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linux/sound/soc/codecs/ak4613.c
Charles Keepax 410e73a533
ASoC: ak*: Remove now redundant non_legacy_dai_naming flag 
The ASoC core has now been changed to default to the non-legacy DAI
naming, as such drivers using the new scheme no longer need to specify
the non_legacy_dai_naming flag.

Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20220623125250.2355471-48-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
2022-06-27 13:16:57 +01:00

937 lines
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// SPDX-License-Identifier: GPL-2.0
//
// ak4613.c -- Asahi Kasei ALSA Soc Audio driver
//
// Copyright (C) 2015 Renesas Electronics Corporation
// Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
//
// Based on ak4642.c by Kuninori Morimoto
// Based on wm8731.c by Richard Purdie
// Based on ak4535.c by Richard Purdie
// Based on wm8753.c by Liam Girdwood
/*
* +-------+
* |AK4613 |
* SDTO1 <-| |
* | |
* SDTI1 ->| |
* SDTI2 ->| |
* SDTI3 ->| |
* +-------+
*
* +---+
* clk | |___________________________________________...
*
* [TDM512]
* SDTO1 [L1][R1][L2][R2]
* SDTI1 [L1][R1][L2][R2][L3][R3][L4][R4][L5][R5][L6][R6]
*
* [TDM256]
* SDTO1 [L1][R1][L2][R2]
* SDTI1 [L1][R1][L2][R2][L3][R3][L4][R4]
* SDTI2 [L5][R5][L6][R6]
*
* [TDM128]
* SDTO1 [L1][R1][L2][R2]
* SDTI1 [L1][R1][L2][R2]
* SDTI2 [L3][R3][L4][R4]
* SDTI3 [L5][R5][L6][R6]
*
* [STEREO]
* Playback 2ch : SDTI1
* Capture 2ch : SDTO1
*
* [TDM512]
* Playback 12ch : SDTI1
* Capture 4ch : SDTO1
*
* [TDM256]
* Playback 12ch : SDTI1 + SDTI2
* Playback 8ch : SDTI1
* Capture 4ch : SDTO1
*
* [TDM128]
* Playback 12ch : SDTI1 + SDTI2 + SDTI3
* Playback 8ch : SDTI1 + SDTI2
* Playback 4ch : SDTI1
* Capture 4ch : SDTO1
*
*
* !!! NOTE !!!
*
* Renesas is the only user of ak4613 on upstream so far,
* but the chip connection is like below.
* Thus, Renesas can't test all connection case.
* Tested TDM is very limited.
*
* +-----+ +-----------+
* | SoC | | AK4613 |
* | |<-----|SDTO1 IN1|<-- Mic
* | | | IN2|
* | | | |
* | |----->|SDTI1 OUT1|--> Headphone
* +-----+ |SDTI2 OUT2|
* |SDTI3 OUT3|
* | OUT4|
* | OUT5|
* | OUT6|
* +-----------+
*
* Renesas SoC can handle [2, 6,8] channels.
* Ak4613 can handle [2,4, 8,12] channels.
*
* Because of above HW connection and available channels number,
* Renesas could test are ...
*
* [STEREO] Playback 2ch : SDTI1
* Capture 2ch : SDTO1
* [TDM256] Playback 8ch : SDTI1 (*)
*
* (*) it used 8ch data between SoC <-> AK4613 on TDM256 mode,
* but could confirm is only first 2ch because only 1
* Headphone is connected.
