/*
   * SoC audio for HTC Magician
   *
   * Copyright (c) 2006 Philipp Zabel <philipp.zabel@gmail.com>
   *
   * based on spitz.c,
   * Authors: Liam Girdwood <lrg@slimlogic.co.uk>
   *          Richard Purdie <richard@openedhand.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/module.h>
  #include <linux/timer.h>
  #include <linux/interrupt.h>
  #include <linux/platform_device.h>
  #include <linux/delay.h>
  #include <linux/gpio.h>
  #include <linux/i2c.h>
  
  #include <sound/core.h>
  #include <sound/pcm.h>
  #include <sound/pcm_params.h>
  #include <sound/soc.h>
  #include <sound/uda1380.h>
  
  #include <mach/magician.h>
  #include <asm/mach-types.h>
  #include "../codecs/uda1380.h"
  #include "pxa2xx-i2s.h"
  #include "pxa-ssp.h"
  
  #define MAGICIAN_MIC       0
  #define MAGICIAN_MIC_EXT   1
  
  static int magician_hp_switch;
  static int magician_spk_switch = 1;
  static int magician_in_sel = MAGICIAN_MIC;
  
  static void magician_ext_control(struct snd_soc_codec *codec)
  {
  	struct snd_soc_dapm_context *dapm = &codec->dapm;
  
  	if (magician_spk_switch)
  		snd_soc_dapm_enable_pin(dapm, "Speaker");
  	else
  		snd_soc_dapm_disable_pin(dapm, "Speaker");
  	if (magician_hp_switch)
  		snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
  	else
  		snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
  
  	switch (magician_in_sel) {
  	case MAGICIAN_MIC:
  		snd_soc_dapm_disable_pin(dapm, "Headset Mic");
  		snd_soc_dapm_enable_pin(dapm, "Call Mic");
  		break;
  	case MAGICIAN_MIC_EXT:
  		snd_soc_dapm_disable_pin(dapm, "Call Mic");
  		snd_soc_dapm_enable_pin(dapm, "Headset Mic");
  		break;
  	}
  
  	snd_soc_dapm_sync(dapm);
  }
  
  static int magician_startup(struct snd_pcm_substream *substream)
  {
  	struct snd_soc_pcm_runtime *rtd = substream->private_data;
  	struct snd_soc_codec *codec = rtd->codec;
  
  	mutex_lock(&codec->mutex);
  
  	/* check the jack status at stream startup */
  	magician_ext_control(codec);
  
  	mutex_unlock(&codec->mutex);
  
  	return 0;
  }
  
  /*
   * Magician uses SSP port for playback.
   */
  static int magician_playback_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_dai *codec_dai = rtd->codec_dai;
  	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
  	unsigned int acps, acds, width;
  	unsigned int div4 = PXA_SSP_CLK_SCDB_4;
  	int ret = 0;
  
  	width = snd_pcm_format_physical_width(params_format(params));
  
