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ASoC: samsung: Add machine driver for Exynos5433 based TM2 board

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This patch adds the sound machine driver for the TM2 and TM2E boards.
Speaker and headphone playback, Main Mic capture, Bluetooth, Voice
call and external accessory are supported.

Signed-off-by: Inha Song <ideal.song@samsung.com>
[k.kozlowski: rebased on 4.1]
Signed-off-by: Krzysztof Kozlowski <krzk@kernel.org>
[s.nawrocki: rebased to 4.7, adjustment to the ASoC core changes,
 removed unused ops and direct calls to the max98504 function,
 added parsing of "audio-amplifier" and "audio-codec"
 properties, added TDM API calls, switched to gpiod API]
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Reviewed-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This commit is contained in:
Sylwester Nawrocki 2016-11-02 17:05:45 +01:00 committed by Mark Brown
parent 48a760279b
commit 1bfbc260a5
3 changed files with 563 additions and 0 deletions
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9
sound/soc/samsung/Kconfig
View File

@ -190,4 +190,13 @@ config SND_SOC_ARNDALE_RT5631_ALC5631
select SND_SAMSUNG_I2S
select SND_SOC_RT5631
config SND_SOC_SAMSUNG_TM2_WM5110
tristate "SoC I2S Audio support for WM5110 on TM2 board"
depends on SND_SOC_SAMSUNG && MFD_ARIZONA && I2C && SPI_MASTER
select SND_SOC_MAX98504
select SND_SOC_WM5110
select SND_SAMSUNG_I2S
help
Say Y if you want to add support for SoC audio on the TM2 board.
endif #SND_SOC_SAMSUNG

2
sound/soc/samsung/Makefile
View File

@ -41,6 +41,7 @@ snd-soc-lowland-objs := lowland.o
snd-soc-littlemill-objs := littlemill.o
snd-soc-bells-objs := bells.o
snd-soc-arndale-rt5631-objs := arndale_rt5631.o
snd-soc-tm2-wm5110-objs := tm2_wm5110.o
obj-$(CONFIG_SND_SOC_SAMSUNG_JIVE_WM8750) += snd-soc-jive-wm8750.o
obj-$(CONFIG_SND_SOC_SAMSUNG_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o
@ -62,3 +63,4 @@ obj-$(CONFIG_SND_SOC_LOWLAND) += snd-soc-lowland.o
obj-$(CONFIG_SND_SOC_LITTLEMILL) += snd-soc-littlemill.o
obj-$(CONFIG_SND_SOC_BELLS) += snd-soc-bells.o
obj-$(CONFIG_SND_SOC_ARNDALE_RT5631_ALC5631) += snd-soc-arndale-rt5631.o
obj-$(CONFIG_SND_SOC_SAMSUNG_TM2_WM5110) += snd-soc-tm2-wm5110.o

552
sound/soc/samsung/tm2_wm5110.c Normal file
View File

@ -0,0 +1,552 @@
/*
* Copyright (C) 2015 - 2016 Samsung Electronics Co., Ltd.
*
* Authors: Inha Song <ideal.song@samsung.com>
* Sylwester Nawrocki <s.nawrocki@samsung.com>
*
* This program is free software; you can redistribute it and/or modify it
* under the terms of the GNU General Public License as published by the
* Free Software Foundation; either version 2 of the License, or (at your
* option) any later version.
*/
#include <linux/clk.h>
#include <linux/gpio.h>
#include <linux/module.h>
#include <linux/of.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include "i2s.h"
#include "../codecs/wm5110.h"
/*
* The source clock is XCLKOUT with its mux set to the external fixed rate
* oscillator (XXTI).
