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-rw-r--r--Documentation/devicetree/bindings/sound/atmel-classd.txt6
-rw-r--r--Documentation/devicetree/bindings/sound/da7218.txt104
-rw-r--r--include/sound/da7218.h109
-rw-r--r--sound/soc/atmel/atmel-classd.c12
-rw-r--r--sound/soc/codecs/Kconfig4
-rw-r--r--sound/soc/codecs/Makefile2
-rw-r--r--sound/soc/codecs/da7218.c3314
-rw-r--r--sound/soc/codecs/da7218.h1414
-rw-r--r--sound/soc/fsl/imx-pcm-dma.c2
-rw-r--r--sound/soc/intel/baytrail/sst-baytrail-pcm.c2
-rw-r--r--sound/soc/omap/omap-hdmi-audio.c2
11 files changed, 4967 insertions, 4 deletions
diff --git a/Documentation/devicetree/bindings/sound/atmel-classd.txt b/Documentation/devicetree/bindings/sound/atmel-classd.txt
index 0018451c4351..549e701cb7a1 100644
--- a/Documentation/devicetree/bindings/sound/atmel-classd.txt
+++ b/Documentation/devicetree/bindings/sound/atmel-classd.txt
@@ -16,6 +16,10 @@ Required properties:
Required elements: "pclk", "gclk" and "aclk".
- clocks
Please refer to clock-bindings.txt.
+- assigned-clocks
+ Should be <&classd_gclk>.
+- assigned-clock-parents
+ Should be <&audio_pll_pmc>.
Optional properties:
- pinctrl-names, pinctrl-0
@@ -43,6 +47,8 @@ classd: classd@fc048000 {
dma-names = "tx";
clocks = <&classd_clk>, <&classd_gclk>, <&audio_pll_pmc>;
clock-names = "pclk", "gclk", "aclk";
+ assigned-clocks = <&classd_gclk>;
+ assigned-clock-parents = <&audio_pll_pmc>;
pinctrl-names = "default";
pinctrl-0 = <&pinctrl_classd_default>;
diff --git a/Documentation/devicetree/bindings/sound/da7218.txt b/Documentation/devicetree/bindings/sound/da7218.txt
new file mode 100644
index 000000000000..5ca5a709b6aa
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/da7218.txt
@@ -0,0 +1,104 @@
+Dialog Semiconductor DA7218 Audio Codec bindings
+
+DA7218 is an audio codec with HP detect feature.
+
+======
+
+Required properties:
+- compatible : Should be "dlg,da7217" or "dlg,da7218"
+- reg: Specifies the I2C slave address
+
+- VDD-supply: VDD power supply for the device
+- VDDMIC-supply: VDDMIC power supply for the device
+- VDDIO-supply: VDDIO power supply for the device
+ (See Documentation/devicetree/bindings/regulator/regulator.txt for further
+ information relating to regulators)
+
+Optional properties:
+- interrupt-parent: Specifies the phandle of the interrupt controller to which
+ the IRQs from DA7218 are delivered to.
+- interrupts: IRQ line info for DA7218 chip.
+ (See Documentation/devicetree/bindings/interrupt-controller/interrupts.txt for
+ further information relating to interrupt properties)
+- interrupt-names : Name associated with interrupt line. Should be "wakeup" if
+ interrupt is to be used to wake system, otherwise "irq" should be used.
+- wakeup-source: Flag to indicate this device can wake system (suspend/resume).
+
+- clocks : phandle and clock specifier for codec MCLK.
+- clock-names : Clock name string for 'clocks' attribute, should be "mclk".
+
+- dlg,micbias1-lvl-millivolt : Voltage (mV) for Mic Bias 1
+ [<1200>, <1600>, <1800>, <2000>, <2200>, <2400>, <2600>, <2800>, <3000>]
+- dlg,micbias2-lvl-millivolt : Voltage (mV) for Mic Bias 2
+ [<1200>, <1600>, <1800>, <2000>, <2200>, <2400>, <2600>, <2800>, <3000>]
+- dlg,mic1-amp-in-sel : Mic1 input source type
+ ["diff", "se_p", "se_n"]
+- dlg,mic2-amp-in-sel : Mic2 input source type
+ ["diff", "se_p", "se_n"]
+- dlg,dmic1-data-sel : DMIC1 channel select based on clock edge.
+ ["lrise_rfall", "lfall_rrise"]
+- dlg,dmic1-samplephase : When to sample audio from DMIC1.
+ ["on_clkedge", "between_clkedge"]
+- dlg,dmic1-clkrate-hz : DMic1 clock frequency (Hz).
+ [<1500000>, <3000000>]
+- dlg,dmic2-data-sel : DMic2 channel select based on clock edge.
+ ["lrise_rfall", "lfall_rrise"]
+- dlg,dmic2-samplephase : When to sample audio from DMic2.
+ ["on_clkedge", "between_clkedge"]
+- dlg,dmic2-clkrate-hz : DMic2 clock frequency (Hz).