  	/*
  	 * rate = SSPSCLK / (2 * width(16 or 32))
  	 * SSPSCLK = (ACPS / ACDS) / SSPSCLKDIV(div4 or div1)
  	 */
  	switch (params_rate(params)) {
  	case 8000:
  		/* off by a factor of 2: bug in the PXA27x audio clock? */
  		acps = 32842000;
  		switch (width) {
  		case 16:
  			/* 513156 Hz ~= _2_ * 8000 Hz * 32 (+0.23%) */
  			acds = PXA_SSP_CLK_AUDIO_DIV_16;
  			break;
  		default: /* 32 */
  			/* 1026312 Hz ~= _2_ * 8000 Hz * 64 (+0.23%) */
  			acds = PXA_SSP_CLK_AUDIO_DIV_8;
  		}
  		break;
  	case 11025:
  		acps = 5622000;
  		switch (width) {
  		case 16:
  			/* 351375 Hz ~= 11025 Hz * 32 (-0.41%) */
  			acds = PXA_SSP_CLK_AUDIO_DIV_4;
  			break;
  		default: /* 32 */
  			/* 702750 Hz ~= 11025 Hz * 64 (-0.41%) */
  			acds = PXA_SSP_CLK_AUDIO_DIV_2;
  		}
  		break;
  	case 22050:
  		acps = 5622000;
  		switch (width) {
  		case 16:
  			/* 702750 Hz ~= 22050 Hz * 32 (-0.41%) */
  			acds = PXA_SSP_CLK_AUDIO_DIV_2;
  			break;
  		default: /* 32 */
  			/* 1405500 Hz ~= 22050 Hz * 64 (-0.41%) */
  			acds = PXA_SSP_CLK_AUDIO_DIV_1;
  		}
  		break;
  	case 44100:
  		acps = 5622000;
  		switch (width) {
  		case 16:
  			/* 1405500 Hz ~= 44100 Hz * 32 (-0.41%) */
  			acds = PXA_SSP_CLK_AUDIO_DIV_2;
  			break;
  		default: /* 32 */
  			/* 2811000 Hz ~= 44100 Hz * 64 (-0.41%) */
  			acds = PXA_SSP_CLK_AUDIO_DIV_1;
  		}
  		break;
  	case 48000:
  		acps = 12235000;
  		switch (width) {
  		case 16:
  			/* 1529375 Hz ~= 48000 Hz * 32 (-0.44%) */
  			acds = PXA_SSP_CLK_AUDIO_DIV_2;
  			break;
  		default: /* 32 */
  			/* 3058750 Hz ~= 48000 Hz * 64 (-0.44%) */
  			acds = PXA_SSP_CLK_AUDIO_DIV_1;
  		}
  		break;
  	case 96000:
  	default:
  		acps = 12235000;
  		switch (width) {
  		case 16:
  			/* 3058750 Hz ~= 96000 Hz * 32 (-0.44%) */
  			acds = PXA_SSP_CLK_AUDIO_DIV_1;
  			break;
  		default: /* 32 */
  			/* 6117500 Hz ~= 96000 Hz * 64 (-0.44%) */
  			acds = PXA_SSP_CLK_AUDIO_DIV_2;
  			div4 = PXA_SSP_CLK_SCDB_1;
  			break;
  		}
  		break;
  	}
  
  	/* set codec DAI configuration */
  	ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_MSB |
  			SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
  	if (ret < 0)
  		return ret;
  
  	/* set cpu DAI configuration */
  	ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A |
  			SND_SOC_DAIFMT_NB_IF | SND_SOC_DAIFMT_CBS_CFS);
  	if (ret < 0)
  		return ret;
  
  	ret = snd_soc_dai_set_tdm_slot(cpu_dai, 1, 0, 1, width);
  	if (ret < 0)
  		return ret;
  
  	/* set audio clock as clock source */
  	ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_AUDIO, 0,
  			SND_SOC_CLOCK_OUT);
  	if (ret < 0)
  		return ret;
  
  	/* set the SSP audio system clock ACDS divider */
  	ret = snd_soc_dai_set_clkdiv(cpu_dai,
  			PXA_SSP_AUDIO_DIV_ACDS, acds);
  	if (ret < 0)
  		return ret;
  
  	/* set the SSP audio system clock SCDB divider4 */
  	ret = snd_soc_dai_set_clkdiv(cpu_dai,
  			PXA_SSP_AUDIO_DIV_SCDB, div4);
  	if (ret < 0)
  		return ret;
  
  	/* set SSP audio pll clock */
  	ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, acps);
  	if (ret < 0)
  		return ret;
  
  	return 0;
  }
  
  /*
   * Magician uses I2S for capture.
   */
  static int magician_capture_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_dai *codec_dai = rtd->codec_dai;
  	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
  	int ret = 0;
  
  	/* set codec DAI configuration */
  	ret = snd_soc_dai_set_fmt(codec_dai,
  			SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF |
  			SND_SOC_DAIFMT_CBS_CFS);
  	if (ret < 0)
  		return ret;
  
