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alistair23-linux/sound/soc/codecs/wm9090.c

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// SPDX-License-Identifier: GPL-2.0-only
/*
* ALSA SoC WM9090 driver
*
* Copyright 2009-12 Wolfson Microelectronics
*
* Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
*/
#include <linux/module.h>
#include <linux/errno.h>
#include <linux/device.h>
#include <linux/i2c.h>
#include <linux/delay.h>
#include <linux/regmap.h>
#include <linux/slab.h>
#include <sound/initval.h>
#include <sound/soc.h>
#include <sound/tlv.h>
#include <sound/wm9090.h>
#include "wm9090.h"
static const struct reg_default wm9090_reg_defaults[] = {
{ 1, 0x0006 }, /* R1 - Power Management (1) */
{ 2, 0x6000 }, /* R2 - Power Management (2) */
{ 3, 0x0000 }, /* R3 - Power Management (3) */
{ 6, 0x01C0 }, /* R6 - Clocking 1 */
{ 22, 0x0003 }, /* R22 - IN1 Line Control */
{ 23, 0x0003 }, /* R23 - IN2 Line Control */
{ 24, 0x0083 }, /* R24 - IN1 Line Input A Volume */
{ 25, 0x0083 }, /* R25 - IN1 Line Input B Volume */
{ 26, 0x0083 }, /* R26 - IN2 Line Input A Volume */
{ 27, 0x0083 }, /* R27 - IN2 Line Input B Volume */
{ 28, 0x002D }, /* R28 - Left Output Volume */
{ 29, 0x002D }, /* R29 - Right Output Volume */
{ 34, 0x0100 }, /* R34 - SPKMIXL Attenuation */
{ 35, 0x0010 }, /* R36 - SPKOUT Mixers */
{ 37, 0x0140 }, /* R37 - ClassD3 */
{ 38, 0x0039 }, /* R38 - Speaker Volume Left */
{ 45, 0x0000 }, /* R45 - Output Mixer1 */
{ 46, 0x0000 }, /* R46 - Output Mixer2 */
{ 47, 0x0100 }, /* R47 - Output Mixer3 */
{ 48, 0x0100 }, /* R48 - Output Mixer4 */
{ 54, 0x0000 }, /* R54 - Speaker Mixer */
{ 57, 0x000D }, /* R57 - AntiPOP2 */
{ 70, 0x0000 }, /* R70 - Write Sequencer 0 */
{ 71, 0x0000 }, /* R71 - Write Sequencer 1 */
{ 72, 0x0000 }, /* R72 - Write Sequencer 2 */
{ 73, 0x0000 }, /* R73 - Write Sequencer 3 */
{ 74, 0x0000 }, /* R74 - Write Sequencer 4 */
{ 75, 0x0000 }, /* R75 - Write Sequencer 5 */
{ 76, 0x1F25 }, /* R76 - Charge Pump 1 */
{ 85, 0x054A }, /* R85 - DC Servo 1 */
{ 87, 0x0000 }, /* R87 - DC Servo 3 */
{ 96, 0x0100 }, /* R96 - Analogue HP 0 */
{ 98, 0x8640 }, /* R98 - AGC Control 0 */
{ 99, 0xC000 }, /* R99 - AGC Control 1 */
{ 100, 0x0200 }, /* R100 - AGC Control 2 */
};
/* This struct is used to save the context */
struct wm9090_priv {
struct wm9090_platform_data pdata;
struct regmap *regmap;
};
static bool wm9090_volatile(struct device *dev, unsigned int reg)
{
switch (reg) {
case WM9090_SOFTWARE_RESET:
case WM9090_DC_SERVO_0:
case WM9090_DC_SERVO_READBACK_0:
case WM9090_DC_SERVO_READBACK_1:
case WM9090_DC_SERVO_READBACK_2:
return true;
default:
return false;
}
}
static bool wm9090_readable(struct device *dev, unsigned int reg)
{
switch (reg) {
case WM9090_SOFTWARE_RESET:
case WM9090_POWER_MANAGEMENT_1:
case WM9090_POWER_MANAGEMENT_2:
case WM9090_POWER_MANAGEMENT_3:
