text: /*
   *  wm1133-ev1.c - Audio for WM1133-EV1 on i.MX31ADS
   *
   *  Copyright (c) 2010 Wolfson Microelectronics plc
   *  Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
   *
   *  Based on an earlier driver for the same hardware by Liam Girdwood.
   *
   *  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/platform_device.h>
  #include <linux/clk.h>
  #include <linux/module.h>
  #include <sound/core.h>
  #include <sound/jack.h>
  #include <sound/pcm.h>
  #include <sound/pcm_params.h>
  #include <sound/soc.h>
  
  #include "imx-ssi.h"
  #include "../codecs/wm8350.h"
  #include "imx-audmux.h"
  
  /* There is a silicon mic on the board optionally connected via a solder pad
   * SP1.  Define this to enable it.
   */
  #undef USE_SIMIC
  
  struct _wm8350_audio {
  	unsigned int channels;
  	snd_pcm_format_t format;
  	unsigned int rate;
  	unsigned int sysclk;
  	unsigned int bclkdiv;
  	unsigned int clkdiv;
  	unsigned int lr_rate;
  };
  
  /* in order of power consumption per rate (lowest first) */
  static const struct _wm8350_audio wm8350_audio[] = {
  	/* 16bit mono modes */
  	{1, SNDRV_PCM_FORMAT_S16_LE, 8000, 12288000 >> 1,
  	 WM8350_BCLK_DIV_48, WM8350_DACDIV_3, 16,},
  
  	/* 16 bit stereo modes */
  	{2, SNDRV_PCM_FORMAT_S16_LE, 8000, 12288000,
  	 WM8350_BCLK_DIV_48, WM8350_DACDIV_6, 32,},
  	{2, SNDRV_PCM_FORMAT_S16_LE, 16000, 12288000,
  	 WM8350_BCLK_DIV_24, WM8350_DACDIV_3, 32,},
  	{2, SNDRV_PCM_FORMAT_S16_LE, 32000, 12288000,
  	 WM8350_BCLK_DIV_12, WM8350_DACDIV_1_5, 32,},
  	{2, SNDRV_PCM_FORMAT_S16_LE, 48000, 12288000,
  	 WM8350_BCLK_DIV_8, WM8350_DACDIV_1, 32,},
  	{2, SNDRV_PCM_FORMAT_S16_LE, 96000, 24576000,
  	 WM8350_BCLK_DIV_8, WM8350_DACDIV_1, 32,},
  	{2, SNDRV_PCM_FORMAT_S16_LE, 11025, 11289600,
  	 WM8350_BCLK_DIV_32, WM8350_DACDIV_4, 32,},
  	{2, SNDRV_PCM_FORMAT_S16_LE, 22050, 11289600,
  	 WM8350_BCLK_DIV_16, WM8350_DACDIV_2, 32,},
  	{2, SNDRV_PCM_FORMAT_S16_LE, 44100, 11289600,
  	 WM8350_BCLK_DIV_8, WM8350_DACDIV_1, 32,},
  	{2, SNDRV_PCM_FORMAT_S16_LE, 88200, 22579200,
  	 WM8350_BCLK_DIV_8, WM8350_DACDIV_1, 32,},
  
  	/* 24bit stereo modes */
  	{2, SNDRV_PCM_FORMAT_S24_LE, 48000, 12288000,
  	 WM8350_BCLK_DIV_4, WM8350_DACDIV_1, 64,},
  	{2, SNDRV_PCM_FORMAT_S24_LE, 96000, 24576000,
  	 WM8350_BCLK_DIV_4, WM8350_DACDIV_1, 64,},
  	{2, SNDRV_PCM_FORMAT_S24_LE, 44100, 11289600,
  	 WM8350_BCLK_DIV_4, WM8350_DACDIV_1, 64,},
  	{2, SNDRV_PCM_FORMAT_S24_LE, 88200, 22579200,
  	 WM8350_BCLK_DIV_4, WM8350_DACDIV_1, 64,},
  };
  
  static int wm1133_ev1_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 i, found = 0;
  	snd_pcm_format_t format = params_format(params);
  	unsigned int rate = params_rate(params);
  	unsigned int channels = params_channels(params);
  
  	/* find the correct audio parameters */
  	for (i = 0; i < ARRAY_SIZE(wm8350_audio); i++) {
  		if (rate == wm8350_audio[i].rate &&
  		    format == wm8350_audio[i].format &&
  		    channels == wm8350_audio[i].channels) {
  			found = 1;
  			break;
  		}
  	}
  	if (!found)
  		return -EINVAL;
  