*
* see
* AK4613_ENABLE_TDM_TEST
*/
#include <linux/clk.h>
#include <linux/delay.h>
#include <linux/i2c.h>
#include <linux/slab.h>
#include <linux/of_device.h>
#include <linux/of_graph.h>
#include <linux/module.h>
#include <linux/regmap.h>
#include <sound/soc.h>
#include <sound/pcm_params.h>
#include <sound/tlv.h>
#define PW_MGMT1 0x00 /* Power Management 1 */
#define PW_MGMT2 0x01 /* Power Management 2 */
#define PW_MGMT3 0x02 /* Power Management 3 */
#define CTRL1 0x03 /* Control 1 */
#define CTRL2 0x04 /* Control 2 */
#define DEMP1 0x05 /* De-emphasis1 */
#define DEMP2 0x06 /* De-emphasis2 */
#define OFD 0x07 /* Overflow Detect */
#define ZRD 0x08 /* Zero Detect */
#define ICTRL 0x09 /* Input Control */
#define OCTRL 0x0a /* Output Control */
#define LOUT1 0x0b /* LOUT1 Volume Control */
#define ROUT1 0x0c /* ROUT1 Volume Control */
#define LOUT2 0x0d /* LOUT2 Volume Control */
#define ROUT2 0x0e /* ROUT2 Volume Control */
#define LOUT3 0x0f /* LOUT3 Volume Control */
#define ROUT3 0x10 /* ROUT3 Volume Control */
#define LOUT4 0x11 /* LOUT4 Volume Control */
#define ROUT4 0x12 /* ROUT4 Volume Control */
#define LOUT5 0x13 /* LOUT5 Volume Control */
#define ROUT5 0x14 /* ROUT5 Volume Control */
#define LOUT6 0x15 /* LOUT6 Volume Control */
#define ROUT6 0x16 /* ROUT6 Volume Control */
/* PW_MGMT1 */
#define RSTN BIT(0)
#define PMDAC BIT(1)
#define PMADC BIT(2)
#define PMVR BIT(3)
/* PW_MGMT2 */
#define PMAD_ALL 0x7
/* PW_MGMT3 */
#define PMDA_ALL 0x3f
/* CTRL1 */
#define DIF0 BIT(3)
#define DIF1 BIT(4)
#define DIF2 BIT(5)
#define TDM0 BIT(6)
#define TDM1 BIT(7)
#define NO_FMT (0xff)
#define FMT_MASK (0xf8)
/* CTRL2 */
#define DFS_MASK (3 << 2)
#define DFS_NORMAL_SPEED (0 << 2)
#define DFS_DOUBLE_SPEED (1 << 2)
#define DFS_QUAD_SPEED (2 << 2)
/* ICTRL */
#define ICTRL_MASK (0x3)
/* OCTRL */
#define OCTRL_MASK (0x3F)
/*
* configs
*
* 0x000000BA
*
* B : AK4613_CONFIG_SDTI_x
* A : AK4613_CONFIG_MODE_x
*/
#define AK4613_CONFIG_SET(priv, x) priv->configs |= AK4613_CONFIG_##x
#define AK4613_CONFIG_GET(priv, x) (priv->configs & AK4613_CONFIG_##x##_MASK)
/*
* AK4613_CONFIG_SDTI_x
*
* It indicates how many SDTIx is connected.
*/
#define AK4613_CONFIG_SDTI_MASK (0xF << 4)
#define AK4613_CONFIG_SDTI(x) (((x) & 0xF) << 4)
#define AK4613_CONFIG_SDTI_set(priv, x) AK4613_CONFIG_SET(priv, SDTI(x))
#define AK4613_CONFIG_SDTI_get(priv) ((AK4613_CONFIG_GET(priv, SDTI) >> 4) & 0xF)
/*
* AK4613_CONFIG_MODE_x
*
* Same as Ctrl1 :: TDM1/TDM0
* No shift is requested
* see
* AK4613_CTRL1_TO_MODE()
* Table 11/12/13/14
*/
#define AK4613_CONFIG_MODE_MASK (0xF)
#define AK4613_CONFIG_MODE_STEREO (0x0)
#define AK4613_CONFIG_MODE_TDM512 (0x1)
#define AK4613_CONFIG_MODE_TDM256 (0x2)
#define AK4613_CONFIG_MODE_TDM128 (0x3)
/*
* !!!! FIXME !!!!
*
* Because of testable HW limitation, TDM256 8ch TDM was only tested.
* This driver uses AK4613_ENABLE_TDM_TEST instead of new DT property so far.
* Don't hesitate to update driver, you don't need to care compatible
* with Renesas.
