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authorGraeme Gregory <gg@opensource.wolfsonmicro.com>2007-05-14 11:03:52 +0200
committerJaroslav Kysela <perex@suse.cz>2007-07-20 11:11:15 +0200
commit74930bb6db56bcc9899723c6c79fe681524e5b62 (patch)
treee3a21928a7fad39c95a169a583703587519c5edd /sound/soc/s3c24xx/neo1973_wm8753.c
parent050f05eaec1c7c5434c78d010ada3cfeb7d0b3b3 (diff)
[ALSA] ASoC S3C24xx machine drivers - Openmoko Neo1973
This patch adds ALSA support for the Openmoko Neo1973 phone. Features:- * HiFi Playback and capture. * Phone calls supported. * Support for BT PCM in WM8753 voice interface. * Support for LM4857 audio amp. Signed-off-by: Graeme Gregory <gg@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Diffstat (limited to 'sound/soc/s3c24xx/neo1973_wm8753.c')
-rw-r--r--sound/soc/s3c24xx/neo1973_wm8753.c670
1 files changed, 670 insertions, 0 deletions
diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c
new file mode 100644
index 00000000000..d5a8fc2cf8d
--- /dev/null
+++ b/sound/soc/s3c24xx/neo1973_wm8753.c
@@ -0,0 +1,670 @@
+/*
+ * neo1973_wm8753.c -- SoC audio for Neo1973
+ *
+ * Copyright 2007 Wolfson Microelectronics PLC.
+ * Author: Graeme Gregory
+ * graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.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.
+ *
+ * Revision history
+ * 20th Jan 2007 Initial version.
+ * 05th Feb 2007 Rename all to Neo1973
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/timer.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+#include <linux/i2c.h>
+#include <sound/driver.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+#include <asm/hardware/scoop.h>
+#include <asm/arch/regs-iis.h>
+#include <asm/arch/regs-clock.h>
+#include <asm/arch/regs-gpio.h>
+#include <asm/hardware.h>
+#include <asm/arch/audio.h>
+#include <asm/io.h>
+#include <asm/arch/spi-gpio.h>
+#include "../codecs/wm8753.h"
+#include "lm4857.h"
+#include "s3c24xx-pcm.h"
+#include "s3c24xx-i2s.h"
+
+/* define the scenarios */
+#define NEO_AUDIO_OFF 0
+#define NEO_GSM_CALL_AUDIO_HANDSET 1
+#define NEO_GSM_CALL_AUDIO_HEADSET 2
+#define NEO_GSM_CALL_AUDIO_BLUETOOTH 3
+#define NEO_STEREO_TO_SPEAKERS 4
+#define NEO_STEREO_TO_HEADPHONES 5
+#define NEO_CAPTURE_HANDSET 6
+#define NEO_CAPTURE_HEADSET 7
+#define NEO_CAPTURE_BLUETOOTH 8
+
+static struct snd_soc_machine neo1973;
+static struct i2c_client *i2c;
+
+static int neo1973_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_codec_dai *codec_dai = rtd->dai->codec_dai;
+ struct snd_soc_cpu_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 = codec_dai->dai_ops.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 = cpu_dai->dai_ops.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 = codec_dai->dai_ops.set_sysclk(codec_dai, WM8753_MCLK, pll_out,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0)
+ return ret;
+
+ /* set MCLK division for sample rate */
+ ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK,
+ S3C2410_IISMOD_32FS );
+ if (ret < 0)
+ return ret;
+
+ /* set codec BCLK division for sample rate */
+ ret = codec_dai->dai_ops.set_clkdiv(codec_dai, WM8753_BCLKDIV, bclk);
+ if (ret < 0)
+ return ret;
+
+ /* set prescaler division for sample rate */
+ ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER,
+ S3C24XX_PRESCALE(4,4));
+ if (ret < 0)
+ return ret;
+
+ /* codec PLL input is PCLK/4 */
+ ret = codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL1,
+ iis_clkrate / 4, pll_out);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+static int neo1973_hifi_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
+
+ /* disable the PLL */
+ return codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL1, 0, 0);
+}
+
+/*
+ * Neo1973 WM8753 HiFi DAI opserations.