  	/* codec FLL input is 14.75 MHz from MCLK */
  	snd_soc_dai_set_pll(codec_dai, 0, 0, 14750000, wm8350_audio[i].sysclk);
  
  	/* TODO: The SSI driver should figure this out for us */
  	switch (channels) {
  	case 2:
  		snd_soc_dai_set_tdm_slot(cpu_dai, 0x3, 0x3, 2, 0);
  		break;
  	case 1:
  		snd_soc_dai_set_tdm_slot(cpu_dai, 0x1, 0x1, 1, 0);
  		break;
  	default:
  		return -EINVAL;
  	}
  
  	/* set MCLK as the codec system clock for DAC and ADC */
  	snd_soc_dai_set_sysclk(codec_dai, WM8350_MCLK_SEL_PLL_MCLK,
  			       wm8350_audio[i].sysclk, SND_SOC_CLOCK_IN);
  
  	/* set codec BCLK division for sample rate */
  	snd_soc_dai_set_clkdiv(codec_dai, WM8350_BCLK_CLKDIV,
  			       wm8350_audio[i].bclkdiv);
  
  	/* DAI is synchronous and clocked with DAC LRCLK & ADC LRC */
  	snd_soc_dai_set_clkdiv(codec_dai,
  			       WM8350_DACLR_CLKDIV, wm8350_audio[i].lr_rate);
  	snd_soc_dai_set_clkdiv(codec_dai,
  			       WM8350_ADCLR_CLKDIV, wm8350_audio[i].lr_rate);
  
  	/* now configure DAC and ADC clocks */
  	snd_soc_dai_set_clkdiv(codec_dai,
  			       WM8350_DAC_CLKDIV, wm8350_audio[i].clkdiv);
  
  	snd_soc_dai_set_clkdiv(codec_dai,
  			       WM8350_ADC_CLKDIV, wm8350_audio[i].clkdiv);
  
  	return 0;
  }
  
  static const struct snd_soc_ops wm1133_ev1_ops = {
  	.hw_params = wm1133_ev1_hw_params,
  };
  
  static const struct snd_soc_dapm_widget wm1133_ev1_widgets[] = {
  #ifdef USE_SIMIC
  	SND_SOC_DAPM_MIC("SiMIC", NULL),
  #endif
  	SND_SOC_DAPM_MIC("Mic1 Jack", NULL),
  	SND_SOC_DAPM_MIC("Mic2 Jack", NULL),
  	SND_SOC_DAPM_LINE("Line In Jack", NULL),
  	SND_SOC_DAPM_LINE("Line Out Jack", NULL),
  	SND_SOC_DAPM_HP("Headphone Jack", NULL),
  };
  
  /* imx32ads soc_card audio map */
  static const struct snd_soc_dapm_route wm1133_ev1_map[] = {
  
  #ifdef USE_SIMIC
  	/* SiMIC --> IN1LN (with automatic bias) via SP1 */
  	{ "IN1LN", NULL, "Mic Bias" },
  	{ "Mic Bias", NULL, "SiMIC" },
  #endif
  
  	/* Mic 1 Jack --> IN1LN and IN1LP (with automatic bias) */
  	{ "IN1LN", NULL, "Mic Bias" },
  	{ "IN1LP", NULL, "Mic1 Jack" },
  	{ "Mic Bias", NULL, "Mic1 Jack" },
  
  	/* Mic 2 Jack --> IN1RN and IN1RP (with automatic bias) */
  	{ "IN1RN", NULL, "Mic Bias" },
  	{ "IN1RP", NULL, "Mic2 Jack" },
  	{ "Mic Bias", NULL, "Mic2 Jack" },
  
  	/* Line in Jack --> AUX (L+R) */
  	{ "IN3R", NULL, "Line In Jack" },
  	{ "IN3L", NULL, "Line In Jack" },
  
  	/* Out1 --> Headphone Jack */
  	{ "Headphone Jack", NULL, "OUT1R" },
  	{ "Headphone Jack", NULL, "OUT1L" },
  
  	/* Out1 --> Line Out Jack */
  	{ "Line Out Jack", NULL, "OUT2R" },
  	{ "Line Out Jack", NULL, "OUT2L" },
  };
  
  static struct snd_soc_jack hp_jack;
  
  static struct snd_soc_jack_pin hp_jack_pins[] = {
  	{ .pin = "Headphone Jack", .mask = SND_JACK_HEADPHONE },
  };
  
  static struct snd_soc_jack mic_jack;
  
  static struct snd_soc_jack_pin mic_jack_pins[] = {
  	{ .pin = "Mic1 Jack", .mask = SND_JACK_MICROPHONE },
  	{ .pin = "Mic2 Jack", .mask = SND_JACK_MICROPHONE },
  };
  
  static int wm1133_ev1_init(struct snd_soc_pcm_runtime *rtd)
  {
  	struct snd_soc_codec *codec = rtd->codec;
  