*
* #define AK4613_ENABLE_TDM_TEST
*/
struct ak4613_interface {
unsigned int width;
unsigned int fmt;
u8 dif;
};
struct ak4613_priv {
struct mutex lock;
struct snd_pcm_hw_constraint_list constraint_rates;
struct snd_pcm_hw_constraint_list constraint_channels;
struct work_struct dummy_write_work;
struct snd_soc_component *component;
unsigned int rate;
unsigned int sysclk;
unsigned int fmt;
unsigned int configs;
int cnt;
u8 ctrl1;
u8 oc;
u8 ic;
};
/*
* Playback Volume
*
* max : 0x00 : 0 dB
* ( 0.5 dB step )
* min : 0xFE : -127.0 dB
* mute: 0xFF
*/
static const DECLARE_TLV_DB_SCALE(out_tlv, -12750, 50, 1);
static const struct snd_kcontrol_new ak4613_snd_controls[] = {
SOC_DOUBLE_R_TLV("Digital Playback Volume1", LOUT1, ROUT1,
0, 0xFF, 1, out_tlv),
SOC_DOUBLE_R_TLV("Digital Playback Volume2", LOUT2, ROUT2,
0, 0xFF, 1, out_tlv),
SOC_DOUBLE_R_TLV("Digital Playback Volume3", LOUT3, ROUT3,
0, 0xFF, 1, out_tlv),
SOC_DOUBLE_R_TLV("Digital Playback Volume4", LOUT4, ROUT4,
0, 0xFF, 1, out_tlv),
SOC_DOUBLE_R_TLV("Digital Playback Volume5", LOUT5, ROUT5,
0, 0xFF, 1, out_tlv),
SOC_DOUBLE_R_TLV("Digital Playback Volume6", LOUT6, ROUT6,
0, 0xFF, 1, out_tlv),
};
static const struct reg_default ak4613_reg[] = {
{ 0x0, 0x0f }, { 0x1, 0x07 }, { 0x2, 0x3f }, { 0x3, 0x20 },
{ 0x4, 0x20 }, { 0x5, 0x55 }, { 0x6, 0x05 }, { 0x7, 0x07 },
{ 0x8, 0x0f }, { 0x9, 0x07 }, { 0xa, 0x3f }, { 0xb, 0x00 },
{ 0xc, 0x00 }, { 0xd, 0x00 }, { 0xe, 0x00 }, { 0xf, 0x00 },
{ 0x10, 0x00 }, { 0x11, 0x00 }, { 0x12, 0x00 }, { 0x13, 0x00 },
{ 0x14, 0x00 }, { 0x15, 0x00 }, { 0x16, 0x00 },
};
/*
* CTRL1 register
* see
* Table 11/12/13/14
*/
#define AUDIO_IFACE(_dif, _width, _fmt) \
{ \
.dif = _dif, \
.width = _width, \
.fmt = SND_SOC_DAIFMT_##_fmt,\
}
static const struct ak4613_interface ak4613_iface[] = {
/* It doesn't support asymmetric format */
AUDIO_IFACE(0x03, 24, LEFT_J),
AUDIO_IFACE(0x04, 24, I2S),
};
#define AK4613_CTRL1_TO_MODE(priv) ((priv)->ctrl1 >> 6) /* AK4613_CONFIG_MODE_x */
static const struct regmap_config ak4613_regmap_cfg = {
.reg_bits = 8,
.val_bits = 8,
.max_register = 0x16,
.reg_defaults = ak4613_reg,
.num_reg_defaults = ARRAY_SIZE(ak4613_reg),
.cache_type = REGCACHE_RBTREE,
};
static const struct of_device_id ak4613_of_match[] = {
{ .compatible = "asahi-kasei,ak4613", .data = &ak4613_regmap_cfg },
{},
};
MODULE_DEVICE_TABLE(of, ak4613_of_match);
static const struct i2c_device_id ak4613_i2c_id[] = {
{ "ak4613", (kernel_ulong_t)&ak4613_regmap_cfg },
{ }
};
MODULE_DEVICE_TABLE(i2c, ak4613_i2c_id);
static const struct snd_soc_dapm_widget ak4613_dapm_widgets[] = {
/* Outputs */
SND_SOC_DAPM_OUTPUT("LOUT1"),
SND_SOC_DAPM_OUTPUT("LOUT2"),
SND_SOC_DAPM_OUTPUT("LOUT3"),
SND_SOC_DAPM_OUTPUT("LOUT4"),
SND_SOC_DAPM_OUTPUT("LOUT5"),
SND_SOC_DAPM_OUTPUT("LOUT6"),
SND_SOC_DAPM_OUTPUT("ROUT1"),
SND_SOC_DAPM_OUTPUT("ROUT2"),
