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85362efb80
Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes the code a bit shorter and cleaner. While we are at it also remove the regcache_cache_only() calls from suspend/resume as there shouldn't be any IO between suspend and resume. Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe() can also be removed as the core will automatically do this after the CODEC has been probed. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@kernel.org>
640 lines
17 KiB
C
640 lines
17 KiB
C
/*
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* File: sound/soc/codecs/ssm2602.c
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* Author: Cliff Cai <Cliff.Cai@analog.com>
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*
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* Created: Tue June 06 2008
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* Description: Driver for ssm2602 sound chip
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*
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* Modified:
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* Copyright 2008 Analog Devices Inc.
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*
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* Bugs: Enter bugs at http://blackfin.uclinux.org/
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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 as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License
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* along with this program; if not, see the file COPYING, or write
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* to the Free Software Foundation, Inc.,
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* 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include <linux/module.h>
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#include <linux/regmap.h>
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#include <linux/slab.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 "ssm2602.h"
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/* codec private data */
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struct ssm2602_priv {
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unsigned int sysclk;
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const struct snd_pcm_hw_constraint_list *sysclk_constraints;
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struct regmap *regmap;
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enum ssm2602_type type;
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unsigned int clk_out_pwr;
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};
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/*
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* ssm2602 register cache
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* We can't read the ssm2602 register space when we are
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* using 2 wire for device control, so we cache them instead.
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* There is no point in caching the reset register
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*/
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static const u16 ssm2602_reg[SSM2602_CACHEREGNUM] = {
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0x0097, 0x0097, 0x0079, 0x0079,
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0x000a, 0x0008, 0x009f, 0x000a,
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0x0000, 0x0000
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};
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/*Appending several "None"s just for OSS mixer use*/
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static const char *ssm2602_input_select[] = {
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"Line", "Mic",
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};
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static const char *ssm2602_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"};
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static const struct soc_enum ssm2602_enum[] = {
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SOC_ENUM_SINGLE(SSM2602_APANA, 2, ARRAY_SIZE(ssm2602_input_select),
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ssm2602_input_select),
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SOC_ENUM_SINGLE(SSM2602_APDIGI, 1, ARRAY_SIZE(ssm2602_deemph),
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ssm2602_deemph),
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};
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static const unsigned int ssm260x_outmix_tlv[] = {
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TLV_DB_RANGE_HEAD(2),
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0, 47, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 0),
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48, 127, TLV_DB_SCALE_ITEM(-7400, 100, 0),
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};
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static const DECLARE_TLV_DB_SCALE(ssm260x_inpga_tlv, -3450, 150, 0);
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static const DECLARE_TLV_DB_SCALE(ssm260x_sidetone_tlv, -1500, 300, 0);
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static const struct snd_kcontrol_new ssm260x_snd_controls[] = {
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SOC_DOUBLE_R_TLV("Capture Volume", SSM2602_LINVOL, SSM2602_RINVOL, 0, 45, 0,
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ssm260x_inpga_tlv),
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SOC_DOUBLE_R("Capture Switch", SSM2602_LINVOL, SSM2602_RINVOL, 7, 1, 1),
