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-rw-r--r--sound/soc/codecs/Kconfig4
-rw-r--r--sound/soc/codecs/Makefile2
-rw-r--r--sound/soc/codecs/max98357a.c138
-rw-r--r--sound/soc/codecs/rt286.c31
-rw-r--r--sound/soc/codecs/rt286.h7
-rw-r--r--sound/soc/codecs/rt5645.c81
-rw-r--r--sound/soc/codecs/rt5645.h72
-rw-r--r--sound/soc/codecs/rt5670.c1
-rw-r--r--sound/soc/intel/Kconfig11
-rw-r--r--sound/soc/intel/Makefile2
-rw-r--r--sound/soc/intel/cht_bsw_rt5645.c326
-rw-r--r--sound/soc/intel/sst-haswell-ipc.c168
-rw-r--r--sound/soc/intel/sst-haswell-ipc.h31
-rw-r--r--sound/soc/intel/sst-haswell-pcm.c70
-rw-r--r--sound/soc/intel/sst/sst.h3
-rw-r--r--sound/soc/intel/sst/sst_acpi.c6
-rw-r--r--sound/soc/jz4740/jz4740-i2s.c21
-rw-r--r--sound/soc/samsung/Kconfig11
-rw-r--r--sound/soc/samsung/Makefile2
-rw-r--r--sound/soc/samsung/goni_wm8994.c289
-rw-r--r--sound/soc/soc-core.c2
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(&params->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;
-
- 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)
@@ -1084,42 +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)
{
@@ -1133,16 +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)
@@ -1451,48 +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;
@@ -1630,30 +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,
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 ad7f4a51e138..78fa01be57f2 100644
--- a/sound/soc/intel/sst-haswell-pcm.c
+++ b/sound/soc/intel/sst-haswell-pcm.c
@@ -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 */
@@ -1035,12 +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;
}
@@ -1051,7 +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);
@@ -1091,7 +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);
@@ -1139,41 +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 < ARRAY_SIZE(mod_map); i++) {
- pcm_data = &pdata->pcm[mod_map[i].dai_id][mod_map[i].stream];
+ 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 < 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, "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/sst.h b/sound/soc/intel/sst/sst.h
index 7f4bbfcbc6f5..562bc483d6b7 100644
--- a/sound/soc/intel/sst/sst.h
+++ b/sound/soc/intel/sst/sst.h
@@ -58,6 +58,7 @@ enum sst_algo_ops {
#define SST_BLOCK_TIMEOUT 1000
#define FW_SIGNATURE_SIZE 4
+#define FW_NAME_SIZE 32
/* stream states */
enum sst_stream_states {
@@ -426,7 +427,7 @@ struct intel_sst_drv {
* Holder for firmware name. Due to async call it needs to be
* persistent till worker thread gets called
*/
- char firmware_name[20];
+ char firmware_name[FW_NAME_SIZE];
};
/* misc definitions */
diff --git a/sound/soc/intel/sst/sst_acpi.c b/sound/soc/intel/sst/sst_acpi.c
index 43bc1c4b9207..b782dfdcdbba 100644
--- a/sound/soc/intel/sst/sst_acpi.c
+++ b/sound/soc/intel/sst/sst_acpi.c
@@ -47,7 +47,7 @@ struct sst_machines {
char board[32];
char machine[32];
void (*machine_quirk)(void);
- char firmware[32];
+ char firmware[FW_NAME_SIZE];
struct sst_platform_info *pdata;
};
@@ -350,9 +350,9 @@ static struct sst_machines sst_acpi_bytcr[] = {
/* Cherryview-based platforms: CherryTrail and Braswell */
static struct sst_machines sst_acpi_chv[] = {
- {"10EC5670", "cht-bsw", "cht-bsw-rt5672", NULL, "fw_sst_22a8.bin",
+ {"10EC5670", "cht-bsw", "cht-bsw-rt5672", NULL, "intel/fw_sst_22a8.bin",
&chv_platform_data },
- {"10EC5645", "cht-bsw", "cht-bsw-rt5645", NULL, "fw_sst_22a8.bin",
+ {"10EC5645", "cht-bsw", "cht-bsw-rt5645", NULL, "intel/fw_sst_22a8.bin",
&chv_platform_data },
{},
};
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)
},
};
diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig
index e817a2f43ea8..3cebf6ca03df 100644
--- a/sound/soc/samsung/Kconfig
+++ b/sound/soc/samsung/Kconfig
@@ -146,17 +146,6 @@ config SND_SOC_SMARTQ
select SND_SAMSUNG_I2S
select SND_SOC_WM8750
-config SND_SOC_GONI_AQUILA_WM8994
- tristate "SoC I2S Audio support for AQUILA/GONI - WM8994"
- depends on SND_SOC_SAMSUNG && (MACH_GONI || MACH_AQUILA)
- depends on I2C=y
- select SND_SAMSUNG_I2S
- select MFD_WM8994
- select SND_SOC_WM8994
- help
- Say Y if you want to add support for SoC audio on goni or aquila
- with the WM8994.