SND_SOC_DAPM_OUTPUT("ROUT3"),
SND_SOC_DAPM_OUTPUT("ROUT4"),
SND_SOC_DAPM_OUTPUT("ROUT5"),
SND_SOC_DAPM_OUTPUT("ROUT6"),
/* Inputs */
SND_SOC_DAPM_INPUT("LIN1"),
SND_SOC_DAPM_INPUT("LIN2"),
SND_SOC_DAPM_INPUT("RIN1"),
SND_SOC_DAPM_INPUT("RIN2"),
/* DAC */
SND_SOC_DAPM_DAC("DAC1", NULL, PW_MGMT3, 0, 0),
SND_SOC_DAPM_DAC("DAC2", NULL, PW_MGMT3, 1, 0),
SND_SOC_DAPM_DAC("DAC3", NULL, PW_MGMT3, 2, 0),
SND_SOC_DAPM_DAC("DAC4", NULL, PW_MGMT3, 3, 0),
SND_SOC_DAPM_DAC("DAC5", NULL, PW_MGMT3, 4, 0),
SND_SOC_DAPM_DAC("DAC6", NULL, PW_MGMT3, 5, 0),
/* ADC */
SND_SOC_DAPM_ADC("ADC1", NULL, PW_MGMT2, 0, 0),
SND_SOC_DAPM_ADC("ADC2", NULL, PW_MGMT2, 1, 0),
};
static const struct snd_soc_dapm_route ak4613_intercon[] = {
{"LOUT1", NULL, "DAC1"},
{"LOUT2", NULL, "DAC2"},
{"LOUT3", NULL, "DAC3"},
{"LOUT4", NULL, "DAC4"},
{"LOUT5", NULL, "DAC5"},
{"LOUT6", NULL, "DAC6"},
{"ROUT1", NULL, "DAC1"},
{"ROUT2", NULL, "DAC2"},
{"ROUT3", NULL, "DAC3"},
{"ROUT4", NULL, "DAC4"},
{"ROUT5", NULL, "DAC5"},
{"ROUT6", NULL, "DAC6"},
{"DAC1", NULL, "Playback"},
{"DAC2", NULL, "Playback"},
{"DAC3", NULL, "Playback"},
{"DAC4", NULL, "Playback"},
{"DAC5", NULL, "Playback"},
{"DAC6", NULL, "Playback"},
{"Capture", NULL, "ADC1"},
{"Capture", NULL, "ADC2"},
{"ADC1", NULL, "LIN1"},
{"ADC2", NULL, "LIN2"},
{"ADC1", NULL, "RIN1"},
{"ADC2", NULL, "RIN2"},
};
static void ak4613_dai_shutdown(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct snd_soc_component *component = dai->component;
struct ak4613_priv *priv = snd_soc_component_get_drvdata(component);
struct device *dev = component->dev;
mutex_lock(&priv->lock);
priv->cnt--;
if (priv->cnt < 0) {
dev_err(dev, "unexpected counter error\n");
priv->cnt = 0;
}
if (!priv->cnt)
priv->ctrl1 = 0;
mutex_unlock(&priv->lock);
}
static void ak4613_hw_constraints(struct ak4613_priv *priv,
struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
static const unsigned int ak4613_rates[] = {
32000,
44100,
48000,
64000,
88200,
96000,
176400,
192000,
};
#define AK4613_CHANNEL_2 0
#define AK4613_CHANNEL_4 1
#define AK4613_CHANNEL_8 2
#define AK4613_CHANNEL_12 3
#define AK4613_CHANNEL_NONE -1
static const unsigned int ak4613_channels[] = {
[AK4613_CHANNEL_2] = 2,
[AK4613_CHANNEL_4] = 4,
[AK4613_CHANNEL_8] = 8,
[AK4613_CHANNEL_12] = 12,
};
#define MODE_MAX 4
#define SDTx_MAX 4
#define MASK(x) (1 << AK4613_CHANNEL_##x)
static const int mask_list[MODE_MAX][SDTx_MAX] = {
/* SDTO SDTIx1 SDTIx2 SDTIx3 */
[AK4613_CONFIG_MODE_STEREO] = { MASK(2), MASK(2), MASK(2), MASK(2)},
[AK4613_CONFIG_MODE_TDM512] = { MASK(4), MASK(12), MASK(12), MASK(12)},
[AK4613_CONFIG_MODE_TDM256] = { MASK(4), MASK(8), MASK(8)|MASK(12), MASK(8)|MASK(12)},
[AK4613_CONFIG_MODE_TDM128] = { MASK(4), MASK(4), MASK(4)|MASK(8), MASK(4)|MASK(8)|MASK(12)},
};
struct snd_pcm_hw_constraint_list *constraint;
unsigned int mask;