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SOC_SINGLE("ADC High Pass Filter Switch", SSM2602_APDIGI, 0, 1, 1),
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SOC_SINGLE("Store DC Offset Switch", SSM2602_APDIGI, 4, 1, 0),
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SOC_ENUM("Playback De-emphasis", ssm2602_enum[1]),
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};
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static const struct snd_kcontrol_new ssm2602_snd_controls[] = {
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SOC_DOUBLE_R_TLV("Master Playback Volume", SSM2602_LOUT1V, SSM2602_ROUT1V,
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0, 127, 0, ssm260x_outmix_tlv),
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SOC_DOUBLE_R("Master Playback ZC Switch", SSM2602_LOUT1V, SSM2602_ROUT1V,
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7, 1, 0),
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SOC_SINGLE_TLV("Sidetone Playback Volume", SSM2602_APANA, 6, 3, 1,
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ssm260x_sidetone_tlv),
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SOC_SINGLE("Mic Boost (+20dB)", SSM2602_APANA, 0, 1, 0),
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SOC_SINGLE("Mic Boost2 (+20dB)", SSM2602_APANA, 8, 1, 0),
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SOC_SINGLE("Mic Switch", SSM2602_APANA, 1, 1, 1),
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};
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/* Output Mixer */
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static const struct snd_kcontrol_new ssm260x_output_mixer_controls[] = {
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SOC_DAPM_SINGLE("Line Bypass Switch", SSM2602_APANA, 3, 1, 0),
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SOC_DAPM_SINGLE("HiFi Playback Switch", SSM2602_APANA, 4, 1, 0),
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SOC_DAPM_SINGLE("Mic Sidetone Switch", SSM2602_APANA, 5, 1, 0),
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};
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/* Input mux */
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static const struct snd_kcontrol_new ssm2602_input_mux_controls =
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SOC_DAPM_ENUM("Input Select", ssm2602_enum[0]);
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static const struct snd_soc_dapm_widget ssm260x_dapm_widgets[] = {
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SND_SOC_DAPM_DAC("DAC", "HiFi Playback", SSM2602_PWR, 3, 1),
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SND_SOC_DAPM_ADC("ADC", "HiFi Capture", SSM2602_PWR, 2, 1),
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SND_SOC_DAPM_PGA("Line Input", SSM2602_PWR, 0, 1, NULL, 0),
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SND_SOC_DAPM_SUPPLY("Digital Core Power", SSM2602_ACTIVE, 0, 0, NULL, 0),
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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("RLINEIN"),
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SND_SOC_DAPM_INPUT("LLINEIN"),
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};
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static const struct snd_soc_dapm_widget ssm2602_dapm_widgets[] = {
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SND_SOC_DAPM_MIXER("Output Mixer", SSM2602_PWR, 4, 1,
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ssm260x_output_mixer_controls,
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ARRAY_SIZE(ssm260x_output_mixer_controls)),
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SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0, &ssm2602_input_mux_controls),
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SND_SOC_DAPM_MICBIAS("Mic Bias", SSM2602_PWR, 1, 1),
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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_INPUT("MICIN"),
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};
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static const struct snd_soc_dapm_widget ssm2604_dapm_widgets[] = {
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SND_SOC_DAPM_MIXER("Output Mixer", SND_SOC_NOPM, 0, 0,
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ssm260x_output_mixer_controls,
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ARRAY_SIZE(ssm260x_output_mixer_controls) - 1), /* Last element is the mic */
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};
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static const struct snd_soc_dapm_route ssm260x_routes[] = {
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{"DAC", NULL, "Digital Core Power"},
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{"ADC", NULL, "Digital Core Power"},
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{"Output Mixer", "Line Bypass Switch", "Line Input"},
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{"Output Mixer", "HiFi Playback Switch", "DAC"},
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{"ROUT", NULL, "Output Mixer"},
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{"LOUT", NULL, "Output Mixer"},
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{"Line Input", NULL, "LLINEIN"},
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{"Line Input", NULL, "RLINEIN"},
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};
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static const struct snd_soc_dapm_route ssm2602_routes[] = {
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{"Output Mixer", "Mic Sidetone Switch", "Mic Bias"},
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{"RHPOUT", NULL, "Output Mixer"},
