -{ - int ret; - - ret = sst_hsw_mixer_get_volume(hsw, stage_id, channel, - &hsw->mute_volume[channel]); - if (ret < 0) - return ret; - - ret = sst_hsw_mixer_set_volume(hsw, stage_id, channel, 0); - if (ret < 0) { - dev_err(hsw->dev, "error: failed to unmute mixer channel %d\n", - channel); - return ret; - } - - hsw->mute[channel] = 1; - return 0; -} - -int sst_hsw_mixer_unmute(struct sst_hsw *hsw, u32 stage_id, u32 channel) -{ - int ret; - - ret = sst_hsw_mixer_set_volume(hsw, stage_id, channel, - hsw->mixer_info.volume_register_address[channel]); - if (ret < 0) { - dev_err(hsw->dev, "error: failed to unmute mixer channel %d\n", - channel); - return ret; - } - - hsw->mute[channel] = 0; - return 0; -} - int sst_hsw_mixer_get_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel, u32 *volume) { @@ -1132,16 +1041,6 @@ int sst_hsw_mixer_get_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel, return 0; } -int sst_hsw_mixer_set_volume_curve(struct sst_hsw *hsw, - u64 curve_duration, enum sst_hsw_volume_curve curve) -{ - /* curve duration in steps of 100ns */ - hsw->curve_duration = curve_duration; - hsw->curve_type = curve; - - return 0; -} - /* global mixer volume */ int sst_hsw_mixer_set_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel, u32 volume) @@ -1208,6 +1107,7 @@ struct sst_hsw_stream *sst_hsw_stream_new(struct sst_hsw *hsw, int id, return NULL; spin_lock_irqsave(&sst->spinlock, flags); + stream->reply.stream_hw_id = INVALID_STREAM_HW_ID; list_add(&stream->node, &hsw->stream_list); stream->notify_position = notify_position; stream->pdata = data; @@ -1449,48 +1349,6 @@ int sst_hsw_stream_commit(struct sst_hsw *hsw, struct sst_hsw_stream *stream) /* Stream Information - these calls could be inline but we want the IPC ABI to be opaque to client PCM drivers to cope with any future ABI changes */ -int sst_hsw_stream_get_hw_id(struct sst_hsw *hsw, - struct sst_hsw_stream *stream) -{ - return stream->reply.stream_hw_id; -} - -int sst_hsw_stream_get_mixer_id(struct sst_hsw *hsw, - struct sst_hsw_stream *stream) -{ - return stream->reply.mixer_hw_id; -} - -u32 sst_hsw_stream_get_read_reg(struct sst_hsw *hsw, - struct sst_hsw_stream *stream) -{ - return stream->reply.read_position_register_address; -} - -u32 sst_hsw_stream_get_pointer_reg(struct sst_hsw *hsw, - struct sst_hsw_stream *stream) -{ - return stream->reply.presentation_position_register_address; -} - -u32 sst_hsw_stream_get_peak_reg(struct sst_hsw *hsw, - struct sst_hsw_stream *stream, u32 channel) -{ - if (channel >= 2) - return 0; - - return stream->reply.peak_meter_register_address[channel]; -} - -u32 sst_hsw_stream_get_vol_reg(struct sst_hsw *hsw, - struct sst_hsw_stream *stream, u32 channel) -{ - if (channel >= 2) - return 0; - - return stream->reply.volume_register_address[channel]; -} - int sst_hsw_mixer_get_info(struct sst_hsw *hsw) { struct sst_hsw_ipc_stream_info_reply *reply; @@ -1628,30 +1486,6 @@ u64 sst_hsw_get_dsp_presentation_position(struct sst_hsw *hsw, return ppos; } -int sst_hsw_stream_set_write_position(struct sst_hsw *hsw, - struct sst_hsw_stream *stream, u32 stage_id, u32 position) -{ - u32 header; - int ret; - - trace_stream_write_position(stream->reply.stream_hw_id, position); - - header = IPC_GLB_TYPE(IPC_GLB_STREAM_MESSAGE) | - IPC_STR_TYPE(IPC_STR_STAGE_MESSAGE); - header |= (stream->reply.stream_hw_id << IPC_STR_ID_SHIFT); - header |= (IPC_STG_SET_WRITE_POSITION << IPC_STG_TYPE_SHIFT); - header |= (stage_id << IPC_STG_ID_SHIFT); - stream->wpos.position = position; - - ret = ipc_tx_message_nowait(hsw, header, &stream->wpos, - sizeof(stream->wpos)); - if (ret < 0) - dev_err(hsw->dev, "error: stream %d set position %d failed\n", - stream->reply.stream_hw_id, position); - - return ret; -} - /* physical BE config */ int sst_hsw_device_set_config(struct sst_hsw *hsw, enum sst_hsw_device_id dev, enum sst_hsw_device_mclk mclk, @@ -2132,7 +1966,6 @@ void sst_hsw_dsp_free(struct device *dev, struct sst_pdata *pdata) dma_free_coherent(hsw->dsp->dma_dev, SST_HSW_DX_CONTEXT_SIZE, hsw->dx_context, hsw->dx_context_paddr); sst_dsp_free(hsw->dsp); - kfree(hsw->scratch); kthread_stop(hsw->tx_thread); kfree(hsw->msg); } diff --git a/sound/soc/intel/sst-haswell-ipc.h b/sound/soc/intel/sst-haswell-ipc.h index 138e894ab413..c1ad901342f2 100644 --- a/sound/soc/intel/sst-haswell-ipc.h +++ b/sound/soc/intel/sst-haswell-ipc.h @@ -376,32 +376,17 @@ int sst_hsw_fw_get_version(struct sst_hsw *hsw, u32 create_channel_map(enum sst_hsw_channel_config config); /* Stream Mixer Controls - */ -int sst_hsw_stream_mute(struct sst_hsw *hsw, struct sst_hsw_stream *stream, - u32 stage_id, u32 channel); -int sst_hsw_stream_unmute(struct sst_hsw *hsw, struct sst_hsw_stream *stream, - u32 stage_id, u32 channel); - int sst_hsw_stream_set_volume(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 stage_id, u32 channel, u32 volume); int sst_hsw_stream_get_volume(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 stage_id, u32 channel, u32 *volume); -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); - /* Global Mixer Controls - */ -int sst_hsw_mixer_mute(struct sst_hsw *hsw, u32 stage_id, u32 channel); -int sst_hsw_mixer_unmute(struct sst_hsw *hsw, u32 stage_id, u32 channel); - int sst_hsw_mixer_set_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel, u32 volume); int sst_hsw_mixer_get_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel, u32 *volume); -int sst_hsw_mixer_set_volume_curve(struct sst_hsw *hsw, - u64 curve_duration, enum sst_hsw_volume_curve curve); - /* Stream API */ struct sst_hsw_stream *sst_hsw_stream_new(struct sst_hsw *hsw, int id, u32 (*get_write_position)(struct sst_hsw_stream *stream, void *data), @@ -440,18 +425,6 @@ int sst_hsw_stream_set_pmemory_info(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 offset, u32 size); int sst_hsw_stream_set_smemory_info(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 offset, u32 size); -int sst_hsw_stream_get_hw_id(struct sst_hsw *hsw, - struct sst_hsw_stream *stream); -int sst_hsw_stream_get_mixer_id(struct sst_hsw *hsw, - struct sst_hsw_stream *stream); -u32 sst_hsw_stream_get_read_reg(struct sst_hsw *hsw, - struct sst_hsw_stream *stream); -u32 sst_hsw_stream_get_pointer_reg(struct sst_hsw *hsw, - struct sst_hsw_stream *stream); -u32 sst_hsw_stream_get_peak_reg(struct sst_hsw *hsw, - struct sst_hsw_stream *stream, u32 channel); -u32 sst_hsw_stream_get_vol_reg(struct sst_hsw *hsw, - struct sst_hsw_stream *stream, u32 channel); int sst_hsw_mixer_get_info(struct sst_hsw *hsw); /* Stream ALSA trigger operations */ @@ -466,8 +439,6 @@ int sst_hsw_stream_get_read_pos(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 *position); int sst_hsw_stream_get_write_pos(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 *position); -int sst_hsw_stream_set_write_position(struct sst_hsw *hsw, - struct sst_hsw_stream *stream, u32 stage_id, u32 position); u32 sst_hsw_get_dsp_position(struct sst_hsw *hsw, struct sst_hsw_stream *stream); u64 sst_hsw_get_dsp_presentation_position(struct sst_hsw *hsw, @@ -481,8 +452,6 @@ int sst_hsw_device_set_config(struct sst_hsw *hsw, /* DX Config */ int sst_hsw_dx_set_state(struct sst_hsw *hsw, enum sst_hsw_dx_state state, struct sst_hsw_ipc_dx_reply *dx); -int sst_hsw_dx_get_state(struct sst_hsw *hsw, u32 item, - u32 *offset, u32 *size, u32 *source); /* init */ int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata); diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c index 619525200705..78fa01be57f2 100644 --- a/sound/soc/intel/sst-haswell-pcm.c +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -78,7 +78,6 @@ static const u32 volume_map[] = { #define HSW_PCM_DAI_ID_OFFLOAD0 1 #define HSW_PCM_DAI_ID_OFFLOAD1 2 #define HSW_PCM_DAI_ID_LOOPBACK 3 -#define HSW_PCM_DAI_ID_CAPTURE 4 static const struct snd_pcm_hardware hsw_pcm_hardware = { @@ -99,6 +98,7 @@ static const struct snd_pcm_hardware hsw_pcm_hardware = { struct hsw_pcm_module_map { int dai_id; + int stream; enum sst_hsw_module_id mod_id; }; @@ -119,8 +119,9 @@ struct hsw_pcm_data { }; enum hsw_pm_state { - HSW_PM_STATE_D3 = 0, - HSW_PM_STATE_D0 = 1, + HSW_PM_STATE_D0 = 0, + HSW_PM_STATE_RTD3 = 1, + HSW_PM_STATE_D3 = 2, }; /* private data for the driver */ @@ -135,7 +136,17 @@ struct hsw_priv_data { struct snd_dma_buffer dmab[HSW_PCM_COUNT][2]; /* DAI data */ - struct hsw_pcm_data pcm[HSW_PCM_COUNT]; + struct hsw_pcm_data pcm[HSW_PCM_COUNT][2]; +}; + + +/* static mappings between PCMs and modules - may be dynamic in future */ +static struct hsw_pcm_module_map mod_map[] = { + {HSW_PCM_DAI_ID_SYSTEM, 0, SST_HSW_MODULE_PCM_SYSTEM}, + {HSW_PCM_DAI_ID_OFFLOAD0, 0, SST_HSW_MODULE_PCM}, + {HSW_PCM_DAI_ID_OFFLOAD1, 0, SST_HSW_MODULE_PCM}, + {HSW_PCM_DAI_ID_LOOPBACK, 1, SST_HSW_MODULE_PCM_REFERENCE}, + {HSW_PCM_DAI_ID_SYSTEM, 1, SST_HSW_MODULE_PCM_CAPTURE}, }; static u32 hsw_notify_pointer(struct sst_hsw_stream *stream, void *data); @@ -168,9 +179,14 @@ static int hsw_stream_volume_put(struct snd_kcontrol *kcontrol, (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 hsw_pcm_data *pcm_data; struct sst_hsw *hsw = pdata->hsw; u32 volume; + int dai, stream; + + dai = mod_map[mc->reg].dai_id; + stream = mod_map[mc->reg].stream; + pcm_data = &pdata->pcm[dai][stream]; mutex_lock(&pcm_data->mutex); pm_runtime_get_sync(pdata->dev); @@ -212,9 +228,14 @@ static int hsw_stream_volume_get(struct snd_kcontrol *kcontrol, (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 hsw_pcm_data *pcm_data; struct sst_hsw *hsw = pdata->hsw; u32 volume; + int dai, stream; + + dai = mod_map[mc->reg].dai_id; + stream = mod_map[mc->reg].stream; + pcm_data = &pdata->pcm[dai][stream]; mutex_lock(&pcm_data->mutex); pm_runtime_get_sync(pdata->dev); @@ -309,7 +330,7 @@ static const struct snd_kcontrol_new hsw_volume_controls[] = { ARRAY_SIZE(volume_map) - 1, 