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https://github.com/AuxXxilium/linux_dsm_epyc7002.git
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e88e495064
codec driver and component driver has duplicated callback functions, and codec side functions are just copied to component side when register timing. This was quick-hack, but no longer needed. This patch moves these functions from codec driver to component driver. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
619 lines
16 KiB
C
619 lines
16 KiB
C
/*
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* ALSA SoC TLV320AIC23 codec driver
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*
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* Author: Arun KS, <arunks@mistralsolutions.com>
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* Copyright: (C) 2008 Mistral Solutions Pvt Ltd.,
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*
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* Based on sound/soc/codecs/wm8731.c by Richard Purdie
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*
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* This program is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License version 2 as
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* published by the Free Software Foundation.
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*
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* Notes:
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* The AIC23 is a driver for a low power stereo audio
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* codec tlv320aic23
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*
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* The machine layer should disable unsupported inputs/outputs by
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* snd_soc_dapm_disable_pin(codec, "LHPOUT"), etc.
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*/
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#include <linux/module.h>
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#include <linux/moduleparam.h>
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#include <linux/init.h>
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#include <linux/delay.h>
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#include <linux/pm.h>
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#include <linux/regmap.h>
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#include <linux/slab.h>
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#include <sound/core.h>
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#include <sound/pcm.h>
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#include <sound/pcm_params.h>
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#include <sound/soc.h>
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#include <sound/tlv.h>
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#include <sound/initval.h>
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#include "tlv320aic23.h"
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/*
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* AIC23 register cache
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*/
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static const struct reg_default tlv320aic23_reg[] = {
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{ 0, 0x0097 },
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{ 1, 0x0097 },
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{ 2, 0x00F9 },
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{ 3, 0x00F9 },
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{ 4, 0x001A },
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{ 5, 0x0004 },
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{ 6, 0x0007 },
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{ 7, 0x0001 },
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{ 8, 0x0020 },
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{ 9, 0x0000 },
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};
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const struct regmap_config tlv320aic23_regmap = {
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.reg_bits = 7,
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.val_bits = 9,
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.max_register = TLV320AIC23_RESET,
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.reg_defaults = tlv320aic23_reg,
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.num_reg_defaults = ARRAY_SIZE(tlv320aic23_reg),
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.cache_type = REGCACHE_RBTREE,
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};
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EXPORT_SYMBOL(tlv320aic23_regmap);
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static const char *rec_src_text[] = { "Line", "Mic" };
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static const char *deemph_text[] = {"None", "32Khz", "44.1Khz", "48Khz"};
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static SOC_ENUM_SINGLE_DECL(rec_src_enum,
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TLV320AIC23_ANLG, 2, rec_src_text);
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static const struct snd_kcontrol_new tlv320aic23_rec_src_mux_controls =
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SOC_DAPM_ENUM("Input Select", rec_src_enum);
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static SOC_ENUM_SINGLE_DECL(tlv320aic23_rec_src,
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TLV320AIC23_ANLG, 2, rec_src_text);
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static SOC_ENUM_SINGLE_DECL(tlv320aic23_deemph,
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TLV320AIC23_DIGT, 1, deemph_text);
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static const DECLARE_TLV_DB_SCALE(out_gain_tlv, -12100, 100, 0);
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static const DECLARE_TLV_DB_SCALE(input_gain_tlv, -1725, 75, 0);
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static const DECLARE_TLV_DB_SCALE(sidetone_vol_tlv, -1800, 300, 0);
