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firmware/br-ext-chip-allwinner/board/v83x/kernel/patches/00000-sound_soc_codecs_acx0...

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diff -drupN a/sound/soc/codecs/acx00.c b/sound/soc/codecs/acx00.c
--- a/sound/soc/codecs/acx00.c 1970-01-01 03:00:00.000000000 +0300
+++ b/sound/soc/codecs/acx00.c 2022-06-12 05:28:14.000000000 +0300
@@ -0,0 +1,1307 @@
+/*
+ * acx00.c -- ACX00 ALSA Soc Audio Codec driver
+ *
+ * (C) Copyright 2010-2016 Allwinnertech Technology., Ltd.
+ *
+ * Author: Wolfgang Huang <huangjinhui@allwinner.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/kernel.h>
+#include <linux/init.h>
+#include <linux/firmware.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/clk.h>
+#include <linux/of_gpio.h>
+#include <linux/sunxi-gpio.h>
+#include <linux/of_device.h>
+#include <linux/of_platform.h>
+#include <linux/debugfs.h>
+#include <linux/slab.h>
+#include <linux/mfd/acx00-mfd.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+#include <linux/workqueue.h>
+
+#include "acx00.h"
+
+
+#define ACX00_DEF_VOL 0x9F9F
+#undef ACX00_DAPM_LINEOUT
+
+struct acx00_priv {
+ struct acx00 *acx00; /* parent mfd device struct */
+ struct snd_soc_codec *codec;
+ struct clk *clk;
+ unsigned int sample_rate;
+ unsigned int fmt;
+ unsigned int enable;
+ unsigned int spk_gpio;
+ bool spk_gpio_used;
+ struct mutex mutex;
+ struct delayed_work spk_work;
+ struct delayed_work resume_work;
+};
+
+struct sample_rate {
+ unsigned int samplerate;
+ unsigned int rate_bit;
+};
+
+static const struct sample_rate sample_rate_conv[] = {
+ {44100, 7},
+ {48000, 8},
+ {8000, 0},
+ {32000, 6},
+ {22050, 4},
+ {24000, 5},
+ {16000, 3},
+ {11025, 1},
+ {12000, 2},
+ {192000, 10},
+ {96000, 9},
+};
+
+void __iomem *io_stat_addr;
+
+static const DECLARE_TLV_DB_SCALE(i2s_mixer_adc_tlv, -600, 600, 1);
+static const DECLARE_TLV_DB_SCALE(i2s_mixer_dac_tlv, -600, 600, 1);
+static const DECLARE_TLV_DB_SCALE(dac_mixer_adc_tlv, -600, 600, 1);
+static const DECLARE_TLV_DB_SCALE(dac_mixer_dac_tlv, -600, 600, 1);
+static const DECLARE_TLV_DB_SCALE(line_out_tlv, -450, 150, 0);
+static const DECLARE_TLV_DB_SCALE(mic_out_tlv, -450, 150, 0);
+static const DECLARE_TLV_DB_SCALE(phoneout_tlv, -450, 150, 0);
+static const DECLARE_TLV_DB_SCALE(adc_input_tlv, -450, 150, 0);
+static const DECLARE_TLV_DB_SCALE(lineout_tlv, -4800, 150, 1);
+static const unsigned int mic_boost_tlv[] = {
+ TLV_DB_RANGE_HEAD(2),
+ 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0),
+ 1, 7, TLV_DB_SCALE_ITEM(2400, 300, 0),
+};
+
+static const struct snd_kcontrol_new acx00_codec_controls[] = {
+ SOC_DOUBLE_TLV("I2S Mixer ADC Volume", AC_I2S_MIXER_GAIN,
+ I2S_MIXERL_GAIN_ADC, I2S_MIXERR_GAIN_ADC,
+ 0x1, 0, i2s_mixer_adc_tlv),
+ SOC_DOUBLE_TLV("I2S Mixer DAC Volume", AC_I2S_MIXER_GAIN,
+ I2S_MIXERR_GAIN_DAC, I2S_MIXERR_GAIN_DAC,
+ 0x1, 0, i2s_mixer_dac_tlv),
+ SOC_DOUBLE_TLV("DAC Mixer ADC Volume", AC_DAC_MIXER_GAIN,
+ DAC_MIXERL_GAIN_ADC, DAC_MIXERR_GAIN_ADC,
+ 0x1, 0, dac_mixer_adc_tlv),
+ SOC_DOUBLE_TLV("DAC Mxier DAC Volume", AC_DAC_MIXER_GAIN,
+ DAC_MIXERL_GAIN_DAC, DAC_MIXERR_GAIN_DAC,
+ 0x1, 0, dac_mixer_dac_tlv),
+ SOC_SINGLE_TLV("Line Out Mixer Volume", AC_OUT_MIXER_CTL,
+ OUT_MIXER_LINE_VOL, 0x7, 0, line_out_tlv),
+ SOC_DOUBLE_TLV("MIC Out Mixer Volume", AC_OUT_MIXER_CTL,
+ OUT_MIXER_MIC1_VOL, OUT_MIXER_MIC2_VOL,
+ 0x7, 0, mic_out_tlv),
+ SOC_SINGLE_TLV("ADC Input Volume", AC_ADC_MIC_CTL,
+ ADC_GAIN, 0x07, 0, adc_input_tlv),
+ SOC_SINGLE_TLV("LINEOUT Volume", AC_LINEOUT_CTL,
+ LINEOUT_VOL, 0x1f, 0, lineout_tlv),
+ SOC_SINGLE_TLV("MIC1 Boost Volume", AC_ADC_MIC_CTL,
+ MIC1_BOOST, 0x07, 0, mic_boost_tlv),
+ SOC_SINGLE_TLV("MIC2 Boost Volume", AC_ADC_MIC_CTL,
+ MIC2_BOOST, 0x07, 0, mic_boost_tlv),
+};
+
+/* Enable I2S & DAC clk, then enable the DAC digital part */
+static int acx00_playback_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *k, int event)
+{
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
+
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ snd_soc_update_bits(codec, AC_SYS_CLK_CTL,
+ (0x1<<SYS_CLK_DAC), (0x1<<SYS_CLK_DAC));
+ snd_soc_update_bits(codec, AC_SYS_MOD_RST,
+ (0x1<<MOD_RST_DAC), (0x1<<MOD_RST_DAC));
+ snd_soc_update_bits(codec, AC_DAC_CTL,
+ (0x1<<DAC_CTL_DAC_EN), (0x1<<DAC_CTL_DAC_EN));
+ break;
+ case SND_SOC_DAPM_POST_PMD:
+ snd_soc_update_bits(codec, AC_SYS_CLK_CTL,
+ (0x1<<SYS_CLK_DAC), (0x0<<SYS_CLK_DAC));
+ snd_soc_update_bits(codec, AC_SYS_MOD_RST,
+ (0x1<<MOD_RST_DAC), (0x0<<MOD_RST_DAC));
+ snd_soc_update_bits(codec, AC_DAC_CTL,
+ (0x1<<DAC_CTL_DAC_EN), (0x0<<DAC_CTL_DAC_EN));
+ break;
+ default:
+ break;
+ }
+ return 0;
+}
+
+/* Enable I2S & ADC clk, then enable the ADC digital part */
+static int acx00_capture_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *k, int event)
+{
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
+
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ snd_soc_update_bits(codec, AC_SYS_CLK_CTL,
+ (0x1<<SYS_CLK_ADC), (0x1<<SYS_CLK_ADC));
+ snd_soc_update_bits(codec, AC_SYS_MOD_RST,
+ (0x1<<MOD_RST_ADC), (0x1<<MOD_RST_ADC));
+ snd_soc_update_bits(codec, AC_ADC_CTL,
+ (0x1<<ADC_EN), (0x1<<ADC_EN));
+ break;
+ case SND_SOC_DAPM_POST_PMD:
+ snd_soc_update_bits(codec, AC_SYS_CLK_CTL,
+ (0x1<<SYS_CLK_ADC), (0x0<<SYS_CLK_ADC));
+ snd_soc_update_bits(codec, AC_SYS_MOD_RST,
+ (0x1<<MOD_RST_ADC), (0x0<<MOD_RST_ADC));
+ snd_soc_update_bits(codec, AC_ADC_CTL,
+ (0x1<<ADC_EN), (0x0<<ADC_EN));
+ break;
+ default:
+ break;
+ }
+ return 0;
+}
+
+/*
+ * we used for three scene:
+ * 1. No external Spker & DAPM LINEOUT used, we just enable the LINEOUT in the
+ * ALSA codec probe(acx00_codec_probe) and resume, and we shutdown the LINEOUT
+ * in device shutdown or suspend.