+ [<1500000>, <3000000>]
+- dlg,hp-diff-single-supply : Boolean flag, use single supply for HP
+ (DA7217 only)
+
+======
+
+Optional Child node - 'da7218_hpldet' (DA7218 only):
+
+Optional properties:
+- dlg,jack-rate-us : Time between jack detect measurements (us)
+ [<5>, <10>, <20>, <40>, <80>, <160>, <320>, <640>]
+- dlg,jack-debounce : Number of debounce measurements taken for jack detect
+ [<0>, <2>, <3>, <4>]
+- dlg,jack-threshold-pct : Threshold level for jack detection (% of VDD)
+ [<84>, <88>, <92>, <96>]
+- dlg,comp-inv : Boolean flag, invert comparator output
+- dlg,hyst : Boolean flag, enable hysteresis
+- dlg,discharge : Boolean flag, auto discharge of Mic Bias on jack removal
+
+======
+
+Example:
+
+ codec: da7218@1a {
+ compatible = "dlg,da7218";
+ reg = <0x1a>;
+ interrupt-parent = <&gpio6>;
+ interrupts = <11 IRQ_TYPE_LEVEL_HIGH>;
+ wakeup-source;
+
+ VDD-supply = <&reg_audio>;
+ VDDMIC-supply = <&reg_audio>;
+ VDDIO-supply = <&reg_audio>;
+
+ clocks = <&clks 201>;
+ clock-names = "mclk";
+
+ dlg,micbias1-lvl-millivolt = <2600>;
+ dlg,micbias2-lvl-millivolt = <2600>;
+ dlg,mic1-amp-in-sel = "diff";
+ dlg,mic2-amp-in-sel = "diff";
+
+ dlg,dmic1-data-sel = "lrise_rfall";
+ dlg,dmic1-samplephase = "on_clkedge";
+ dlg,dmic1-clkrate-hz = <3000000>;
+ dlg,dmic2-data-sel = "lrise_rfall";
+ dlg,dmic2-samplephase = "on_clkedge";
+ dlg,dmic2-clkrate-hz = <3000000>;
+
+ da7218_hpldet {
+ dlg,jack-rate-us = <40>;
+ dlg,jack-debounce = <2>;
+ dlg,jack-threshold-pct = <84>;
+ dlg,hyst;
+ };
+ };
diff --git a/include/sound/da7218.h b/include/sound/da7218.h
new file mode 100644
index 000000000000..0dbb818ac116
--- /dev/null
+++ b/include/sound/da7218.h
@@ -0,0 +1,109 @@
+/*
+ * da7218.h - DA7218 ASoC Codec Driver Platform Data
+ *
+ * Copyright (c) 2015 Dialog Semiconductor
+ *
+ * Author: Adam Thomson <Adam.Thomson.Opensource@diasemi.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.
+ */
+
+#ifndef _DA7218_PDATA_H
+#define _DA7218_PDATA_H
+
+/* Mic Bias */
+enum da7218_micbias_voltage {
+ DA7218_MICBIAS_1_2V = -1,
+ DA7218_MICBIAS_1_6V,
+ DA7218_MICBIAS_1_8V,
+ DA7218_MICBIAS_2_0V,
+ DA7218_MICBIAS_2_2V,
+ DA7218_MICBIAS_2_4V,
+ DA7218_MICBIAS_2_6V,
+ DA7218_MICBIAS_2_8V,
+ DA7218_MICBIAS_3_0V,
+};
+
+enum da7218_mic_amp_in_sel {
+ DA7218_MIC_AMP_IN_SEL_DIFF = 0,
+ DA7218_MIC_AMP_IN_SEL_SE_P,
+ DA7218_MIC_AMP_IN_SEL_SE_N,
+};
+
+/* DMIC */
+enum da7218_dmic_data_sel {
+ DA7218_DMIC_DATA_LRISE_RFALL = 0,
+ DA7218_DMIC_DATA_LFALL_RRISE,
+};
+
+enum da7218_dmic_samplephase {
+ DA7218_DMIC_SAMPLE_ON_CLKEDGE = 0,
+ DA7218_DMIC_SAMPLE_BETWEEN_CLKEDGE,
+};
+
+enum da7218_dmic_clk_rate {
+ DA7218_DMIC_CLK_3_0MHZ = 0,
+ DA7218_DMIC_CLK_1_5MHZ,
+};
+
+/* Headphone Detect */
+enum da7218_hpldet_jack_rate {
+ DA7218_HPLDET_JACK_RATE_5US = 0,
+ DA7218_HPLDET_JACK_RATE_10US,
+ DA7218_HPLDET_JACK_RATE_20US,
+ DA7218_HPLDET_JACK_RATE_40US,
+ DA7218_HPLDET_JACK_RATE_80US,
+ DA7218_HPLDET_JACK_RATE_160US,
+ DA7218_HPLDET_JACK_RATE_320US,
+ DA7218_HPLDET_JACK_RATE_640US,
+};
+
+enum da7218_hpldet_jack_debounce {
+ DA7218_HPLDET_JACK_DEBOUNCE_OFF = 0,
+ DA7218_HPLDET_JACK_DEBOUNCE_2,
+ DA7218_HPLDET_JACK_DEBOUNCE_3,
+ DA7218_HPLDET_JACK_DEBOUNCE_4,
+};
+
+enum da7218_hpldet_jack_thr {
+ DA7218_HPLDET_JACK_THR_84PCT = 0,
+ DA7218_HPLDET_JACK_THR_88PCT,
+ DA7218_HPLDET_JACK_THR_92PCT,
+ DA7218_HPLDET_JACK_THR_96PCT,
+};
+
+struct da7218_hpldet_pdata {
+ enum da7218_hpldet_jack_rate jack_rate;
+ enum da7218_hpldet_jack_debounce jack_debounce;
+ enum da7218_hpldet_jack_thr jack_thr;
+ bool comp_inv;
+ bool hyst;
+ bool discharge;
+};
+
+struct da7218_pdata {
+ /* Mic */
+ enum da7218_micbias_voltage micbias1_lvl;
+ enum da7218_micbias_voltage micbias2_lvl;
+ enum da7218_mic_amp_in_sel mic1_amp_in_sel;
+ enum da7218_mic_amp_in_sel mic2_amp_in_sel;
