This patch adds the sound machine driver for TM2 and TM2E board. 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 onto v4.1] Signed-off-by: Krzysztof Kozlowski k.kozlowski@samsung.com [s.nawrocki: rebased onto v4.8, 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 ---

Changes since v5: - dropped requesting and managing of the CODEC's clocks, - removed driver remove() handler, - changed pm_ops to use prepare/complete rather than late_suspend/early_resume.

Changes since v4 (addressing review comments from Charles): - changed the order of WM5110_FLL{1,2}, WM5110_FLL{1,2}_REFCLK setting, - ARIZONA_CLK_SYSCLK, ARIZONA_CLK_ASYNCCLK setting moved to late_probe, - added tm2_aif2_hw_free callback for disabling FLL2, - removed unneded card->dapm.bias_level assignment in tm2_mic_bias callback, - suspend_late, resume_early dev_pm_ops used instead of suspend_post, resume_pre struct snd_soc_card callbacks.

Changes since v3: - removed SND_SOC_SAMSUNG_AUDSS from Kconfig.

Changes since v2: - added missing Kconfig dependencies.

Changes since initial version: - added PDM Tx channels setup through TDM API - adaptation to renamed 'samsung,model', 'samsung,i2s-controller', 'samsung,speaker-amplifier' properties, - removed some dev_dbg() calls, - cleaned up mic-bias GPIO handling and switched to gpiod API, - added parsing of 'audio-codec' property, - initialized codec_of_node of dai_link instead of codec_name, - switched to using clock, clock-names properties from the wm5110 codec node, - fixed error paths in probe() (of_node reference counting). --- sound/soc/samsung/Kconfig | 9 + sound/soc/samsung/Makefile | 2 + sound/soc/samsung/tm2_wm5110.c | 553 +++++++++++++++++++++++++++++++++++++++++ 3 files changed, 564 insertions(+) create mode 100644 sound/soc/samsung/tm2_wm5110.c

diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 78baa26..a711605 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -237,3 +237,12 @@ config SND_SOC_ARNDALE_RT5631_ALC5631 depends on SND_SOC_SAMSUNG && I2C 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. diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile index 052fe71..c15a759 100644 --- a/sound/soc/samsung/Makefile +++ b/sound/soc/samsung/Makefile @@ -45,6 +45,7 @@ snd-soc-littlemill-objs := littlemill.o snd-soc-bells-objs := bells.o snd-soc-odroidx2-max98090-objs := odroidx2_max98090.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 @@ -71,3 +72,4 @@ obj-$(CONFIG_SND_SOC_LITTLEMILL) += snd-soc-littlemill.o obj-$(CONFIG_SND_SOC_BELLS) += snd-soc-bells.o obj-$(CONFIG_SND_SOC_ODROIDX2) += snd-soc-odroidx2-max98090.o obj-$(CONFIG_SND_SOC_ARNDALE_RT5631_ALC5631) += snd-soc-arndale-rt5631.o +obj-$(CONFIG_SND_SOC_SAMSUNG_TM2_WM5110) += snd-soc-tm2-wm5110.o diff --git a/sound/soc/samsung/tm2_wm5110.c b/sound/soc/samsung/tm2_wm5110.c new file mode 100644 index 0000000..7950350 --- /dev/null +++ b/sound/soc/samsung/tm2_wm5110.c @@ -0,0 +1,553 @@ +/* + * 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; + struct tm2_machine_priv *priv = snd_soc_card_get_drvdata(rtd->card); + 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 int tm2_pm_complete(struct device *dev) +{ + struct snd_soc_card *card = dev_get_drvdata(dev); + + return 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"); -- 1.9.1