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static int snd_soc_tlv320aic23_put_volsw(struct snd_kcontrol *kcontrol,
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struct snd_ctl_elem_value *ucontrol)
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{
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struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
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u16 val, reg;
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val = (ucontrol->value.integer.value[0] & 0x07);
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/* linear conversion to userspace
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* 000 = -6db
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* 001 = -9db
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* 010 = -12db
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* 011 = -18db (Min)
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* 100 = 0db (Max)
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*/
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val = (val >= 4) ? 4 : (3 - val);
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reg = snd_soc_read(codec, TLV320AIC23_ANLG) & (~0x1C0);
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snd_soc_write(codec, TLV320AIC23_ANLG, reg | (val << 6));
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return 0;
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}
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static int snd_soc_tlv320aic23_get_volsw(struct snd_kcontrol *kcontrol,
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struct snd_ctl_elem_value *ucontrol)
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{
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struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
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u16 val;
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val = snd_soc_read(codec, TLV320AIC23_ANLG) & (0x1C0);
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val = val >> 6;
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val = (val >= 4) ? 4 : (3 - val);
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ucontrol->value.integer.value[0] = val;
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return 0;
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}
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static const struct snd_kcontrol_new tlv320aic23_snd_controls[] = {
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SOC_DOUBLE_R_TLV("Digital Playback Volume", TLV320AIC23_LCHNVOL,
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TLV320AIC23_RCHNVOL, 0, 127, 0, out_gain_tlv),
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SOC_SINGLE("Digital Playback Switch", TLV320AIC23_DIGT, 3, 1, 1),
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SOC_DOUBLE_R("Line Input Switch", TLV320AIC23_LINVOL,
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TLV320AIC23_RINVOL, 7, 1, 0),
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SOC_DOUBLE_R_TLV("Line Input Volume", TLV320AIC23_LINVOL,
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TLV320AIC23_RINVOL, 0, 31, 0, input_gain_tlv),
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SOC_SINGLE("Mic Input Switch", TLV320AIC23_ANLG, 1, 1, 1),
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SOC_SINGLE("Mic Booster Switch", TLV320AIC23_ANLG, 0, 1, 0),
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SOC_SINGLE_EXT_TLV("Sidetone Volume", TLV320AIC23_ANLG, 6, 4, 0,
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snd_soc_tlv320aic23_get_volsw,
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snd_soc_tlv320aic23_put_volsw, sidetone_vol_tlv),
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SOC_ENUM("Playback De-emphasis", tlv320aic23_deemph),
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};
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/* PGA Mixer controls for Line and Mic switch */
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static const struct snd_kcontrol_new tlv320aic23_output_mixer_controls[] = {
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SOC_DAPM_SINGLE("Line Bypass Switch", TLV320AIC23_ANLG, 3, 1, 0),
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SOC_DAPM_SINGLE("Mic Sidetone Switch", TLV320AIC23_ANLG, 5, 1, 0),
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SOC_DAPM_SINGLE("Playback Switch", TLV320AIC23_ANLG, 4, 1, 0),
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};
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static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = {
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SND_SOC_DAPM_DAC("DAC", "Playback", TLV320AIC23_PWR, 3, 1),
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SND_SOC_DAPM_ADC("ADC", "Capture", TLV320AIC23_PWR, 2, 1),
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SND_SOC_DAPM_MUX("Capture Source", SND_SOC_NOPM, 0, 0,
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&tlv320aic23_rec_src_mux_controls),
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SND_SOC_DAPM_MIXER("Output Mixer", TLV320AIC23_PWR, 4, 1,
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&tlv320aic23_output_mixer_controls[0],
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ARRAY_SIZE(tlv320aic23_output_mixer_controls)),
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SND_SOC_DAPM_PGA("Line Input", TLV320AIC23_PWR, 0, 1, NULL, 0),
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SND_SOC_DAPM_PGA("Mic Input", TLV320AIC23_PWR, 1, 1, NULL, 0),
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SND_SOC_DAPM_OUTPUT("LHPOUT"),
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SND_SOC_DAPM_OUTPUT("RHPOUT"),
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SND_SOC_DAPM_OUTPUT("LOUT"),
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SND_SOC_DAPM_OUTPUT("ROUT"),
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SND_SOC_DAPM_INPUT("LLINEIN"),
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SND_SOC_DAPM_INPUT("RLINEIN"),
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SND_SOC_DAPM_INPUT("MICIN"),