*/
#define MCLK_RATE 24000000U
#define TM2_DAI_AIF1 0
#define TM2_DAI_AIF2 1
struct tm2_machine_priv {
struct snd_soc_codec *codec;
unsigned int sysclk_rate;
struct gpio_desc *gpio_mic_bias;
};
static int tm2_start_sysclk(struct snd_soc_card *card)
{
struct tm2_machine_priv *priv = snd_soc_card_get_drvdata(card);
struct snd_soc_codec *codec = priv->codec;
int ret;
ret = snd_soc_codec_set_pll(codec, WM5110_FLL1_REFCLK,
ARIZONA_FLL_SRC_MCLK1,
MCLK_RATE,
priv->sysclk_rate);
if (ret < 0) {
dev_err(codec->dev, "Failed to set FLL1 source: %d\n", ret);
return ret;
}
ret = snd_soc_codec_set_pll(codec, WM5110_FLL1,
ARIZONA_FLL_SRC_MCLK1,
MCLK_RATE,
priv->sysclk_rate);
if (ret < 0) {
dev_err(codec->dev, "Failed to start FLL1: %d\n", ret);
return ret;
}
ret = snd_soc_codec_set_sysclk(codec, ARIZONA_CLK_SYSCLK,
ARIZONA_CLK_SRC_FLL1,
priv->sysclk_rate,
SND_SOC_CLOCK_IN);
if (ret < 0) {
dev_err(codec->dev, "Failed to set SYSCLK source: %d\n", ret);
return ret;
}
return 0;
}
static int tm2_stop_sysclk(struct snd_soc_card *card)
{
struct tm2_machine_priv *priv = snd_soc_card_get_drvdata(card);
struct snd_soc_codec *codec = priv->codec;
int ret;
ret = snd_soc_codec_set_pll(codec, WM5110_FLL1, 0, 0, 0);
if (ret < 0) {
dev_err(codec->dev, "Failed to stop FLL1: %d\n", ret);
return ret;
}
ret = snd_soc_codec_set_sysclk(codec, ARIZONA_CLK_SYSCLK,
ARIZONA_CLK_SRC_FLL1, 0, 0);
if (ret < 0) {
dev_err(codec->dev, "Failed to stop SYSCLK: %d\n", ret);
return ret;
}
return 0;
}
static int tm2_aif1_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_codec *codec = rtd->codec;
struct tm2_machine_priv *priv = snd_soc_card_get_drvdata(rtd->card);
switch (params_rate(params)) {
case 4000:
case 8000:
case 12000:
case 16000:
case 24000:
case 32000:
case 48000:
case 96000:
case 192000:
/* Highest possible SYSCLK frequency: 147.456MHz */
priv->sysclk_rate = 147456000U;
break;
case 11025:
case 22050:
case 44100:
case 88200:
case 176400:
/* Highest possible SYSCLK frequency: 135.4752 MHz */
priv->sysclk_rate = 135475200U;
break;
default:
dev_err(codec->dev, "Not supported sample rate: %d\n",
params_rate(params));
return -EINVAL;
}
return tm2_start_sysclk(rtd->card);
}
static struct snd_soc_ops tm2_aif1_ops = {
.hw_params = tm2_aif1_hw_params,
};
static int tm2_aif2_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_codec *codec = rtd->codec;
unsigned int asyncclk_rate;
int ret;
switch (params_rate(params)) {
case 8000:
case 12000:
case 16000:
/* Highest possible ASYNCCLK frequency: 49.152MHz */
asyncclk_rate = 49152000U;
break;
case 11025:
/* Highest possible ASYNCCLK frequency: 45.1584 MHz */
asyncclk_rate = 45158400U;
break;
default:
dev_err(codec->dev, "Not supported sample rate: %d\n",
params_rate(params));
return -EINVAL;
}
ret = snd_soc_codec_set_pll(codec, WM5110_FLL2_REFCLK,
ARIZONA_FLL_SRC_MCLK1,
MCLK_RATE,
asyncclk_rate);
if (ret < 0) {
dev_err(codec->dev, "Failed to set FLL2 source: %d\n", ret);
return ret;
}