+ * 2. No external Spker, but DAPM LINEOUT used, we just using the LINEOUT
+ * enable or disable throught the DAPM control.
+ * 3. External Spker & DAPM LINEOUT used, we just using the LINEOUT and
+ * External Spker control GPIO enable or disable through DAPM control.
+ */
+static unsigned int spk_delay = 100;
+module_param(spk_delay, int, 0644);
+MODULE_PARM_DESC(spk_delay, "ACX00-Codec spk mute delay time");
+
+static void acx00_spk_enable(struct work_struct *work)
+{
+ struct acx00_priv *priv = container_of(work,
+ struct acx00_priv, spk_work.work);
+ gpio_set_value(priv->spk_gpio, 1);
+}
+
+static int acx00_lineout_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *k, int event)
+{
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
+ struct acx00_priv *priv = snd_soc_codec_get_drvdata(codec);
+
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ if (!priv->enable) {
+ snd_soc_update_bits(priv->codec, AC_LINEOUT_CTL,
+ (1<<LINEL_SRC_EN), (1<<LINEL_SRC_EN));
+ snd_soc_update_bits(priv->codec, AC_LINEOUT_CTL,
+ (1<<LINER_SRC_EN), (1<<LINER_SRC_EN));
+ msleep(100);
+ priv->enable = 1;
+ }
+#ifdef ACX00_DAPM_LINEOUT
+ snd_soc_update_bits(codec, AC_LINEOUT_CTL,
+ (1<<LINEOUT_EN), (1<<LINEOUT_EN));
+ mdelay(50);
+#endif
+ if (priv->spk_gpio_used) {
+ if (spk_delay == 0) {
+ gpio_set_value(priv->spk_gpio, 1);
+ /*
+ * time delay to wait spk pa work fine,
+ * general setting 50ms
+ */
+ mdelay(50);
+ } else
+ schedule_delayed_work(&priv->spk_work,
+ msecs_to_jiffies(spk_delay));
+ }
+ break;
+ case SND_SOC_DAPM_PRE_PMD:
+ mdelay(50);
+ if (priv->spk_gpio_used) {
+ gpio_set_value(priv->spk_gpio, 0);
+ msleep(50);
+ }
+#ifdef ACX00_DAPM_LINEOUT
+ snd_soc_update_bits(codec, AC_LINEOUT_CTL,
+ (1<<LINEOUT_EN), (0<<LINEOUT_EN));
+#endif
+ break;
+ default:
+ break;
+ }
+ return 0;
+}
+
+/* AC_I2S_MIXER_SRC : 0x2114 */
+static const struct snd_kcontrol_new i2sl_mixer_src[] = {
+ SOC_DAPM_SINGLE("I2SDACL Switch", AC_I2S_MIXER_SRC,
+ I2S_MIXERL_SRC_DAC, 1, 0),
+ SOC_DAPM_SINGLE("ADCL Switch", AC_I2S_MIXER_SRC,
+ I2S_MIXERL_SRC_ADC, 1, 0),
+};
+
+static const struct snd_kcontrol_new i2sr_mixer_src[] = {
+ SOC_DAPM_SINGLE("I2SDACR Switch", AC_I2S_MIXER_SRC,
+ I2S_MIXERR_SRC_DAC, 1, 0),
+ SOC_DAPM_SINGLE("ADCR Switch", AC_I2S_MIXER_SRC,
+ I2S_MIXERR_SRC_ADC, 1, 0),
+};
+
+/* AC_DAC_MIXER_SRC : 0x2202 */
+static const struct snd_kcontrol_new dacl_mixer_src[] = {
+ SOC_DAPM_SINGLE("I2SDACL Switch", AC_DAC_MIXER_SRC,
+ DAC_MIXERL_SRC_DAC, 1, 0),
+ SOC_DAPM_SINGLE("ADCL Switch", AC_DAC_MIXER_SRC,
+ DAC_MIXERL_SRC_ADC, 1, 0),
+};
+
+static const struct snd_kcontrol_new dacr_mixer_src[] = {
+ SOC_DAPM_SINGLE("I2SDACR Switch", AC_DAC_MIXER_SRC,
+ DAC_MIXERR_SRC_DAC, 1, 0),
+ SOC_DAPM_SINGLE("ADCR Switch", AC_DAC_MIXER_SRC,
+ DAC_MIXERR_SRC_ADC, 1, 0),
+};
+
+/* AC_OUT_MIXER_SRC : 0x2222 */
+static const struct snd_kcontrol_new left_output_mixer[] = {
+ SOC_DAPM_SINGLE("MIC1 Switch", AC_OUT_MIXER_SRC,
+ OUT_MIXERL_SRC_MIC1, 1, 0),
+ SOC_DAPM_SINGLE("MIC2 Switch", AC_OUT_MIXER_SRC,
+ OUT_MIXERL_SRC_MIC2, 1, 0),
+ SOC_DAPM_SINGLE("PhonePN Switch", AC_OUT_MIXER_SRC,
+ OUT_MIXERL_SRC_PHPN, 1, 0),
+ SOC_DAPM_SINGLE("PhoneN Switch", AC_OUT_MIXER_SRC,
+ OUT_MIXERL_SRC_PHN, 1, 0),
+ SOC_DAPM_SINGLE("LINEINL Switch", AC_OUT_MIXER_SRC,
+ OUT_MIXERL_SRC_LINEL, 1, 0),
+ SOC_DAPM_SINGLE("DACL Switch", AC_OUT_MIXER_SRC,
+ OUT_MIXERL_SRC_DACL, 1, 0),
+ SOC_DAPM_SINGLE("DACR Switch", AC_OUT_MIXER_SRC,
+ OUT_MIXERL_SRC_DACR, 1, 0),
+};
+
+static const struct snd_kcontrol_new right_output_mixer[] = {
+ SOC_DAPM_SINGLE("MIC1 Switch", AC_OUT_MIXER_SRC,
+ OUT_MIXERR_SRC_MIC1, 1, 0),
+ SOC_DAPM_SINGLE("MIC2 Switch", AC_OUT_MIXER_SRC,
+ OUT_MIXERR_SRC_MIC2, 1, 0),
+ SOC_DAPM_SINGLE("PhonePN Switch", AC_OUT_MIXER_SRC,
+ OUT_MIXERR_SRC_PHPN, 1, 0),
+ SOC_DAPM_SINGLE("PhoneP Switch", AC_OUT_MIXER_SRC,
+ OUT_MIXERR_SRC_PHP, 1, 0),
+ SOC_DAPM_SINGLE("LINEINR Switch", AC_OUT_MIXER_SRC,
+ OUT_MIXERR_SRC_LINER, 1, 0),
+ SOC_DAPM_SINGLE("DACR Switch", AC_OUT_MIXER_SRC,
+ OUT_MIXERR_SRC_DACR, 1, 0),
+ SOC_DAPM_SINGLE("DACL Switch", AC_OUT_MIXER_SRC,
+ OUT_MIXERR_SRC_DACL, 1, 0),
+};
+
+/* AC_LINEOUT_CTL : 0x2224 */
+const char * const left_lineout_text[] = {
+ "Left OMixer", "LR OMixer",
+};
+
+static const struct soc_enum left_lineout_enum =
+ SOC_ENUM_SINGLE(AC_LINEOUT_CTL, LINEL_SRC,
+ ARRAY_SIZE(left_lineout_text), left_lineout_text);
+
+static const struct snd_kcontrol_new left_lineout_mux =
+ SOC_DAPM_ENUM("Left LINEOUT Mux", left_lineout_enum);
+
+const char * const right_lineout_text[] = {
+ "Right OMixer", "LR OMixer",
+};
+
+static const struct soc_enum right_lineout_enum =