unsigned int mode;
unsigned int fs;
int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
int sdti_num;
int i;
constraint = &priv->constraint_rates;
constraint->list = ak4613_rates;
constraint->mask = 0;
constraint->count = 0;
/*
* Slave Mode
* Normal: [32kHz, 48kHz] : 256fs,384fs or 512fs
* Double: [64kHz, 96kHz] : 256fs
* Quad : [128kHz,192kHz]: 128fs
*
* Master mode
* Normal: [32kHz, 48kHz] : 256fs or 512fs
* Double: [64kHz, 96kHz] : 256fs
* Quad : [128kHz,192kHz]: 128fs
*/
for (i = 0; i < ARRAY_SIZE(ak4613_rates); i++) {
/* minimum fs on each range */
fs = (ak4613_rates[i] <= 96000) ? 256 : 128;
if (priv->sysclk >= ak4613_rates[i] * fs)
constraint->count = i + 1;
}
snd_pcm_hw_constraint_list(runtime, 0,
SNDRV_PCM_HW_PARAM_RATE, constraint);
sdti_num = AK4613_CONFIG_SDTI_get(priv);
if (WARN_ON(sdti_num >= SDTx_MAX))
return;
if (priv->cnt) {
/*
* If it was already working,
* the constraint is same as working mode.
*/
mode = AK4613_CTRL1_TO_MODE(priv);
mask = 0; /* no default */
} else {
/*
* It is not yet working,
* the constraint is based on board configs.
* STEREO mask is default
*/
mode = AK4613_CONFIG_GET(priv, MODE);
mask = mask_list[AK4613_CONFIG_MODE_STEREO][is_play * sdti_num];
}
if (WARN_ON(mode >= MODE_MAX))
return;
/* add each mode mask */
mask |= mask_list[mode][is_play * sdti_num];
constraint = &priv->constraint_channels;
constraint->list = ak4613_channels;
constraint->mask = mask;
constraint->count = sizeof(ak4613_channels);
snd_pcm_hw_constraint_list(runtime, 0,
SNDRV_PCM_HW_PARAM_CHANNELS, constraint);
}
static int ak4613_dai_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct snd_soc_component *component = dai->component;
struct ak4613_priv *priv = snd_soc_component_get_drvdata(component);
mutex_lock(&priv->lock);
ak4613_hw_constraints(priv, substream);
priv->cnt++;
mutex_unlock(&priv->lock);
return 0;
}
static int ak4613_dai_set_sysclk(struct snd_soc_dai *codec_dai,
int clk_id, unsigned int freq, int dir)
{
struct snd_soc_component *component = codec_dai->component;
struct ak4613_priv *priv = snd_soc_component_get_drvdata(component);
priv->sysclk = freq;
return 0;
}
static int ak4613_dai_set_fmt(struct snd_soc_dai *dai, unsigned int format)
{
struct snd_soc_component *component = dai->component;
struct ak4613_priv *priv = snd_soc_component_get_drvdata(component);
unsigned int fmt;
fmt = format & SND_SOC_DAIFMT_FORMAT_MASK;
switch (fmt) {
case SND_SOC_DAIFMT_LEFT_J:
case SND_SOC_DAIFMT_I2S:
priv->fmt = fmt;
break;
default:
return -EINVAL;
}
fmt = format & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK;
switch (fmt) {
case SND_SOC_DAIFMT_CBC_CFC:
break;
default:
/*
* SUPPORTME
*
* "clock provider" is not yet supperted
*/
return -EINVAL;
}
return 0;
}
static int ak4613_dai_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