0, hsw_stream_volume_get, hsw_stream_volume_put, hsw_vol_tlv), /* Mic Capture volume */ - SOC_DOUBLE_EXT_TLV("Mic Capture Volume", 0, 0, 8, + SOC_DOUBLE_EXT_TLV("Mic Capture Volume", 4, 0, 8, ARRAY_SIZE(volume_map) - 1, 0, hsw_stream_volume_get, hsw_stream_volume_put, hsw_vol_tlv), }; @@ -353,7 +374,7 @@ static int hsw_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime = substream->runtime; struct hsw_priv_data *pdata = snd_soc_platform_get_drvdata(rtd->platform); - struct hsw_pcm_data *pcm_data = snd_soc_pcm_get_drvdata(rtd); + struct hsw_pcm_data *pcm_data; struct sst_hsw *hsw = pdata->hsw; struct sst_module *module_data; struct sst_dsp *dsp; @@ -362,7 +383,10 @@ static int hsw_pcm_hw_params(struct snd_pcm_substream *substream, enum sst_hsw_stream_path_id path_id; u32 rate, bits, map, pages, module_id; u8 channels; - int ret; + int ret, dai; + + dai = mod_map[rtd->cpu_dai->id].dai_id; + pcm_data = &pdata->pcm[dai][substream->stream]; /* check if we are being called a subsequent time */ if (pcm_data->allocated) { @@ -552,8 +576,12 @@ static int hsw_pcm_trigger(struct snd_pcm_substream *substream, int cmd) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct hsw_priv_data *pdata = snd_soc_platform_get_drvdata(rtd->platform); - struct hsw_pcm_data *pcm_data = snd_soc_pcm_get_drvdata(rtd); + struct hsw_pcm_data *pcm_data; struct sst_hsw *hsw = pdata->hsw; + int dai; + + dai = mod_map[rtd->cpu_dai->id].dai_id; + pcm_data = &pdata->pcm[dai][substream->stream]; switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -597,11 +625,16 @@ static snd_pcm_uframes_t hsw_pcm_pointer(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct hsw_priv_data *pdata = snd_soc_platform_get_drvdata(rtd->platform); - struct hsw_pcm_data *pcm_data = snd_soc_pcm_get_drvdata(rtd); + struct hsw_pcm_data *pcm_data; struct sst_hsw *hsw = pdata->hsw; snd_pcm_uframes_t offset; uint64_t ppos; - u32 position = sst_hsw_get_dsp_position(hsw, pcm_data->stream); + u32 position; + int dai; + + dai = mod_map[rtd->cpu_dai->id].dai_id; + pcm_data = &pdata->pcm[dai][substream->stream]; + position = sst_hsw_get_dsp_position(hsw, pcm_data->stream); offset = bytes_to_frames(runtime, position); ppos = sst_hsw_get_dsp_presentation_position(hsw, pcm_data->stream); @@ -618,8 +651,10 @@ static int hsw_pcm_open(struct snd_pcm_substream *substream) snd_soc_platform_get_drvdata(rtd->platform); struct hsw_pcm_data *pcm_data; struct sst_hsw *hsw = pdata->hsw; + int dai; - pcm_data = &pdata->pcm[rtd->cpu_dai->id]; + dai = mod_map[rtd->cpu_dai->id].dai_id; + pcm_data = &pdata->pcm[dai][substream->stream]; mutex_lock(&pcm_data->mutex); pm_runtime_get_sync(pdata->dev); @@ -648,9 +683,12 @@ static int hsw_pcm_close(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct hsw_priv_data *pdata = snd_soc_platform_get_drvdata(rtd->platform); - struct hsw_pcm_data *pcm_data = snd_soc_pcm_get_drvdata(rtd); + struct hsw_pcm_data *pcm_data; struct sst_hsw *hsw = pdata->hsw; - int ret; + int ret, dai; + + dai = mod_map[rtd->cpu_dai->id].dai_id; + pcm_data = &pdata->pcm[dai][substream->stream]; mutex_lock(&pcm_data->mutex); ret = sst_hsw_stream_reset(hsw, pcm_data->stream); @@ -685,15 +723,6 @@ static struct snd_pcm_ops hsw_pcm_ops = { .page = snd_pcm_sgbuf_ops_page, }; -/* static mappings between PCMs and modules - may be