+
+ /* DMIC */
+ enum da7218_dmic_data_sel dmic1_data_sel;
+ enum da7218_dmic_data_sel dmic2_data_sel;
+ enum da7218_dmic_samplephase dmic1_samplephase;
+ enum da7218_dmic_samplephase dmic2_samplephase;
+ enum da7218_dmic_clk_rate dmic1_clk_rate;
+ enum da7218_dmic_clk_rate dmic2_clk_rate;
+
+ /* HP Diff Supply - DA7217 only */
+ bool hp_diff_single_supply;
+
+ /* HP Detect - DA7218 only */
+ struct da7218_hpldet_pdata *hpldet_pdata;
+};
+
+#endif /* _DA7218_PDATA_H */
diff --git a/sound/soc/atmel/atmel-classd.c b/sound/soc/atmel/atmel-classd.c
index f3ffb39bfe27..6107de9c538b 100644
--- a/sound/soc/atmel/atmel-classd.c
+++ b/sound/soc/atmel/atmel-classd.c
@@ -106,7 +106,7 @@ static const struct snd_pcm_hardware atmel_classd_hw = {
.rates = ATMEL_CLASSD_RATES,
.rate_min = 8000,
.rate_max = 96000,
- .channels_min = 2,
+ .channels_min = 1,
.channels_max = 2,
.buffer_bytes_max = 64 * 1024,
.period_bytes_min = 256,
@@ -145,7 +145,7 @@ static const struct snd_soc_dai_ops atmel_classd_cpu_dai_ops = {
static struct snd_soc_dai_driver atmel_classd_cpu_dai = {
.playback = {
- .channels_min = 2,
+ .channels_min = 1,
.channels_max = 2,
.rates = ATMEL_CLASSD_RATES,
.formats = SNDRV_PCM_FMTBIT_S16_LE,},
@@ -171,9 +171,13 @@ atmel_classd_platform_configure_dma(struct snd_pcm_substream *substream,
return -EINVAL;
}
+ if (params_channels(params) == 1)
+ slave_config->dst_addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES;
+ else
+ slave_config->dst_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES;
+
slave_config->direction = DMA_MEM_TO_DEV;
slave_config->dst_addr = dd->phy_base + CLASSD_THR;
- slave_config->dst_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES;
slave_config->dst_maxburst = 1;
slave_config->src_maxburst = 1;
slave_config->device_fc = false;
@@ -486,7 +490,7 @@ static struct snd_soc_dai_driver atmel_classd_codec_dai = {
.name = ATMEL_CLASSD_CODEC_DAI_NAME,
.playback = {
.stream_name = "Playback",
- .channels_min = 2,
+ .channels_min = 1,
.channels_max = 2,
.rates = ATMEL_CLASSD_RATES,
.formats = SNDRV_PCM_FMTBIT_S16_LE,
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index efdd0f8e2e9c..05fb938c7704 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -59,6 +59,7 @@ config SND_SOC_ALL_CODECS
select SND_SOC_CX20442 if TTY
select SND_SOC_DA7210 if SND_SOC_I2C_AND_SPI
select SND_SOC_DA7213 if I2C
+ select SND_SOC_DA7218 if I2C
select SND_SOC_DA7219 if I2C
select SND_SOC_DA732X if I2C
select SND_SOC_DA9055 if I2C
@@ -450,6 +451,9 @@ config SND_SOC_DA7210
config SND_SOC_DA7213
tristate
+config SND_SOC_DA7218
+ tristate
+
config SND_SOC_DA7219
tristate
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 463d7d0880b7..266c6be0c7f5 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -51,6 +51,7 @@ snd-soc-cs47l24-objs := cs47l24.o
snd-soc-cx20442-objs := cx20442.o
snd-soc-da7210-objs := da7210.o
snd-soc-da7213-objs := da7213.o
+snd-soc-da7218-objs := da7218.o
snd-soc-da7219-objs := da7219.o da7219-aad.o
snd-soc-da732x-objs := da732x.o
snd-soc-da9055-objs := da9055.o
@@ -250,6 +251,7 @@ obj-$(CONFIG_SND_SOC_CS47L24) += snd-soc-cs47l24.o
obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o
obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o
obj-$(CONFIG_SND_SOC_DA7213) += snd-soc-da7213.o
+obj-$(CONFIG_SND_SOC_DA7218) += snd-soc-da7218.o
obj-$(CONFIG_SND_SOC_DA7219) += snd-soc-da7219.o
obj-$(CONFIG_SND_SOC_DA732X) += snd-soc-da732x.o
obj-$(CONFIG_SND_SOC_DA9055) += snd-soc-da9055.o
diff --git a/sound/soc/codecs/da7218.c b/sound/soc/codecs/da7218.c
new file mode 100644
index 000000000000..72686517ff54
--- /dev/null
+++ b/sound/soc/codecs/da7218.c
@@ -0,0 +1,3314 @@
+/*
+ * da7218.c - DA7218 ALSA SoC Codec Driver
+ *
+ * Copyright (c) 2015 Dialog Semiconductor
+ *
