magician.c

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/*
 * SoC audio for HTC Magician
 *
 * Copyright (c) 2006 Philipp Zabel <[email protected]>
 *
 * based on spitz.c,
 * Authors: Liam Girdwood <[email protected]>
 *          Richard Purdie <[email protected]>
 *
 *  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");