struct snd_soc_component *component = dai->component;
struct ak4613_priv *priv = snd_soc_component_get_drvdata(component);
struct device *dev = component->dev;
unsigned int width = params_width(params);
unsigned int fmt = priv->fmt;
unsigned int rate;
int i, ret;
u8 ctrl2;
rate = params_rate(params);
switch (rate) {
case 32000:
case 44100:
case 48000:
ctrl2 = DFS_NORMAL_SPEED;
break;
case 64000:
case 88200:
case 96000:
ctrl2 = DFS_DOUBLE_SPEED;
break;
case 176400:
case 192000:
ctrl2 = DFS_QUAD_SPEED;
break;
default:
return -EINVAL;
}
priv->rate = rate;
/*
* FIXME
*
* It doesn't have full TDM suppert yet
*/
ret = -EINVAL;
mutex_lock(&priv->lock);
if (priv->cnt > 1) {
/*
* If it was already working, use current priv->ctrl1
*/
ret = 0;
} else {
/*
* It is not yet working,
*/
unsigned int channel = params_channels(params);
u8 tdm;
/* STEREO or TDM */
if (channel == 2)
tdm = AK4613_CONFIG_MODE_STEREO;
else
tdm = AK4613_CONFIG_GET(priv, MODE);
for (i = ARRAY_SIZE(ak4613_iface) - 1; i >= 0; i--) {
const struct ak4613_interface *iface = ak4613_iface + i;
if ((iface->fmt == fmt) && (iface->width == width)) {
/*
* Ctrl1
* | D7 | D6 | D5 | D4 | D3 | D2 | D1 | D0 |
* |TDM1|TDM0|DIF2|DIF1|DIF0|ATS1|ATS0|SMUTE|
* < tdm > < iface->dif >
*/
priv->ctrl1 = (tdm << 6) | (iface->dif << 3);
ret = 0;
break;
}
}
}
mutex_unlock(&priv->lock);
if (ret < 0)
goto hw_params_end;
snd_soc_component_update_bits(component, CTRL1, FMT_MASK, priv->ctrl1);
snd_soc_component_update_bits(component, CTRL2, DFS_MASK, ctrl2);
snd_soc_component_update_bits(component, ICTRL, ICTRL_MASK, priv->ic);
snd_soc_component_update_bits(component, OCTRL, OCTRL_MASK, priv->oc);
hw_params_end:
if (ret < 0)
dev_warn(dev, "unsupported data width/format combination\n");
return ret;
}
static int ak4613_set_bias_level(struct snd_soc_component *component,
enum snd_soc_bias_level level)
{
u8 mgmt1 = 0;
switch (level) {
case SND_SOC_BIAS_ON:
mgmt1 |= RSTN;
fallthrough;
case SND_SOC_BIAS_PREPARE:
mgmt1 |= PMADC | PMDAC;
fallthrough;
case SND_SOC_BIAS_STANDBY:
mgmt1 |= PMVR;
fallthrough;
case SND_SOC_BIAS_OFF:
default:
break;
}
snd_soc_component_write(component, PW_MGMT1, mgmt1);
return 0;
}
static void ak4613_dummy_write(struct work_struct *work)
{
struct ak4613_priv *priv = container_of(work,
struct ak4613_priv,
dummy_write_work);
struct snd_soc_component *component = priv->component;
unsigned int mgmt1;
unsigned int mgmt3;
/*
* PW_MGMT1 / PW_MGMT3 needs dummy write at least after 5 LR clocks
*
* Note
*
* To avoid extra delay, we want to avoid preemption here,
* but we can't. Because it uses I2C access which is using IRQ
* and sleep. Thus, delay might be more than 5 LR clocks
* see also
* ak4613_dai_trigger()
*/
udelay(5000000 / priv->rate);
mgmt1 = snd_soc_component_read(component, PW_MGMT1);
mgmt3 = snd_soc_component_read(component, PW_MGMT3);