case WM9090_CLOCKING_1:
case WM9090_IN1_LINE_CONTROL:
case WM9090_IN2_LINE_CONTROL:
case WM9090_IN1_LINE_INPUT_A_VOLUME:
case WM9090_IN1_LINE_INPUT_B_VOLUME:
case WM9090_IN2_LINE_INPUT_A_VOLUME:
case WM9090_IN2_LINE_INPUT_B_VOLUME:
case WM9090_LEFT_OUTPUT_VOLUME:
case WM9090_RIGHT_OUTPUT_VOLUME:
case WM9090_SPKMIXL_ATTENUATION:
case WM9090_SPKOUT_MIXERS:
case WM9090_CLASSD3:
case WM9090_SPEAKER_VOLUME_LEFT:
case WM9090_OUTPUT_MIXER1:
case WM9090_OUTPUT_MIXER2:
case WM9090_OUTPUT_MIXER3:
case WM9090_OUTPUT_MIXER4:
case WM9090_SPEAKER_MIXER:
case WM9090_ANTIPOP2:
case WM9090_WRITE_SEQUENCER_0:
case WM9090_WRITE_SEQUENCER_1:
case WM9090_WRITE_SEQUENCER_2:
case WM9090_WRITE_SEQUENCER_3:
case WM9090_WRITE_SEQUENCER_4:
case WM9090_WRITE_SEQUENCER_5:
case WM9090_CHARGE_PUMP_1:
case WM9090_DC_SERVO_0:
case WM9090_DC_SERVO_1:
case WM9090_DC_SERVO_3:
case WM9090_DC_SERVO_READBACK_0:
case WM9090_DC_SERVO_READBACK_1:
case WM9090_DC_SERVO_READBACK_2:
case WM9090_ANALOGUE_HP_0:
case WM9090_AGC_CONTROL_0:
case WM9090_AGC_CONTROL_1:
case WM9090_AGC_CONTROL_2:
return true;
default:
return false;
}
}
static void wait_for_dc_servo(struct snd_soc_component *component)
{
unsigned int reg;
int count = 0;
dev_dbg(component->dev, "Waiting for DC servo...\n");
do {
count++;
msleep(1);
reg = snd_soc_component_read32(component, WM9090_DC_SERVO_READBACK_0);
dev_dbg(component->dev, "DC servo status: %x\n", reg);
} while ((reg & WM9090_DCS_CAL_COMPLETE_MASK)
!= WM9090_DCS_CAL_COMPLETE_MASK && count < 1000);
if ((reg & WM9090_DCS_CAL_COMPLETE_MASK)
!= WM9090_DCS_CAL_COMPLETE_MASK)
dev_err(component->dev, "Timed out waiting for DC Servo\n");
}
static const DECLARE_TLV_DB_RANGE(in_tlv,
0, 0, TLV_DB_SCALE_ITEM(-600, 0, 0),
1, 3, TLV_DB_SCALE_ITEM(-350, 350, 0),
4, 6, TLV_DB_SCALE_ITEM(600, 600, 0)
);
static const DECLARE_TLV_DB_RANGE(mix_tlv,
0, 2, TLV_DB_SCALE_ITEM(-1200, 300, 0),
3, 3, TLV_DB_SCALE_ITEM(0, 0, 0)
);
static const DECLARE_TLV_DB_SCALE(out_tlv, -5700, 100, 0);
static const DECLARE_TLV_DB_RANGE(spkboost_tlv,
0, 6, TLV_DB_SCALE_ITEM(0, 150, 0),
7, 7, TLV_DB_SCALE_ITEM(1200, 0, 0)
);
static const struct snd_kcontrol_new wm9090_controls[] = {
SOC_SINGLE_TLV("IN1A Volume", WM9090_IN1_LINE_INPUT_A_VOLUME, 0, 6, 0,
in_tlv),
SOC_SINGLE("IN1A Switch", WM9090_IN1_LINE_INPUT_A_VOLUME, 7, 1, 1),
SOC_SINGLE("IN1A ZC Switch", WM9090_IN1_LINE_INPUT_A_VOLUME, 6, 1, 0),
SOC_SINGLE_TLV("IN2A Volume", WM9090_IN2_LINE_INPUT_A_VOLUME, 0, 6, 0,
in_tlv),
SOC_SINGLE("IN2A Switch", WM9090_IN2_LINE_INPUT_A_VOLUME, 7, 1, 1),
SOC_SINGLE("IN2A ZC Switch", WM9090_IN2_LINE_INPUT_A_VOLUME, 6, 1, 0),
SOC_SINGLE("MIXOUTL Switch", WM9090_OUTPUT_MIXER3, 8, 1, 1),
SOC_SINGLE_TLV("MIXOUTL IN1A Volume", WM9090_OUTPUT_MIXER3, 6, 3, 1,
mix_tlv),
SOC_SINGLE_TLV("MIXOUTL IN2A Volume", WM9090_OUTPUT_MIXER3, 2, 3, 1,
mix_tlv),
SOC_SINGLE("MIXOUTR Switch", WM9090_OUTPUT_MIXER4, 8, 1, 1),