  	/* set cpu DAI configuration */
  	ret = snd_soc_dai_set_fmt(cpu_dai,
  			SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF |
  			SND_SOC_DAIFMT_CBS_CFS);
  	if (ret < 0)
  		return ret;
  
  	/* set the I2S system clock as output */
  	ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0,
  			SND_SOC_CLOCK_OUT);
  	if (ret < 0)
  		return ret;
  
  	return 0;
  }
  
  static struct snd_soc_ops magician_capture_ops = {
  	.startup = magician_startup,
  	.hw_params = magician_capture_hw_params,
  };
  
  static struct snd_soc_ops magician_playback_ops = {
  	.startup = magician_startup,
  	.hw_params = magician_playback_hw_params,
  };
  
  static int magician_get_hp(struct snd_kcontrol *kcontrol,
  			     struct snd_ctl_elem_value *ucontrol)
  {
  	ucontrol->value.integer.value[0] = magician_hp_switch;
  	return 0;
  }
  
  static int magician_set_hp(struct snd_kcontrol *kcontrol,
  			     struct snd_ctl_elem_value *ucontrol)
  {
  	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
  
  	if (magician_hp_switch == ucontrol->value.integer.value[0])
  		return 0;
  
  	magician_hp_switch = ucontrol->value.integer.value[0];
  	magician_ext_control(codec);
  	return 1;
  }
  
  static int magician_get_spk(struct snd_kcontrol *kcontrol,
  			    struct snd_ctl_elem_value *ucontrol)
  {
  	ucontrol->value.integer.value[0] = magician_spk_switch;
  	return 0;
  }
  
  static int magician_set_spk(struct snd_kcontrol *kcontrol,
  			    struct snd_ctl_elem_value *ucontrol)
  {
  	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
  
  	if (magician_spk_switch == ucontrol->value.integer.value[0])
  		return 0;
  
  	magician_spk_switch = ucontrol->value.integer.value[0];
  	magician_ext_control(codec);
  	return 1;
  }
  
  static int magician_get_input(struct snd_kcontrol *kcontrol,
  			      struct snd_ctl_elem_value *ucontrol)
  {
  	ucontrol->value.integer.value[0] = magician_in_sel;
  	return 0;
  }
  
  static int magician_set_input(struct snd_kcontrol *kcontrol,
  			      struct snd_ctl_elem_value *ucontrol)
  {
  	if (magician_in_sel == ucontrol->value.integer.value[0])
  		return 0;
  
  	magician_in_sel = ucontrol->value.integer.value[0];
  
  	switch (magician_in_sel) {
  	case MAGICIAN_MIC:
  		gpio_set_value(EGPIO_MAGICIAN_IN_SEL1, 1);
  		break;
  	case MAGICIAN_MIC_EXT:
  		gpio_set_value(EGPIO_MAGICIAN_IN_SEL1, 0);
  	}
  
  	return 1;
  }
  
  static int magician_spk_power(struct snd_soc_dapm_widget *w,
  				struct snd_kcontrol *k, int event)
  {
  	gpio_set_value(EGPIO_MAGICIAN_SPK_POWER, SND_SOC_DAPM_EVENT_ON(event));
  	return 0;
  }
  
  static int magician_hp_power(struct snd_soc_dapm_widget *w,
  				struct snd_kcontrol *k, int event)
  {
  	gpio_set_value(EGPIO_MAGICIAN_EP_POWER, SND_SOC_DAPM_EVENT_ON(event));
  	return 0;
  }
  
  static int magician_mic_bias(struct snd_soc_dapm_widget *w,
  				struct snd_kcontrol *k, int event)
  {
  	gpio_set_value(EGPIO_MAGICIAN_MIC_POWER, SND_SOC_DAPM_EVENT_ON(event));
  	return 0;
  }
  
  /* magician machine dapm widgets */
  static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = {
  	SND_SOC_DAPM_HP("Headphone Jack", magician_hp_power),
  	SND_SOC_DAPM_SPK("Speaker", magician_spk_power),
  	SND_SOC_DAPM_MIC("Call Mic", magician_mic_bias),
  	SND_SOC_DAPM_MIC("Headset Mic", magician_mic_bias),
  };
  