snd_soc_component_write(component, PW_MGMT1, mgmt1);
snd_soc_component_write(component, PW_MGMT3, mgmt3);
}
static int ak4613_dai_trigger(struct snd_pcm_substream *substream, int cmd,
struct snd_soc_dai *dai)
{
struct snd_soc_component *component = dai->component;
struct ak4613_priv *priv = snd_soc_component_get_drvdata(component);
/*
* FIXME
*
* PW_MGMT1 / PW_MGMT3 needs dummy write at least after 5 LR clocks
* from Power Down Release. Otherwise, Playback volume will be 0dB.
* To avoid complex multiple delay/dummy_write method from
* ak4613_set_bias_level() / SND_SOC_DAPM_DAC_E("DACx", ...),
* call it once here.
*
* But, unfortunately, we can't "write" here because here is atomic
* context (It uses I2C access for writing).
* Thus, use schedule_work() to switching to normal context
* immediately.
*
* Note
*
* Calling ak4613_dummy_write() function might be delayed.
* In such case, ak4613 volume might be temporarily 0dB when
* beggining of playback.
* see also
* ak4613_dummy_write()
*/
if ((cmd != SNDRV_PCM_TRIGGER_START) &&
(cmd != SNDRV_PCM_TRIGGER_RESUME))
return 0;
if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)
return 0;
priv->component = component;
schedule_work(&priv->dummy_write_work);
return 0;
}
/*
* Select below from Sound Card, not Auto
* SND_SOC_DAIFMT_CBC_CFC
* SND_SOC_DAIFMT_CBP_CFP
*/
static u64 ak4613_dai_formats =
SND_SOC_POSSIBLE_DAIFMT_I2S |
SND_SOC_POSSIBLE_DAIFMT_LEFT_J;
static const struct snd_soc_dai_ops ak4613_dai_ops = {
.startup = ak4613_dai_startup,
.shutdown = ak4613_dai_shutdown,
.set_sysclk = ak4613_dai_set_sysclk,
.set_fmt = ak4613_dai_set_fmt,
.trigger = ak4613_dai_trigger,
.hw_params = ak4613_dai_hw_params,
.auto_selectable_formats = &ak4613_dai_formats,
.num_auto_selectable_formats = 1,
};
#define AK4613_PCM_RATE (SNDRV_PCM_RATE_32000 |\
SNDRV_PCM_RATE_44100 |\
SNDRV_PCM_RATE_48000 |\
SNDRV_PCM_RATE_64000 |\
SNDRV_PCM_RATE_88200 |\
SNDRV_PCM_RATE_96000 |\
SNDRV_PCM_RATE_176400 |\
SNDRV_PCM_RATE_192000)
#define AK4613_PCM_FMTBIT (SNDRV_PCM_FMTBIT_S24_LE)
static struct snd_soc_dai_driver ak4613_dai = {
.name = "ak4613-hifi",
.playback = {
.stream_name = "Playback",
.channels_min = 2,
.channels_max = 12,
.rates = AK4613_PCM_RATE,
.formats = AK4613_PCM_FMTBIT,
},
.capture = {
.stream_name = "Capture",
.channels_min = 2,
.channels_max = 4,
.rates = AK4613_PCM_RATE,
.formats = AK4613_PCM_FMTBIT,
},
.ops = &ak4613_dai_ops,
.symmetric_rate = 1,
};
static int ak4613_suspend(struct snd_soc_component *component)
{
struct regmap *regmap = dev_get_regmap(component->dev, NULL);
regcache_cache_only(regmap, true);
regcache_mark_dirty(regmap);
return 0;
}
static int ak4613_resume(struct snd_soc_component *component)
{
struct regmap *regmap = dev_get_regmap(component->dev, NULL);
regcache_cache_only(regmap, false);
return regcache_sync(regmap);
}