+ SOC_ENUM_SINGLE(AC_LINEOUT_CTL, LINER_SRC,
+ ARRAY_SIZE(right_lineout_text), right_lineout_text);
+
+static const struct snd_kcontrol_new right_lineout_mux =
+ SOC_DAPM_ENUM("Right LINEOUT Mux", right_lineout_enum);
+
+/* AC_ADC_MIXER_SRC : 0x2322 */
+static const struct snd_kcontrol_new left_input_mixer[] = {
+ SOC_DAPM_SINGLE("MIC1 Switch", AC_ADC_MIXER_SRC,
+ ADC_MIXERL_MIC1, 1, 0),
+ SOC_DAPM_SINGLE("MIC2 Switch", AC_ADC_MIXER_SRC,
+ ADC_MIXERL_MIC2, 1, 0),
+ SOC_DAPM_SINGLE("PhonePN Switch", AC_ADC_MIXER_SRC,
+ ADC_MIXERL_PHPN, 1, 0),
+ SOC_DAPM_SINGLE("PhoneN Switch", AC_ADC_MIXER_SRC,
+ ADC_MIXERL_PHN, 1, 0),
+ SOC_DAPM_SINGLE("LINEINL Switch", AC_ADC_MIXER_SRC,
+ ADC_MIXERL_LINEL, 1, 0),
+ SOC_DAPM_SINGLE("OMixerL Switch", AC_ADC_MIXER_SRC,
+ ADC_MIXERL_MIXL, 1, 0),
+ SOC_DAPM_SINGLE("OMixerR Switch", AC_ADC_MIXER_SRC,
+ ADC_MIXERL_MIXR, 1, 0),
+};
+
+static const struct snd_kcontrol_new right_input_mixer[] = {
+ SOC_DAPM_SINGLE("MIC1 Switch", AC_ADC_MIXER_SRC,
+ ADC_MIXERR_MIC1, 1, 0),
+ SOC_DAPM_SINGLE("MIC2 Switch", AC_ADC_MIXER_SRC,
+ ADC_MIXERR_MIC2, 1, 0),
+ SOC_DAPM_SINGLE("PhonePN Switch", AC_ADC_MIXER_SRC,
+ ADC_MIXERR_PHPN, 1, 0),
+ SOC_DAPM_SINGLE("PhoneP Switch", AC_ADC_MIXER_SRC,
+ ADC_MIXERR_PHP, 1, 0),
+ SOC_DAPM_SINGLE("LINEINR Switch", AC_ADC_MIXER_SRC,
+ ADC_MIXERR_LINER, 1, 0),
+ SOC_DAPM_SINGLE("OMixerR Switch", AC_ADC_MIXER_SRC,
+ ADC_MIXERR_MIXR, 1, 0),
+ SOC_DAPM_SINGLE("OMixerL Switch", AC_ADC_MIXER_SRC,
+ ADC_MIXERR_MIXL, 1, 0),
+};
+
+static const struct snd_soc_dapm_widget acx00_codec_dapm_widgets[] = {
+ SND_SOC_DAPM_AIF_IN_E("DACL", "Playback", 0, AC_DAC_CTL,
+ OUT_MIXER_DACL_EN, 0,
+ acx00_playback_event,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_AIF_IN_E("DACR", "Playback", 0,
+ AC_DAC_CTL, OUT_MIXER_DACR_EN, 0,
+ acx00_playback_event,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+ SND_SOC_DAPM_AIF_OUT_E("ADCL", "Capture", 0,
+ AC_ADC_MIC_CTL, ADCL_EN, 0,
+ acx00_capture_event,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_AIF_OUT_E("ADCR", "Capture", 0,
+ AC_ADC_MIC_CTL, ADCR_EN, 0,
+ acx00_capture_event,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+ SND_SOC_DAPM_MIXER("Left Output Mixer", AC_OUT_MIXER_CTL,
+ OUT_MIXER_LMIX_EN, 0,
+ left_output_mixer, ARRAY_SIZE(left_output_mixer)),
+
+ SND_SOC_DAPM_MIXER("Right Output Mixer", AC_OUT_MIXER_CTL,
+ OUT_MIXER_RMIX_EN, 0, right_output_mixer,
+ ARRAY_SIZE(right_output_mixer)),
+
+ SND_SOC_DAPM_MIXER("Left Input Mixer", SND_SOC_NOPM, 0, 0,
+ left_input_mixer, ARRAY_SIZE(left_input_mixer)),
+ SND_SOC_DAPM_MIXER("Right Input Mixer", SND_SOC_NOPM, 0, 0,
+ right_input_mixer, ARRAY_SIZE(right_input_mixer)),
+
+ SND_SOC_DAPM_MIXER("Left DAC Mixer", AC_OUT_MIXER_CTL,
+ OUT_MIXER_DACL_EN, 0, dacl_mixer_src,
+ ARRAY_SIZE(dacl_mixer_src)),
+ SND_SOC_DAPM_MIXER("Right DAC Mixer", AC_OUT_MIXER_CTL,
+ OUT_MIXER_DACR_EN, 0, dacr_mixer_src,
+ ARRAY_SIZE(dacr_mixer_src)),
+
+ SND_SOC_DAPM_MIXER("Left I2S Mixer", SND_SOC_NOPM,
+ 0, 0, i2sl_mixer_src, ARRAY_SIZE(i2sl_mixer_src)),
+ SND_SOC_DAPM_MIXER("Right I2S Mixer", SND_SOC_NOPM,
+ 0, 0, i2sr_mixer_src, ARRAY_SIZE(i2sr_mixer_src)),
+
+ SND_SOC_DAPM_MUX("Left LINEOUT Mux", SND_SOC_NOPM,
+ 0, 0, &left_lineout_mux),
+ SND_SOC_DAPM_MUX("Right LINEOUT Mux", SND_SOC_NOPM,
+ 0, 0, &right_lineout_mux),
+
+ SND_SOC_DAPM_PGA("MIC1 PGA", AC_ADC_MIC_CTL,
+ MIC1_GAIN_EN, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("MIC2 PGA", AC_ADC_MIC_CTL,
+ MIC2_GAIN_EN, 0, NULL, 0),
+
+ SND_SOC_DAPM_MICBIAS("MIC Bias", AC_MICBIAS_CTL,
+ MMBIAS_EN, 0),
+
+ /* PHONEIN & PHONEOUT not enable in pin assign */
+ SND_SOC_DAPM_INPUT("PHONEINP"),
+ SND_SOC_DAPM_INPUT("PHONEINN"),
+ SND_SOC_DAPM_INPUT("PHONEINPN"),
+
+ /* endpoint define */
+ SND_SOC_DAPM_LINE("LINEIN", NULL),
+ SND_SOC_DAPM_LINE("LINEOUT", acx00_lineout_event),
+ SND_SOC_DAPM_MIC("MIC1", NULL),
+ SND_SOC_DAPM_MIC("MIC2", NULL),
+};
+
+static const struct snd_soc_dapm_route acx00_codec_dapm_routes[] = {
+ {"Left Output Mixer", "MIC1 Switch", "MIC1 PGA"},
+ {"Left Output Mixer", "MIC2 Switch", "MIC2 PGA"},
+ {"Left Output Mixer", "PhonePN Switch", "PHONEINPN"},
+ {"Left Output Mixer", "PhoneN Switch", "PHONEINN"},
+ {"Left Output Mixer", "LINEINL Switch", "LINEIN"},
+ {"Left Output Mixer", "DACR Switch", "Right DAC Mixer"},
+ {"Left Output Mixer", "DACL Switch", "Left DAC Mixer"},
+
+ {"Right Output Mixer", "MIC1 Switch", "MIC1 PGA"},
+ {"Right Output Mixer", "MIC2 Switch", "MIC2 PGA"},
+ {"Right Output Mixer", "PhonePN Switch", "PHONEINPN"},
+ {"Right Output Mixer", "PhoneP Switch", "PHONEINP"},
+ {"Right Output Mixer", "LINEINR Switch", "LINEIN"},
