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magician.c

/*
 * 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/soc-dapm.h>
#include <sound/uda1380.h>

#include <mach/magician.h>
#include <asm/mach-types.h>
#include "../codecs/uda1380.h"
#include "pxa2xx-pcm.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)
{
      if (magician_spk_switch)
            snd_soc_dapm_enable_pin(codec, "Speaker");
      else
            snd_soc_dapm_disable_pin(codec, "Speaker");
      if (magician_hp_switch)
            snd_soc_dapm_enable_pin(codec, "Headphone Jack");
      else
            snd_soc_dapm_disable_pin(codec, "Headphone Jack");

      switch (magician_in_sel) {
      case MAGICIAN_MIC:
            snd_soc_dapm_disable_pin(codec, "Headset Mic");
            snd_soc_dapm_enable_pin(codec, "Call Mic");
            break;
      case MAGICIAN_MIC_EXT:
            snd_soc_dapm_disable_pin(codec, "Call Mic");
            snd_soc_dapm_enable_pin(codec, "Headset Mic");
            break;
      }

      snd_soc_dapm_sync(codec);
}

static int magician_startup(struct snd_pcm_substream *substream)
{
      struct snd_soc_pcm_runtime *rtd = substream->private_data;
      struct snd_soc_codec *codec = rtd->socdev->card->codec;

      /* check the jack status at stream startup */
      magician_ext_control(codec);

      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->dai->codec_dai;
      struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
      unsigned int acps, acds, width, rate;
      unsigned int div4 = PXA_SSP_CLK_SCDB_4;
      int ret = 0;

      rate = params_rate(params);
      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, 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->dai->codec_dai;
      struct snd_soc_dai *cpu_dai = rtd->dai->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_codec *codec)
{
      int err;

      /* NC codec pins */
      snd_soc_dapm_nc_pin(codec, "VOUTLHP");
      snd_soc_dapm_nc_pin(codec, "VOUTRHP");

      /* FIXME: is anything connected here? */
      snd_soc_dapm_nc_pin(codec, "VINL");
      snd_soc_dapm_nc_pin(codec, "VINR");

      /* Add magician specific controls */
      err = snd_soc_add_controls(codec, uda1380_magician_controls,
                        ARRAY_SIZE(uda1380_magician_controls));
      if (err < 0)
            return err;

      /* Add magician specific widgets */
      snd_soc_dapm_new_controls(codec, uda1380_dapm_widgets,
                          ARRAY_SIZE(uda1380_dapm_widgets));

      /* Set up magician specific audio path interconnects */
      snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));

      snd_soc_dapm_sync(codec);
      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 = &pxa_ssp_dai[PXA_DAI_SSP1],
      .codec_dai = &uda1380_dai[UDA1380_DAI_PLAYBACK],
      .init = magician_uda1380_init,
      .ops = &magician_playback_ops,
},
{
      .name = "uda1380",
      .stream_name = "UDA1380 Capture",
      .cpu_dai = &pxa_i2s_dai,
      .codec_dai = &uda1380_dai[UDA1380_DAI_CAPTURE],
      .ops = &magician_capture_ops,
}
};

/* magician audio machine driver */
static struct snd_soc_card snd_soc_card_magician = {
      .name = "Magician",
      .dai_link = magician_dai,
      .num_links = ARRAY_SIZE(magician_dai),
      .platform = &pxa2xx_soc_platform,
};

/* magician audio subsystem */
static struct snd_soc_device magician_snd_devdata = {
      .card = &snd_soc_card_magician,
      .codec_dev = &soc_codec_dev_uda1380,
};

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, &magician_snd_devdata);
      magician_snd_devdata.dev = &magician_snd_device->dev;
      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");

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