+ * Author: Adam Thomson <Adam.Thomson.Opensource@diasemi.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/clk.h>
+#include <linux/i2c.h>
+#include <linux/of_device.h>
+#include <linux/regmap.h>
+#include <linux/slab.h>
+#include <linux/pm.h>
+#include <linux/module.h>
+#include <linux/delay.h>
+#include <linux/regulator/consumer.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/jack.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+#include <asm/div64.h>
+
+#include <sound/da7218.h>
+#include "da7218.h"
+
+
+/*
+ * TLVs and Enums
+ */
+
+/* Input TLVs */
+static const DECLARE_TLV_DB_SCALE(da7218_mic_gain_tlv, -600, 600, 0);
+static const DECLARE_TLV_DB_SCALE(da7218_mixin_gain_tlv, -450, 150, 0);
+static const DECLARE_TLV_DB_SCALE(da7218_in_dig_gain_tlv, -8325, 75, 0);
+static const DECLARE_TLV_DB_SCALE(da7218_ags_trigger_tlv, -9000, 600, 0);
+static const DECLARE_TLV_DB_SCALE(da7218_ags_att_max_tlv, 0, 600, 0);
+static const DECLARE_TLV_DB_SCALE(da7218_alc_threshold_tlv, -9450, 150, 0);
+static const DECLARE_TLV_DB_SCALE(da7218_alc_gain_tlv, 0, 600, 0);
+static const DECLARE_TLV_DB_SCALE(da7218_alc_ana_gain_tlv, 0, 600, 0);
+
+/* Input/Output TLVs */
+static const DECLARE_TLV_DB_SCALE(da7218_dmix_gain_tlv, -4200, 150, 0);
+
+/* Output TLVs */
+static const DECLARE_TLV_DB_SCALE(da7218_dgs_trigger_tlv, -9450, 150, 0);
+static const DECLARE_TLV_DB_SCALE(da7218_dgs_anticlip_tlv, -4200, 600, 0);
+static const DECLARE_TLV_DB_SCALE(da7218_dgs_signal_tlv, -9000, 600, 0);
+static const DECLARE_TLV_DB_SCALE(da7218_out_eq_band_tlv, -1050, 150, 0);
+static const DECLARE_TLV_DB_SCALE(da7218_out_dig_gain_tlv, -8325, 75, 0);
+static const DECLARE_TLV_DB_SCALE(da7218_dac_ng_threshold_tlv, -10200, 600, 0);
+static const DECLARE_TLV_DB_SCALE(da7218_mixout_gain_tlv, -100, 50, 0);
+static const DECLARE_TLV_DB_SCALE(da7218_hp_gain_tlv, -5700, 150, 0);
+
+/* Input Enums */
+static const char * const da7218_alc_attack_rate_txt[] = {
+ "7.33/fs", "14.66/fs", "29.32/fs", "58.64/fs", "117.3/fs", "234.6/fs",
+ "469.1/fs", "938.2/fs", "1876/fs", "3753/fs", "7506/fs", "15012/fs",
+ "30024/fs",
+};
+
+static const struct soc_enum da7218_alc_attack_rate =
+ SOC_ENUM_SINGLE(DA7218_ALC_CTRL2, DA7218_ALC_ATTACK_SHIFT,
+ DA7218_ALC_ATTACK_MAX, da7218_alc_attack_rate_txt);
+
+static const char * const da7218_alc_release_rate_txt[] = {
+ "28.66/fs", "57.33/fs", "114.6/fs", "229.3/fs", "458.6/fs", "917.1/fs",
+ "1834/fs", "3668/fs", "7337/fs", "14674/fs", "29348/fs",
+};
+
+static const struct soc_enum da7218_alc_release_rate =
+ SOC_ENUM_SINGLE(DA7218_ALC_CTRL2, DA7218_ALC_RELEASE_SHIFT,
+ DA7218_ALC_RELEASE_MAX, da7218_alc_release_rate_txt);
+
+static const char * const da7218_alc_hold_time_txt[] = {
+ "62/fs", "124/fs", "248/fs", "496/fs", "992/fs", "1984/fs", "3968/fs",
+ "7936/fs", "15872/fs", "31744/fs", "63488/fs", "126976/fs",
+ "253952/fs", "507904/fs", "1015808/fs", "2031616/fs"
+};
+
+static const struct soc_enum da7218_alc_hold_time =
+ SOC_ENUM_SINGLE(DA7218_ALC_CTRL3, DA7218_ALC_HOLD_SHIFT,
+ DA7218_ALC_HOLD_MAX, da7218_alc_hold_time_txt);
+
+static const char * const da7218_alc_anticlip_step_txt[] = {
+ "0.034dB/fs", "0.068dB/fs", "0.136dB/fs", "0.272dB/fs",
+};
+
+static const struct soc_enum da7218_alc_anticlip_step =
+ SOC_ENUM_SINGLE(DA7218_ALC_ANTICLIP_CTRL,
+ DA7218_ALC_ANTICLIP_STEP_SHIFT,
+ DA7218_ALC_ANTICLIP_STEP_MAX,
+ da7218_alc_anticlip_step_txt);
+
+static const char * const da7218_integ_rate_txt[] = {
+ "1/4", "1/16", "1/256", "1/65536"
+};
+
+static const struct soc_enum da7218_integ_attack_rate =
+ SOC_ENUM_SINGLE(DA7218_ENV_TRACK_CTRL, DA7218_INTEG_ATTACK_SHIFT,
+ DA7218_INTEG_MAX, da7218_integ_rate_txt);
+
+static const struct soc_enum da7218_integ_release_rate =
+ SOC_ENUM_SINGLE(DA7218_ENV_TRACK_CTRL, DA7218_INTEG_RELEASE_SHIFT,
+ DA7218_INTEG_MAX, da7218_integ_rate_txt);