ret = snd_soc_codec_set_pll(codec, WM5110_FLL2,
ARIZONA_FLL_SRC_MCLK1,
MCLK_RATE,
asyncclk_rate);
if (ret < 0) {
dev_err(codec->dev, "Failed to start FLL2: %d\n", ret);
return ret;
}
ret = snd_soc_codec_set_sysclk(codec, ARIZONA_CLK_ASYNCCLK,
ARIZONA_CLK_SRC_FLL2,
asyncclk_rate,
SND_SOC_CLOCK_IN);
if (ret < 0) {
dev_err(codec->dev, "Failed to set ASYNCCLK source: %d\n", ret);
return ret;
}
return 0;
}
static int tm2_aif2_hw_free(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_codec *codec = rtd->codec;
int ret;
/* disable FLL2 */
ret = snd_soc_codec_set_pll(codec, WM5110_FLL2, ARIZONA_FLL_SRC_MCLK1,
0, 0);
if (ret < 0)
dev_err(codec->dev, "Failed to stop FLL2: %d\n", ret);
return ret;
}
static struct snd_soc_ops tm2_aif2_ops = {
.hw_params = tm2_aif2_hw_params,
.hw_free = tm2_aif2_hw_free,
};
static int tm2_mic_bias(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
struct snd_soc_card *card = w->dapm->card;
struct tm2_machine_priv *priv = snd_soc_card_get_drvdata(card);
switch (event) {
case SND_SOC_DAPM_PRE_PMU:
gpiod_set_value_cansleep(priv->gpio_mic_bias, 1);
break;
case SND_SOC_DAPM_POST_PMD:
gpiod_set_value_cansleep(priv->gpio_mic_bias, 0);
break;
}
return 0;
}
static int tm2_set_bias_level(struct snd_soc_card *card,
struct snd_soc_dapm_context *dapm,
enum snd_soc_bias_level level)
{
struct snd_soc_pcm_runtime *rtd;
rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name);
if (dapm->dev != rtd->codec_dai->dev)
return 0;
switch (level) {
case SND_SOC_BIAS_STANDBY:
if (card->dapm.bias_level == SND_SOC_BIAS_OFF)
tm2_start_sysclk(card);
break;
case SND_SOC_BIAS_OFF:
tm2_stop_sysclk(card);
break;
default:
break;
}
return 0;
}
static struct snd_soc_aux_dev tm2_speaker_amp_dev;
static int tm2_late_probe(struct snd_soc_card *card)
{
struct tm2_machine_priv *priv = snd_soc_card_get_drvdata(card);
struct snd_soc_dai_link_component dlc = { 0 };
unsigned int ch_map[] = { 0, 1 };
struct snd_soc_dai *amp_pdm_dai;
struct snd_soc_pcm_runtime *rtd;
struct snd_soc_dai *aif1_dai;
struct snd_soc_dai *aif2_dai;
int ret;
rtd = snd_soc_get_pcm_runtime(card, card->dai_link[TM2_DAI_AIF1].name);
aif1_dai = rtd->codec_dai;
priv->codec = rtd->codec;
ret = snd_soc_dai_set_sysclk(aif1_dai, ARIZONA_CLK_SYSCLK, 0, 0);
if (ret < 0) {
dev_err(aif1_dai->dev, "Failed to set SYSCLK: %d\n", ret);
return ret;
}
rtd = snd_soc_get_pcm_runtime(card, card->dai_link[TM2_DAI_AIF2].name);
aif2_dai = rtd->codec_dai;
ret = snd_soc_dai_set_sysclk(aif2_dai, ARIZONA_CLK_ASYNCCLK, 0, 0);
if (ret < 0) {
dev_err(aif2_dai->dev, "Failed to set ASYNCCLK: %d\n", ret);
return ret;
}
dlc.of_node = tm2_speaker_amp_dev.codec_of_node;
amp_pdm_dai = snd_soc_find_dai(&dlc);
if (!amp_pdm_dai)
return -ENODEV;
/* Set the MAX98504 V/I sense PDM Tx DAI channel mapping */
ret = snd_soc_dai_set_channel_map(amp_pdm_dai, ARRAY_SIZE(ch_map),
ch_map, 0, NULL);
if (ret < 0)
return ret;
ret = snd_soc_dai_set_tdm_slot(amp_pdm_dai, 0x3, 0x0, 2, 16);
if (ret < 0)
return ret;
return 0;