SOC_SINGLE_TLV("MIXOUTR IN1A Volume", WM9090_OUTPUT_MIXER4, 6, 3, 1,
mix_tlv),
SOC_SINGLE_TLV("MIXOUTR IN2A Volume", WM9090_OUTPUT_MIXER4, 2, 3, 1,
mix_tlv),
SOC_SINGLE("SPKMIX Switch", WM9090_SPKMIXL_ATTENUATION, 8, 1, 1),
SOC_SINGLE_TLV("SPKMIX IN1A Volume", WM9090_SPKMIXL_ATTENUATION, 6, 3, 1,
mix_tlv),
SOC_SINGLE_TLV("SPKMIX IN2A Volume", WM9090_SPKMIXL_ATTENUATION, 2, 3, 1,
mix_tlv),
SOC_DOUBLE_R_TLV("Headphone Volume", WM9090_LEFT_OUTPUT_VOLUME,
WM9090_RIGHT_OUTPUT_VOLUME, 0, 63, 0, out_tlv),
SOC_DOUBLE_R("Headphone Switch", WM9090_LEFT_OUTPUT_VOLUME,
WM9090_RIGHT_OUTPUT_VOLUME, 6, 1, 1),
SOC_DOUBLE_R("Headphone ZC Switch", WM9090_LEFT_OUTPUT_VOLUME,
WM9090_RIGHT_OUTPUT_VOLUME, 7, 1, 0),
SOC_SINGLE_TLV("Speaker Volume", WM9090_SPEAKER_VOLUME_LEFT, 0, 63, 0,
out_tlv),
SOC_SINGLE("Speaker Switch", WM9090_SPEAKER_VOLUME_LEFT, 6, 1, 1),
SOC_SINGLE("Speaker ZC Switch", WM9090_SPEAKER_VOLUME_LEFT, 7, 1, 0),
SOC_SINGLE_TLV("Speaker Boost Volume", WM9090_CLASSD3, 3, 7, 0, spkboost_tlv),
};
static const struct snd_kcontrol_new wm9090_in1_se_controls[] = {
SOC_SINGLE_TLV("IN1B Volume", WM9090_IN1_LINE_INPUT_B_VOLUME, 0, 6, 0,
in_tlv),
SOC_SINGLE("IN1B Switch", WM9090_IN1_LINE_INPUT_B_VOLUME, 7, 1, 1),
SOC_SINGLE("IN1B ZC Switch", WM9090_IN1_LINE_INPUT_B_VOLUME, 6, 1, 0),
SOC_SINGLE_TLV("SPKMIX IN1B Volume", WM9090_SPKMIXL_ATTENUATION, 4, 3, 1,
mix_tlv),
SOC_SINGLE_TLV("MIXOUTL IN1B Volume", WM9090_OUTPUT_MIXER3, 4, 3, 1,
mix_tlv),
SOC_SINGLE_TLV("MIXOUTR IN1B Volume", WM9090_OUTPUT_MIXER4, 4, 3, 1,
mix_tlv),
};
static const struct snd_kcontrol_new wm9090_in2_se_controls[] = {
SOC_SINGLE_TLV("IN2B Volume", WM9090_IN2_LINE_INPUT_B_VOLUME, 0, 6, 0,
in_tlv),
SOC_SINGLE("IN2B Switch", WM9090_IN2_LINE_INPUT_B_VOLUME, 7, 1, 1),
SOC_SINGLE("IN2B ZC Switch", WM9090_IN2_LINE_INPUT_B_VOLUME, 6, 1, 0),
SOC_SINGLE_TLV("SPKMIX IN2B Volume", WM9090_SPKMIXL_ATTENUATION, 0, 3, 1,
mix_tlv),
SOC_SINGLE_TLV("MIXOUTL IN2B Volume", WM9090_OUTPUT_MIXER3, 0, 3, 1,
mix_tlv),
SOC_SINGLE_TLV("MIXOUTR IN2B Volume", WM9090_OUTPUT_MIXER4, 0, 3, 1,
mix_tlv),
};
static int hp_ev(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm);
unsigned int reg = snd_soc_component_read32(component, WM9090_ANALOGUE_HP_0);
switch (event) {
case SND_SOC_DAPM_POST_PMU:
snd_soc_component_update_bits(component, WM9090_CHARGE_PUMP_1,
WM9090_CP_ENA, WM9090_CP_ENA);
msleep(5);
snd_soc_component_update_bits(component, WM9090_POWER_MANAGEMENT_1,
WM9090_HPOUT1L_ENA | WM9090_HPOUT1R_ENA,
WM9090_HPOUT1L_ENA | WM9090_HPOUT1R_ENA);
reg |= WM9090_HPOUT1L_DLY | WM9090_HPOUT1R_DLY;
snd_soc_component_write(component, WM9090_ANALOGUE_HP_0, reg);
/* Start the DC servo. We don't currently use the
* ability to save the state since we don't have full
* control of the analogue paths and they can change
* DC offsets; see the WM8904 driver for an example of
* doing so.