+
+/* Input/Output Enums */
+static const char * const da7218_gain_ramp_rate_txt[] = {
+ "Nominal Rate * 8", "Nominal Rate", "Nominal Rate / 8",
+ "Nominal Rate / 16",
+};
+
+static const struct soc_enum da7218_gain_ramp_rate =
+ SOC_ENUM_SINGLE(DA7218_GAIN_RAMP_CTRL, DA7218_GAIN_RAMP_RATE_SHIFT,
+ DA7218_GAIN_RAMP_RATE_MAX, da7218_gain_ramp_rate_txt);
+
+static const char * const da7218_hpf_mode_txt[] = {
+ "Disabled", "Audio", "Voice",
+};
+
+static const unsigned int da7218_hpf_mode_val[] = {
+ DA7218_HPF_DISABLED, DA7218_HPF_AUDIO_EN, DA7218_HPF_VOICE_EN,
+};
+
+static const struct soc_enum da7218_in1_hpf_mode =
+ SOC_VALUE_ENUM_SINGLE(DA7218_IN_1_HPF_FILTER_CTRL,
+ DA7218_HPF_MODE_SHIFT, DA7218_HPF_MODE_MASK,
+ DA7218_HPF_MODE_MAX, da7218_hpf_mode_txt,
+ da7218_hpf_mode_val);
+
+static const struct soc_enum da7218_in2_hpf_mode =
+ SOC_VALUE_ENUM_SINGLE(DA7218_IN_2_HPF_FILTER_CTRL,
+ DA7218_HPF_MODE_SHIFT, DA7218_HPF_MODE_MASK,
+ DA7218_HPF_MODE_MAX, da7218_hpf_mode_txt,
+ da7218_hpf_mode_val);
+
+static const struct soc_enum da7218_out1_hpf_mode =
+ SOC_VALUE_ENUM_SINGLE(DA7218_OUT_1_HPF_FILTER_CTRL,
+ DA7218_HPF_MODE_SHIFT, DA7218_HPF_MODE_MASK,
+ DA7218_HPF_MODE_MAX, da7218_hpf_mode_txt,
+ da7218_hpf_mode_val);
+
+static const char * const da7218_audio_hpf_corner_txt[] = {
+ "2Hz", "4Hz", "8Hz", "16Hz",
+};
+
+static const struct soc_enum da7218_in1_audio_hpf_corner =
+ SOC_ENUM_SINGLE(DA7218_IN_1_HPF_FILTER_CTRL,
+ DA7218_IN_1_AUDIO_HPF_CORNER_SHIFT,
+ DA7218_AUDIO_HPF_CORNER_MAX,
+ da7218_audio_hpf_corner_txt);
+
+static const struct soc_enum da7218_in2_audio_hpf_corner =
+ SOC_ENUM_SINGLE(DA7218_IN_2_HPF_FILTER_CTRL,
+ DA7218_IN_2_AUDIO_HPF_CORNER_SHIFT,
+ DA7218_AUDIO_HPF_CORNER_MAX,
+ da7218_audio_hpf_corner_txt);
+
+static const struct soc_enum da7218_out1_audio_hpf_corner =
+ SOC_ENUM_SINGLE(DA7218_OUT_1_HPF_FILTER_CTRL,
+ DA7218_OUT_1_AUDIO_HPF_CORNER_SHIFT,
+ DA7218_AUDIO_HPF_CORNER_MAX,
+ da7218_audio_hpf_corner_txt);
+
+static const char * const da7218_voice_hpf_corner_txt[] = {
+ "2.5Hz", "25Hz", "50Hz", "100Hz", "150Hz", "200Hz", "300Hz", "400Hz",
+};
+
+static const struct soc_enum da7218_in1_voice_hpf_corner =
+ SOC_ENUM_SINGLE(DA7218_IN_1_HPF_FILTER_CTRL,
+ DA7218_IN_1_VOICE_HPF_CORNER_SHIFT,
+ DA7218_VOICE_HPF_CORNER_MAX,
+ da7218_voice_hpf_corner_txt);
+
+static const struct soc_enum da7218_in2_voice_hpf_corner =
+ SOC_ENUM_SINGLE(DA7218_IN_2_HPF_FILTER_CTRL,
+ DA7218_IN_2_VOICE_HPF_CORNER_SHIFT,
+ DA7218_VOICE_HPF_CORNER_MAX,
+ da7218_voice_hpf_corner_txt);
+
+static const struct soc_enum da7218_out1_voice_hpf_corner =
+ SOC_ENUM_SINGLE(DA7218_OUT_1_HPF_FILTER_CTRL,
+ DA7218_OUT_1_VOICE_HPF_CORNER_SHIFT,
+ DA7218_VOICE_HPF_CORNER_MAX,
+ da7218_voice_hpf_corner_txt);
+
+static const char * const da7218_tonegen_dtmf_key_txt[] = {
+ "0", "1", "2", "3", "4", "5", "6", "7", "8", "9", "A", "B", "C", "D",
+ "*", "#"
+};
+
+static const struct soc_enum da7218_tonegen_dtmf_key =
+ SOC_ENUM_SINGLE(DA7218_TONE_GEN_CFG1, DA7218_DTMF_REG_SHIFT,
+ DA7218_DTMF_REG_MAX, da7218_tonegen_dtmf_key_txt);
+
+static const char * const da7218_tonegen_swg_sel_txt[] = {
+ "Sum", "SWG1", "SWG2", "SWG1_1-Cos"
+};
+
+static const struct soc_enum da7218_tonegen_swg_sel =
+ SOC_ENUM_SINGLE(DA7218_TONE_GEN_CFG2, DA7218_SWG_SEL_SHIFT,
+ DA7218_SWG_SEL_MAX, da7218_tonegen_swg_sel_txt);
+
+/* Output Enums */
+static const char * const da7218_dgs_rise_coeff_txt[] = {
+ "1/1", "1/16", "1/64", "1/256", "1/1024", "1/4096", "1/16384",
+};
+
+static const struct soc_enum da7218_dgs_rise_coeff =
+ SOC_ENUM_SINGLE(DA7218_DGS_RISE_FALL, DA7218_DGS_RISE_COEFF_SHIFT,
+ DA7218_DGS_RISE_COEFF_MAX, da7218_dgs_rise_coeff_txt);
+
+static const char * const da7218_dgs_fall_coeff_txt[] = {
+ "1/4", "1/16", "1/64", "1/256", "1/1024", "1/4096", "1/16384", "1/65536",