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};
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static const struct snd_soc_dapm_route tlv320aic23_intercon[] = {
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/* Output Mixer */
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{"Output Mixer", "Line Bypass Switch", "Line Input"},
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{"Output Mixer", "Playback Switch", "DAC"},
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{"Output Mixer", "Mic Sidetone Switch", "Mic Input"},
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/* Outputs */
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{"RHPOUT", NULL, "Output Mixer"},
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{"LHPOUT", NULL, "Output Mixer"},
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{"LOUT", NULL, "Output Mixer"},
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{"ROUT", NULL, "Output Mixer"},
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/* Inputs */
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{"Line Input", "NULL", "LLINEIN"},
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{"Line Input", "NULL", "RLINEIN"},
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{"Mic Input", "NULL", "MICIN"},
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/* input mux */
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{"Capture Source", "Line", "Line Input"},
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{"Capture Source", "Mic", "Mic Input"},
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{"ADC", NULL, "Capture Source"},
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};
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/* AIC23 driver data */
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struct aic23 {
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struct regmap *regmap;
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int mclk;
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int requested_adc;
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int requested_dac;
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};
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/*
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* Common Crystals used
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* 11.2896 Mhz /128 = *88.2k /192 = 58.8k
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* 12.0000 Mhz /125 = *96k /136 = 88.235K
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* 12.2880 Mhz /128 = *96k /192 = 64k
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* 16.9344 Mhz /128 = 132.3k /192 = *88.2k
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* 18.4320 Mhz /128 = 144k /192 = *96k
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*/
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/*
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* Normal BOSR 0-256/2 = 128, 1-384/2 = 192
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* USB BOSR 0-250/2 = 125, 1-272/2 = 136
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*/
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static const int bosr_usb_divisor_table[] = {
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128, 125, 192, 136
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};
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#define LOWER_GROUP ((1<<0) | (1<<1) | (1<<2) | (1<<3) | (1<<6) | (1<<7))
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#define UPPER_GROUP ((1<<8) | (1<<9) | (1<<10) | (1<<11) | (1<<15))
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static const unsigned short sr_valid_mask[] = {
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LOWER_GROUP|UPPER_GROUP, /* Normal, bosr - 0*/
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LOWER_GROUP, /* Usb, bosr - 0*/
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LOWER_GROUP|UPPER_GROUP, /* Normal, bosr - 1*/
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UPPER_GROUP, /* Usb, bosr - 1*/
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};
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/*
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* Every divisor is a factor of 11*12
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*/
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#define SR_MULT (11*12)
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#define A(x) (SR_MULT/x)
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static const unsigned char sr_adc_mult_table[] = {
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A(2), A(2), A(12), A(12), 0, 0, A(3), A(1),
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A(2), A(2), A(11), A(11), 0, 0, 0, A(1)
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};
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static const unsigned char sr_dac_mult_table[] = {
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A(2), A(12), A(2), A(12), 0, 0, A(3), A(1),
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A(2), A(11), A(2), A(11), 0, 0, 0, A(1)
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};
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static unsigned get_score(int adc, int adc_l, int adc_h, int need_adc,
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int dac, int dac_l, int dac_h, int need_dac)
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{
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if ((adc >= adc_l) && (adc <= adc_h) &&
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(dac >= dac_l) && (dac <= dac_h)) {
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int diff_adc = need_adc - adc;
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int diff_dac = need_dac - dac;
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return abs(diff_adc) + abs(diff_dac);
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}
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return UINT_MAX;
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}
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static int find_rate(int mclk, u32 need_adc, u32 need_dac)
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{
