ASoC: Add Openmoko Neo FreeRunner (GTA02) audio driver

This driver supports the audio subsystem on the Openmoko Neo FreeRunner
smartphone, often known by its codename GTA02.  The system has a WM8753
connected to a Samsung S3C2442 with an external GPIO controlled speaker
amplifier.

The driver was originally written by Graeme Gregory and has recieved
contributions from Openmoko, myself and members of the Openmoko
community.  For much of this time the primary Openmoko kernel maintainer
was Andy Green.

Signed-off-by: Graeme Gregory <graeme@openmoko.com>
Signed-off-by: Andy Green <andy@openmoko.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

3 files changed, 492 insertions, 0 deletions

diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig
index df494d1e346f..d1ed0f537c3a 100644
--- a/sound/soc/s3c24xx/Kconfig
+++ b/sound/soc/s3c24xx/Kconfig
@@ -38,6 +38,15 @@ config SND_S3C24XX_SOC_NEO1973_WM8753
           Say Y if you want to add support for SoC audio on smdk2440
           with the WM8753.
 
+config SND_S3C24XX_SOC_NEO1973_GTA02_WM8753
+        tristate "Audio support for the Openmoko Neo FreeRunner (GTA02)"
+        depends on SND_S3C24XX_SOC && MACH_NEO1973_GTA02
+        select SND_S3C24XX_SOC_I2S
+        select SND_SOC_WM8753
+        help
+          This driver provides audio support for the Openmoko Neo FreeRunner
+          smartphone.
+          
 config SND_S3C24XX_SOC_JIVE_WM8750
         tristate "SoC I2S Audio support for Jive"
         depends on SND_S3C24XX_SOC && MACH_JIVE
diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile
index 07a93a2ebe5f..eb219b016499 100644
--- a/sound/soc/s3c24xx/Makefile
+++ b/sound/soc/s3c24xx/Makefile
@@ -16,12 +16,14 @@ obj-$(CONFIG_SND_S3C_I2SV2_SOC) += snd-soc-s3c-i2s-v2.o
 # S3C24XX Machine Support
 snd-soc-jive-wm8750-objs := jive_wm8750.o
 snd-soc-neo1973-wm8753-objs := neo1973_wm8753.o
+snd-soc-neo1973-gta02-wm8753-objs := neo1973_gta02_wm8753.o
 snd-soc-smdk2443-wm9710-objs := smdk2443_wm9710.o
 snd-soc-ln2440sbc-alc650-objs := ln2440sbc_alc650.o
 snd-soc-s3c24xx-uda134x-objs := s3c24xx_uda134x.o
 
 obj-$(CONFIG_SND_S3C24XX_SOC_JIVE_WM8750) += snd-soc-jive-wm8750.o
 obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o
+obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_GTA02_WM8753) += snd-soc-neo1973-gta02-wm8753.o
 obj-$(CONFIG_SND_S3C24XX_SOC_SMDK2443_WM9710) += snd-soc-smdk2443-wm9710.o
 obj-$(CONFIG_SND_S3C24XX_SOC_LN2440SBC_ALC650) += snd-soc-ln2440sbc-alc650.o
 obj-$(CONFIG_SND_S3C24XX_SOC_S3C24XX_UDA134X) += snd-soc-s3c24xx-uda134x.o
diff --git a/sound/soc/s3c24xx/neo1973_gta02_wm8753.c b/sound/soc/s3c24xx/neo1973_gta02_wm8753.c
new file mode 100644
index 000000000000..1358f6fe8201
--- /dev/null
+++ b/sound/soc/s3c24xx/neo1973_gta02_wm8753.c
@@ -0,0 +1,481 @@
+/*
+ * neo1973_gta02_wm8753.c  --  SoC audio for Neo1973
+ *
+ * Copyright 2007 Openmoko Inc
+ * Author: Graeme Gregory <graeme@openmoko.org>
+ * Copyright 2007 Wolfson Microelectronics PLC.