*/
snd_soc_component_write(component, WM9090_DC_SERVO_0,
WM9090_DCS_ENA_CHAN_0 |
WM9090_DCS_ENA_CHAN_1 |
WM9090_DCS_TRIG_STARTUP_1 |
WM9090_DCS_TRIG_STARTUP_0);
wait_for_dc_servo(component);
reg |= WM9090_HPOUT1R_OUTP | WM9090_HPOUT1R_RMV_SHORT |
WM9090_HPOUT1L_OUTP | WM9090_HPOUT1L_RMV_SHORT;
snd_soc_component_write(component, WM9090_ANALOGUE_HP_0, reg);
break;
case SND_SOC_DAPM_PRE_PMD:
reg &= ~(WM9090_HPOUT1L_RMV_SHORT |
WM9090_HPOUT1L_DLY |
WM9090_HPOUT1L_OUTP |
WM9090_HPOUT1R_RMV_SHORT |
WM9090_HPOUT1R_DLY |
WM9090_HPOUT1R_OUTP);
snd_soc_component_write(component, WM9090_ANALOGUE_HP_0, reg);
snd_soc_component_write(component, WM9090_DC_SERVO_0, 0);
snd_soc_component_update_bits(component, WM9090_POWER_MANAGEMENT_1,
WM9090_HPOUT1L_ENA | WM9090_HPOUT1R_ENA,
0);
snd_soc_component_update_bits(component, WM9090_CHARGE_PUMP_1,
WM9090_CP_ENA, 0);
break;
}
return 0;
}
static const struct snd_kcontrol_new spkmix[] = {
SOC_DAPM_SINGLE("IN1A Switch", WM9090_SPEAKER_MIXER, 6, 1, 0),
SOC_DAPM_SINGLE("IN1B Switch", WM9090_SPEAKER_MIXER, 4, 1, 0),
SOC_DAPM_SINGLE("IN2A Switch", WM9090_SPEAKER_MIXER, 2, 1, 0),
SOC_DAPM_SINGLE("IN2B Switch", WM9090_SPEAKER_MIXER, 0, 1, 0),
};
static const struct snd_kcontrol_new spkout[] = {
SOC_DAPM_SINGLE("Mixer Switch", WM9090_SPKOUT_MIXERS, 4, 1, 0),
};
static const struct snd_kcontrol_new mixoutl[] = {
SOC_DAPM_SINGLE("IN1A Switch", WM9090_OUTPUT_MIXER1, 6, 1, 0),
SOC_DAPM_SINGLE("IN1B Switch", WM9090_OUTPUT_MIXER1, 4, 1, 0),
SOC_DAPM_SINGLE("IN2A Switch", WM9090_OUTPUT_MIXER1, 2, 1, 0),
SOC_DAPM_SINGLE("IN2B Switch", WM9090_OUTPUT_MIXER1, 0, 1, 0),
};
static const struct snd_kcontrol_new mixoutr[] = {
SOC_DAPM_SINGLE("IN1A Switch", WM9090_OUTPUT_MIXER2, 6, 1, 0),
SOC_DAPM_SINGLE("IN1B Switch", WM9090_OUTPUT_MIXER2, 4, 1, 0),
SOC_DAPM_SINGLE("IN2A Switch", WM9090_OUTPUT_MIXER2, 2, 1, 0),
SOC_DAPM_SINGLE("IN2B Switch", WM9090_OUTPUT_MIXER2, 0, 1, 0),
};
static const struct snd_soc_dapm_widget wm9090_dapm_widgets[] = {
SND_SOC_DAPM_INPUT("IN1+"),
SND_SOC_DAPM_INPUT("IN1-"),
SND_SOC_DAPM_INPUT("IN2+"),
SND_SOC_DAPM_INPUT("IN2-"),
SND_SOC_DAPM_SUPPLY("OSC", WM9090_POWER_MANAGEMENT_1, 3, 0, NULL, 0),
SND_SOC_DAPM_PGA("IN1A PGA", WM9090_POWER_MANAGEMENT_2, 7, 0, NULL, 0),
SND_SOC_DAPM_PGA("IN1B PGA", WM9090_POWER_MANAGEMENT_2, 6, 0, NULL, 0),
SND_SOC_DAPM_PGA("IN2A PGA", WM9090_POWER_MANAGEMENT_2, 5, 0, NULL, 0),
SND_SOC_DAPM_PGA("IN2B PGA", WM9090_POWER_MANAGEMENT_2, 4, 0, NULL, 0),
SND_SOC_DAPM_MIXER("SPKMIX", WM9090_POWER_MANAGEMENT_3, 3, 0,
spkmix, ARRAY_SIZE(spkmix)),
SND_SOC_DAPM_MIXER("MIXOUTL", WM9090_POWER_MANAGEMENT_3, 5, 0,
mixoutl, ARRAY_SIZE(mixoutl)),
SND_SOC_DAPM_MIXER("MIXOUTR", WM9090_POWER_MANAGEMENT_3, 4, 0,
mixoutr, ARRAY_SIZE(mixoutr)),
SND_SOC_DAPM_PGA_E("HP PGA", SND_SOC_NOPM, 0, 0, NULL, 0,
hp_ev, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
SND_SOC_DAPM_PGA("SPKPGA", WM9090_POWER_MANAGEMENT_3, 8, 0, NULL, 0),
SND_SOC_DAPM_MIXER("SPKOUT", WM9090_POWER_MANAGEMENT_1, 12, 0,
spkout, ARRAY_SIZE(spkout)),
SND_SOC_DAPM_OUTPUT("HPR"),
SND_SOC_DAPM_OUTPUT("HPL"),
SND_SOC_DAPM_OUTPUT("Speaker"),
};
static const struct snd_soc_dapm_route audio_map[] = {
{ "IN1A PGA", NULL, "IN1+" },
{ "IN2A PGA", NULL, "IN2+" },
{ "SPKMIX", "IN1A Switch", "IN1A PGA" },
{ "SPKMIX", "IN2A Switch", "IN2A PGA" },
{ "MIXOUTL", "IN1A Switch", "IN1A PGA" },
{ "MIXOUTL", "IN2A Switch", "IN2A PGA" },
{ "MIXOUTR", "IN1A Switch", "IN1A PGA" },
{ "MIXOUTR", "IN2A Switch", "IN2A PGA" },
{ "HP PGA", NULL, "OSC" },
{ "HP PGA", NULL, "MIXOUTL" },
{ "HP PGA", NULL, "MIXOUTR" },
{ "HPL", NULL, "HP PGA" },
{ "HPR", NULL, "HP PGA" },
{ "SPKPGA", NULL, "OSC" },
{ "SPKPGA", NULL, "SPKMIX" },
{ "SPKOUT", "Mixer Switch", "SPKPGA" },
{ "Speaker", NULL, "SPKOUT" },
};
static const struct snd_soc_dapm_route audio_map_in1_se[] = {
{ "IN1B PGA", NULL, "IN1-" },
{ "SPKMIX", "IN1B Switch", "IN1B PGA" },
{ "MIXOUTL", "IN1B Switch", "IN1B PGA" },
{ "MIXOUTR", "IN1B Switch", "IN1B PGA" },
};
static const struct snd_soc_dapm_route audio_map_in1_diff[] = {
{ "IN1A PGA", NULL, "IN1-" },
};