+};
+
+static const struct soc_enum da7218_dgs_fall_coeff =
+ SOC_ENUM_SINGLE(DA7218_DGS_RISE_FALL, DA7218_DGS_FALL_COEFF_SHIFT,
+ DA7218_DGS_FALL_COEFF_MAX, da7218_dgs_fall_coeff_txt);
+
+static const char * const da7218_dac_ng_setup_time_txt[] = {
+ "256 Samples", "512 Samples", "1024 Samples", "2048 Samples"
+};
+
+static const struct soc_enum da7218_dac_ng_setup_time =
+ SOC_ENUM_SINGLE(DA7218_DAC_NG_SETUP_TIME,
+ DA7218_DAC_NG_SETUP_TIME_SHIFT,
+ DA7218_DAC_NG_SETUP_TIME_MAX,
+ da7218_dac_ng_setup_time_txt);
+
+static const char * const da7218_dac_ng_rampup_txt[] = {
+ "0.22ms/dB", "0.0138ms/dB"
+};
+
+static const struct soc_enum da7218_dac_ng_rampup_rate =
+ SOC_ENUM_SINGLE(DA7218_DAC_NG_SETUP_TIME,
+ DA7218_DAC_NG_RAMPUP_RATE_SHIFT,
+ DA7218_DAC_NG_RAMPUP_RATE_MAX,
+ da7218_dac_ng_rampup_txt);
+
+static const char * const da7218_dac_ng_rampdown_txt[] = {
+ "0.88ms/dB", "14.08ms/dB"
+};
+
+static const struct soc_enum da7218_dac_ng_rampdown_rate =
+ SOC_ENUM_SINGLE(DA7218_DAC_NG_SETUP_TIME,
+ DA7218_DAC_NG_RAMPDN_RATE_SHIFT,
+ DA7218_DAC_NG_RAMPDN_RATE_MAX,
+ da7218_dac_ng_rampdown_txt);
+
+static const char * const da7218_cp_mchange_txt[] = {
+ "Largest Volume", "DAC Volume", "Signal Magnitude"
+};
+
+static const unsigned int da7218_cp_mchange_val[] = {
+ DA7218_CP_MCHANGE_LARGEST_VOL, DA7218_CP_MCHANGE_DAC_VOL,
+ DA7218_CP_MCHANGE_SIG_MAG
+};
+
+static const struct soc_enum da7218_cp_mchange =
+ SOC_VALUE_ENUM_SINGLE(DA7218_CP_CTRL, DA7218_CP_MCHANGE_SHIFT,
+ DA7218_CP_MCHANGE_REL_MASK, DA7218_CP_MCHANGE_MAX,
+ da7218_cp_mchange_txt, da7218_cp_mchange_val);
+
+static const char * const da7218_cp_fcontrol_txt[] = {
+ "1MHz", "500KHz", "250KHz", "125KHz", "63KHz", "0KHz"
+};
+
+static const struct soc_enum da7218_cp_fcontrol =
+ SOC_ENUM_SINGLE(DA7218_CP_DELAY, DA7218_CP_FCONTROL_SHIFT,
+ DA7218_CP_FCONTROL_MAX, da7218_cp_fcontrol_txt);
+
+static const char * const da7218_cp_tau_delay_txt[] = {
+ "0ms", "2ms", "4ms", "16ms", "64ms", "128ms", "256ms", "512ms"
+};
+
+static const struct soc_enum da7218_cp_tau_delay =
+ SOC_ENUM_SINGLE(DA7218_CP_DELAY, DA7218_CP_TAU_DELAY_SHIFT,
+ DA7218_CP_TAU_DELAY_MAX, da7218_cp_tau_delay_txt);
+
+/*
+ * Control Functions
+ */
+
+/* ALC */
+static void da7218_alc_calib(struct snd_soc_codec *codec)
+{
+ u8 mic_1_ctrl, mic_2_ctrl;
+ u8 mixin_1_ctrl, mixin_2_ctrl;
+ u8 in_1l_filt_ctrl, in_1r_filt_ctrl, in_2l_filt_ctrl, in_2r_filt_ctrl;
+ u8 in_1_hpf_ctrl, in_2_hpf_ctrl;
+ u8 calib_ctrl;
+ int i = 0;
+ bool calibrated = false;
+
+ /* Save current state of MIC control registers */
+ mic_1_ctrl = snd_soc_read(codec, DA7218_MIC_1_CTRL);
+ mic_2_ctrl = snd_soc_read(codec, DA7218_MIC_2_CTRL);
+
+ /* Save current state of input mixer control registers */
+ mixin_1_ctrl = snd_soc_read(codec, DA7218_MIXIN_1_CTRL);
+ mixin_2_ctrl = snd_soc_read(codec, DA7218_MIXIN_2_CTRL);
+
+ /* Save current state of input filter control registers */
+ in_1l_filt_ctrl = snd_soc_read(codec, DA7218_IN_1L_FILTER_CTRL);
+ in_1r_filt_ctrl = snd_soc_read(codec, DA7218_IN_1R_FILTER_CTRL);
+ in_2l_filt_ctrl = snd_soc_read(codec, DA7218_IN_2L_FILTER_CTRL);
+ in_2r_filt_ctrl = snd_soc_read(codec, DA7218_IN_2R_FILTER_CTRL);
+
+ /* Save current state of input HPF control registers */
+ in_1_hpf_ctrl = snd_soc_read(codec, DA7218_IN_1_HPF_FILTER_CTRL);
+ in_2_hpf_ctrl = snd_soc_read(codec, DA7218_IN_2_HPF_FILTER_CTRL);
+
+ /* Enable then Mute MIC PGAs */
+ snd_soc_update_bits(codec, DA7218_MIC_1_CTRL, DA7218_MIC_1_AMP_EN_MASK,
+ DA7218_MIC_1_AMP_EN_MASK);
+ snd_soc_update_bits(codec, DA7218_MIC_2_CTRL, DA7218_MIC_2_AMP_EN_MASK,
+ DA7218_MIC_2_AMP_EN_MASK);
+ snd_soc_update_bits(codec, DA7218_MIC_1_CTRL,
+ DA7218_MIC_1_AMP_MUTE_EN_MASK,
+ DA7218_MIC_1_AMP_MUTE_EN_MASK);
+ snd_soc_update_bits(codec, DA7218_MIC_2_CTRL,