static const struct snd_soc_component_driver soc_component_dev_ak4613 = {
.suspend = ak4613_suspend,
.resume = ak4613_resume,
.set_bias_level = ak4613_set_bias_level,
.controls = ak4613_snd_controls,
.num_controls = ARRAY_SIZE(ak4613_snd_controls),
.dapm_widgets = ak4613_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(ak4613_dapm_widgets),
.dapm_routes = ak4613_intercon,
.num_dapm_routes = ARRAY_SIZE(ak4613_intercon),
.idle_bias_on = 1,
.endianness = 1,
};
static void ak4613_parse_of(struct ak4613_priv *priv,
struct device *dev)
{
struct device_node *np = dev->of_node;
char prop[32];
int sdti_num;
int i;
/* Input 1 - 2 */
for (i = 0; i < 2; i++) {
snprintf(prop, sizeof(prop), "asahi-kasei,in%d-single-end", i + 1);
if (!of_get_property(np, prop, NULL))
priv->ic |= 1 << i;
}
/* Output 1 - 6 */
for (i = 0; i < 6; i++) {
snprintf(prop, sizeof(prop), "asahi-kasei,out%d-single-end", i + 1);
if (!of_get_property(np, prop, NULL))
priv->oc |= 1 << i;
}
/*
* enable TDM256 test
*
* !!! FIXME !!!
*
* It should be configured by DT or other way
* if it was full supported.
* But it is using ifdef style for now for test
* purpose.
*/
#if defined(AK4613_ENABLE_TDM_TEST)
AK4613_CONFIG_SET(priv, MODE_TDM256);
#endif
/*
* connected STDI
* TDM support is assuming it is probed via Audio-Graph-Card style here.
* Default is SDTIx1 if it was probed via Simple-Audio-Card for now.
*/
sdti_num = of_graph_get_endpoint_count(np);
if ((sdti_num >= SDTx_MAX) || (sdti_num < 1))
sdti_num = 1;
AK4613_CONFIG_SDTI_set(priv, sdti_num);
}
static int ak4613_i2c_probe(struct i2c_client *i2c)
{
struct device *dev = &i2c->dev;
struct device_node *np = dev->of_node;
const struct regmap_config *regmap_cfg;
struct regmap *regmap;
struct ak4613_priv *priv;
regmap_cfg = NULL;
if (np)
regmap_cfg = of_device_get_match_data(dev);
else {
const struct i2c_device_id *id =
i2c_match_id(ak4613_i2c_id, i2c);
regmap_cfg = (const struct regmap_config *)id->driver_data;
}
if (!regmap_cfg)
return -EINVAL;
priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
if (!priv)
return -ENOMEM;
ak4613_parse_of(priv, dev);
priv->ctrl1 = 0;
priv->cnt = 0;
priv->sysclk = 0;
INIT_WORK(&priv->dummy_write_work, ak4613_dummy_write);
mutex_init(&priv->lock);
i2c_set_clientdata(i2c, priv);
regmap = devm_regmap_init_i2c(i2c, regmap_cfg);
if (IS_ERR(regmap))
return PTR_ERR(regmap);
return devm_snd_soc_register_component(dev, &soc_component_dev_ak4613,
&ak4613_dai, 1);
}
static struct i2c_driver ak4613_i2c_driver = {
.driver = {
.name = "ak4613-codec",
.of_match_table = ak4613_of_match,
},
.probe_new = ak4613_i2c_probe,
.id_table = ak4613_i2c_id,
};
module_i2c_driver(ak4613_i2c_driver);
MODULE_DESCRIPTION("Soc AK4613 driver");
MODULE_AUTHOR("Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>");
MODULE_LICENSE("GPL v2");
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