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