+ */
+static struct snd_soc_ops neo1973_hifi_ops = {
+ .hw_params = neo1973_hifi_hw_params,
+ .hw_free = neo1973_hifi_hw_free,
+};
+
+static int neo1973_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_codec_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 = codec_dai->dai_ops.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 = codec_dai->dai_ops.set_sysclk(codec_dai, WM8753_PCMCLK, 12288000,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0)
+ return ret;
+
+ /* set codec PCM division for sample rate */
+ ret = codec_dai->dai_ops.set_clkdiv(codec_dai, WM8753_PCMDIV, pcmdiv);
+ if (ret < 0)
+ return ret;
+
+ /* configue and enable PLL for 12.288MHz output */
+ ret = codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL2,
+ iis_clkrate / 4, 12288000);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+static int neo1973_voice_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
+
+ /* disable the PLL */
+ return codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL2, 0, 0);
+}
+
+static struct snd_soc_ops neo1973_voice_ops = {
+ .hw_params = neo1973_voice_hw_params,
+ .hw_free = neo1973_voice_hw_free,
+};
+
+static int neo1973_scenario = 0;
+
+static int neo1973_get_scenario(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = neo1973_scenario;
+ return 0;
+}
+
+static int set_scenario_endpoints(struct snd_soc_codec *codec, int scenario)
+{
+ switch(neo1973_scenario) {
+ case NEO_AUDIO_OFF:
+ snd_soc_dapm_set_endpoint(codec, "Audio Out", 0);
+ snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0);
+ snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0);
+ snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0);
+ snd_soc_dapm_set_endpoint(codec, "Call Mic", 0);
+ break;
+ case NEO_GSM_CALL_AUDIO_HANDSET:
+ snd_soc_dapm_set_endpoint(codec, "Audio Out", 1);
+ snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 1);
+ snd_soc_dapm_set_endpoint(codec, "GSM Line In", 1);
+ snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0);
+ snd_soc_dapm_set_endpoint(codec, "Call Mic", 1);
+ break;
+ case NEO_GSM_CALL_AUDIO_HEADSET:
+ snd_soc_dapm_set_endpoint(codec, "Audio Out", 1);
+ snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 1);
+ snd_soc_dapm_set_endpoint(codec, "GSM Line In", 1);
+ snd_soc_dapm_set_endpoint(codec, "Headset Mic", 1);
+ snd_soc_dapm_set_endpoint(codec, "Call Mic", 0);
+ break;
+ case NEO_GSM_CALL_AUDIO_BLUETOOTH:
+ snd_soc_dapm_set_endpoint(codec, "Audio Out", 0);
+ snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 1);
+ snd_soc_dapm_set_endpoint(codec, "GSM Line In", 1);
+ snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0);
+ snd_soc_dapm_set_endpoint(codec, "Call Mic", 0);
+ break;
+ case NEO_STEREO_TO_SPEAKERS:
+ snd_soc_dapm_set_endpoint(codec, "Audio Out", 1);
+ snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0);
+ snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0);
+ snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0);
+ snd_soc_dapm_set_endpoint(codec, "Call Mic", 0);
+ break;
+ case NEO_STEREO_TO_HEADPHONES:
+ snd_soc_dapm_set_endpoint(codec, "Audio Out", 1);
+ snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0);
+ snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0);
+ snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0);
+ snd_soc_dapm_set_endpoint(codec, "Call Mic", 0);
+ break;
+ case NEO_CAPTURE_HANDSET:
+ snd_soc_dapm_set_endpoint(codec, "Audio Out", 0);
+ snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0);
+ snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0);
+ snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0);
+ snd_soc_dapm_set_endpoint(codec, "Call Mic", 1);
+ break;
+ case NEO_CAPTURE_HEADSET:
+ snd_soc_dapm_set_endpoint(codec, "Audio Out", 0);
+ snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0);
+ snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0);
+ snd_soc_dapm_set_endpoint(codec, "Headset Mic", 1);
+ snd_soc_dapm_set_endpoint(codec, "Call Mic", 0);
+ break;
+ case NEO_CAPTURE_BLUETOOTH:
+ snd_soc_dapm_set_endpoint(codec, "Audio Out", 0);
+ snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0);
+ snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0);
+ snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0);
+ snd_soc_dapm_set_endpoint(codec, "Call Mic", 0);
+ break;
+ default:
+ snd_soc_dapm_set_endpoint(codec, "Audio Out", 0);
+ snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0);
+ snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0);
+ snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0);
+ snd_soc_dapm_set_endpoint(codec, "Call Mic", 0);
+ }
+
+ snd_soc_dapm_sync_endpoints(codec);
+
+ return 0;
+}
+
+static int neo1973_set_scenario(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+ if (neo1973_scenario == ucontrol->value.integer.value[0])
+ return 0;
+
+ neo1973_scenario = ucontrol->value.integer.value[0];
+ set_scenario_endpoints(codec, neo1973_scenario);
+ return 1;
+}
+
+static u8 lm4857_regs[4] = {0x00, 0x40, 0x80, 0xC0};
+
+static void lm4857_write_regs(void)
+{
+ if (i2c_master_send(i2c, lm4857_regs, 4) != 4)
+ printk(KERN_ERR "lm4857: i2c write failed\n");
+}
+
+static int lm4857_get_reg(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int reg=kcontrol->private_value & 0xFF;
+ int shift = (kcontrol->private_value >> 8) & 0x0F;
+ int mask = (kcontrol->private_value >> 16) & 0xFF;
+
+ ucontrol->value.integer.value[0] = (lm4857_regs[reg] >> shift) & mask;
+ return 0;
+}
+
+static int lm4857_set_reg(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int reg = kcontrol->private_value & 0xFF;
+ int shift = (kcontrol->private_value >> 8) & 0x0F;
+ int mask = (kcontrol->private_value >> 16) & 0xFF;
+
+ if (((lm4857_regs[reg] >> shift ) & mask) ==
+ ucontrol->value.integer.value[0])
+ return 0;
+
+ lm4857_regs[reg] &= ~ (mask << shift);
+ lm4857_regs[reg] |= ucontrol->value.integer.value[0] << shift;
+ lm4857_write_regs();
+ return 1;
+}
+
+static int lm4857_get_mode(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ u8 value = lm4857_regs[LM4857_CTRL] & 0x0F;
+
+ if (value)
+ value -= 5;
+
+ ucontrol->value.integer.value[0] = value;
+ return 0;
+}
+
+static int lm4857_set_mode(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ u8 value = ucontrol->value.integer.value[0];
+
+ if (value)
+ value += 5;
+
+ if ((lm4857_regs[LM4857_CTRL] & 0x0F) == value)
+ return 0;
+
+ lm4857_regs[LM4857_CTRL] &= 0xF0;
+ lm4857_regs[LM4857_CTRL] |= value;
+ lm4857_write_regs();
+ return 1;
+}
+