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{"LHPOUT", NULL, "Output Mixer"},
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{"Input Mux", "Line", "Line Input"},
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{"Input Mux", "Mic", "Mic Bias"},
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{"ADC", NULL, "Input Mux"},
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{"Mic Bias", NULL, "MICIN"},
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};
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static const struct snd_soc_dapm_route ssm2604_routes[] = {
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{"ADC", NULL, "Line Input"},
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};
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static const unsigned int ssm2602_rates_12288000[] = {
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8000, 16000, 32000, 48000, 96000,
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};
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static const struct snd_pcm_hw_constraint_list ssm2602_constraints_12288000 = {
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.list = ssm2602_rates_12288000,
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.count = ARRAY_SIZE(ssm2602_rates_12288000),
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};
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static const unsigned int ssm2602_rates_11289600[] = {
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8000, 44100, 88200,
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};
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static const struct snd_pcm_hw_constraint_list ssm2602_constraints_11289600 = {
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.list = ssm2602_rates_11289600,
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.count = ARRAY_SIZE(ssm2602_rates_11289600),
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};
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struct ssm2602_coeff {
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u32 mclk;
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u32 rate;
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u8 srate;
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};
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#define SSM2602_COEFF_SRATE(sr, bosr, usb) (((sr) << 2) | ((bosr) << 1) | (usb))
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/* codec mclk clock coefficients */
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static const struct ssm2602_coeff ssm2602_coeff_table[] = {
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/* 48k */
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{12288000, 48000, SSM2602_COEFF_SRATE(0x0, 0x0, 0x0)},
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{18432000, 48000, SSM2602_COEFF_SRATE(0x0, 0x1, 0x0)},
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{12000000, 48000, SSM2602_COEFF_SRATE(0x0, 0x0, 0x1)},
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/* 32k */
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{12288000, 32000, SSM2602_COEFF_SRATE(0x6, 0x0, 0x0)},
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{18432000, 32000, SSM2602_COEFF_SRATE(0x6, 0x1, 0x0)},
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{12000000, 32000, SSM2602_COEFF_SRATE(0x6, 0x0, 0x1)},
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/* 16k */
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{12288000, 16000, SSM2602_COEFF_SRATE(0x5, 0x0, 0x0)},
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{18432000, 16000, SSM2602_COEFF_SRATE(0x5, 0x1, 0x0)},
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{12000000, 16000, SSM2602_COEFF_SRATE(0xa, 0x0, 0x1)},
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/* 8k */
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{12288000, 8000, SSM2602_COEFF_SRATE(0x3, 0x0, 0x0)},
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{18432000, 8000, SSM2602_COEFF_SRATE(0x3, 0x1, 0x0)},
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{11289600, 8000, SSM2602_COEFF_SRATE(0xb, 0x0, 0x0)},
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{16934400, 8000, SSM2602_COEFF_SRATE(0xb, 0x1, 0x0)},
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{12000000, 8000, SSM2602_COEFF_SRATE(0x3, 0x0, 0x1)},
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/* 96k */
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{12288000, 96000, SSM2602_COEFF_SRATE(0x7, 0x0, 0x0)},
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{18432000, 96000, SSM2602_COEFF_SRATE(0x7, 0x1, 0x0)},
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{12000000, 96000, SSM2602_COEFF_SRATE(0x7, 0x0, 0x1)},
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/* 44.1k */
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{11289600, 44100, SSM2602_COEFF_SRATE(0x8, 0x0, 0x0)},
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{16934400, 44100, SSM2602_COEFF_SRATE(0x8, 0x1, 0x0)},
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{12000000, 44100, SSM2602_COEFF_SRATE(0x8, 0x1, 0x1)},
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/* 88.2k */
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{11289600, 88200, SSM2602_COEFF_SRATE(0xf, 0x0, 0x0)},
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{16934400, 88200, SSM2602_COEFF_SRATE(0xf, 0x1, 0x0)},
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{12000000, 88200, SSM2602_COEFF_SRATE(0xf, 0x1, 0x1)},
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};
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static inline int ssm2602_get_coeff(int mclk, int rate)
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{