dynamic in future */ -static struct hsw_pcm_module_map mod_map[] = { - {HSW_PCM_DAI_ID_SYSTEM, SST_HSW_MODULE_PCM_SYSTEM}, - {HSW_PCM_DAI_ID_OFFLOAD0, SST_HSW_MODULE_PCM}, - {HSW_PCM_DAI_ID_OFFLOAD1, SST_HSW_MODULE_PCM}, - {HSW_PCM_DAI_ID_LOOPBACK, SST_HSW_MODULE_PCM_REFERENCE}, - {HSW_PCM_DAI_ID_CAPTURE, SST_HSW_MODULE_PCM_CAPTURE}, -}; - static int hsw_pcm_create_modules(struct hsw_priv_data *pdata) { struct sst_hsw *hsw = pdata->hsw; @@ -701,7 +730,7 @@ static int hsw_pcm_create_modules(struct hsw_priv_data *pdata) int i; for (i = 0; i < ARRAY_SIZE(mod_map); i++) { - pcm_data = &pdata->pcm[i]; + pcm_data = &pdata->pcm[mod_map[i].dai_id][mod_map[i].stream]; /* create new runtime module, use same offset if recreated */ pcm_data->runtime = sst_hsw_runtime_module_create(hsw, @@ -716,7 +745,7 @@ static int hsw_pcm_create_modules(struct hsw_priv_data *pdata) err: for (--i; i >= 0; i--) { - pcm_data = &pdata->pcm[i]; + pcm_data = &pdata->pcm[mod_map[i].dai_id][mod_map[i].stream]; sst_hsw_runtime_module_free(pcm_data->runtime); } @@ -729,17 +758,12 @@ static void hsw_pcm_free_modules(struct hsw_priv_data *pdata) int i; for (i = 0; i < ARRAY_SIZE(mod_map); i++) { - pcm_data = &pdata->pcm[i]; + pcm_data = &pdata->pcm[mod_map[i].dai_id][mod_map[i].stream]; sst_hsw_runtime_module_free(pcm_data->runtime); } } -static void hsw_pcm_free(struct snd_pcm *pcm) -{ - snd_pcm_lib_preallocate_free_for_all(pcm); -} - static int hsw_pcm_new(struct snd_soc_pcm_runtime *rtd) { struct snd_pcm *pcm = rtd->pcm; @@ -762,7 +786,10 @@ static int hsw_pcm_new(struct snd_soc_pcm_runtime *rtd) return ret; } } - priv_data->pcm[rtd->cpu_dai->id].hsw_pcm = pcm; + if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) + priv_data->pcm[rtd->cpu_dai->id][SNDRV_PCM_STREAM_PLAYBACK].hsw_pcm = pcm; + if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) + priv_data->pcm[rtd->cpu_dai->id][SNDRV_PCM_STREAM_CAPTURE].hsw_pcm = pcm; return ret; } @@ -871,10 +898,9 @@ static int hsw_pcm_probe(struct snd_soc_platform *platform) /* allocate DSP buffer page tables */ for (i = 0; i < ARRAY_SIZE(hsw_dais); i++) { - mutex_init(&priv_data->pcm[i].mutex); - /* playback */ if (hsw_dais[i].playback.channels_min) { + mutex_init(&priv_data->pcm[i][SNDRV_PCM_STREAM_PLAYBACK].mutex); ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, dma_dev, PAGE_SIZE, &priv_data->dmab[i][0]); if (ret < 0) @@ -883,6 +909,7 @@ static int hsw_pcm_probe(struct snd_soc_platform *platform) /* capture */ if (hsw_dais[i].capture.channels_min) { + mutex_init(&priv_data->pcm[i][SNDRV_PCM_STREAM_CAPTURE].mutex); ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, dma_dev, PAGE_SIZE, &priv_data->dmab[i][1]); if (ret < 0) @@ -936,7 +963,6 @@ static struct snd_soc_platform_driver hsw_soc_platform = { .remove = hsw_pcm_remove, .ops = &hsw_pcm_ops, .pcm_new = hsw_pcm_new, - .pcm_free = hsw_pcm_free, }; static const struct snd_soc_component_driver hsw_dai_component = { @@ -1010,12 +1036,12 @@ static int hsw_pcm_runtime_suspend(struct device *dev) struct hsw_priv_data *pdata = dev_get_drvdata(dev); struct sst_hsw *hsw = pdata->hsw; - if (pdata->pm_state == HSW_PM_STATE_D3) + if (pdata->pm_state >= HSW_PM_STATE_RTD3) return 0; sst_hsw_dsp_runtime_suspend(hsw); sst_hsw_dsp_runtime_sleep(hsw); - pdata->pm_state = HSW_PM_STATE_D3; + pdata->pm_state = HSW_PM_STATE_RTD3; return 0; } @@ -1026,7 +1052,7 @@ static int hsw_pcm_runtime_resume(struct device *dev) struct sst_hsw *hsw = pdata->hsw; int ret; - if (pdata->pm_state == HSW_PM_STATE_D0) + if (pdata->pm_state != HSW_PM_STATE_RTD3) return 0; ret = sst_hsw_dsp_load(hsw); @@ -1066,7 +1092,7 @@ static void hsw_pcm_complete(struct device *dev) struct hsw_pcm_data *pcm_data; int i, err; - if (pdata->pm_state == HSW_PM_STATE_D0) + if (pdata->pm_state != HSW_PM_STATE_D3) return; err = sst_hsw_dsp_load(hsw); @@ -1081,8 +1107,8 @@ static void hsw_pcm_complete(struct device *dev) return; } - for (i = 0; i < HSW_PCM_DAI_ID_CAPTURE + 1; i++) { - pcm_data = &pdata->pcm[i]; + for (i = 0; i < ARRAY_SIZE(mod_map); i++) { + pcm_data = &pdata->pcm[mod_map[i].dai_id][mod_map[i].stream]; if (!pcm_data->substream) continue; @@ -1114,41 +1140,42 @@ static int hsw_pcm_prepare(struct device *dev) if (pdata->pm_state == HSW_PM_STATE_D3) return 0; - /* suspend all active streams */ - for (i = 0; i < HSW_PCM_DAI_ID_CAPTURE + 1; i++) { - pcm_data = &pdata->pcm[i]; + else if (pdata->pm_state == HSW_PM_STATE_D0) { + /* suspend all active streams */ + for (i = 0; i < ARRAY_SIZE(mod_map); i++) { + pcm_data = &pdata->pcm[mod_map[i].dai_id][mod_map[i].stream]; + + if (!pcm_data->substream) + continue; + dev_dbg(dev, "suspending pcm %d\n", i); + snd_pcm_suspend_all(pcm_data->hsw_pcm); + + /* We need to wait until the DSP FW stops the streams */ + msleep(2); + } - if (!pcm_data->substream) - continue; - dev_dbg(dev, "suspending pcm %d\n", i); - snd_pcm_suspend_all(pcm_data->hsw_pcm); + /* preserve persistent memory */ + for (i = 0; i < ARRAY_SIZE(mod_map); i++) { + pcm_data = &pdata->pcm[mod_map[i].dai_id][mod_map[i].stream]; + + if (!pcm_data->substream) + continue; - /* We need to wait until the DSP FW stops the streams */ - msleep(2); + dev_dbg(dev, "saving context pcm %d\n", i); + err = sst_module_runtime_save(pcm_data->runtime, + &pcm_data->context); + if (err < 0) + dev_err(dev, "failed to save context for PCM %d\n", i); + } + /* enter D3 state and stall */ + sst_hsw_dsp_runtime_suspend(hsw); + /* put the DSP to sleep */ + sst_hsw_dsp_runtime_sleep(hsw); } snd_soc_suspend(pdata->soc_card->dev); snd_soc_poweroff(pdata->soc_card->dev); - /* enter D3 state and stall */ - sst_hsw_dsp_runtime_suspend(hsw); - - /* preserve persistent memory */ - for (i = 0; i < HSW_PCM_DAI_ID_CAPTURE + 1; i++) { - pcm_data = &pdata->pcm[i]; - - if (!pcm_data->substream) - continue; - - dev_dbg(dev, "saving context pcm %d\n", i); - err = sst_module_runtime_save(pcm_data->runtime, - &pcm_data->context); - if (err < 0) - dev_err(dev, "failed to save context for PCM %d\n", i); - } - - /* put the DSP to sleep */ - sst_hsw_dsp_runtime_sleep(hsw); pdata->pm_state = HSW_PM_STATE_D3; return 0; diff --git a/sound/soc/intel/sst-mfld-platform-pcm.c b/sound/soc/intel/sst-mfld-platform-pcm.c index a1a8d9d91539..7523cbef8780 100644 --- a/sound/soc/intel/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/sst-mfld-platform-pcm.c @@ -643,12 +643,6 @@ static struct snd_pcm_ops sst_platform_ops = { .pointer = sst_platform_pcm_pointer, }; -static void sst_pcm_free(struct snd_pcm *pcm) -{ - dev_dbg(pcm->dev, "sst_pcm_free called\n"); - snd_pcm_lib_preallocate_free_for_all(pcm); -} - static