  /* magician machine audio_map */
  static const struct snd_soc_dapm_route audio_map[] = {
  
  	/* Headphone connected to VOUTL, VOUTR */
  	{"Headphone Jack", NULL, "VOUTL"},
  	{"Headphone Jack", NULL, "VOUTR"},
  
  	/* Speaker connected to VOUTL, VOUTR */
  	{"Speaker", NULL, "VOUTL"},
  	{"Speaker", NULL, "VOUTR"},
  
  	/* Mics are connected to VINM */
  	{"VINM", NULL, "Headset Mic"},
  	{"VINM", NULL, "Call Mic"},
  };
  
  static const char *input_select[] = {"Call Mic", "Headset Mic"};
  static const struct soc_enum magician_in_sel_enum =
  	SOC_ENUM_SINGLE_EXT(2, input_select);
  
  static const struct snd_kcontrol_new uda1380_magician_controls[] = {
  	SOC_SINGLE_BOOL_EXT("Headphone Switch",
  			(unsigned long)&magician_hp_switch,
  			magician_get_hp, magician_set_hp),
  	SOC_SINGLE_BOOL_EXT("Speaker Switch",
  			(unsigned long)&magician_spk_switch,
  			magician_get_spk, magician_set_spk),
  	SOC_ENUM_EXT("Input Select", magician_in_sel_enum,
  			magician_get_input, magician_set_input),
  };
  
  /*
   * Logic for a uda1380 as connected on a HTC Magician
   */
  static int magician_uda1380_init(struct snd_soc_pcm_runtime *rtd)
  {
  	struct snd_soc_codec *codec = rtd->codec;
  	struct snd_soc_dapm_context *dapm = &codec->dapm;
  	int err;
  
  	/* NC codec pins */
  	snd_soc_dapm_nc_pin(dapm, "VOUTLHP");
  	snd_soc_dapm_nc_pin(dapm, "VOUTRHP");
  
  	/* FIXME: is anything connected here? */
  	snd_soc_dapm_nc_pin(dapm, "VINL");
  	snd_soc_dapm_nc_pin(dapm, "VINR");
  
  	/* Add magician specific controls */
  	err = snd_soc_add_codec_controls(codec, uda1380_magician_controls,
  				ARRAY_SIZE(uda1380_magician_controls));
  	if (err < 0)
  		return err;
  
  	/* Add magician specific widgets */
  	snd_soc_dapm_new_controls(dapm, uda1380_dapm_widgets,
  				  ARRAY_SIZE(uda1380_dapm_widgets));
  
  	/* Set up magician specific audio path interconnects */
  	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
  
  	return 0;
  }
  
  /* magician digital audio interface glue - connects codec <--> CPU */
  static struct snd_soc_dai_link magician_dai[] = {
  {
  	.name = "uda1380",
  	.stream_name = "UDA1380 Playback",
  	.cpu_dai_name = "pxa-ssp-dai.0",
  	.codec_dai_name = "uda1380-hifi-playback",
  	.platform_name = "pxa-pcm-audio",
  	.codec_name = "uda1380-codec.0-0018",
  	.init = magician_uda1380_init,
  	.ops = &magician_playback_ops,
  },
  {
  	.name = "uda1380",
  	.stream_name = "UDA1380 Capture",
  	.cpu_dai_name = "pxa2xx-i2s",
  	.codec_dai_name = "uda1380-hifi-capture",
  	.platform_name = "pxa-pcm-audio",
  	.codec_name = "uda1380-codec.0-0018",
  	.ops = &magician_capture_ops,
  }
  };
  
  /* magician audio machine driver */
  static struct snd_soc_card snd_soc_card_magician = {
  	.name = "Magician",
  	.owner = THIS_MODULE,
  	.dai_link = magician_dai,
  	.num_links = ARRAY_SIZE(magician_dai),
  
  };
  
  static struct platform_device *magician_snd_device;
  