+ {"Right Output Mixer", "DACR Switch", "Right DAC Mixer"},
+ {"Right Output Mixer", "DACL Switch", "Left DAC Mixer"},
+
+ {"Left LINEOUT Mux", NULL, "Left Output Mixer"},
+ {"Left LINEOUT Mux", "LR OMixer", "Right Output Mixer"},
+ {"Right LINEOUT Mux", NULL, "Right Output Mixer"},
+ {"Right LINEOUT Mux", "LR OMixer", "Left Output Mixer"},
+
+ {"Left Input Mixer", "MIC1 Switch", "MIC1 PGA"},
+ {"Left Input Mixer", "MIC2 Switch", "MIC2 PGA"},
+ {"Left Input Mixer", "PhonePN Switch", "PHONEINPN"},
+ {"Left Input Mixer", "PhoneN Switch", "PHONEINN"},
+ {"Left Input Mixer", "LINEINL Switch", "LINEIN"},
+ {"Left Input Mixer", "OMixerL Switch", "Left Output Mixer"},
+ {"Left Input Mixer", "OMixerR Switch", "Right Output Mixer"},
+
+ {"Right Input Mixer", "MIC1 Switch", "MIC1 PGA"},
+ {"Right Input Mixer", "MIC2 Switch", "MIC2 PGA"},
+ {"Right Input Mixer", "PhonePN Switch", "PHONEINPN"},
+ {"Right Input Mixer", "PhoneP Switch", "PHONEINP"},
+ {"Right Input Mixer", "LINEINR Switch", "LINEIN"},
+ {"Right Input Mixer", "OMixerR Switch", "Right Output Mixer"},
+ {"Right Input Mixer", "OMixerL Switch", "Left Output Mixer"},
+
+ {"Left I2S Mixer", "I2SDACL Switch", "DACL"},
+ {"Left I2S Mixer", "ADCL Switch", "Left Input Mixer"},
+
+ {"Right I2S Mixer", "I2SDACR Switch", "DACR"},
+ {"Right I2S Mixer", "ADCR Switch", "Right Input Mixer"},
+
+ {"Left DAC Mixer", "I2SDACL Switch", "DACL"},
+ {"Left DAC Mixer", "ADCL Switch", "Left Input Mixer"},
+
+ {"Right DAC Mixer", "I2SDACR Switch", "DACR"},
+ {"Right DAC Mixer", "ADCR Switch", "Right Input Mixer"},
+
+ {"ADCL", NULL, "Left I2S Mixer"},
+ {"ADCR", NULL, "Right I2S Mixer"},
+
+ {"LINEOUT", NULL, "Left LINEOUT Mux"},
+ {"LINEOUT", NULL, "Right LINEOUT Mux"},
+
+ {"MIC Bias", NULL, "MIC1"},
+ {"MIC Bias", NULL, "MIC2"},
+ {"MIC1 PGA", NULL, "MIC Bias"},
+ {"MIC2 PGA", NULL, "MIC Bias"},
+};
+
+static void acx00_codec_txctrl_enable(struct snd_soc_codec *codec,
+ int enable)
+{
+ pr_debug("Enter %s, enable %d\n", __func__, enable);
+ if (enable) {
+ snd_soc_update_bits(codec, AC_I2S_CTL,
+ (1<<I2S_RX_EN), (1<<I2S_RX_EN));
+ } else {
+ snd_soc_update_bits(codec, AC_I2S_CTL,
+ (1<<I2S_RX_EN), (0<<I2S_RX_EN));
+ }
+ pr_debug("End %s, enable %d\n", __func__, enable);
+}
+
+static void acx00_codec_rxctrl_enable(struct snd_soc_codec *codec,
+ int enable)
+{
+ pr_debug("Enter %s, enable %d\n", __func__, enable);
+ if (enable) {
+ snd_soc_update_bits(codec, AC_I2S_CTL,
+ (1<<I2S_TX_EN), (1<<I2S_TX_EN));
+ } else {
+ snd_soc_update_bits(codec, AC_I2S_CTL,
+ (1<<I2S_TX_EN), (0<<I2S_TX_EN));
+ }
+ pr_debug("End %s, enable %d\n", __func__, enable);
+}
+
+static void acx00_codec_init(struct snd_soc_codec *codec)
+{
+ struct acx00_priv *priv = snd_soc_codec_get_drvdata(codec);
+
+ /* acx00_codec sysctl init */
+ acx00_reg_write(priv->acx00, 0x0010, 0x03);
+ acx00_reg_write(priv->acx00, 0x0012, 0x01);
+ /* The bit3 need to setup to 1 for bias current. */
+ snd_soc_update_bits(codec, AC_MICBIAS_CTL,
+ (0x1 << ADDA_BIAS_CUR), (0x1 << ADDA_BIAS_CUR));
+
+ /* enable the output & global enable bit */
+ snd_soc_update_bits(codec, AC_I2S_CTL,
+ (1<<I2S_SDO0_EN), (1<<I2S_SDO0_EN));
+ snd_soc_update_bits(codec, AC_I2S_CTL, (1<<I2S_GEN), (1<<I2S_GEN));
+
+ /* Default setting slot width as 32 bit for I2S */
+ snd_soc_update_bits(codec, AC_I2S_FMT0,
+ (7<<I2S_FMT_SLOT_WIDTH), (7<<I2S_FMT_SLOT_WIDTH));
+
+ /* default setting 0xA0A0 for ADC & DAC Volume */
+ snd_soc_write(codec, AC_I2S_DAC_VOL, ACX00_DEF_VOL);
+ snd_soc_write(codec, AC_I2S_ADC_VOL, ACX00_DEF_VOL);
+
+ /* Enable HPF for high pass filter */
+ snd_soc_update_bits(codec, AC_DAC_CTL,
+ (1<<DAC_CTL_HPF_EN), (1<<DAC_CTL_HPF_EN));
+
+ /* LINEOUT ANTI POP & Click noise */
+ snd_soc_update_bits(codec, AC_LINEOUT_CTL,
+ (0x7<<LINE_ANTI_TIME), (0x3<<LINE_ANTI_TIME));
+ snd_soc_update_bits(codec, AC_LINEOUT_CTL,
+ (0x3<<LINE_SLOPE_SEL), (0x3<<LINE_SLOPE_SEL));
+
+ /* enable & setting adc convert delay time */
+ snd_soc_update_bits(codec, AC_ADC_CTL, (0x3<<ADC_DELAY_TIME),
+ (0x3<<ADC_DELAY_TIME));
+ snd_soc_update_bits(codec, AC_ADC_CTL, (1<<ADC_DELAY_EN),
+ (1<<ADC_DELAY_EN));
+
+
+ if (priv->spk_gpio_used) {
+ snd_soc_update_bits(priv->codec, AC_LINEOUT_CTL,
+ (1<<LINEL_SRC_EN), (1<<LINEL_SRC_EN));
+ snd_soc_update_bits(priv->codec, AC_LINEOUT_CTL,
+ (1<<LINER_SRC_EN), (1<<LINER_SRC_EN));
+ priv->enable = 1;
+ }
+#ifndef ACX00_DAPM_LINEOUT
+ snd_soc_update_bits(codec, AC_LINEOUT_CTL, (1<<LINEOUT_EN),
+ (1<<LINEOUT_EN));
+#endif
+}
+
+static int acx00_codec_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ int i;
+