+ DA7218_MIC_2_AMP_MUTE_EN_MASK,
+ DA7218_MIC_2_AMP_MUTE_EN_MASK);
+
+ /* Enable input mixers unmuted */
+ snd_soc_update_bits(codec, DA7218_MIXIN_1_CTRL,
+ DA7218_MIXIN_1_AMP_EN_MASK |
+ DA7218_MIXIN_1_AMP_MUTE_EN_MASK,
+ DA7218_MIXIN_1_AMP_EN_MASK);
+ snd_soc_update_bits(codec, DA7218_MIXIN_2_CTRL,
+ DA7218_MIXIN_2_AMP_EN_MASK |
+ DA7218_MIXIN_2_AMP_MUTE_EN_MASK,
+ DA7218_MIXIN_2_AMP_EN_MASK);
+
+ /* Enable input filters unmuted */
+ snd_soc_update_bits(codec, DA7218_IN_1L_FILTER_CTRL,
+ DA7218_IN_1L_FILTER_EN_MASK |
+ DA7218_IN_1L_MUTE_EN_MASK,
+ DA7218_IN_1L_FILTER_EN_MASK);
+ snd_soc_update_bits(codec, DA7218_IN_1R_FILTER_CTRL,
+ DA7218_IN_1R_FILTER_EN_MASK |
+ DA7218_IN_1R_MUTE_EN_MASK,
+ DA7218_IN_1R_FILTER_EN_MASK);
+ snd_soc_update_bits(codec, DA7218_IN_2L_FILTER_CTRL,
+ DA7218_IN_2L_FILTER_EN_MASK |
+ DA7218_IN_2L_MUTE_EN_MASK,
+ DA7218_IN_2L_FILTER_EN_MASK);
+ snd_soc_update_bits(codec, DA7218_IN_2R_FILTER_CTRL,
+ DA7218_IN_2R_FILTER_EN_MASK |
+ DA7218_IN_2R_MUTE_EN_MASK,
+ DA7218_IN_2R_FILTER_EN_MASK);
+
+ /*
+ * Make sure input HPFs voice mode is disabled, otherwise for sampling
+ * rates above 32KHz the ADC signals will be stopped and will cause
+ * calibration to lock up.
+ */
+ snd_soc_update_bits(codec, DA7218_IN_1_HPF_FILTER_CTRL,
+ DA7218_IN_1_VOICE_EN_MASK, 0);
+ snd_soc_update_bits(codec, DA7218_IN_2_HPF_FILTER_CTRL,
+ DA7218_IN_2_VOICE_EN_MASK, 0);
+
+ /* Perform auto calibration */
+ snd_soc_update_bits(codec, DA7218_CALIB_CTRL, DA7218_CALIB_AUTO_EN_MASK,
+ DA7218_CALIB_AUTO_EN_MASK);
+ do {
+ calib_ctrl = snd_soc_read(codec, DA7218_CALIB_CTRL);
+ if (calib_ctrl & DA7218_CALIB_AUTO_EN_MASK) {
+ ++i;
+ usleep_range(DA7218_ALC_CALIB_DELAY_MIN,
+ DA7218_ALC_CALIB_DELAY_MAX);
+ } else {
+ calibrated = true;
+ }
+
+ } while ((i < DA7218_ALC_CALIB_MAX_TRIES) && (!calibrated));
+
+ /* If auto calibration fails, disable DC offset, hybrid ALC */
+ if ((!calibrated) || (calib_ctrl & DA7218_CALIB_OVERFLOW_MASK)) {
+ dev_warn(codec->dev,
+ "ALC auto calibration failed - %s\n",
+ (calibrated) ? "overflow" : "timeout");
+ snd_soc_update_bits(codec, DA7218_CALIB_CTRL,
+ DA7218_CALIB_OFFSET_EN_MASK, 0);
+ snd_soc_update_bits(codec, DA7218_ALC_CTRL1,
+ DA7218_ALC_SYNC_MODE_MASK, 0);
+
+ } else {
+ /* Enable DC offset cancellation */
+ snd_soc_update_bits(codec, DA7218_CALIB_CTRL,
+ DA7218_CALIB_OFFSET_EN_MASK,
+ DA7218_CALIB_OFFSET_EN_MASK);
+
+ /* Enable ALC hybrid mode */
+ snd_soc_update_bits(codec, DA7218_ALC_CTRL1,
+ DA7218_ALC_SYNC_MODE_MASK,
+ DA7218_ALC_SYNC_MODE_CH1 |
+ DA7218_ALC_SYNC_MODE_CH2);
+ }
+
+ /* Restore input HPF control registers to original states */
+ snd_soc_write(codec, DA7218_IN_1_HPF_FILTER_CTRL, in_1_hpf_ctrl);
+ snd_soc_write(codec, DA7218_IN_2_HPF_FILTER_CTRL, in_2_hpf_ctrl);
+
+ /* Restore input filter control registers to original states */
+ snd_soc_write(codec, DA7218_IN_1L_FILTER_CTRL, in_1l_filt_ctrl);
+ snd_soc_write(codec, DA7218_IN_1R_FILTER_CTRL, in_1r_filt_ctrl);
+ snd_soc_write(codec, DA7218_IN_2L_FILTER_CTRL, in_2l_filt_ctrl);
+ snd_soc_write(codec, DA7218_IN_2R_FILTER_CTRL, in_2r_filt_ctrl);
+
+ /* Restore input mixer control registers to original state */
+ snd_soc_write(codec, DA7218_MIXIN_1_CTRL, mixin_1_ctrl);
+ snd_soc_write(codec, DA7218_MIXIN_2_CTRL, mixin_2_ctrl);
+
+ /* Restore MIC control registers to original states */
+ snd_soc_write(codec, DA7218_MIC_1_CTRL, mic_1_ctrl);
+ snd_soc_write(codec, DA7218_MIC_2_CTRL, mic_2_ctrl);
+}
+
+static int da7218_mixin_gain_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
+ struct da7218_priv *da7218 = snd_soc_codec_get_drvdata(codec);
+ int ret;
+
+ ret = snd_soc_put_volsw(kcontrol, ucontrol);
+
+ /*
+ * If ALC in operation and value of control has been updated,
+ * make sure calibrated offsets are updated.