  	/* Headphone jack detection */
  	snd_soc_card_jack_new(rtd->card, "Headphone", SND_JACK_HEADPHONE,
  			      &hp_jack, hp_jack_pins, ARRAY_SIZE(hp_jack_pins));
  	wm8350_hp_jack_detect(codec, WM8350_JDR, &hp_jack, SND_JACK_HEADPHONE);
  
  	/* Microphone jack detection */
  	snd_soc_card_jack_new(rtd->card, "Microphone",
  			      SND_JACK_MICROPHONE | SND_JACK_BTN_0, &mic_jack,
  			      mic_jack_pins, ARRAY_SIZE(mic_jack_pins));
  	wm8350_mic_jack_detect(codec, &mic_jack, SND_JACK_MICROPHONE,
  			       SND_JACK_BTN_0);
  
  	snd_soc_dapm_force_enable_pin(&rtd->card->dapm, "Mic Bias");
  
  	return 0;
  }
  
  
  static struct snd_soc_dai_link wm1133_ev1_dai = {
  	.name = "WM1133-EV1",
  	.stream_name = "Audio",
  	.cpu_dai_name = "imx-ssi.0",
  	.codec_dai_name = "wm8350-hifi",
  	.platform_name = "imx-ssi.0",
  	.codec_name = "wm8350-codec.0-0x1a",
  	.init = wm1133_ev1_init,
  	.ops = &wm1133_ev1_ops,
  	.symmetric_rates = 1,
  	.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
  		   SND_SOC_DAIFMT_CBM_CFM,
  };
  
  static struct snd_soc_card wm1133_ev1 = {
  	.name = "WM1133-EV1",
  	.owner = THIS_MODULE,
  	.dai_link = &wm1133_ev1_dai,
  	.num_links = 1,
  
  	.dapm_widgets = wm1133_ev1_widgets,
  	.num_dapm_widgets = ARRAY_SIZE(wm1133_ev1_widgets),
  	.dapm_routes = wm1133_ev1_map,
  	.num_dapm_routes = ARRAY_SIZE(wm1133_ev1_map),
  };
  
  static struct platform_device *wm1133_ev1_snd_device;
  
  static int __init wm1133_ev1_audio_init(void)
  {
  	int ret;
  	unsigned int ptcr, pdcr;
  
  	/* SSI0 mastered by port 5 */
  	ptcr = IMX_AUDMUX_V2_PTCR_SYN |
  		IMX_AUDMUX_V2_PTCR_TFSDIR |
  		IMX_AUDMUX_V2_PTCR_TFSEL(MX31_AUDMUX_PORT5_SSI_PINS_5) |
  		IMX_AUDMUX_V2_PTCR_TCLKDIR |
  		IMX_AUDMUX_V2_PTCR_TCSEL(MX31_AUDMUX_PORT5_SSI_PINS_5);
  	pdcr = IMX_AUDMUX_V2_PDCR_RXDSEL(MX31_AUDMUX_PORT5_SSI_PINS_5);
  	imx_audmux_v2_configure_port(MX31_AUDMUX_PORT1_SSI0, ptcr, pdcr);
  
  	ptcr = IMX_AUDMUX_V2_PTCR_SYN;
  	pdcr = IMX_AUDMUX_V2_PDCR_RXDSEL(MX31_AUDMUX_PORT1_SSI0);
  	imx_audmux_v2_configure_port(MX31_AUDMUX_PORT5_SSI_PINS_5, ptcr, pdcr);
  
  	wm1133_ev1_snd_device = platform_device_alloc("soc-audio", -1);
  	if (!wm1133_ev1_snd_device)
  		return -ENOMEM;
  
  	platform_set_drvdata(wm1133_ev1_snd_device, &wm1133_ev1);
  	ret = platform_device_add(wm1133_ev1_snd_device);
  
  	if (ret)
  		platform_device_put(wm1133_ev1_snd_device);
  
  	return ret;
  }
  module_init(wm1133_ev1_audio_init);
  
  static void __exit wm1133_ev1_audio_exit(void)
  {
  	platform_device_unregister(wm1133_ev1_snd_device);
  }
  module_exit(wm1133_ev1_audio_exit);
  
  MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
  MODULE_DESCRIPTION("Audio for WM1133-EV1 on i.MX31ADS");
  MODULE_LICENSE("GPL");
  