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int i, j;
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int best_i = -1;
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int best_j = -1;
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int best_div = 0;
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unsigned best_score = UINT_MAX;
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int adc_l, adc_h, dac_l, dac_h;
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need_adc *= SR_MULT;
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need_dac *= SR_MULT;
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/*
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* rates given are +/- 1/32
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*/
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adc_l = need_adc - (need_adc >> 5);
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adc_h = need_adc + (need_adc >> 5);
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dac_l = need_dac - (need_dac >> 5);
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dac_h = need_dac + (need_dac >> 5);
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for (i = 0; i < ARRAY_SIZE(bosr_usb_divisor_table); i++) {
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int base = mclk / bosr_usb_divisor_table[i];
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int mask = sr_valid_mask[i];
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for (j = 0; j < ARRAY_SIZE(sr_adc_mult_table);
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j++, mask >>= 1) {
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int adc;
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int dac;
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int score;
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if ((mask & 1) == 0)
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continue;
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adc = base * sr_adc_mult_table[j];
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dac = base * sr_dac_mult_table[j];
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score = get_score(adc, adc_l, adc_h, need_adc,
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dac, dac_l, dac_h, need_dac);
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if (best_score > score) {
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best_score = score;
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best_i = i;
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best_j = j;
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best_div = 0;
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}
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score = get_score((adc >> 1), adc_l, adc_h, need_adc,
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(dac >> 1), dac_l, dac_h, need_dac);
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/* prefer to have a /2 */
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if ((score != UINT_MAX) && (best_score >= score)) {
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best_score = score;
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best_i = i;
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best_j = j;
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best_div = 1;
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}
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}
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}
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return (best_j << 2) | best_i | (best_div << TLV320AIC23_CLKIN_SHIFT);
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}
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#ifdef DEBUG
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static void get_current_sample_rates(struct snd_soc_codec *codec, int mclk,
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u32 *sample_rate_adc, u32 *sample_rate_dac)
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{
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int src = snd_soc_read(codec, TLV320AIC23_SRATE);
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int sr = (src >> 2) & 0x0f;
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int val = (mclk / bosr_usb_divisor_table[src & 3]);
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int adc = (val * sr_adc_mult_table[sr]) / SR_MULT;
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int dac = (val * sr_dac_mult_table[sr]) / SR_MULT;
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if (src & TLV320AIC23_CLKIN_HALF) {
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adc >>= 1;
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dac >>= 1;
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}
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*sample_rate_adc = adc;
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*sample_rate_dac = dac;
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}
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#endif
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static int set_sample_rate_control(struct snd_soc_codec *codec, int mclk,
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u32 sample_rate_adc, u32 sample_rate_dac)
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{
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/* Search for the right sample rate */
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int data = find_rate(mclk, sample_rate_adc, sample_rate_dac);
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if (data < 0) {
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printk(KERN_ERR "%s:Invalid rate %u,%u requested\n",
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__func__, sample_rate_adc, sample_rate_dac);
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return -EINVAL;
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}
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snd_soc_write(codec, TLV320AIC23_SRATE, data);
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#ifdef DEBUG
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{
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u32 adc, dac;