-
config SND_SOC_SAMSUNG_SMDK_SPDIF
tristate "SoC S/PDIF Audio support for SMDK"
depends on SND_SOC_SAMSUNG
diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile
index 31e3dba7e3b5..052fe71be518 100644
--- a/sound/soc/samsung/Makefile
+++ b/sound/soc/samsung/Makefile
@@ -35,7 +35,6 @@ snd-soc-smdk-wm8994-objs := smdk_wm8994.o
snd-soc-snow-objs := snow.o
snd-soc-smdk-wm9713-objs := smdk_wm9713.o
snd-soc-s3c64xx-smartq-wm8987-objs := smartq_wm8987.o
-snd-soc-goni-wm8994-objs := goni_wm8994.o
snd-soc-smdk-spdif-objs := smdk_spdif.o
snd-soc-smdk-wm8580pcm-objs := smdk_wm8580pcm.o
snd-soc-smdk-wm8994pcm-objs := smdk_wm8994pcm.o
@@ -63,7 +62,6 @@ obj-$(CONFIG_SND_SOC_SNOW) += snd-soc-snow.o
obj-$(CONFIG_SND_SOC_SAMSUNG_SMDK_WM9713) += snd-soc-smdk-wm9713.o
obj-$(CONFIG_SND_SOC_SMARTQ) += snd-soc-s3c64xx-smartq-wm8987.o
obj-$(CONFIG_SND_SOC_SAMSUNG_SMDK_SPDIF) += snd-soc-smdk-spdif.o
-obj-$(CONFIG_SND_SOC_GONI_AQUILA_WM8994) += snd-soc-goni-wm8994.o
obj-$(CONFIG_SND_SOC_SMDK_WM8580_PCM) += snd-soc-smdk-wm8580pcm.o
obj-$(CONFIG_SND_SOC_SMDK_WM8994_PCM) += snd-soc-smdk-wm8994pcm.o
obj-$(CONFIG_SND_SOC_SPEYSIDE) += snd-soc-speyside.o
diff --git a/sound/soc/samsung/goni_wm8994.c b/sound/soc/samsung/goni_wm8994.c
deleted file mode 100644
index fad56b9e7369..000000000000
--- a/sound/soc/samsung/goni_wm8994.c
+++ /dev/null
@@ -1,289 +0,0 @@
-/*
- * goni_wm8994.c
- *
- * Copyright (C) 2010 Samsung Electronics Co.Ltd
- * Author: Chanwoo Choi <cw00.choi@samsung.com>
- *
- * This program is free software; you can redistribute it and/or modify it
- * under the terms of the GNU General Public License as published by the
- * Free Software Foundation; either version 2 of the License, or (at your
- * option) any later version.