static const struct snd_soc_dapm_route audio_map_in2_se[] = {
{ "IN2B PGA", NULL, "IN2-" },
{ "SPKMIX", "IN2B Switch", "IN2B PGA" },
{ "MIXOUTL", "IN2B Switch", "IN2B PGA" },
{ "MIXOUTR", "IN2B Switch", "IN2B PGA" },
};
static const struct snd_soc_dapm_route audio_map_in2_diff[] = {
{ "IN2A PGA", NULL, "IN2-" },
};
static int wm9090_add_controls(struct snd_soc_component *component)
{
struct wm9090_priv *wm9090 = snd_soc_component_get_drvdata(component);
struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component);
int i;
ASoC: Decouple DAPM from CODECs Decoupling Dynamic Audio Power Management (DAPM) from codec devices is required when developing ASoC further. Such as for other ASoC components to have DAPM widgets or when extending DAPM to handle cross-device paths. This patch decouples DAPM related variables from struct snd_soc_codec and moves them to new struct snd_soc_dapm_context that is used to encapsulate DAPM context of a device. ASoC core and API of DAPM functions are modified to use DAPM context instead of codec. This patch does not change current functionality and a large part of changes come because of structure and internal API changes. Core implementation is from Liam Girdwood <lrg@slimlogic.co.uk> with some minor core changes, codecs and machine driver conversions from Jarkko Nikula <jhnikula@gmail.com>. Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Cc: Nicolas Ferre <nicolas.ferre@atmel.com> Cc: Manuel Lauss <manuel.lauss@googlemail.com> Cc: Mike Frysinger <vapier.adi@gmail.com> Cc: Cliff Cai <cliff.cai@analog.com> Cc: Kevin Hilman <khilman@deeprootsystems.com> Cc: Ryan Mallon <ryan@bluewatersys.com> Cc: Timur Tabi <timur@freescale.com> Cc: Sascha Hauer <s.hauer@pengutronix.de> Cc: Lars-Peter Clausen <lars@metafoo.de> Cc: Arnaud Patard (Rtp) <arnaud.patard@rtp-net.org> Cc: Wan ZongShun <mcuos.com@gmail.com> Cc: Eric Miao <eric.y.miao@gmail.com> Cc: Jassi Brar <jassi.brar@samsung.com> Cc: Daniel Gloeckner <dg@emlix.com> Cc: Kuninori Morimoto <morimoto.kuninori@renesas.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-11-05 07:53:46 -06:00
snd_soc_dapm_new_controls(dapm, wm9090_dapm_widgets,
ARRAY_SIZE(wm9090_dapm_widgets));
ASoC: Decouple DAPM from CODECs Decoupling Dynamic Audio Power Management (DAPM) from codec devices is required when developing ASoC further. Such as for other ASoC components to have DAPM widgets or when extending DAPM to handle cross-device paths. This patch decouples DAPM related variables from struct snd_soc_codec and moves them to new struct snd_soc_dapm_context that is used to encapsulate DAPM context of a device. ASoC core and API of DAPM functions are modified to use DAPM context instead of codec. This patch does not change current functionality and a large part of changes come because of structure and internal API changes. Core implementation is from Liam Girdwood <lrg@slimlogic.co.uk> with some minor core changes, codecs and machine driver conversions from Jarkko Nikula <jhnikula@gmail.com>. Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Cc: Nicolas Ferre <nicolas.ferre@atmel.com> Cc: Manuel Lauss <manuel.lauss@googlemail.com> Cc: Mike Frysinger <vapier.adi@gmail.com> Cc: Cliff Cai <cliff.cai@analog.com> Cc: Kevin Hilman <khilman@deeprootsystems.com> Cc: Ryan Mallon <ryan@bluewatersys.com> Cc: Timur Tabi <timur@freescale.com> Cc: Sascha Hauer <s.hauer@pengutronix.de> Cc: Lars-Peter Clausen <lars@metafoo.de> Cc: Arnaud Patard (Rtp) <arnaud.patard@rtp-net.org> Cc: Wan ZongShun <mcuos.com@gmail.com> Cc: Eric Miao <eric.y.miao@gmail.com> Cc: Jassi Brar <jassi.brar@samsung.com> Cc: Daniel Gloeckner <dg@emlix.com> Cc: Kuninori Morimoto <morimoto.kuninori@renesas.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-11-05 07:53:46 -06:00
snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
snd_soc_add_component_controls(component, wm9090_controls,
ARRAY_SIZE(wm9090_controls));
if (wm9090->pdata.lin1_diff) {
