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+/*
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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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+
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+#include <linux/clk.h>
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+#include <linux/gpio.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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+
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+#include "i2s.h"
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+#include "../codecs/wm5110.h"
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+
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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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+
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+#define TM2_DAI_AIF1 0
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+#define TM2_DAI_AIF2 1
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+
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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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+
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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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+
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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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+
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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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+
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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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+
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+ return 0;
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+}
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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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+
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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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+
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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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+
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+ return 0;
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+}
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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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+
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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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+
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+ return tm2_start_sysclk(rtd->card);
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+}
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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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+
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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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+
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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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+
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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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+
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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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+
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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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+
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+ return 0;
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+}
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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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+
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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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+
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+ return ret;
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+}
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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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+
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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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+
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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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+
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+ return 0;
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+}
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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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+
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+ rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name);
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+
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+ if (dapm->dev != rtd->codec_dai->dev)
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+ return 0;
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+
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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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+
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+ return 0;
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+}
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+
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+static struct snd_soc_aux_dev tm2_speaker_amp_dev;
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+
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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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+
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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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+
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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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+
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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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+
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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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+
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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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+
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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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+
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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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+
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+ return 0;
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+}
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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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+
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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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+
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+ SOC_DAPM_PIN_SWITCH("Headset Mic"),
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+};
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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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+
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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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+
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+ SND_SOC_DAPM_MIC("Headset Mic", NULL),
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+};
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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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+
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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 = {
|
|
|
|
+ .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");
|