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wm1133-ev1.c

/*
 *  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 <sound/core.h>
#include <sound/jack.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>

#include <mach/audmux.h>

#include "imx-ssi.h"
#include "../codecs/wm8350.h"

/* There is a silicon mic on the board optionally connected via a solder pad
 * SP1.  Define this to enable it.
 */
#undef USE_SIMIC

00033 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);
      u32 dai_format;

      /* 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);

      dai_format = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
            SND_SOC_DAIFMT_CBM_CFM;

      /* set codec DAI configuration */
      snd_soc_dai_set_fmt(codec_dai, dai_format);

      /* set cpu DAI configuration */
      snd_soc_dai_set_fmt(cpu_dai, dai_format);

      /* TODO: The SSI driver should figure this out for us */
      switch (channels) {
      case 2:
            snd_soc_dai_set_tdm_slot(cpu_dai, 0xffffffc, 0xffffffc, 2, 0);
            break;
      case 1:
            snd_soc_dai_set_tdm_slot(cpu_dai, 0xffffffe, 0xffffffe, 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 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;
      struct snd_soc_dapm_context *dapm = &codec->dapm;

      snd_soc_dapm_new_controls(dapm, wm1133_ev1_widgets,
                          ARRAY_SIZE(wm1133_ev1_widgets));

      snd_soc_dapm_add_routes(dapm, wm1133_ev1_map,
                        ARRAY_SIZE(wm1133_ev1_map));

      /* Headphone jack detection */
      snd_soc_jack_new(codec, "Headphone", SND_JACK_HEADPHONE, &hp_jack);
      snd_soc_jack_add_pins(&hp_jack, ARRAY_SIZE(hp_jack_pins),
                        hp_jack_pins);
      wm8350_hp_jack_detect(codec, WM8350_JDR, &hp_jack, SND_JACK_HEADPHONE);

      /* Microphone jack detection */
      snd_soc_jack_new(codec, "Microphone",
                   SND_JACK_MICROPHONE | SND_JACK_BTN_0, &mic_jack);
      snd_soc_jack_add_pins(&mic_jack, ARRAY_SIZE(mic_jack_pins),
                        mic_jack_pins);
      wm8350_mic_jack_detect(codec, &mic_jack, SND_JACK_MICROPHONE,
                         SND_JACK_BTN_0);

      snd_soc_dapm_force_enable_pin(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-fiq-pcm-audio.0",
      .codec_name = "wm8350-codec.0-0x1a",
      .init = wm1133_ev1_init,
      .ops = &wm1133_ev1_ops,
      .symmetric_rates = 1,
};

static struct snd_soc_card wm1133_ev1 = {
      .name = "WM1133-EV1",
      .dai_link = &wm1133_ev1_dai,
      .num_links = 1,
};

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 = MXC_AUDMUX_V2_PTCR_SYN |
            MXC_AUDMUX_V2_PTCR_TFSDIR |
            MXC_AUDMUX_V2_PTCR_TFSEL(MX31_AUDMUX_PORT5_SSI_PINS_5) |
            MXC_AUDMUX_V2_PTCR_TCLKDIR |
            MXC_AUDMUX_V2_PTCR_TCSEL(MX31_AUDMUX_PORT5_SSI_PINS_5);
      pdcr = MXC_AUDMUX_V2_PDCR_RXDSEL(MX31_AUDMUX_PORT5_SSI_PINS_5);
      mxc_audmux_v2_configure_port(MX31_AUDMUX_PORT1_SSI0, ptcr, pdcr);

      ptcr = MXC_AUDMUX_V2_PTCR_SYN;
      pdcr = MXC_AUDMUX_V2_PDCR_RXDSEL(MX31_AUDMUX_PORT1_SSI0);
      mxc_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");

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