ASoC: Decouple DAPM from CODECs Decoupling Dynamic Audio Power Management (DAPM) from codec devices is required when developing ASoC further. Such as for other ASoC components to have DAPM widgets or when extending DAPM to handle cross-device paths. This patch decouples DAPM related variables from struct snd_soc_codec and moves them to new struct snd_soc_dapm_context that is used to encapsulate DAPM context of a device. ASoC core and API of DAPM functions are modified to use DAPM context instead of codec. This patch does not change current functionality and a large part of changes come because of structure and internal API changes. Core implementation is from Liam Girdwood <lrg@slimlogic.co.uk> with some minor core changes, codecs and machine driver conversions from Jarkko Nikula <jhnikula@gmail.com>. Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Cc: Nicolas Ferre <nicolas.ferre@atmel.com> Cc: Manuel Lauss <manuel.lauss@googlemail.com> Cc: Mike Frysinger <vapier.adi@gmail.com> Cc: Cliff Cai <cliff.cai@analog.com> Cc: Kevin Hilman <khilman@deeprootsystems.com> Cc: Ryan Mallon <ryan@bluewatersys.com> Cc: Timur Tabi <timur@freescale.com> Cc: Sascha Hauer <s.hauer@pengutronix.de> Cc: Lars-Peter Clausen <lars@metafoo.de> Cc: Arnaud Patard (Rtp) <arnaud.patard@rtp-net.org> Cc: Wan ZongShun <mcuos.com@gmail.com> Cc: Eric Miao <eric.y.miao@gmail.com> Cc: Jassi Brar <jassi.brar@samsung.com> Cc: Daniel Gloeckner <dg@emlix.com> Cc: Kuninori Morimoto <morimoto.kuninori@renesas.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-11-05 07:53:46 -06:00
snd_soc_dapm_add_routes(dapm, audio_map_in1_diff,
ARRAY_SIZE(audio_map_in1_diff));
} else {
ASoC: Decouple DAPM from CODECs Decoupling Dynamic Audio Power Management (DAPM) from codec devices is required when developing ASoC further. Such as for other ASoC components to have DAPM widgets or when extending DAPM to handle cross-device paths. This patch decouples DAPM related variables from struct snd_soc_codec and moves them to new struct snd_soc_dapm_context that is used to encapsulate DAPM context of a device. ASoC core and API of DAPM functions are modified to use DAPM context instead of codec. This patch does not change current functionality and a large part of changes come because of structure and internal API changes. Core implementation is from Liam Girdwood <lrg@slimlogic.co.uk> with some minor core changes, codecs and machine driver conversions from Jarkko Nikula <jhnikula@gmail.com>. Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Cc: Nicolas Ferre <nicolas.ferre@atmel.com> Cc: Manuel Lauss <manuel.lauss@googlemail.com> Cc: Mike Frysinger <vapier.adi@gmail.com> Cc: Cliff Cai <cliff.cai@analog.com> Cc: Kevin Hilman <khilman@deeprootsystems.com> Cc: Ryan Mallon <ryan@bluewatersys.com> Cc: Timur Tabi <timur@freescale.com> Cc: Sascha Hauer <s.hauer@pengutronix.de> Cc: Lars-Peter Clausen <lars@metafoo.de> Cc: Arnaud Patard (Rtp) <arnaud.patard@rtp-net.org> Cc: Wan ZongShun <mcuos.com@gmail.com> Cc: Eric Miao <eric.y.miao@gmail.com> Cc: Jassi Brar <jassi.brar@samsung.com> Cc: Daniel Gloeckner <dg@emlix.com> Cc: Kuninori Morimoto <morimoto.kuninori@renesas.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-11-05 07:53:46 -06:00
snd_soc_dapm_add_routes(dapm, audio_map_in1_se,
ARRAY_SIZE(audio_map_in1_se));
snd_soc_add_component_controls(component, wm9090_in1_se_controls,
ARRAY_SIZE(wm9090_in1_se_controls));
}
if (wm9090->pdata.lin2_diff) {
ASoC: Decouple DAPM from CODECs Decoupling Dynamic Audio Power Management (DAPM) from codec devices is required when developing ASoC further. Such as for other ASoC components to have DAPM widgets or when extending DAPM to handle cross-device paths. This patch decouples DAPM related variables from struct snd_soc_codec and moves them to new struct snd_soc_dapm_context that is used to encapsulate DAPM context of a device. ASoC core and API of DAPM functions are modified to use DAPM context instead of codec. This patch does not change current functionality and a large part of changes come because of structure and internal API changes. Core implementation is from Liam Girdwood <lrg@slimlogic.co.uk> with some minor core changes, codecs and machine driver conversions from Jarkko Nikula <jhnikula@gmail.com>. Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Cc: Nicolas Ferre <nicolas.ferre@atmel.com> Cc: Manuel Lauss <manuel.lauss@googlemail.com> Cc: Mike Frysinger <vapier.adi@gmail.com> Cc: Cliff Cai <cliff.cai@analog.com> Cc: Kevin Hilman <khilman@deeprootsystems.com> Cc: Ryan Mallon <ryan@bluewatersys.com> Cc: Timur Tabi <timur@freescale.com> Cc: Sascha Hauer <s.hauer@pengutronix.de> Cc: Lars-Peter Clausen <lars@metafoo.de> Cc: Arnaud Patard (Rtp) <arnaud.patard@rtp-net.org> Cc: Wan ZongShun <mcuos.com@gmail.com> Cc: Eric Miao <eric.y.miao@gmail.com> Cc: Jassi Brar <jassi.brar@samsung.com> Cc: Daniel Gloeckner <dg@emlix.com> Cc: Kuninori Morimoto <morimoto.kuninori@renesas.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-11-05 07:53:46 -06:00
snd_soc_dapm_add_routes(dapm, audio_map_in2_diff,
ARRAY_SIZE(audio_map_in2_diff));
} else {
ASoC: Decouple DAPM from CODECs Decoupling Dynamic Audio Power Management (DAPM) from codec devices is required when developing ASoC further. Such as for other ASoC components to have DAPM widgets or when extending DAPM to handle cross-device paths. This patch decouples DAPM related variables from struct snd_soc_codec and moves them to new struct snd_soc_dapm_context that is used to encapsulate DAPM context of a device. ASoC core and API of DAPM functions are modified to use DAPM context instead of codec. This patch does not change current functionality and a large part of changes come because of structure and internal API changes. Core implementation is from Liam Girdwood <lrg@slimlogic.co.uk> with some minor core changes, codecs and machine driver conversions from Jarkko Nikula <jhnikula@gmail.com>. Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Cc: Nicolas Ferre <nicolas.ferre@atmel.com> Cc: Manuel Lauss <manuel.lauss@googlemail.com> Cc: Mike Frysinger <vapier.adi@gmail.com> Cc: Cliff Cai <cliff.cai@analog.com> Cc: Kevin Hilman <khilman@deeprootsystems.com> Cc: Ryan Mallon <ryan@bluewatersys.com> Cc: Timur Tabi <timur@freescale.com> Cc: Sascha Hauer <s.hauer@pengutronix.de> Cc: Lars-Peter Clausen <lars@metafoo.de> Cc: Arnaud Patard (Rtp) <arnaud.patard@rtp-net.org> Cc: Wan ZongShun <mcuos.com@gmail.com> Cc: Eric Miao <eric.y.miao@gmail.com> Cc: Jassi Brar <jassi.brar@samsung.com> Cc: Daniel Gloeckner <dg@emlix.com> Cc: Kuninori Morimoto <morimoto.kuninori@renesas.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-11-05 07:53:46 -06:00
snd_soc_dapm_add_routes(dapm, audio_map_in2_se,
ARRAY_SIZE(audio_map_in2_se));
snd_soc_add_component_controls(component, wm9090_in2_se_controls,
ARRAY_SIZE(wm9090_in2_se_controls));
}
if (wm9090->pdata.agc_ena) {
for (i = 0; i < ARRAY_SIZE(wm9090->pdata.agc); i++)
snd_soc_component_write(component, WM9090_AGC_CONTROL_0 + i,
wm9090->pdata.agc[i]);
snd_soc_component_update_bits(component, WM9090_POWER_MANAGEMENT_3,
WM9090_AGC_ENA, WM9090_AGC_ENA);
} else {
snd_soc_component_update_bits(component, WM9090_POWER_MANAGEMENT_3,
WM9090_AGC_ENA, 0);
}
return 0;
}
/*
* The machine driver should call this from their set_bias_level; if there
* isn't one then this can just be set as the set_bias_level function.
*/
static int wm9090_set_bias_level(struct snd_soc_component *component,
enum snd_soc_bias_level level)
{
struct wm9090_priv *wm9090 = snd_soc_component_get_drvdata(component);
switch (level) {
case SND_SOC_BIAS_ON:
break;
case SND_SOC_BIAS_PREPARE:
snd_soc_component_update_bits(component, WM9090_ANTIPOP2, WM9090_VMID_ENA,
WM9090_VMID_ENA);
snd_soc_component_update_bits(component, WM9090_POWER_MANAGEMENT_1,
WM9090_BIAS_ENA |
WM9090_VMID_RES_MASK,
WM9090_BIAS_ENA |
1 << WM9090_VMID_RES_SHIFT);
msleep(1); /* Probably an overestimate */
break;
case SND_SOC_BIAS_STANDBY:
if (snd_soc_component_get_bias_level(component) == SND_SOC_BIAS_OFF) {
/* Restore the register cache */
regcache_sync(wm9090->regmap);
}
/* We keep VMID off during standby since the combination of
* ground referenced outputs and class D speaker mean that
* latency is not an issue.