+ * Author: Graeme Gregory <linux@wolfsonmicro.com>
+ * Copyright 2009 Wolfson Microelectronics
+ *
+ *  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/moduleparam.h>
+#include <linux/timer.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+
+#include <plat/regs-iis.h>
+
+#include <mach/regs-clock.h>
+#include <mach/regs-gpio.h>
+#include <mach/hardware.h>
+#include <asm/io.h>
+#include <mach/regs-gpioj.h>
+#include <mach/gta02.h>
+#include "../codecs/wm8753.h"
+#include "s3c24xx-pcm.h"
+#include "s3c24xx-i2s.h"
+
+static struct snd_soc_card neo1973_gta02;
+
+static int neo1973_gta02_hifi_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 pll_out = 0, bclk = 0;
+        int ret = 0;
+        unsigned long iis_clkrate;
+
+        iis_clkrate = s3c24xx_i2s_get_clockrate();
+
+        switch (params_rate(params)) {
+        case 8000:
+        case 16000:
+                pll_out = 12288000;
+                break;
+        case 48000:
+                bclk = WM8753_BCLK_DIV_4;
+                pll_out = 12288000;
+                break;
+        case 96000:
+                bclk = WM8753_BCLK_DIV_2;
+                pll_out = 12288000;
+                break;
+        case 11025:
+                bclk = WM8753_BCLK_DIV_16;
+                pll_out = 11289600;
+                break;
+        case 22050:
+                bclk = WM8753_BCLK_DIV_8;
+                pll_out = 11289600;
+                break;
+        case 44100:
+                bclk = WM8753_BCLK_DIV_4;
+                pll_out = 11289600;
+                break;
+        case 88200:
+                bclk = WM8753_BCLK_DIV_2;
+                pll_out = 11289600;
+                break;
+        }
+
+        /* set codec DAI configuration */
+        ret = snd_soc_dai_set_fmt(codec_dai,
+                SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+                SND_SOC_DAIFMT_CBM_CFM);
+        if (ret < 0)
+                return ret;
+
+        /* set cpu DAI configuration */
+        ret = snd_soc_dai_set_fmt(cpu_dai,
+                SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+                SND_SOC_DAIFMT_CBM_CFM);
+        if (ret < 0)
+                return ret;
+
+        /* set the codec system clock for DAC and ADC */
+        ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_MCLK, pll_out,
+                SND_SOC_CLOCK_IN);
+        if (ret < 0)
+                return ret;
+
+        /* set MCLK division for sample rate */
+        ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK,
+                S3C2410_IISMOD_32FS);
+        if (ret < 0)
+                return ret;
+
+        /* set codec BCLK division for sample rate */
+        ret = snd_soc_dai_set_clkdiv(codec_dai,
+                                        WM8753_BCLKDIV, bclk);
+        if (ret < 0)
+                return ret;
+
+        /* set prescaler division for sample rate */
+        ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER,
+                S3C24XX_PRESCALE(4, 4));
+        if (ret < 0)
+                return ret;
+
+        /* codec PLL input is PCLK/4 */
+        ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL1,
+                iis_clkrate / 4, pll_out);
+        if (ret < 0)
+                return ret;
+
+        return 0;
+}
+
+static int neo1973_gta02_hifi_hw_free(struct snd_pcm_substream *substream)
+{
+        struct snd_soc_pcm_runtime *rtd = substream->private_data;
+        struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+
+        /* disable the PLL */
+        return snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, 0);
+}
+
+/*
+ * Neo1973 WM8753 HiFi DAI opserations.
