wm1133-ev1.c

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/*
 *  wm1133-ev1.c - Audio for WM1133-EV1 on i.MX31ADS
 *
 *  Copyright (c) 2010 Wolfson Microelectronics plc
 *  Author: Mark Brown <[email protected]>
 *
 *  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);
    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",
    .owner = THIS_MODULE,
    .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 = 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 <[email protected]>");
MODULE_DESCRIPTION("Audio for WM1133-EV1 on i.MX31ADS");
MODULE_LICENSE("GPL");