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ snd_soc_update_bits(codec, AC_I2S_FMT0,
+ (7<<I2S_FMT_SAMPLE), (3<<I2S_FMT_SAMPLE));
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ snd_soc_update_bits(codec, AC_I2S_FMT0,
+ (7<<I2S_FMT_SAMPLE), (5<<I2S_FMT_SAMPLE));
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ snd_soc_update_bits(codec, AC_I2S_FMT0,
+ (7<<I2S_FMT_SAMPLE), (7<<I2S_FMT_SAMPLE));
+ break;
+ default:
+ dev_err(codec->dev, "unrecognized format support\n");
+ break;
+ }
+ for (i = 0; i < ARRAY_SIZE(sample_rate_conv); i++) {
+ if (sample_rate_conv[i].samplerate == params_rate(params)) {
+ snd_soc_update_bits(codec, AC_SYS_SR_CTL,
+ (SYS_SR_MASK<<SYS_SR_BIT),
+ (sample_rate_conv[i].rate_bit<<SYS_SR_BIT));
+ }
+ }
+
+ return 0;
+}
+
+static int acx00_codec_dai_set_sysclk(struct snd_soc_dai *codec_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ return 0;
+}
+
+static int acx00_codec_dai_set_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct acx00_priv *priv = snd_soc_dai_get_drvdata(codec_dai);
+ struct snd_soc_codec *codec = priv->codec;
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ /* codec clk & FRM master */
+ case SND_SOC_DAIFMT_CBM_CFM:
+ snd_soc_update_bits(codec, AC_I2S_CLK,
+ (0x1<<I2S_BCLK_OUT), (0x1<<I2S_BCLK_OUT));
+ snd_soc_update_bits(codec, AC_I2S_CLK,
+ (0x1<<I2S_LRCK_OUT), (0x1<<I2S_LRCK_OUT));
+ break;
+ /* codec clk & FRM slave */
+ case SND_SOC_DAIFMT_CBS_CFS:
+ snd_soc_update_bits(codec, AC_I2S_CLK,
+ (0x1<<I2S_BCLK_OUT), 0x0<<I2S_BCLK_OUT);
+ snd_soc_update_bits(codec, AC_I2S_CLK,
+ (0x1<<I2S_LRCK_OUT), 0x0<<I2S_LRCK_OUT);
+ break;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ snd_soc_update_bits(codec, AC_I2S_CLK,
+ (0x3FF<<I2S_LRCK_PERIOD),
+ (0x1F<<I2S_LRCK_PERIOD));
+ snd_soc_update_bits(codec, AC_I2S_FMT0,
+ (0x3<<I2S_FMT_MODE), (0x1<<I2S_FMT_MODE));
+ snd_soc_update_bits(codec, AC_I2S_FMT0,
+ (0x1<<I2S_FMT_TX_OFFSET),
+ (0x1<<I2S_FMT_TX_OFFSET));
+ snd_soc_update_bits(codec, AC_I2S_FMT0,
+ (0x1<<I2S_FMT_RX_OFFSET),
+ (0x1<<I2S_FMT_RX_OFFSET));
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ snd_soc_update_bits(codec, AC_I2S_CLK,
+ (0x3FF<<I2S_LRCK_PERIOD),
+ (0x1F<<I2S_LRCK_PERIOD));
+ snd_soc_update_bits(codec, AC_I2S_FMT0,
+ (0x3<<I2S_FMT_MODE), (0x2<<I2S_FMT_MODE));
+ snd_soc_update_bits(codec, AC_I2S_FMT0,
+ (0x1<<I2S_FMT_TX_OFFSET),
+ (0x0<<I2S_FMT_TX_OFFSET));
+ snd_soc_update_bits(codec, AC_I2S_FMT0,
+ (0x1<<I2S_FMT_TX_OFFSET),
+ (0x0<<I2S_FMT_RX_OFFSET));
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ snd_soc_update_bits(codec, AC_I2S_CLK,
+ (0x3FF<<I2S_LRCK_PERIOD),
+ (0x1F<<I2S_LRCK_PERIOD));
+ snd_soc_update_bits(codec, AC_I2S_FMT0,
+ (0x3<<I2S_FMT_MODE), (0x1<<I2S_FMT_MODE));
+ snd_soc_update_bits(codec, AC_I2S_FMT0,
+ (0x1<<I2S_FMT_TX_OFFSET),
+ (0x0<<I2S_FMT_TX_OFFSET));
+ snd_soc_update_bits(codec, AC_I2S_FMT0,
+ (0x1<<I2S_FMT_RX_OFFSET),
+ (0x0<<I2S_FMT_RX_OFFSET));
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ snd_soc_update_bits(codec, AC_I2S_CLK,
+ (0x3FF<<I2S_LRCK_PERIOD),
+ (0x3F<<I2S_LRCK_PERIOD));
+ snd_soc_update_bits(codec, AC_I2S_FMT0,
+ (0x3<<I2S_FMT_MODE), (0x0<<I2S_FMT_MODE));
+ snd_soc_update_bits(codec, AC_I2S_FMT0,
+ (0x1<<I2S_FMT_TX_OFFSET),
+ (0x1<<I2S_FMT_TX_OFFSET));
+ snd_soc_update_bits(codec, AC_I2S_FMT0,
+ (0x1<<I2S_FMT_RX_OFFSET),
+ (0x1<<I2S_FMT_RX_OFFSET));
+ break;
+ case SND_SOC_DAIFMT_DSP_B:
+ snd_soc_update_bits(codec, AC_I2S_CLK,
+ (0x3FF<<I2S_LRCK_PERIOD),
+ (0x3F<<I2S_LRCK_PERIOD));
+ snd_soc_update_bits(codec, AC_I2S_FMT0,
+ (0x3<<I2S_FMT_MODE), (0x0<<I2S_FMT_MODE));
+ snd_soc_update_bits(codec, AC_I2S_FMT0,
+ (0x1<<I2S_FMT_TX_OFFSET),
+ (0x0<<I2S_FMT_TX_OFFSET));
+ snd_soc_update_bits(codec, AC_I2S_FMT0,
+ (0x1<<I2S_FMT_RX_OFFSET),
+ (0x0<<I2S_FMT_RX_OFFSET));
+ break;
+ default:
+ dev_err(codec->dev, "format setting failed\n");
+ break;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ snd_soc_update_bits(codec, AC_I2S_FMT1,
+ (0x1<<I2S_FMT_BCLK_POLAR),
+ (0x0<<I2S_FMT_BCLK_POLAR));
+ snd_soc_update_bits(codec, AC_I2S_FMT1,
+ (0x1<<I2S_FMT_LRCK_POLAR),
+ (0x0<<I2S_FMT_LRCK_POLAR));
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ snd_soc_update_bits(codec, AC_I2S_FMT1,
+ (0x1<<I2S_FMT_BCLK_POLAR),
+ (0x0<<I2S_FMT_BCLK_POLAR));
+ snd_soc_update_bits(codec, AC_I2S_FMT1,
+ (0x1<<I2S_FMT_LRCK_POLAR),
+ (0x1<<I2S_FMT_LRCK_POLAR));
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ snd_soc_update_bits(codec, AC_I2S_FMT1,
+ (0x1<<I2S_FMT_BCLK_POLAR),
+ (0x1<<I2S_FMT_BCLK_POLAR));
+ snd_soc_update_bits(codec, AC_I2S_FMT1,
+ (0x1<<I2S_FMT_LRCK_POLAR),
+ (0x0<<I2S_FMT_LRCK_POLAR));