  /*
   * FIXME: move into magician board file once merged into the pxa tree
   */
  static struct uda1380_platform_data uda1380_info = {
  	.gpio_power = EGPIO_MAGICIAN_CODEC_POWER,
  	.gpio_reset = EGPIO_MAGICIAN_CODEC_RESET,
  	.dac_clk    = UDA1380_DAC_CLK_WSPLL,
  };
  
  static struct i2c_board_info i2c_board_info[] = {
  	{
  		I2C_BOARD_INFO("uda1380", 0x18),
  		.platform_data = &uda1380_info,
  	},
  };
  
  static int __init magician_init(void)
  {
  	int ret;
  	struct i2c_adapter *adapter;
  	struct i2c_client *client;
  
  	if (!machine_is_magician())
  		return -ENODEV;
  
  	adapter = i2c_get_adapter(0);
  	if (!adapter)
  		return -ENODEV;
  	client = i2c_new_device(adapter, i2c_board_info);
  	i2c_put_adapter(adapter);
  	if (!client)
  		return -ENODEV;
  
  	ret = gpio_request(EGPIO_MAGICIAN_SPK_POWER, "SPK_POWER");
  	if (ret)
  		goto err_request_spk;
  	ret = gpio_request(EGPIO_MAGICIAN_EP_POWER, "EP_POWER");
  	if (ret)
  		goto err_request_ep;
  	ret = gpio_request(EGPIO_MAGICIAN_MIC_POWER, "MIC_POWER");
  	if (ret)
  		goto err_request_mic;
  	ret = gpio_request(EGPIO_MAGICIAN_IN_SEL0, "IN_SEL0");
  	if (ret)
  		goto err_request_in_sel0;
  	ret = gpio_request(EGPIO_MAGICIAN_IN_SEL1, "IN_SEL1");
  	if (ret)
  		goto err_request_in_sel1;
  
  	gpio_set_value(EGPIO_MAGICIAN_IN_SEL0, 0);
  
  	magician_snd_device = platform_device_alloc("soc-audio", -1);
  	if (!magician_snd_device) {
  		ret = -ENOMEM;
  		goto err_pdev;
  	}
  
  	platform_set_drvdata(magician_snd_device, &snd_soc_card_magician);
  	ret = platform_device_add(magician_snd_device);
  	if (ret) {
  		platform_device_put(magician_snd_device);
  		goto err_pdev;
  	}
  
  	return 0;
  
  err_pdev:
  	gpio_free(EGPIO_MAGICIAN_IN_SEL1);
  err_request_in_sel1:
  	gpio_free(EGPIO_MAGICIAN_IN_SEL0);
  err_request_in_sel0:
  	gpio_free(EGPIO_MAGICIAN_MIC_POWER);
  err_request_mic:
  	gpio_free(EGPIO_MAGICIAN_EP_POWER);
  err_request_ep:
  	gpio_free(EGPIO_MAGICIAN_SPK_POWER);
  err_request_spk:
  	return ret;
  }
  
  static void __exit magician_exit(void)
  {
  	platform_device_unregister(magician_snd_device);
  
  	gpio_set_value(EGPIO_MAGICIAN_SPK_POWER, 0);
  	gpio_set_value(EGPIO_MAGICIAN_EP_POWER, 0);
  	gpio_set_value(EGPIO_MAGICIAN_MIC_POWER, 0);
  
  	gpio_free(EGPIO_MAGICIAN_IN_SEL1);
  	gpio_free(EGPIO_MAGICIAN_IN_SEL0);
  	gpio_free(EGPIO_MAGICIAN_MIC_POWER);
  	gpio_free(EGPIO_MAGICIAN_EP_POWER);
  	gpio_free(EGPIO_MAGICIAN_SPK_POWER);
  }
  
  module_init(magician_init);
  module_exit(magician_exit);
  
  MODULE_AUTHOR("Philipp Zabel");
  MODULE_DESCRIPTION("ALSA SoC Magician");
  MODULE_LICENSE("GPL");