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int i;
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for (i = 0; i < ARRAY_SIZE(ssm2602_coeff_table); i++) {
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if (ssm2602_coeff_table[i].rate == rate &&
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ssm2602_coeff_table[i].mclk == mclk)
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return ssm2602_coeff_table[i].srate;
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}
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return -EINVAL;
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}
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static int ssm2602_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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struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec);
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int srate = ssm2602_get_coeff(ssm2602->sysclk, params_rate(params));
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unsigned int iface;
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if (srate < 0)
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return srate;
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regmap_write(ssm2602->regmap, SSM2602_SRATE, srate);
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/* bit size */
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switch (params_width(params)) {
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case 16:
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iface = 0x0;
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break;
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case 20:
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iface = 0x4;
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break;
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case 24:
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iface = 0x8;
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break;
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case 32:
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iface = 0xc;
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break;
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default:
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return -EINVAL;
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}
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regmap_update_bits(ssm2602->regmap, SSM2602_IFACE,
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IFACE_AUDIO_DATA_LEN, iface);
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return 0;
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}
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static int ssm2602_startup(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 ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec);
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if (ssm2602->sysclk_constraints) {
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snd_pcm_hw_constraint_list(substream->runtime, 0,
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SNDRV_PCM_HW_PARAM_RATE,
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ssm2602->sysclk_constraints);
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}
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return 0;
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}
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static int ssm2602_mute(struct snd_soc_dai *dai, int mute)
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{
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struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(dai->codec);
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if (mute)
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regmap_update_bits(ssm2602->regmap, SSM2602_APDIGI,
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APDIGI_ENABLE_DAC_MUTE,
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APDIGI_ENABLE_DAC_MUTE);
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else
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regmap_update_bits(ssm2602->regmap, SSM2602_APDIGI,
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APDIGI_ENABLE_DAC_MUTE, 0);
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return 0;
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}
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static int ssm2602_set_dai_sysclk(struct snd_soc_dai *codec_dai,
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int clk_id, unsigned int freq, int dir)
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{
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struct snd_soc_codec *codec = codec_dai->codec;
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struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec);
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if (dir == SND_SOC_CLOCK_IN) {
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if (clk_id != SSM2602_SYSCLK)
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return -EINVAL;
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switch (freq) {
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case 12288000:
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case 18432000:
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ssm2602->sysclk_constraints = &ssm2602_constraints_12288000;
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break;
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case 11289600:
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case 16934400:
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ssm2602->sysclk_constraints = &ssm2602_constraints_11289600;
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break;