+ */
+static struct snd_soc_ops neo1973_gta02_hifi_ops = {
+        .hw_params = neo1973_gta02_hifi_hw_params,
+        .hw_free = neo1973_gta02_hifi_hw_free,
+};
+
+static int neo1973_gta02_voice_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;
+        unsigned int pcmdiv = 0;
+        int ret = 0;
+        unsigned long iis_clkrate;
+
+        iis_clkrate = s3c24xx_i2s_get_clockrate();
+
+        if (params_rate(params) != 8000)
+                return -EINVAL;
+        if (params_channels(params) != 1)
+                return -EINVAL;
+
+        pcmdiv = WM8753_PCM_DIV_6; /* 2.048 MHz */
+
+        /* todo: gg check mode (DSP_B) against CSR datasheet */
+        /* set codec DAI configuration */
+        ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_B |
+                SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+        if (ret < 0)
+                return ret;
+
+        /* set the codec system clock for DAC and ADC */
+        ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_PCMCLK,
+                12288000, SND_SOC_CLOCK_IN);
+        if (ret < 0)
+                return ret;
+
+        /* set codec PCM division for sample rate */
+        ret = snd_soc_dai_set_clkdiv(codec_dai, WM8753_PCMDIV,
+                                        pcmdiv);
+        if (ret < 0)
+                return ret;
+
+        /* configue and enable PLL for 12.288MHz output */
+        ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL2,
+                iis_clkrate / 4, 12288000);
+        if (ret < 0)
+                return ret;
+
+        return 0;
+}
+
+static int neo1973_gta02_voice_hw_free(struct snd_pcm_substream *substream)
+{
+        struct snd_soc_pcm_runtime *rtd = substream->private_data;
+        struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+
+        /* disable the PLL */
+        return snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, 0);
+}
+
+static struct snd_soc_ops neo1973_gta02_voice_ops = {
+        .hw_params = neo1973_gta02_voice_hw_params,
+        .hw_free = neo1973_gta02_voice_hw_free,
+};
+
+#define LM4853_AMP 1
+#define LM4853_SPK 2
+
+static u8 lm4853_state;
+
+/* This has no effect, it exists only to maintain compatibility with
+ * existing ALSA state files.
+ */
+static int lm4853_set_state(struct snd_kcontrol *kcontrol,
+        struct snd_ctl_elem_value *ucontrol)
+{
+        int val = ucontrol->value.integer.value[0];
+
+        if (val)
+                lm4853_state |= LM4853_AMP;
+        else
+                lm4853_state &= ~LM4853_AMP;
+
+        return 0;
+}
+
+static int lm4853_get_state(struct snd_kcontrol *kcontrol,
+        struct snd_ctl_elem_value *ucontrol)
+{
+        ucontrol->value.integer.value[0] = lm4853_state & LM4853_AMP;
+
+        return 0;
+}
+
+static int lm4853_set_spk(struct snd_kcontrol *kcontrol,
+        struct snd_ctl_elem_value *ucontrol)
+{
+        int val = ucontrol->value.integer.value[0];
+
+        if (val) {
+                lm4853_state |= LM4853_SPK;
+                s3c2410_gpio_setpin(GTA02_GPIO_HP_IN, 0);
+        } else {
+                lm4853_state &= ~LM4853_SPK;
+                s3c2410_gpio_setpin(GTA02_GPIO_HP_IN, 1);
+        }
+
+        return 0;
+}
+
+static int lm4853_get_spk(struct snd_kcontrol *kcontrol,
+        struct snd_ctl_elem_value *ucontrol)
+{
+        ucontrol->value.integer.value[0] = (lm4853_state & LM4853_SPK) >> 1;
+
+        return 0;
+}
+
+static int lm4853_event(struct snd_soc_dapm_widget *w,
+                        struct snd_kcontrol *k,