- *
- */
-
-#include <linux/module.h>
-#include <sound/soc.h>
-#include <sound/jack.h>
-
-#include <asm/mach-types.h>
-#include <mach/gpio-samsung.h>
-
-#include "../codecs/wm8994.h"
-
-#define MACHINE_NAME 0
-#define CPU_VOICE_DAI 1
-
-static const char *aquila_str[] = {
- [MACHINE_NAME] = "aquila",
- [CPU_VOICE_DAI] = "aquila-voice-dai",
-};
-
-static struct snd_soc_card goni;
-static struct platform_device *goni_snd_device;
-
-/* 3.5 pie jack */
-static struct snd_soc_jack jack;
-
-/* 3.5 pie jack detection DAPM pins */
-static struct snd_soc_jack_pin jack_pins[] = {
- {
- .pin = "Headset Mic",
- .mask = SND_JACK_MICROPHONE,
- }, {
- .pin = "Headset Stereophone",
- .mask = SND_JACK_HEADPHONE | SND_JACK_MECHANICAL |
- SND_JACK_AVOUT,
- },
-};
-
-/* 3.5 pie jack detection gpios */
-static struct snd_soc_jack_gpio jack_gpios[] = {
- {
- .gpio = S5PV210_GPH0(6),
- .name = "DET_3.5",
- .report = SND_JACK_HEADSET | SND_JACK_MECHANICAL |
- SND_JACK_AVOUT,
- .debounce_time = 200,
- },
-};
-
-static const struct snd_soc_dapm_widget goni_dapm_widgets[] = {
- SND_SOC_DAPM_SPK("Ext Left Spk", NULL),
- SND_SOC_DAPM_SPK("Ext Right Spk", NULL),
- SND_SOC_DAPM_SPK("Ext Rcv", NULL),
- SND_SOC_DAPM_HP("Headset Stereophone", NULL),
- SND_SOC_DAPM_MIC("Headset Mic", NULL),
- SND_SOC_DAPM_MIC("Main Mic", NULL),
- SND_SOC_DAPM_MIC("2nd Mic", NULL),
- SND_SOC_DAPM_LINE("Radio In", NULL),
-};
-
-static const struct snd_soc_dapm_route goni_dapm_routes[] = {
- {"Ext Left Spk", NULL, "SPKOUTLP"},
- {"Ext Left Spk", NULL, "SPKOUTLN"},
-
- {"Ext Right Spk", NULL, "SPKOUTRP"},
- {"Ext Right Spk", NULL, "SPKOUTRN"},
-
- {"Ext Rcv", NULL, "HPOUT2N"},
- {"Ext Rcv", NULL, "HPOUT2P"},
-
- {"Headset Stereophone", NULL, "HPOUT1L"},
- {"Headset Stereophone", NULL, "HPOUT1R"},
-
- {"IN1RN", NULL, "Headset Mic"},
- {"IN1RP", NULL, "Headset Mic"},
-
- {"IN1RN", NULL, "2nd Mic"},
- {"IN1RP", NULL, "2nd Mic"},
-
- {"IN1LN", NULL, "Main Mic"},
- {"IN1LP", NULL, "Main Mic"},
-
- {"IN2LN", NULL, "Radio In"},
- {"IN2RN", NULL, "Radio In"},
-};
-
-static int goni_wm8994_init(struct snd_soc_pcm_runtime *rtd)
-{
- struct snd_soc_codec *codec = rtd->codec;
- struct snd_soc_dapm_context *dapm = &codec->dapm;
- int ret;
-
- /* set endpoints to not connected */
- snd_soc_dapm_nc_pin(dapm, "IN2LP:VXRN");
- snd_soc_dapm_nc_pin(dapm, "IN2RP:VXRP");
- snd_soc_dapm_nc_pin(dapm, "LINEOUT1N");
- snd_soc_dapm_nc_pin(dapm, "LINEOUT1P");
- snd_soc_dapm_nc_pin(dapm, "LINEOUT2N");
- snd_soc_dapm_nc_pin(dapm, "LINEOUT2P");
-
- if (machine_is_aquila()) {
- snd_soc_dapm_nc_pin(dapm, "SPKOUTRN");
- snd_soc_dapm_nc_pin(dapm, "SPKOUTRP");
- }
-
- /* Headset jack detection */
- ret = snd_soc_jack_new(codec, "Headset Jack",
- SND_JACK_HEADSET | SND_JACK_MECHANICAL | SND_JACK_AVOUT,
- &jack);
- if (ret)
- return ret;
-
- ret = snd_soc_jack_add_pins(&jack, ARRAY_SIZE(jack_pins), jack_pins);
- if (ret)
- return ret;
-
- ret = snd_soc_jack_add_gpios(&jack, ARRAY_SIZE(jack_gpios), jack_gpios);
- if (ret)
- return ret;
-
- return 0;
-}
-
-static int goni_hifi_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;
- unsigned int pll_out = 24000000;
- int ret = 0;
-
- /* set the codec FLL */
- ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL1, 0, pll_out,
- params_rate(params) * 256);
- if (ret < 0)
- return ret;
-
- /* set the codec system clock */
- ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_FLL1,
- params_rate(params) * 256, SND_SOC_CLOCK_IN);
- if (ret < 0)
- return ret;
-
- return 0;
-}
-
-static struct snd_soc_ops goni_hifi_ops = {
- .hw_params = goni_hifi_hw_params,
-};
-
-static int goni_voice_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;