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case 12000000:
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ssm2602->sysclk_constraints = NULL;
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break;
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default:
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return -EINVAL;
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}
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ssm2602->sysclk = freq;
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} else {
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unsigned int mask;
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switch (clk_id) {
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case SSM2602_CLK_CLKOUT:
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mask = PWR_CLK_OUT_PDN;
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break;
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case SSM2602_CLK_XTO:
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mask = PWR_OSC_PDN;
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break;
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default:
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return -EINVAL;
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}
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if (freq == 0)
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ssm2602->clk_out_pwr |= mask;
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else
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ssm2602->clk_out_pwr &= ~mask;
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regmap_update_bits(ssm2602->regmap, SSM2602_PWR,
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PWR_CLK_OUT_PDN | PWR_OSC_PDN, ssm2602->clk_out_pwr);
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}
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return 0;
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}
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static int ssm2602_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 ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec_dai->codec);
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unsigned int iface = 0;
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/* set master/slave audio interface */
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switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
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case SND_SOC_DAIFMT_CBM_CFM:
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iface |= 0x0040;
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break;
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case SND_SOC_DAIFMT_CBS_CFS:
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break;
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default:
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return -EINVAL;
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}
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/* interface format */
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switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
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case SND_SOC_DAIFMT_I2S:
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iface |= 0x0002;
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break;
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case SND_SOC_DAIFMT_RIGHT_J:
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break;
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case SND_SOC_DAIFMT_LEFT_J:
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iface |= 0x0001;
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break;
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case SND_SOC_DAIFMT_DSP_A:
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iface |= 0x0013;
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break;
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case SND_SOC_DAIFMT_DSP_B:
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iface |= 0x0003;
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break;
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default:
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return -EINVAL;
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}
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/* clock inversion */
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switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
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case SND_SOC_DAIFMT_NB_NF:
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break;
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case SND_SOC_DAIFMT_IB_IF:
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iface |= 0x0090;
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break;
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case SND_SOC_DAIFMT_IB_NF:
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iface |= 0x0080;
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break;
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case SND_SOC_DAIFMT_NB_IF:
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iface |= 0x0010;
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break;
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default:
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return -EINVAL;
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}
|
|
|
|
/* set iface */
|
|
regmap_write(ssm2602->regmap, SSM2602_IFACE, iface);
|
|
return 0;
|
|
}
|
|
|
|
static int ssm2602_set_bias_level(struct snd_soc_codec *codec,
|
|
enum snd_soc_bias_level level)
|
|
{
|
|
struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec);