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get_current_sample_rates(codec, mclk, &adc, &dac);
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printk(KERN_DEBUG "actual samplerate = %u,%u reg=%x\n",
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adc, dac, data);
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}
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#endif
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return 0;
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}
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static int tlv320aic23_hw_params(struct snd_pcm_substream *substream,
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struct snd_pcm_hw_params *params,
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struct snd_soc_dai *dai)
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{
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struct snd_soc_codec *codec = dai->codec;
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u16 iface_reg;
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int ret;
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struct aic23 *aic23 = snd_soc_codec_get_drvdata(codec);
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u32 sample_rate_adc = aic23->requested_adc;
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u32 sample_rate_dac = aic23->requested_dac;
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u32 sample_rate = params_rate(params);
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if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
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aic23->requested_dac = sample_rate_dac = sample_rate;
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if (!sample_rate_adc)
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sample_rate_adc = sample_rate;
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} else {
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aic23->requested_adc = sample_rate_adc = sample_rate;
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if (!sample_rate_dac)
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sample_rate_dac = sample_rate;
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}
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ret = set_sample_rate_control(codec, aic23->mclk, sample_rate_adc,
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sample_rate_dac);
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if (ret < 0)
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return ret;
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iface_reg = snd_soc_read(codec, TLV320AIC23_DIGT_FMT) & ~(0x03 << 2);
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switch (params_width(params)) {
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case 16:
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break;
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case 20:
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iface_reg |= (0x01 << 2);
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break;
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case 24:
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iface_reg |= (0x02 << 2);
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break;
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case 32:
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iface_reg |= (0x03 << 2);
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break;
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}
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snd_soc_write(codec, TLV320AIC23_DIGT_FMT, iface_reg);
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return 0;
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}
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static int tlv320aic23_pcm_prepare(struct snd_pcm_substream *substream,
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struct snd_soc_dai *dai)
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{
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struct snd_soc_codec *codec = dai->codec;
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/* set active */
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snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x0001);
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return 0;
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}
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static void tlv320aic23_shutdown(struct snd_pcm_substream *substream,
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struct snd_soc_dai *dai)
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{
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struct snd_soc_codec *codec = dai->codec;
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struct aic23 *aic23 = snd_soc_codec_get_drvdata(codec);
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/* deactivate */
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if (!snd_soc_codec_is_active(codec)) {
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udelay(50);
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snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x0);
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}
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if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
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aic23->requested_dac = 0;
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else
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aic23->requested_adc = 0;
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}
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static int tlv320aic23_mute(struct snd_soc_dai *dai, int mute)
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{
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struct snd_soc_codec *codec = dai->codec;
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u16 reg;
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reg = snd_soc_read(codec, TLV320AIC23_DIGT);
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if (mute)
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reg |= TLV320AIC23_DACM_MUTE;