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ snd_soc_update_bits(codec, AC_I2S_FMT1,
+ (0x1<<I2S_FMT_BCLK_POLAR),
+ (0x1<<I2S_FMT_BCLK_POLAR));
+ snd_soc_update_bits(codec, AC_I2S_FMT1,
+ (0x1<<I2S_FMT_LRCK_POLAR),
+ (0x1<<I2S_FMT_LRCK_POLAR));
+ break;
+ default:
+ dev_err(codec->dev, "invert clk setting failed\n");
+ return -EINVAL;
+ }
+ return 0;
+}
+
+static int acx00_codec_dai_set_clkdiv(struct snd_soc_dai *codec_dai,
+ int clk_id, int clk_div)
+{
+ struct acx00_priv *priv = snd_soc_dai_get_drvdata(codec_dai);
+ struct snd_soc_codec *codec = priv->codec;
+ unsigned int bclk_div;
+ /*
+ * when PCM mode, setting as 64fs, when I2S mode as 32fs,
+ * then two channel, then just as 64fs
+ */
+ unsigned int div_ratio = clk_div / 64;
+
+ switch (div_ratio) {
+ case 1:
+ bclk_div = I2S_BCLK_DIV_1;
+ break;
+ case 2:
+ bclk_div = I2S_BCLK_DIV_2;
+ break;
+ case 4:
+ bclk_div = I2S_BCLK_DIV_3;
+ break;
+ case 6:
+ bclk_div = I2S_BCLK_DIV_4;
+ break;
+ case 8:
+ bclk_div = I2S_BCLK_DIV_5;
+ break;
+ case 12:
+ bclk_div = I2S_BCLK_DIV_6;
+ break;
+ case 16:
+ bclk_div = I2S_BCLK_DIV_7;
+ break;
+ case 24:
+ bclk_div = I2S_BCLK_DIV_8;
+ break;
+ case 32:
+ bclk_div = I2S_BCLK_DIV_9;
+ break;
+ case 48:
+ bclk_div = I2S_BCLK_DIV_10;
+ break;
+ case 64:
+ bclk_div = I2S_BCLK_DIV_11;
+ break;
+ case 96:
+ bclk_div = I2S_BCLK_DIV_12;
+ break;
+ case 128:
+ bclk_div = I2S_BCLK_DIV_13;
+ break;
+ case 176:
+ bclk_div = I2S_BCLK_DIV_14;
+ break;
+ case 192:
+ bclk_div = I2S_BCLK_DIV_15;
+ break;
+ default:
+ dev_err(codec->dev, "setting blck div failed\n");
+ break;
+ }
+
+ snd_soc_update_bits(codec, AC_I2S_CLK,
+ (I2S_BCLK_DIV_MASK<<I2S_BLCK_DIV),
+ (bclk_div<<I2S_BLCK_DIV));
+ return 0;
+}
+
+static int acx00_codec_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *codec_dai)
+{
+ return 0;
+}
+
+static bool acx00_loop_en;
+module_param(acx00_loop_en, bool, 0644);
+MODULE_PARM_DESC(acx00_loop_en, "ACX00-Codec audio loopback debug(Y=enable, N=disable)");
+
+static int acx00_codec_trigger(struct snd_pcm_substream *substream,
+ int cmd, struct snd_soc_dai *codec_dai)
+{
+ return 0;
+}
+
+static int acx00_codec_prepare(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *codec_dai)
+{
+ snd_soc_update_bits(codec_dai->codec, AC_SYS_CLK_CTL,
+ (0x1<<SYS_CLK_I2S), (0x1<<SYS_CLK_I2S));
+ snd_soc_update_bits(codec_dai->codec, AC_SYS_MOD_RST,
+ (0x1<<MOD_RST_I2S), (0x1<<MOD_RST_I2S));
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ if (acx00_loop_en)
+ snd_soc_update_bits(codec_dai->codec, AC_I2S_FMT0,
+ (0x1<<I2S_FMT_LOOP),
+ (0x1<<I2S_FMT_LOOP));
+ else
+ snd_soc_update_bits(codec_dai->codec, AC_I2S_FMT0,
+ (0x1<<I2S_FMT_LOOP),
+ (0x0<<I2S_FMT_LOOP));
+ acx00_codec_txctrl_enable(codec_dai->codec, 1);
+ } else
+ acx00_codec_rxctrl_enable(codec_dai->codec, 1);
+ return 0;
+}
+
+static int acx00_codec_digital_mute(struct snd_soc_dai *codec_dai,
+ int mute)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+
+ if (mute)
+ snd_soc_write(codec, AC_I2S_DAC_VOL, 0);
+ else
+ snd_soc_write(codec, AC_I2S_DAC_VOL, ACX00_DEF_VOL);
+ return 0;
+}
+
+static void acx00_codec_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ acx00_codec_txctrl_enable(codec, 0);
+ else
+ acx00_codec_rxctrl_enable(codec, 0);
+}
+
+static const struct snd_soc_dai_ops acx00_codec_dai_ops = {
+ .hw_params = acx00_codec_hw_params,
+ .shutdown = acx00_codec_shutdown,
+ .digital_mute = acx00_codec_digital_mute,
+ .set_sysclk = acx00_codec_dai_set_sysclk,
+ .set_fmt = acx00_codec_dai_set_fmt,
+ .set_clkdiv = acx00_codec_dai_set_clkdiv,
+ .startup = acx00_codec_startup,
+ .trigger = acx00_codec_trigger,
+ .prepare = acx00_codec_prepare,
+};
+
+static struct snd_soc_dai_driver acx00_codec_dai[] = {
+ {
+ .name = "acx00-dai",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_192000
+ | SNDRV_PCM_RATE_KNOT,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE
+ | SNDRV_PCM_FMTBIT_S24_LE
+ | SNDRV_PCM_FMTBIT_S32_LE,
+ },
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_192000
+ | SNDRV_PCM_RATE_KNOT,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE
+ | SNDRV_PCM_FMTBIT_S24_LE
+ | SNDRV_PCM_FMTBIT_S32_LE,
+ },
+ .ops = &acx00_codec_dai_ops,
+ },
+};
+
+static void acx00_codec_resume_work(struct work_struct *work)
+{
+ struct acx00_priv *priv = container_of(work,
+ struct acx00_priv, resume_work.work);
+
+ acx00_codec_init(priv->codec);
+}
+
+static int acx00_codec_probe(struct snd_soc_codec *codec)
+{
+ struct acx00_priv *priv = snd_soc_codec_get_drvdata(codec);
+ struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
+ int ret = 0;
+
+ mutex_init(&priv->mutex);
+