- unsigned int pll_out = 24000000;
- int ret = 0;
-
- if (params_rate(params) != 8000)
- return -EINVAL;
-
- /* set the codec FLL */
- ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL2, 0, pll_out,
- params_rate(params) * 256);
- if (ret < 0)
- return ret;
-
- /* set the codec system clock */
- ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_FLL2,
- params_rate(params) * 256, SND_SOC_CLOCK_IN);
- if (ret < 0)
- return ret;
-
- return 0;
-}
-
-static struct snd_soc_dai_driver voice_dai = {
- .name = "goni-voice-dai",
- .id = 0,
- .playback = {
- .channels_min = 1,
- .channels_max = 2,
- .rates = SNDRV_PCM_RATE_8000,
- .formats = SNDRV_PCM_FMTBIT_S16_LE,},
- .capture = {
- .channels_min = 1,
- .channels_max = 2,
- .rates = SNDRV_PCM_RATE_8000,
- .formats = SNDRV_PCM_FMTBIT_S16_LE,},
-};
-
-static const struct snd_soc_component_driver voice_component = {
- .name = "goni-voice",
-};
-
-static struct snd_soc_ops goni_voice_ops = {
- .hw_params = goni_voice_hw_params,
-};
-
-static struct snd_soc_dai_link goni_dai[] = {
-{
- .name = "WM8994",
- .stream_name = "WM8994 HiFi",
- .cpu_dai_name = "samsung-i2s.0",
- .codec_dai_name = "wm8994-aif1",
- .platform_name = "samsung-i2s.0",
- .codec_name = "wm8994-codec.0-001a",
- .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
- SND_SOC_DAIFMT_CBM_CFM,
- .init = goni_wm8994_init,
- .ops = &goni_hifi_ops,
-}, {
- .name = "WM8994 Voice",
- .stream_name = "Voice",
- .cpu_dai_name = "goni-voice-dai",
- .codec_dai_name = "wm8994-aif2",
- .codec_name = "wm8994-codec.0-001a",
- .dai_fmt = SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_IB_IF |
- SND_SOC_DAIFMT_CBM_CFM,
- .ops = &goni_voice_ops,
-},
-};
-
-static struct snd_soc_card goni = {
- .name = "goni",
- .owner = THIS_MODULE,
- .dai_link = goni_dai,
- .num_links = ARRAY_SIZE(goni_dai),
-
- .dapm_widgets = goni_dapm_widgets,
- .num_dapm_widgets = ARRAY_SIZE(goni_dapm_widgets),
- .dapm_routes = goni_dapm_routes,
- .num_dapm_routes = ARRAY_SIZE(goni_dapm_routes),
-};
-
-static int __init goni_init(void)
-{
- int ret;
-
- if (machine_is_aquila()) {
- voice_dai.name = aquila_str[CPU_VOICE_DAI];
- goni_dai[1].cpu_dai_name = aquila_str[CPU_VOICE_DAI];
- goni.name = aquila_str[MACHINE_NAME];
- } else if (!machine_is_goni())
- return -ENODEV;
-
- goni_snd_device = platform_device_alloc("soc-audio", -1);
- if (!goni_snd_device)
- return -ENOMEM;
-
- /* register voice DAI here */
- ret = devm_snd_soc_register_component(&goni_snd_device->dev,
- &voice_component, &voice_dai, 1);
- if (ret) {
- platform_device_put(goni_snd_device);
- return ret;
- }
-
- platform_set_drvdata(goni_snd_device, &goni);
- ret = platform_device_add(goni_snd_device);
-
- if (ret)
- platform_device_put(goni_snd_device);
-
- return ret;
-}
-
-static void __exit goni_exit(void)
-{
- platform_device_unregister(goni_snd_device);
-}
-
-module_init(goni_init);
-module_exit(goni_exit);
-
-/* Module information */
-MODULE_DESCRIPTION("ALSA SoC WM8994 GONI(S5PV210)");
-MODULE_AUTHOR("Chanwoo Choi <cw00.choi@samsung.com>");
-MODULE_LICENSE("GPL");
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index d94434d6138e..30579ca5bacb 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -2419,8 +2419,8 @@ int snd_soc_unregister_card(struct snd_soc_card *card)
card->instantiated = false;
snd_soc_dapm_shutdown(card);
soc_cleanup_card_resources(card);
+ dev_dbg(card->dev, "ASoC: Unregistered card '%s'\n", card->name);
}
- dev_dbg(card->dev, "ASoC: Unregistered card '%s'\n", card->name);
return 0;
}
generated by cgit v1.2.3-55-g7522 (git 2.39.0) at 2024-07-19 21:53:38 +0200