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else
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reg &= ~TLV320AIC23_DACM_MUTE;
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snd_soc_write(codec, TLV320AIC23_DIGT, reg);
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return 0;
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}
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static int tlv320aic23_set_dai_fmt(struct snd_soc_dai *codec_dai,
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unsigned int fmt)
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{
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struct snd_soc_codec *codec = codec_dai->codec;
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u16 iface_reg;
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|
|
iface_reg = snd_soc_read(codec, TLV320AIC23_DIGT_FMT) & (~0x03);
|
|
|
|
/* set master/slave audio interface */
|
|
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
|
|
case SND_SOC_DAIFMT_CBM_CFM:
|
|
iface_reg |= TLV320AIC23_MS_MASTER;
|
|
break;
|
|
case SND_SOC_DAIFMT_CBS_CFS:
|
|
iface_reg &= ~TLV320AIC23_MS_MASTER;
|
|
break;
|
|
default:
|
|
return -EINVAL;
|
|
|
|
}
|
|
|
|
/* interface format */
|
|
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
|
|
case SND_SOC_DAIFMT_I2S:
|
|
iface_reg |= TLV320AIC23_FOR_I2S;
|
|
break;
|
|
case SND_SOC_DAIFMT_DSP_A:
|
|
iface_reg |= TLV320AIC23_LRP_ON;
|
|
case SND_SOC_DAIFMT_DSP_B:
|
|
iface_reg |= TLV320AIC23_FOR_DSP;
|
|
break;
|
|
case SND_SOC_DAIFMT_RIGHT_J:
|
|
break;
|
|
case SND_SOC_DAIFMT_LEFT_J:
|
|
iface_reg |= TLV320AIC23_FOR_LJUST;
|
|
break;
|
|
default:
|
|
return -EINVAL;
|
|
|
|
}
|
|
|
|
snd_soc_write(codec, TLV320AIC23_DIGT_FMT, iface_reg);
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int tlv320aic23_set_dai_sysclk(struct snd_soc_dai *codec_dai,
|
|
int clk_id, unsigned int freq, int dir)
|
|
{
|
|
struct aic23 *aic23 = snd_soc_dai_get_drvdata(codec_dai);
|
|
aic23->mclk = freq;
|
|
return 0;
|
|
}
|
|
|
|
static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec,
|
|
enum snd_soc_bias_level level)
|
|
{
|
|
u16 reg = snd_soc_read(codec, TLV320AIC23_PWR) & 0x17f;
|
|
|
|
switch (level) {
|
|
case SND_SOC_BIAS_ON:
|
|
/* vref/mid, osc on, dac unmute */
|
|
reg &= ~(TLV320AIC23_DEVICE_PWR_OFF | TLV320AIC23_OSC_OFF | \
|
|
TLV320AIC23_DAC_OFF);
|
|
snd_soc_write(codec, TLV320AIC23_PWR, reg);
|
|
break;
|
|
case SND_SOC_BIAS_PREPARE:
|
|
break;
|
|
case SND_SOC_BIAS_STANDBY:
|
|
/* everything off except vref/vmid, */
|
|
snd_soc_write(codec, TLV320AIC23_PWR,
|
|
reg | TLV320AIC23_CLK_OFF);
|
|
break;
|
|
case SND_SOC_BIAS_OFF:
|
|
/* everything off, dac mute, inactive */
|
|
snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x0);
|
|
snd_soc_write(codec, TLV320AIC23_PWR, 0x1ff);
|
|
break;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
#define AIC23_RATES SNDRV_PCM_RATE_8000_96000
|
|
#define AIC23_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
|
|
SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE)
|
|
|
|
static const struct snd_soc_dai_ops tlv320aic23_dai_ops = {
|
|
.prepare = tlv320aic23_pcm_prepare,
|
|
.hw_params = tlv320aic23_hw_params,
|
|
.shutdown = tlv320aic23_shutdown,
|
|
.digital_mute = tlv320aic23_mute,
|
|
.set_fmt = tlv320aic23_set_dai_fmt,
|
|
.set_sysclk = tlv320aic23_set_dai_sysclk,
|
|
};
|
|
|
|
static struct snd_soc_dai_driver tlv320aic23_dai = {
|
|
.name = "tlv320aic23-hifi",
|
|
.playback = {
|
|
.stream_name = "Playback",
|
|
.channels_min = 2,
|
|
.channels_max = 2,
|
|
.rates = AIC23_RATES,
|
|
.formats = AIC23_FORMATS,},
|
|
.capture = {
|
|
.stream_name = "Capture",
|
|
.channels_min = 2,
|
|
.channels_max = 2,
|
|
.rates = AIC23_RATES,
|
|
.formats = AIC23_FORMATS,},
|
|
.ops = &tlv320aic23_dai_ops,
|
|
};
|
|
|
|
static int tlv320aic23_resume(struct snd_soc_codec *codec)
|
|
{
|
|
struct aic23 *aic23 = snd_soc_codec_get_drvdata(codec);
|
|
regcache_mark_dirty(aic23->regmap);
|
|
regcache_sync(aic23->regmap);
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int tlv320aic23_codec_probe(struct snd_soc_codec *codec)
|
|
{
|
|
/* Reset codec */
|
|
snd_soc_write(codec, TLV320AIC23_RESET, 0);
|
|
|
|
snd_soc_write(codec, TLV320AIC23_DIGT, TLV320AIC23_DEEMP_44K);
|
|
|
|
/* Unmute input */
|
|
snd_soc_update_bits(codec, TLV320AIC23_LINVOL,
|
|
TLV320AIC23_LIM_MUTED, TLV320AIC23_LRS_ENABLED);
|
|
|
|
snd_soc_update_bits(codec, TLV320AIC23_RINVOL,
|
|
TLV320AIC23_LIM_MUTED, TLV320AIC23_LRS_ENABLED);
|
|
|
|
snd_soc_update_bits(codec, TLV320AIC23_ANLG,
|
|
TLV320AIC23_BYPASS_ON | TLV320AIC23_MICM_MUTED,
|
|
0);
|
|
|
|
/* Default output volume */
|
|
snd_soc_write(codec, TLV320AIC23_LCHNVOL,
|
|
TLV320AIC23_DEFAULT_OUT_VOL & TLV320AIC23_OUT_VOL_MASK);
|
|
snd_soc_write(codec, TLV320AIC23_RCHNVOL,
|
|
TLV320AIC23_DEFAULT_OUT_VOL & TLV320AIC23_OUT_VOL_MASK);
|
|
|
|
snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x1);
|
|
|
|
return 0;
|
|
}
|
|
|
|
static struct snd_soc_codec_driver soc_codec_dev_tlv320aic23 = {
|
|
.probe = tlv320aic23_codec_probe,
|
|
.resume = tlv320aic23_resume,
|
|
.set_bias_level = tlv320aic23_set_bias_level,
|
|
.suspend_bias_off = true,
|
|
|
|
.component_driver = {
|
|
.controls = tlv320aic23_snd_controls,
|
|
.num_controls = ARRAY_SIZE(tlv320aic23_snd_controls),
|
|
.dapm_widgets = tlv320aic23_dapm_widgets,
|
|
.num_dapm_widgets = ARRAY_SIZE(tlv320aic23_dapm_widgets),
|
|
.dapm_routes = tlv320aic23_intercon,
|
|
.num_dapm_routes = ARRAY_SIZE(tlv320aic23_intercon),
|
|
},
|
|
};
|
|
|
|
int tlv320aic23_probe(struct device *dev, struct regmap *regmap)
|
|
{
|
|
struct aic23 *aic23;
|
|
|
|
if (IS_ERR(regmap))
|
|
return PTR_ERR(regmap);
|
|
|
|
aic23 = devm_kzalloc(dev, sizeof(struct aic23), GFP_KERNEL);
|
|
if (aic23 == NULL)
|
|
return -ENOMEM;
|
|
|
|
aic23->regmap = regmap;
|
|
|
|
dev_set_drvdata(dev, aic23);
|
|
|
|
return snd_soc_register_codec(dev, &soc_codec_dev_tlv320aic23,
|
|
&tlv320aic23_dai, 1);
|
|
}
|
|
EXPORT_SYMBOL(tlv320aic23_probe);
|
|
|
|
MODULE_DESCRIPTION("ASoC TLV320AIC23 codec driver");
|
|
MODULE_AUTHOR("Arun KS <arunks@mistralsolutions.com>");
|
|
MODULE_LICENSE("GPL");
|