int sst_pcm_new(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_dai *dai = rtd->cpu_dai; @@ -679,7 +673,6 @@ static struct snd_soc_platform_driver sst_soc_platform_drv = { .ops = &sst_platform_ops, .compr_ops = &sst_platform_compr_ops, .pcm_new = sst_pcm_new, - .pcm_free = sst_pcm_free, }; static const struct snd_soc_component_driver sst_component = { diff --git a/sound/soc/intel/sst/sst_acpi.c b/sound/soc/intel/sst/sst_acpi.c index e541d0e69ea2..b782dfdcdbba 100644 --- a/sound/soc/intel/sst/sst_acpi.c +++ b/sound/soc/intel/sst/sst_acpi.c @@ -352,6 +352,8 @@ static struct sst_machines sst_acpi_bytcr[] = { static struct sst_machines sst_acpi_chv[] = { {"10EC5670", "cht-bsw", "cht-bsw-rt5672", NULL, "intel/fw_sst_22a8.bin", &chv_platform_data }, + {"10EC5645", "cht-bsw", "cht-bsw-rt5645", NULL, "intel/fw_sst_22a8.bin", + &chv_platform_data }, {}, }; diff --git a/sound/soc/intel/sst/sst_loader.c b/sound/soc/intel/sst/sst_loader.c index b580f96e25e5..7888cd707853 100644 --- a/sound/soc/intel/sst/sst_loader.c +++ b/sound/soc/intel/sst/sst_loader.c @@ -324,8 +324,7 @@ void sst_firmware_load_cb(const struct firmware *fw, void *context) if (ctx->sst_state != SST_RESET || ctx->fw_in_mem != NULL) { - if (fw != NULL) - release_firmware(fw); + release_firmware(fw); mutex_unlock(&ctx->sst_lock); return; } diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c index d3d45c6f064f..07f77815a586 100644 --- a/sound/soc/jz4740/jz4740-i2s.c +++ b/sound/soc/jz4740/jz4740-i2s.c @@ -14,6 +14,8 @@ #include <linux/init.h> #include <linux/io.h> +#include <linux/of.h> +#include <linux/of_device.h> #include <linux/kernel.h> #include <linux/module.h> #include <linux/platform_device.h> @@ -83,6 +85,8 @@ #define JZ_AIC_I2S_STATUS_BUSY BIT(2) #define JZ_AIC_CLK_DIV_MASK 0xf +#define I2SDIV_DV_SHIFT 8 +#define I2SDIV_DV_MASK (0xf << I2SDIV_DV_SHIFT) struct jz4740_i2s { struct resource *mem; @@ -237,10 +241,14 @@ static int jz4740_i2s_hw_params(struct snd_pcm_substream *substream, { struct jz4740_i2s *i2s = snd_soc_dai_get_drvdata(dai); unsigned int sample_size; - uint32_t ctrl; + uint32_t ctrl, div_reg; + int div; ctrl = jz4740_i2s_read(i2s, JZ_REG_AIC_CTRL); + div_reg = jz4740_i2s_read(i2s, JZ_REG_AIC_CLK_DIV); + div = clk_get_rate(i2s->clk_i2s) / (64 * params_rate(params)); + switch (params_format(params)) { case SNDRV_PCM_FORMAT_S8: sample_size = 0; @@ -264,7 +272,10 @@ static int jz4740_i2s_hw_params(struct snd_pcm_substream *substream, ctrl |= sample_size << JZ_AIC_CTRL_INPUT_SAMPLE_SIZE_OFFSET; } + div_reg &= ~I2SDIV_DV_MASK; + div_reg |= (div - 1) << I2SDIV_DV_SHIFT; jz4740_i2s_write(i2s, JZ_REG_AIC_CTRL, ctrl); + jz4740_i2s_write(i2s, JZ_REG_AIC_CLK_DIV, div_reg); return 0; } @@ -415,6 +426,13 @@ static const struct snd_soc_component_driver jz4740_i2s_component = { .name = "jz4740-i2s", }; +#ifdef CONFIG_OF +static const struct of_device_id jz4740_of_matches[] = { + { .compatible = "ingenic,jz4740-i2s" }, + { /* sentinel */ } +}; +#endif + static int jz4740_i2s_dev_probe(struct platform_device *pdev) { struct jz4740_i2s *i2s; @@ -455,6 +473,7 @@ static struct platform_driver jz4740_i2s_driver = { .probe = jz4740_i2s_dev_probe, .driver = { .name = "jz4740-i2s", + .of_match_table = of_match_ptr(jz4740_of_matches) }, }; |