}
static const struct snd_kcontrol_new tm2_controls[] = {
SOC_DAPM_PIN_SWITCH("HP"),
SOC_DAPM_PIN_SWITCH("SPK"),
SOC_DAPM_PIN_SWITCH("RCV"),
SOC_DAPM_PIN_SWITCH("VPS"),
SOC_DAPM_PIN_SWITCH("HDMI"),
SOC_DAPM_PIN_SWITCH("Main Mic"),
SOC_DAPM_PIN_SWITCH("Sub Mic"),
SOC_DAPM_PIN_SWITCH("Third Mic"),
SOC_DAPM_PIN_SWITCH("Headset Mic"),
};
const struct snd_soc_dapm_widget tm2_dapm_widgets[] = {
SND_SOC_DAPM_HP("HP", NULL),
SND_SOC_DAPM_SPK("SPK", NULL),
SND_SOC_DAPM_SPK("RCV", NULL),
SND_SOC_DAPM_LINE("VPS", NULL),
SND_SOC_DAPM_LINE("HDMI", NULL),
SND_SOC_DAPM_MIC("Main Mic", tm2_mic_bias),
SND_SOC_DAPM_MIC("Sub Mic", NULL),
SND_SOC_DAPM_MIC("Third Mic", NULL),
SND_SOC_DAPM_MIC("Headset Mic", NULL),
};
static const struct snd_soc_component_driver tm2_component = {
.name = "tm2-audio",
};
static struct snd_soc_dai_driver tm2_ext_dai[] = {
{
.name = "Voice call",
.playback = {
.channels_min = 1,
.channels_max = 4,
.rate_min = 8000,
.rate_max = 48000,
.rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |
SNDRV_PCM_RATE_48000),
.formats = SNDRV_PCM_FMTBIT_S16_LE,
},
.capture = {
.channels_min = 1,
.channels_max = 4,
.rate_min = 8000,
.rate_max = 48000,
.rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |
SNDRV_PCM_RATE_48000),
.formats = SNDRV_PCM_FMTBIT_S16_LE,
},
},
{
.name = "Bluetooth",
.playback = {
.channels_min = 1,
.channels_max = 4,
.rate_min = 8000,
.rate_max = 16000,
.rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000),
.formats = SNDRV_PCM_FMTBIT_S16_LE,
},
.capture = {
.channels_min = 1,
.channels_max = 2,
.rate_min = 8000,
.rate_max = 16000,
.rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000),
.formats = SNDRV_PCM_FMTBIT_S16_LE,
},
},
};
static struct snd_soc_dai_link tm2_dai_links[] = {
{
.name = "WM5110 AIF1",
.stream_name = "HiFi Primary",
.codec_dai_name = "wm5110-aif1",
.ops = &tm2_aif1_ops,
.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBM_CFM,
}, {
.name = "WM5110 Voice",
.stream_name = "Voice call",
.codec_dai_name = "wm5110-aif2",
.ops = &tm2_aif2_ops,
.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBM_CFM,
.ignore_suspend = 1,
}, {
.name = "WM5110 BT",
.stream_name = "Bluetooth",
.codec_dai_name = "wm5110-aif3",
.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBM_CFM,
.ignore_suspend = 1,
}
};
static struct snd_soc_card tm2_card = {
.owner = THIS_MODULE,
.dai_link = tm2_dai_links,
.num_links = ARRAY_SIZE(tm2_dai_links),
.controls = tm2_controls,
.num_controls = ARRAY_SIZE(tm2_controls),
.dapm_widgets = tm2_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(tm2_dapm_widgets),
.aux_dev = &tm2_speaker_amp_dev,
.num_aux_devs = 1,
.late_probe = tm2_late_probe,
.set_bias_level = tm2_set_bias_level,
};
static int tm2_probe(struct platform_device *pdev)
{
struct device *dev = &pdev->dev;
struct snd_soc_card *card = &tm2_card;
struct tm2_machine_priv *priv;
struct device_node *cpu_dai_node, *codec_dai_node;
int ret, i;
priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