+static const struct snd_soc_dapm_widget wm8753_dapm_widgets[] = {
+ SND_SOC_DAPM_LINE("Audio Out", NULL),
+ 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("Call Mic", NULL),
+};
+
+
+/* example machine audio_mapnections */
+static const char* audio_map[][3] = {
+
+ /* Connections to the lm4857 amp */
+ {"Audio Out", NULL, "LOUT1"},
+ {"Audio 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, "Call Mic"},
+
+ /* Connect the ALC pins */
+ {"ACIN", NULL, "ACOP"},
+
+ {NULL, NULL, NULL},
+};
+
+static const char *lm4857_mode[] = {
+ "Off",
+ "Call Speaker",
+ "Stereo Speakers",
+ "Stereo Speakers + Headphones",
+ "Headphones"
+};
+
+static const struct soc_enum lm4857_mode_enum[] = {
+ SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(lm4857_mode), lm4857_mode),
+};
+
+static const char *neo_scenarios[] = {
+ "Off",
+ "GSM Handset",
+ "GSM Headset",
+ "GSM Bluetooth",
+ "Speakers",
+ "Headphones",
+ "Capture Handset",
+ "Capture Headset",
+ "Capture Bluetooth"
+};
+
+static const struct soc_enum neo_scenario_enum[] = {
+ SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(neo_scenarios),neo_scenarios),
+};
+
+static const struct snd_kcontrol_new wm8753_neo1973_controls[] = {
+ SOC_SINGLE_EXT("Amp Left Playback Volume", LM4857_LVOL, 0, 31, 0,
+ lm4857_get_reg, lm4857_set_reg),
+ SOC_SINGLE_EXT("Amp Right Playback Volume", LM4857_RVOL, 0, 31, 0,
+ lm4857_get_reg, lm4857_set_reg),
+ SOC_SINGLE_EXT("Amp Mono Playback Volume", LM4857_MVOL, 0, 31, 0,
+ lm4857_get_reg, lm4857_set_reg),
+ SOC_ENUM_EXT("Amp Mode", lm4857_mode_enum[0],
+ lm4857_get_mode, lm4857_set_mode),
+ SOC_ENUM_EXT("Neo Mode", neo_scenario_enum[0],
+ neo1973_get_scenario, neo1973_set_scenario),
+ SOC_SINGLE_EXT("Amp Spk 3D Playback Switch", LM4857_LVOL, 5, 1, 0,
+ lm4857_get_reg, lm4857_set_reg),
+ SOC_SINGLE_EXT("Amp HP 3d Playback Switch", LM4857_RVOL, 5, 1, 0,
+ lm4857_get_reg, lm4857_set_reg),
+ SOC_SINGLE_EXT("Amp Fast Wakeup Playback Switch", LM4857_CTRL, 5, 1, 0,
+ lm4857_get_reg, lm4857_set_reg),
+ SOC_SINGLE_EXT("Amp Earpiece 6dB Playback Switch", LM4857_CTRL, 4, 1, 0,
+ lm4857_get_reg, lm4857_set_reg),
+};
+
+/*
+ * This is an example machine initialisation for a wm8753 connected to a
+ * neo1973 II. It is missing logic to detect hp/mic insertions and logic
+ * to re-route the audio in such an event.
+ */
+static int neo1973_wm8753_init(struct snd_soc_codec *codec)
+{
+ int i, err;
+
+ /* set up NC codec pins */
+ snd_soc_dapm_set_endpoint(codec, "LOUT2", 0);
+ snd_soc_dapm_set_endpoint(codec, "ROUT2", 0);
+ snd_soc_dapm_set_endpoint(codec, "OUT3", 0);
+ snd_soc_dapm_set_endpoint(codec, "OUT4", 0);
+ snd_soc_dapm_set_endpoint(codec, "LINE1", 0);
+ snd_soc_dapm_set_endpoint(codec, "LINE2", 0);
+
+
+ /* set endpoints to default mode */
+ set_scenario_endpoints(codec, NEO_AUDIO_OFF);
+
+ /* Add neo1973 specific widgets */
+ for (i = 0; i < ARRAY_SIZE(wm8753_dapm_widgets); i++)
+ snd_soc_dapm_new_control(codec, &wm8753_dapm_widgets[i]);
+
+ /* add neo1973 specific controls */
+ for (i = 0; i < ARRAY_SIZE(wm8753_neo1973_controls); i++) {
+ err = snd_ctl_add(codec->card,
+ snd_soc_cnew(&wm8753_neo1973_controls[i],
+ codec, NULL));
+ if (err < 0)
+ return err;
+ }
+
+ /* set up neo1973 specific audio path audio_mapnects */