*/
snd_soc_component_update_bits(component, WM9090_POWER_MANAGEMENT_1,
WM9090_BIAS_ENA | WM9090_VMID_RES_MASK, 0);
snd_soc_component_update_bits(component, WM9090_ANTIPOP2,
WM9090_VMID_ENA, 0);
break;
case SND_SOC_BIAS_OFF:
break;
}
return 0;
}
static int wm9090_probe(struct snd_soc_component *component)
{
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 14:15:21 -06:00
/* Configure some defaults; they will be written out when we
* bring the bias up.
*/
snd_soc_component_update_bits(component, WM9090_IN1_LINE_INPUT_A_VOLUME,
WM9090_IN1_VU | WM9090_IN1A_ZC,
WM9090_IN1_VU | WM9090_IN1A_ZC);
snd_soc_component_update_bits(component, WM9090_IN1_LINE_INPUT_B_VOLUME,
WM9090_IN1_VU | WM9090_IN1B_ZC,
WM9090_IN1_VU | WM9090_IN1B_ZC);
snd_soc_component_update_bits(component, WM9090_IN2_LINE_INPUT_A_VOLUME,
WM9090_IN2_VU | WM9090_IN2A_ZC,
WM9090_IN2_VU | WM9090_IN2A_ZC);
snd_soc_component_update_bits(component, WM9090_IN2_LINE_INPUT_B_VOLUME,
WM9090_IN2_VU | WM9090_IN2B_ZC,
WM9090_IN2_VU | WM9090_IN2B_ZC);
snd_soc_component_update_bits(component, WM9090_SPEAKER_VOLUME_LEFT,
WM9090_SPKOUT_VU | WM9090_SPKOUTL_ZC,
WM9090_SPKOUT_VU | WM9090_SPKOUTL_ZC);
snd_soc_component_update_bits(component, WM9090_LEFT_OUTPUT_VOLUME,
WM9090_HPOUT1_VU | WM9090_HPOUT1L_ZC,
WM9090_HPOUT1_VU | WM9090_HPOUT1L_ZC);
snd_soc_component_update_bits(component, WM9090_RIGHT_OUTPUT_VOLUME,
WM9090_HPOUT1_VU | WM9090_HPOUT1R_ZC,
WM9090_HPOUT1_VU | WM9090_HPOUT1R_ZC);
snd_soc_component_update_bits(component, WM9090_CLOCKING_1,
WM9090_TOCLK_ENA, WM9090_TOCLK_ENA);
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 14:15:21 -06:00
wm9090_add_controls(component);
return 0;
}
static const struct snd_soc_component_driver soc_component_dev_wm9090 = {
.probe = wm9090_probe,
.set_bias_level = wm9090_set_bias_level,
.suspend_bias_off = 1,
.idle_bias_on = 1,
.use_pmdown_time = 1,
.endianness = 1,
.non_legacy_dai_naming = 1,
};
static const struct regmap_config wm9090_regmap = {
.reg_bits = 8,
.val_bits = 16,
.max_register = WM9090_MAX_REGISTER,
.volatile_reg = wm9090_volatile,
.readable_reg = wm9090_readable,
.cache_type = REGCACHE_RBTREE,
.reg_defaults = wm9090_reg_defaults,
.num_reg_defaults = ARRAY_SIZE(wm9090_reg_defaults),
};
static int wm9090_i2c_probe(struct i2c_client *i2c,
const struct i2c_device_id *id)
{
struct wm9090_priv *wm9090;
unsigned int reg;
int ret;
wm9090 = devm_kzalloc(&i2c->dev, sizeof(*wm9090), GFP_KERNEL);
if (!wm9090)
return -ENOMEM;
wm9090->regmap = devm_regmap_init_i2c(i2c, &wm9090_regmap);
if (IS_ERR(wm9090->regmap)) {
ret = PTR_ERR(wm9090->regmap);
dev_err(&i2c->dev, "Failed to allocate regmap: %d\n", ret);
return ret;
}
ret = regmap_read(wm9090->regmap, WM9090_SOFTWARE_RESET, &reg);
if (ret < 0)
return ret;
if (reg != 0x9093) {
dev_err(&i2c->dev, "Device is not a WM9090, ID=%x\n", reg);
return -ENODEV;
}
ret = regmap_write(wm9090->regmap, WM9090_SOFTWARE_RESET, 0);
if (ret < 0)
return ret;
if (i2c->dev.platform_data)
memcpy(&wm9090->pdata, i2c->dev.platform_data,
sizeof(wm9090->pdata));
i2c_set_clientdata(i2c, wm9090);
ret = devm_snd_soc_register_component(&i2c->dev,
&soc_component_dev_wm9090, NULL, 0);
if (ret != 0) {
dev_err(&i2c->dev, "Failed to register CODEC: %d\n", ret);
return ret;
}
return 0;
}
static const struct i2c_device_id wm9090_id[] = {
{ "wm9090", 0 },
{ "wm9093", 0 },
{ }
};
MODULE_DEVICE_TABLE(i2c, wm9090_id);
static struct i2c_driver wm9090_i2c_driver = {
.driver = {
.name = "wm9090",
},
.probe = wm9090_i2c_probe,
.id_table = wm9090_id,
};
module_i2c_driver(wm9090_i2c_driver);
MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
MODULE_DESCRIPTION("WM9090 ASoC driver");
MODULE_LICENSE("GPL");