if (!priv)
return -ENOMEM;
snd_soc_card_set_drvdata(card, priv);
card->dev = dev;
priv->gpio_mic_bias = devm_gpiod_get(dev, "mic-bias",
GPIOF_OUT_INIT_LOW);
if (IS_ERR(priv->gpio_mic_bias)) {
dev_err(dev, "Failed to get mic bias gpio\n");
return PTR_ERR(priv->gpio_mic_bias);
}
ret = snd_soc_of_parse_card_name(card, "model");
if (ret < 0) {
dev_err(dev, "Card name is not specified\n");
return ret;
}
ret = snd_soc_of_parse_audio_routing(card, "samsung,audio-routing");
if (ret < 0) {
dev_err(dev, "Audio routing is not specified or invalid\n");
return ret;
}
card->aux_dev[0].codec_of_node = of_parse_phandle(dev->of_node,
"audio-amplifier", 0);
if (!card->aux_dev[0].codec_of_node) {
dev_err(dev, "audio-amplifier property invalid or missing\n");
return -EINVAL;
}
cpu_dai_node = of_parse_phandle(dev->of_node, "i2s-controller", 0);
if (!cpu_dai_node) {
dev_err(dev, "i2s-controllers property invalid or missing\n");
ret = -EINVAL;
goto amp_node_put;
}
codec_dai_node = of_parse_phandle(dev->of_node, "audio-codec", 0);
if (!codec_dai_node) {
dev_err(dev, "audio-codec property invalid or missing\n");
ret = -EINVAL;
goto cpu_dai_node_put;
}
for (i = 0; i < card->num_links; i++) {
card->dai_link[i].cpu_dai_name = NULL;
card->dai_link[i].cpu_name = NULL;
card->dai_link[i].platform_name = NULL;
card->dai_link[i].codec_of_node = codec_dai_node;
card->dai_link[i].cpu_of_node = cpu_dai_node;
card->dai_link[i].platform_of_node = cpu_dai_node;
}
ret = devm_snd_soc_register_component(dev, &tm2_component,
tm2_ext_dai, ARRAY_SIZE(tm2_ext_dai));
if (ret < 0) {
dev_err(dev, "Failed to register component: %d\n", ret);
goto codec_dai_node_put;
}
ret = devm_snd_soc_register_card(dev, card);
if (ret < 0) {
dev_err(dev, "Failed to register card: %d\n", ret);
goto codec_dai_node_put;
}
codec_dai_node_put:
of_node_put(codec_dai_node);
cpu_dai_node_put:
of_node_put(cpu_dai_node);
amp_node_put:
of_node_put(card->aux_dev[0].codec_of_node);
return ret;
}
static int tm2_pm_prepare(struct device *dev)
{
struct snd_soc_card *card = dev_get_drvdata(dev);
return tm2_stop_sysclk(card);
}
static void tm2_pm_complete(struct device *dev)
{
struct snd_soc_card *card = dev_get_drvdata(dev);
tm2_start_sysclk(card);
}
const struct dev_pm_ops tm2_pm_ops = {
.prepare = tm2_pm_prepare,
.suspend = snd_soc_suspend,
.resume = snd_soc_resume,
.complete = tm2_pm_complete,
.freeze = snd_soc_suspend,
.thaw = snd_soc_resume,
.poweroff = snd_soc_poweroff,
.restore = snd_soc_resume,
};
static const struct of_device_id tm2_of_match[] = {
{ .compatible = "samsung,tm2-audio" },
{ },
};
MODULE_DEVICE_TABLE(of, tm2_of_match);
static struct platform_driver tm2_driver = {
.driver = {
.name = "tm2-audio",
.pm = &tm2_pm_ops,
.of_match_table = tm2_of_match,
},
.probe = tm2_probe,
};
module_platform_driver(tm2_driver);
MODULE_AUTHOR("Inha Song <ideal.song@samsung.com>");
MODULE_DESCRIPTION("ALSA SoC Exynos TM2 Audio Support");
MODULE_LICENSE("GPL v2");