+ for (i = 0; audio_map[i][0] != NULL; i++) {
+ snd_soc_dapm_connect_input(codec, audio_map[i][0],
+ audio_map[i][1], audio_map[i][2]);
+ }
+
+ snd_soc_dapm_sync_endpoints(codec);
+ return 0;
+}
+
+/*
+ * BT Codec DAI
+ */
+static struct snd_soc_cpu_dai bt_dai =
+{ .name = "Bluetooth",
+ .id = 0,
+ .type = SND_SOC_DAI_PCM,
+ .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_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_wm8753_init,
+ .ops = &neo1973_hifi_ops,
+},
+{ /* Voice via BT */
+ .name = "Bluetooth",
+ .stream_name = "Voice",
+ .cpu_dai = &bt_dai,
+ .codec_dai = &wm8753_dai[WM8753_DAI_VOICE],
+ .ops = &neo1973_voice_ops,
+},
+};
+
+static struct snd_soc_machine neo1973 = {
+ .name = "neo1973",
+ .dai_link = neo1973_dai,
+ .num_links = ARRAY_SIZE(neo1973_dai),
+};
+
+static struct wm8753_setup_data neo1973_wm8753_setup = {
+ .i2c_address = 0x1a,
+};
+
+static struct snd_soc_device neo1973_snd_devdata = {
+ .machine = &neo1973,
+ .platform = &s3c24xx_soc_platform,
+ .codec_dev = &soc_codec_dev_wm8753,
+ .codec_data = &neo1973_wm8753_setup,
+};
+
+static struct i2c_client client_template;
+
+static unsigned short normal_i2c[] = { 0x7C, I2C_CLIENT_END };
+
+/* Magic definition of all other variables and things */
+I2C_CLIENT_INSMOD;
+
+static int lm4857_amp_probe(struct i2c_adapter *adap, int addr, int kind)
+{
+ int ret;
+
+ client_template.adapter = adap;
+ client_template.addr = addr;
+
+ i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL);
+ if (i2c == NULL)
+ return -ENOMEM;
+
+ ret = i2c_attach_client(i2c);
+ if (ret < 0) {
+ printk(KERN_ERR "LM4857 failed to attach at addr %x\n", addr);
+ goto exit_err;
+ }
+
+ lm4857_write_regs();
+ return ret;
+
+exit_err:
+ kfree(i2c);
+ return ret;
+}
+
+static int lm4857_i2c_detach(struct i2c_client *client)
+{
+ i2c_detach_client(client);
+ kfree(client);
+ return 0;
+}
+
+static int lm4857_i2c_attach(struct i2c_adapter *adap)
+{
+ return i2c_probe(adap, &addr_data, lm4857_amp_probe);
+}
+
+/* corgi i2c codec control layer */
+static struct i2c_driver lm4857_i2c_driver = {
+ .driver = {
+ .name = "LM4857 I2C Amp",
+ .owner = THIS_MODULE,
+ },
+ .id = I2C_DRIVERID_LM4857,
+ .attach_adapter = lm4857_i2c_attach,
+ .detach_client = lm4857_i2c_detach,
+ .command = NULL,
+};
+
+static struct i2c_client client_template = {
+ .name = "LM4857",
+ .driver = &lm4857_i2c_driver,
+};
+
+static struct platform_device *neo1973_snd_device;
+
+static int __init neo1973_init(void)
+{
+ int ret;
+
+ neo1973_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!neo1973_snd_device)
+ return -ENOMEM;
+
+ platform_set_drvdata(neo1973_snd_device, &neo1973_snd_devdata);
+ neo1973_snd_devdata.dev = &neo1973_snd_device->dev;
+ ret = platform_device_add(neo1973_snd_device);
+
+ if (ret)
+ platform_device_put(neo1973_snd_device);
+
+ ret = i2c_add_driver(&lm4857_i2c_driver);
+ if (ret != 0)
+ printk(KERN_ERR "can't add i2c driver");
+
+ return ret;
+}
+
+static void __exit neo1973_exit(void)
+{
+ platform_device_unregister(neo1973_snd_device);
+}
+
+module_init(neo1973_init);
+module_exit(neo1973_exit);
+
+/* Module information */
+MODULE_AUTHOR("Graeme Gregory, graeme.gregory@wolfsonmicro.com, www.wolfsonmicro.com");
+MODULE_DESCRIPTION("ALSA SoC WM8753 Neo1973");
+MODULE_LICENSE("GPL");