+                        int event)
+{
+        if (SND_SOC_DAPM_EVENT_ON(event))
+                s3c2410_gpio_setpin(GTA02_GPIO_AMP_SHUT, 0);
+
+        if (SND_SOC_DAPM_EVENT_OFF(event))
+                s3c2410_gpio_setpin(GTA02_GPIO_AMP_SHUT, 1);
+
+        return 0;
+}
+
+static const struct snd_soc_dapm_widget wm8753_dapm_widgets[] = {
+        SND_SOC_DAPM_SPK("Stereo Out", lm4853_event),
+        SND_SOC_DAPM_LINE("GSM Line Out", NULL),
+        SND_SOC_DAPM_LINE("GSM Line In", NULL),
+        SND_SOC_DAPM_MIC("Headset Mic", NULL),
+        SND_SOC_DAPM_MIC("Handset Mic", NULL),
+        SND_SOC_DAPM_SPK("Handset Spk", NULL),
+};
+
+
+/* example machine audio_mapnections */
+static const struct snd_soc_dapm_route audio_map[] = {
+
+        /* Connections to the lm4853 amp */
+        {"Stereo Out", NULL, "LOUT1"},
+        {"Stereo Out", NULL, "ROUT1"},
+
+        /* Connections to the GSM Module */
+        {"GSM Line Out", NULL, "MONO1"},
+        {"GSM Line Out", NULL, "MONO2"},
+        {"RXP", NULL, "GSM Line In"},
+        {"RXN", NULL, "GSM Line In"},
+
+        /* Connections to Headset */
+        {"MIC1", NULL, "Mic Bias"},
+        {"Mic Bias", NULL, "Headset Mic"},
+
+        /* Call Mic */
+        {"MIC2", NULL, "Mic Bias"},
+        {"MIC2N", NULL, "Mic Bias"},
+        {"Mic Bias", NULL, "Handset Mic"},
+
+        /* Call Speaker */
+        {"Handset Spk", NULL, "LOUT2"},
+        {"Handset Spk", NULL, "ROUT2"},
+
+        /* Connect the ALC pins */
+        {"ACIN", NULL, "ACOP"},
+};
+
+static const struct snd_kcontrol_new wm8753_neo1973_gta02_controls[] = {
+        SOC_DAPM_PIN_SWITCH("Stereo Out"),
+        SOC_DAPM_PIN_SWITCH("GSM Line Out"),
+        SOC_DAPM_PIN_SWITCH("GSM Line In"),
+        SOC_DAPM_PIN_SWITCH("Headset Mic"),
+        SOC_DAPM_PIN_SWITCH("Handset Mic"),
+        SOC_DAPM_PIN_SWITCH("Handset Spk"),
+
+        /* This has no effect, it exists only to maintain compatibility with
+         * existing ALSA state files.
+         */
+        SOC_SINGLE_EXT("Amp State Switch", 6, 0, 1, 0,
+                lm4853_get_state,
+                lm4853_set_state),
+        SOC_SINGLE_EXT("Amp Spk Switch", 7, 0, 1, 0,
+                lm4853_get_spk,
+                lm4853_set_spk),
+};
+
+/*
+ * This is an example machine initialisation for a wm8753 connected to a
+ * neo1973 GTA02.
+ */
+static int neo1973_gta02_wm8753_init(struct snd_soc_codec *codec)
+{
+        int i, err;
+
+        /* set up NC codec pins */
+        snd_soc_dapm_nc_pin(codec, "OUT3");
+        snd_soc_dapm_nc_pin(codec, "OUT4");
+        snd_soc_dapm_nc_pin(codec, "LINE1");
+        snd_soc_dapm_nc_pin(codec, "LINE2");
+
+        /* Add neo1973 gta02 specific widgets */
+        snd_soc_dapm_new_controls(codec, wm8753_dapm_widgets,
+                                  ARRAY_SIZE(wm8753_dapm_widgets));
+
+        /* add neo1973 gta02 specific controls */
+        for (i = 0; i < ARRAY_SIZE(wm8753_neo1973_gta02_controls); i++) {
+                err = snd_ctl_add(codec->card,
+                        snd_soc_cnew(&wm8753_neo1973_gta02_controls[i],
+                        codec, NULL));
+                if (err < 0)
+                        return err;
+        }
+
+        /* set up neo1973 gta02 specific audio path audio_map */
+        snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+        /* set endpoints to default off mode */
+        snd_soc_dapm_disable_pin(codec, "Stereo Out");
+        snd_soc_dapm_disable_pin(codec, "GSM Line Out");