+ priv->codec = codec;
+
+ /* Add virtual switch */
+ ret = snd_soc_add_codec_controls(codec, acx00_codec_controls,
+ ARRAY_SIZE(acx00_codec_controls));
+ if (ret) {
+ pr_err("[audio-codec] Failed to register audio mode control, will continue without it.\n");
+ }
+ snd_soc_dapm_new_controls(dapm, acx00_codec_dapm_widgets, ARRAY_SIZE(acx00_codec_dapm_widgets));
+ snd_soc_dapm_add_routes(dapm, acx00_codec_dapm_routes, ARRAY_SIZE(acx00_codec_dapm_routes));
+
+ /* using late_initcall to wait 120ms acx00-core to make chip reset */
+ acx00_codec_init(codec);
+ INIT_DELAYED_WORK(&priv->spk_work, acx00_spk_enable);
+ INIT_DELAYED_WORK(&priv->resume_work, acx00_codec_resume_work);
+ return 0;
+}
+
+static int acx00_codec_remove(struct snd_soc_codec *codec)
+{
+ struct acx00_priv *priv = snd_soc_codec_get_drvdata(codec);
+
+ cancel_delayed_work_sync(&priv->spk_work);
+ cancel_delayed_work_sync(&priv->resume_work);
+ return 0;
+}
+
+static unsigned int acx00_codec_read(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ unsigned int data;
+ struct acx00_priv *priv = snd_soc_codec_get_drvdata(codec);
+
+ /* Device I/O API */
+ data = acx00_reg_read(priv->acx00, reg);
+ return data;
+}
+
+static int acx00_codec_write(struct snd_soc_codec *codec,
+ unsigned int reg, unsigned int value)
+{
+ struct acx00_priv *priv = snd_soc_codec_get_drvdata(codec);
+
+ return acx00_reg_write(priv->acx00, reg, value);
+}
+
+static int sunxi_gpio_iodisable(u32 gpio)
+{
+ char pin_name[8];
+ u32 config, ret;
+
+ sunxi_gpio_to_name(gpio, pin_name);
+ config = 7 << 16;
+ ret = pin_config_set(SUNXI_PINCTRL, pin_name, config);
+ return ret;
+}
+
+static int acx00_codec_suspend(struct snd_soc_codec *codec)
+{
+ struct acx00_priv *priv = snd_soc_codec_get_drvdata(codec);
+
+ pr_debug("Enter %s\n", __func__);
+
+ clk_disable_unprepare(priv->clk);
+
+ /* PA_CTRL first setting low state, then make it iodisabled */
+ if (priv->spk_gpio_used) {
+ sunxi_gpio_iodisable(priv->spk_gpio);
+ msleep(30);
+ }
+
+ /*
+ * when codec suspend, then the register reset, if auto reset produce
+ * Pop & Click noise, then we should cut down the LINEOUT in this town.
+ */
+ if (priv->enable) {
+ snd_soc_update_bits(codec, AC_LINEOUT_CTL,
+ (1<<LINEOUT_EN), (0<<LINEOUT_EN));
+ snd_soc_update_bits(priv->codec, AC_LINEOUT_CTL,
+ (1<<LINEL_SRC_EN), (0<<LINEL_SRC_EN));
+ snd_soc_update_bits(priv->codec, AC_LINEOUT_CTL,
+ (1<<LINER_SRC_EN), (0<<LINER_SRC_EN));
+ priv->enable = 0;
+ }
+
+ pr_debug("Exit %s\n", __func__);
+
+ return 0;
+}
+
+static int acx00_codec_resume(struct snd_soc_codec *codec)
+{
+ struct acx00_priv *priv = snd_soc_codec_get_drvdata(codec);
+
+ pr_debug("Enter %s\n", __func__);
+
+ if (clk_prepare_enable(priv->clk)) {
+ dev_err(codec->dev, "codec resume clk failed\n");
+ return -EBUSY;
+ }
+
+ schedule_delayed_work(&priv->resume_work, msecs_to_jiffies(300));
+
+ if (priv->spk_gpio_used) {
+ gpio_direction_output(priv->spk_gpio, 1);
+ gpio_set_value(priv->spk_gpio, 0);
+ }
+
+ pr_debug("Exit %s\n", __func__);
+
+ return 0;
+}
+
+
+static int acx00_codec_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ codec->component.dapm.bias_level = level;
+ return 0;
+}
+
+struct label {
+ const char *name;
+ int value;
+};
+
+#define LABEL(constant) { #constant, constant }
+#define LABEL_END { NULL, -1 }
+
+static struct label reg_labels[] = {
+ LABEL(AC_SYS_CLK_CTL),
+ LABEL(AC_SYS_MOD_RST),
+ LABEL(AC_SYS_SR_CTL),
+ LABEL(AC_I2S_CTL),
+ LABEL(AC_I2S_CLK),
+ LABEL(AC_I2S_FMT0),
+ LABEL(AC_I2S_FMT1),
+ LABEL(AC_I2S_MIXER_SRC),
+ LABEL(AC_I2S_MIXER_GAIN),
+ LABEL(AC_I2S_DAC_VOL),
+ LABEL(AC_I2S_ADC_VOL),
+ LABEL(AC_DAC_CTL),
+ LABEL(AC_DAC_MIXER_SRC),
+ LABEL(AC_DAC_MIXER_GAIN),
+ LABEL(AC_OUT_MIXER_CTL),
+ LABEL(AC_OUT_MIXER_SRC),
+ LABEL(AC_LINEOUT_CTL),
+ LABEL(AC_ADC_CTL),
+ LABEL(AC_MICBIAS_CTL),
+ LABEL(AC_ADC_MIC_CTL),
+ LABEL(AC_ADC_MIXER_SRC),
+ LABEL(AC_BIAS_CTL),
+ LABEL(AC_ANALOG_PROF_CTL),
+ LABEL_END,
+};
+
+static ssize_t show_audio_reg(struct device *dev,
+ struct device_attribute *attr, char *buf)
+{
+ struct acx00_priv *priv = dev_get_drvdata(dev);
+ int count = 0, i = 0;
+ unsigned int reg_val;
+
+ count += sprintf(buf, "dump audio reg:\n");
+
+ while (reg_labels[i].name != NULL) {
+ reg_val = acx00_reg_read(priv->acx00, reg_labels[i].value);
+ count += sprintf(buf + count, "%s 0x%x: 0x%04x\n",
+ reg_labels[i].name, (reg_labels[i].value), reg_val);
+ i++;
+ }
+
+ return count;
+}
+
+/*
+ * param 1: 0 read;1 write
+ * param 2: 1 digital reg; 2 analog reg
+ * param 3: reg value;
+ * param 4: write value;
+ * read:
+ * echo 0,1,0x00> audio_reg
+ * echo 0,2,0x00> audio_reg