|
|
|
|
switch (level) {
|
|
case SND_SOC_BIAS_ON:
|
|
/* vref/mid on, osc and clkout on if enabled */
|
|
regmap_update_bits(ssm2602->regmap, SSM2602_PWR,
|
|
PWR_POWER_OFF | PWR_CLK_OUT_PDN | PWR_OSC_PDN,
|
|
ssm2602->clk_out_pwr);
|
|
break;
|
|
case SND_SOC_BIAS_PREPARE:
|
|
break;
|
|
case SND_SOC_BIAS_STANDBY:
|
|
/* everything off except vref/vmid, */
|
|
regmap_update_bits(ssm2602->regmap, SSM2602_PWR,
|
|
PWR_POWER_OFF | PWR_CLK_OUT_PDN | PWR_OSC_PDN,
|
|
PWR_CLK_OUT_PDN | PWR_OSC_PDN);
|
|
break;
|
|
case SND_SOC_BIAS_OFF:
|
|
/* everything off */
|
|
regmap_update_bits(ssm2602->regmap, SSM2602_PWR,
|
|
PWR_POWER_OFF, PWR_POWER_OFF);
|
|
break;
|
|
|
|
}
|
|
codec->dapm.bias_level = level;
|
|
return 0;
|
|
}
|
|
|
|
#define SSM2602_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\
|
|
SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\
|
|
SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\
|
|
SNDRV_PCM_RATE_96000)
|
|
|
|
#define SSM2602_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
|
|
SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
|
|
|
|
static const struct snd_soc_dai_ops ssm2602_dai_ops = {
|
|
.startup = ssm2602_startup,
|
|
.hw_params = ssm2602_hw_params,
|
|
.digital_mute = ssm2602_mute,
|
|
.set_sysclk = ssm2602_set_dai_sysclk,
|
|
.set_fmt = ssm2602_set_dai_fmt,
|
|
};
|
|
|
|
static struct snd_soc_dai_driver ssm2602_dai = {
|
|
.name = "ssm2602-hifi",
|
|
.playback = {
|
|
.stream_name = "Playback",
|
|
.channels_min = 2,
|
|
.channels_max = 2,
|
|
.rates = SSM2602_RATES,
|
|
.formats = SSM2602_FORMATS,},
|
|
.capture = {
|
|
.stream_name = "Capture",
|
|
.channels_min = 2,
|
|
.channels_max = 2,
|
|
.rates = SSM2602_RATES,
|
|
.formats = SSM2602_FORMATS,},
|
|
.ops = &ssm2602_dai_ops,
|
|
.symmetric_rates = 1,
|
|
.symmetric_samplebits = 1,
|
|
};
|
|
|
|
static int ssm2602_resume(struct snd_soc_codec *codec)
|
|
{
|
|
struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec);
|
|
|
|
regcache_sync(ssm2602->regmap);
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int ssm2602_codec_probe(struct snd_soc_codec *codec)
|
|
{
|
|
struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec);
|
|
struct snd_soc_dapm_context *dapm = &codec->dapm;
|
|
int ret;
|
|
|
|
regmap_update_bits(ssm2602->regmap, SSM2602_LOUT1V,
|
|
LOUT1V_LRHP_BOTH, LOUT1V_LRHP_BOTH);
|
|
regmap_update_bits(ssm2602->regmap, SSM2602_ROUT1V,
|
|
ROUT1V_RLHP_BOTH, ROUT1V_RLHP_BOTH);
|
|
|
|
ret = snd_soc_add_codec_controls(codec, ssm2602_snd_controls,
|
|
ARRAY_SIZE(ssm2602_snd_controls));
|
|
if (ret)
|
|
return ret;
|
|
|
|
ret = snd_soc_dapm_new_controls(dapm, ssm2602_dapm_widgets,
|
|
ARRAY_SIZE(ssm2602_dapm_widgets));
|
|
if (ret)
|
|
return ret;
|
|
|
|
return snd_soc_dapm_add_routes(dapm, ssm2602_routes,
|
|
ARRAY_SIZE(ssm2602_routes));
|
|
}
|
|
|
|
static int ssm2604_codec_probe(struct snd_soc_codec *codec)
|
|
{
|
|
struct snd_soc_dapm_context *dapm = &codec->dapm;
|
|
int ret;
|
|
|
|
ret = snd_soc_dapm_new_controls(dapm, ssm2604_dapm_widgets,
|
|
ARRAY_SIZE(ssm2604_dapm_widgets));
|
|
if (ret)
|
|
return ret;
|
|
|
|
return snd_soc_dapm_add_routes(dapm, ssm2604_routes,
|
|
ARRAY_SIZE(ssm2604_routes));
|
|
}
|
|
|
|
static int ssm260x_codec_probe(struct snd_soc_codec *codec)
|
|
{
|
|
struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec);
|
|
int ret;
|
|
|
|
ret = regmap_write(ssm2602->regmap, SSM2602_RESET, 0);
|
|
if (ret < 0) {
|
|
dev_err(codec->dev, "Failed to issue reset: %d\n", ret);
|
|
return ret;
|
|
}
|
|
|
|
/* set the update bits */
|
|
regmap_update_bits(ssm2602->regmap, SSM2602_LINVOL,
|
|
LINVOL_LRIN_BOTH, LINVOL_LRIN_BOTH);
|
|
regmap_update_bits(ssm2602->regmap, SSM2602_RINVOL,
|
|
RINVOL_RLIN_BOTH, RINVOL_RLIN_BOTH);
|
|
/*select Line in as default input*/
|
|
regmap_write(ssm2602->regmap, SSM2602_APANA, APANA_SELECT_DAC |
|
|
APANA_ENABLE_MIC_BOOST);
|
|
|
|
switch (ssm2602->type) {
|
|
case SSM2602:
|
|
ret = ssm2602_codec_probe(codec);
|
|
break;
|
|
case SSM2604:
|
|
ret = ssm2604_codec_probe(codec);
|
|
break;
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
static struct snd_soc_codec_driver soc_codec_dev_ssm2602 = {
|
|
.probe = ssm260x_codec_probe,
|
|
.resume = ssm2602_resume,
|
|
.set_bias_level = ssm2602_set_bias_level,
|
|
.suspend_bias_off = true,
|
|
|
|
.controls = ssm260x_snd_controls,
|
|
.num_controls = ARRAY_SIZE(ssm260x_snd_controls),
|
|
.dapm_widgets = ssm260x_dapm_widgets,
|
|
.num_dapm_widgets = ARRAY_SIZE(ssm260x_dapm_widgets),
|
|
.dapm_routes = ssm260x_routes,
|
|
.num_dapm_routes = ARRAY_SIZE(ssm260x_routes),
|
|
};
|
|
|
|
static bool ssm2602_register_volatile(struct device *dev, unsigned int reg)
|
|
{
|
|
return reg == SSM2602_RESET;
|
|
}
|
|
|
|
const struct regmap_config ssm2602_regmap_config = {
|
|
.val_bits = 9,
|
|
.reg_bits = 7,
|
|
|
|
.max_register = SSM2602_RESET,
|
|
.volatile_reg = ssm2602_register_volatile,
|
|
|
|
.cache_type = REGCACHE_RBTREE,
|
|
.reg_defaults_raw = ssm2602_reg,
|
|
.num_reg_defaults_raw = ARRAY_SIZE(ssm2602_reg),
|
|
};
|
|
EXPORT_SYMBOL_GPL(ssm2602_regmap_config);
|
|
|
|
int ssm2602_probe(struct device *dev, enum ssm2602_type type,
|
|
struct regmap *regmap)
|
|
{
|
|
struct ssm2602_priv *ssm2602;
|
|
|
|
if (IS_ERR(regmap))
|
|
return PTR_ERR(regmap);
|
|
|
|
ssm2602 = devm_kzalloc(dev, sizeof(*ssm2602), GFP_KERNEL);
|
|
if (ssm2602 == NULL)
|
|
return -ENOMEM;
|
|
|
|
dev_set_drvdata(dev, ssm2602);
|
|
ssm2602->type = SSM2602;
|
|
ssm2602->regmap = regmap;
|
|
|
|
return snd_soc_register_codec(dev, &soc_codec_dev_ssm2602,
|
|
&ssm2602_dai, 1);
|
|
}
|
|
EXPORT_SYMBOL_GPL(ssm2602_probe);
|
|
|
|
MODULE_DESCRIPTION("ASoC SSM2602/SSM2603/SSM2604 driver");
|
|
MODULE_AUTHOR("Cliff Cai");
|
|
MODULE_LICENSE("GPL");
|