+        snd_soc_dapm_disable_pin(codec, "GSM Line In");
+        snd_soc_dapm_disable_pin(codec, "Headset Mic");
+        snd_soc_dapm_disable_pin(codec, "Handset Mic");
+        snd_soc_dapm_disable_pin(codec, "Handset Spk");
+
+        snd_soc_dapm_sync(codec);
+
+        return 0;
+}
+
+/*
+ * BT Codec DAI
+ */
+static struct snd_soc_dai bt_dai = {
+        .name = "Bluetooth",
+        .id = 0,
+        .playback = {
+                .channels_min = 1,
+                .channels_max = 1,
+                .rates = SNDRV_PCM_RATE_8000,
+                .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+        .capture = {
+                .channels_min = 1,
+                .channels_max = 1,
+                .rates = SNDRV_PCM_RATE_8000,
+                .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+};
+
+static struct snd_soc_dai_link neo1973_gta02_dai[] = {
+{ /* Hifi Playback - for similatious use with voice below */
+        .name = "WM8753",
+        .stream_name = "WM8753 HiFi",
+        .cpu_dai = &s3c24xx_i2s_dai,
+        .codec_dai = &wm8753_dai[WM8753_DAI_HIFI],
+        .init = neo1973_gta02_wm8753_init,
+        .ops = &neo1973_gta02_hifi_ops,
+},
+{ /* Voice via BT */
+        .name = "Bluetooth",
+        .stream_name = "Voice",
+        .cpu_dai = &bt_dai,
+        .codec_dai = &wm8753_dai[WM8753_DAI_VOICE],
+        .ops = &neo1973_gta02_voice_ops,
+},
+};
+
+static struct snd_soc_card neo1973_gta02 = {
+        .name = "neo1973-gta02",
+        .platform = &s3c24xx_soc_platform,
+        .dai_link = neo1973_gta02_dai,
+        .num_links = ARRAY_SIZE(neo1973_gta02_dai),
+};
+
+static struct snd_soc_device neo1973_gta02_snd_devdata = {
+        .card = &neo1973_gta02,
+        .codec_dev = &soc_codec_dev_wm8753,
+};
+
+static struct platform_device *neo1973_gta02_snd_device;
+
+static int __init neo1973_gta02_init(void)
+{
+        int ret;
+
+        if (!machine_is_neo1973_gta02()) {
+                printk(KERN_INFO
+                       "Only GTA02 is supported by this ASoC driver\n");
+                return -ENODEV;
+        }
+
+        /* register bluetooth DAI here */
+        ret = snd_soc_register_dai(&bt_dai);
+        if (ret)
+                return ret;
+
+        neo1973_gta02_snd_device = platform_device_alloc("soc-audio", -1);
+        if (!neo1973_gta02_snd_device)
+                return -ENOMEM;
+
+        platform_set_drvdata(neo1973_gta02_snd_device,
+                        &neo1973_gta02_snd_devdata);
+        neo1973_gta02_snd_devdata.dev = &neo1973_gta02_snd_device->dev;
+        ret = platform_device_add(neo1973_gta02_snd_device);
+
+        if (ret) {
+                platform_device_put(neo1973_gta02_snd_device);
+                return ret;
+        }
+
+        /* Initialise GPIOs used by amp */
+        s3c2410_gpio_cfgpin(GTA02_GPIO_HP_IN, S3C2410_GPIO_OUTPUT);
+        s3c2410_gpio_cfgpin(GTA02_GPIO_AMP_SHUT, S3C2410_GPIO_OUTPUT);
+
+        /* Amp off by default */
+        s3c2410_gpio_setpin(GTA02_GPIO_AMP_SHUT, 1);
+
+        /* Speaker off by default */
+        s3c2410_gpio_setpin(GTA02_GPIO_HP_IN, 1);
+
+        return ret;
+}
+module_init(neo1973_gta02_init);
+
+static void __exit neo1973_gta02_exit(void)
+{
+        snd_soc_unregister_dai(&bt_dai);
+        platform_device_unregister(neo1973_gta02_snd_device);
+}
+module_exit(neo1973_gta02_exit);
+
+/* Module information */
+MODULE_AUTHOR("Graeme Gregory, graeme@openmoko.org");
+MODULE_DESCRIPTION("ALSA SoC WM8753 Neo1973 GTA02");
+MODULE_LICENSE("GPL");