+ */
+ if ((ret == 1) && (da7218->alc_en))
+ da7218_alc_calib(codec);
+
+ return ret;
+}
+
+static int da7218_alc_sw_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *) kcontrol->private_value;
+ struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
+ struct da7218_priv *da7218 = snd_soc_codec_get_drvdata(codec);
+ unsigned int lvalue = ucontrol->value.integer.value[0];
+ unsigned int rvalue = ucontrol->value.integer.value[1];
+ unsigned int lshift = mc->shift;
+ unsigned int rshift = mc->rshift;
+ unsigned int mask = (mc->max << lshift) | (mc->max << rshift);
+
+ /* Force ALC offset calibration if enabling ALC */
+ if ((lvalue || rvalue) && (!da7218->alc_en))
+ da7218_alc_calib(codec);
+
+ /* Update bits to detail which channels are enabled/disabled */
+ da7218->alc_en &= ~mask;
+ da7218->alc_en |= (lvalue << lshift) | (rvalue << rshift);
+
+ return snd_soc_put_volsw(kcontrol, ucontrol);
+}
+
+/* ToneGen */
+static int da7218_tonegen_freq_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
+ struct da7218_priv *da7218 = snd_soc_codec_get_drvdata(codec);
+ struct soc_mixer_control *mixer_ctrl =
+ (struct soc_mixer_control *) kcontrol->private_value;
+ unsigned int reg = mixer_ctrl->reg;
+ u16 val;
+ int ret;
+
+ /*
+ * Frequency value spans two 8-bit registers, lower then upper byte.
+ * Therefore we need to convert to host endianness here.
+ */
+ ret = regmap_raw_read(da7218->regmap, reg, &val, 2);
+ if (ret)
+ return ret;
+
+ ucontrol->value.integer.value[0] = le16_to_cpu(val);
+
+ return 0;
+}
+
+static int da7218_tonegen_freq_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
+ struct da7218_priv *da7218 = snd_soc_codec_get_drvdata(codec);
+ struct soc_mixer_control *mixer_ctrl =
+ (struct soc_mixer_control *) kcontrol->private_value;
+ unsigned int reg = mixer_ctrl->reg;
+ u16 val;
+
+ /*
+ * Frequency value spans two 8-bit registers, lower then upper byte.
+ * Therefore we need to convert to little endian here to align with
+ * HW registers.
+ */
+ val = cpu_to_le16(ucontrol->value.integer.value[0]);
+
+ return regmap_raw_write(da7218->regmap, reg, &val, 2);
+}
+
+static int da7218_mic_lvl_det_sw_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
+ struct da7218_priv *da7218 = snd_soc_codec_get_drvdata(codec);
+ struct soc_mixer_control *mixer_ctrl =
+ (struct soc_mixer_control *) kcontrol->private_value;
+ unsigned int lvalue = ucontrol->value.integer.value[0];
+ unsigned int rvalue = ucontrol->value.integer.value[1];
+ unsigned int lshift = mixer_ctrl->shift;
+ unsigned int rshift = mixer_ctrl->rshift;
+ unsigned int mask = (mixer_ctrl->max << lshift) |
+ (mixer_ctrl->max << rshift);
+ da7218->mic_lvl_det_en &= ~mask;
+ da7218->mic_lvl_det_en |= (lvalue << lshift) | (rvalue << rshift);
+
+ /*
+ * Here we only enable the feature on paths which are already
+ * powered. If a channel is enabled here for level detect, but that path
+ * isn't powered, then the channel will actually be enabled when we do
+ * power the path (IN_FILTER widget events). This handling avoids
+ * unwanted level detect events.
+ */
+ return snd_soc_write(codec, mixer_ctrl->reg,
+ (da7218->in_filt_en & da7218->mic_lvl_det_en));
+}
+
+static int da7218_mic_lvl_det_sw_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
+ struct da7218_priv *da7218 = snd_soc_codec_get_drvdata(codec);
+ struct soc_mixer_control *mixer_ctrl =
+ (struct soc_mixer_control *) kcontrol->private_value;
+ unsigned int lshift = mixer_ctrl->shift;
+ unsigned int rshift = mixer_ctrl->rshift;
+ unsigned int lmask = (mixer_ctrl->max << lshift);
+ unsigned int rmask = (mixer_ctrl->max << rshift);
+
+ ucontrol->value.integer.value[0] =
+ (da7218->mic_lvl_det_en & lmask) >> lshift;
+ ucontrol->value.integer.value[1] =
+ (da7218->mic_lvl_det_en & rmask) >> rshift;
+
+ return 0;
+}
+
+static int da7218_biquad_coeff_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
+ struct da7218_priv *da7218 = snd_soc_codec_get_drvdata(codec);
+ struct soc_bytes_ext *bytes_ext =
+ (struct soc_bytes_ext *) kcontrol->private_value;
+
+ /* Determine which BiQuads we're setting based on size of config data */
+ switch (bytes_ext->max) {
+ case DA7218_OUT_1_BIQ_5STAGE_CFG_SIZE:
+ memcpy(ucontrol->value.bytes.data, da7218->biq_5stage_coeff,
+ bytes_ext->max);
+ break;
+ case DA7218_SIDETONE_BIQ_3STAGE_CFG_SIZE:
+ memcpy(ucontrol->value.bytes.data, da7218->stbiq_3stage_coeff,
+ bytes_ext->max);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int da7218_biquad_coeff_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)</