+ * write:
+ * echo 1,1,0x00,0xa > audio_reg
+ * echo 1,2,0x00,0xff > audio_reg
+*/
+static ssize_t store_audio_reg(struct device *dev,
+ struct device_attribute *attr, const char *buf, size_t count)
+{
+ int ret;
+ int rw_flag;
+ unsigned int input_reg_val = 0;
+ int input_reg_group = 0;
+ unsigned int input_reg_offset = 0;
+ struct acx00_priv *priv = dev_get_drvdata(dev);
+
+ ret = sscanf(buf, "%d,%d,0x%x,0x%x", &rw_flag, &input_reg_group,
+ &input_reg_offset, &input_reg_val);
+ dev_info(dev, "ret:%d, reg_group:%d, reg_offset:%d, reg_val:0x%x\n",
+ ret, input_reg_group, input_reg_offset, input_reg_val);
+
+ if (input_reg_group != 1) {
+ pr_err("not exist reg group\n");
+ ret = count;
+ goto out;
+ }
+ if (!(rw_flag == 1 || rw_flag == 0)) {
+ pr_err("not rw_flag\n");
+ ret = count;
+ goto out;
+ }
+
+ if (rw_flag) {
+ acx00_reg_write(priv->acx00, input_reg_offset, input_reg_val);
+ } else {
+ input_reg_val = acx00_reg_read(priv->acx00, input_reg_offset);
+ dev_info(dev, "\n\n Reg[0x%x] : 0x%04x\n\n",
+ input_reg_offset, input_reg_val);
+ }
+ ret = count;
+
+out:
+ return ret;
+}
+
+static DEVICE_ATTR(audio_reg, 0644, show_audio_reg, store_audio_reg);
+
+static struct attribute *audio_debug_attrs[] = {
+ &dev_attr_audio_reg.attr,
+ NULL,
+};
+
+static struct attribute_group audio_debug_attr_group = {
+ .name = "audio_reg_debug",
+ .attrs = audio_debug_attrs,
+};
+
+static struct snd_soc_codec_driver soc_codec_driver_acx00 = {
+ .probe = acx00_codec_probe,
+ .remove = acx00_codec_remove,
+ .suspend = acx00_codec_suspend,
+ .resume = acx00_codec_resume,
+ .read = acx00_codec_read,
+ .write = acx00_codec_write,
+ .ignore_pmdown_time = 1,
+ .set_bias_level = acx00_codec_set_bias_level,
+};
+
+/* through acx00 is part of mfd devices, after the mfd */
+static int acx00_codec_dev_probe(struct platform_device *pdev)
+{
+ struct acx00_priv *priv;
+ int ret;
+ struct device_node *np = of_find_compatible_node(NULL, NULL,
+ "allwinner,ac200_codec");
+
+ priv = devm_kzalloc(&pdev->dev, sizeof(struct acx00_priv), GFP_KERNEL);
+ if (!priv) {
+ dev_err(&pdev->dev, "acx00 codec priv mem alloc failed\n");
+ return -ENOMEM;
+ }
+
+ platform_set_drvdata(pdev, priv);
+ priv->acx00 = dev_get_drvdata(pdev->dev.parent);
+
+ if (np) {
+ ret = of_get_named_gpio(np, "gpio-spk", 0);
+ if (ret >= 0) {
+ priv->spk_gpio_used = 1;
+ priv->spk_gpio = ret;
+ if (!gpio_is_valid(priv->spk_gpio)) {
+ dev_err(&pdev->dev, "gpio-spk is valid\n");
+ ret = -EINVAL;
+ goto err_devm_kfree;
+ } else {
+ ret = devm_gpio_request(&pdev->dev,
+ priv->spk_gpio, "SPK");
+ if (ret) {
+ dev_err(&pdev->dev,
+ "failed request gpio-spk\n");
+ ret = -EBUSY;
+ goto err_devm_kfree;
+ } else {
+ gpio_direction_output(priv->spk_gpio,
+ 1);
+ gpio_set_value(priv->spk_gpio, 0);
+ }
+ }
+ } else {
+ priv->spk_gpio_used = 0;
+ }
+ }
+
+ ret = snd_soc_register_codec(&pdev->dev, &soc_codec_driver_acx00,
+ acx00_codec_dai, ARRAY_SIZE(acx00_codec_dai));
+
+ if (ret < 0)
+ dev_err(&pdev->dev, "Failed register acx00: %d\n", ret);
+
+ ret = sysfs_create_group(&pdev->dev.kobj, &audio_debug_attr_group);
+ if (ret)
+ dev_warn(&pdev->dev, "failed to create attr group\n");
+
+ return 0;
+
+err_devm_kfree:
+ devm_kfree(&pdev->dev, priv);
+ return ret;
+}
+
+/* Mark this space to clear the LINEOUT & gpio */
+static void acx00_codec_dev_shutdown(struct platform_device *pdev)
+{
+ struct acx00_priv *priv = platform_get_drvdata(pdev);
+
+ if (priv->spk_gpio_used)
+ gpio_set_value(priv->spk_gpio, 0);
+}
+
+static int acx00_codec_dev_remove(struct platform_device *pdev)
+{
+ struct acx00_priv *priv = platform_get_drvdata(pdev);
+
+#ifndef ACX00_DAPM_LINEOUT
+ snd_soc_update_bits(priv->codec, AC_LINEOUT_CTL,
+ (1<<LINEOUT_EN), (0<<LINEOUT_EN));
+#endif
+ snd_soc_unregister_codec(&pdev->dev);
+ clk_disable_unprepare(priv->clk);
+ devm_kfree(&pdev->dev, priv);
+ return 0;
+}
+
+static struct platform_driver acx00_codec_driver = {
+ .driver = {
+ .name = "acx00-codec",
+ .owner = THIS_MODULE,
+ },
+ .probe = acx00_codec_dev_probe,
+ .remove = acx00_codec_dev_remove,
+ .shutdown = acx00_codec_dev_shutdown,
+};
+
+static int __init acx00_codec_driver_init(void)
+{
+ return platform_driver_register(&acx00_codec_driver);
+}
+
+static void __exit acx00_codec_driver_exit(void)
+{
+ platform_driver_unregister(&acx00_codec_driver);
+}
+late_initcall(acx00_codec_driver_init);
+module_exit(acx00_codec_driver_exit);
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("SUNXI ASoC ACX00 Codec Driver");
+MODULE_AUTHOR("wolfgang huang");
+MODULE_ALIAS("platform:acx00-codec");
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