ASOC Audio driver for TLVaic23b audio codec

Signed-off-by: Arun KS arunks@mistralsolutions.com --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tlv320aic23.c | 634 ++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/tlv320aic23.h | 143 +++++++++ 4 files changed, 783 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/tlv320aic23.c create mode 100644 sound/soc/codecs/tlv320aic23.h

diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index b1a5eed..7be0b37 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -55,3 +55,7 @@ config SND_SOC_TWL4030 tristate depends on SND_SOC && TWL4030_CORE

+config SND_SOC_TLV320AIC23 + tristate + depends on I2C + diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index a519ced..adb5aa6 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -11,6 +11,7 @@ snd-soc-wm9713-objs := wm9713.o snd-soc-cs4270-objs := cs4270.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o snd-soc-twl4030-objs := twl4030.o +snd-soc-tlv320aic23-objs := tlv320aic23.o

obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o @@ -25,3 +26,4 @@ obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o obj-$(CONFIG_SND_SOC_TWL4030) += snd-soc-twl4030.o +obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c new file mode 100644 index 0000000..eefeb2d --- /dev/null +++ b/sound/soc/codecs/tlv320aic23.c @@ -0,0 +1,634 @@ +/* + * ALSA SoC TLV320AIC23 codec driver + * + * Author: Arun KS, arunks@mistralsolutions.com + * Copyright: (C) 2008 Mistral Solutions Pvt Ltd., + * + * Based on sound/soc/codecs/tlv320aic23.c by Vladimir Barinov + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * Notes: + * The AIC23 is a driver for a low power stereo audio + * codec aic23 + * + * The machine layer should disable unsupported inputs/outputs by + * snd_soc_dapm_disable_pin(codec, "LHPOUT"), etc. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.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/initval.h> + +#include "tlv320aic23.h" + +#define AUDIO_NAME "aic23" +#define AIC23_VERSION "0.1" + +/* codec private data */ +struct aic23_priv { + unsigned int sysclk; + int master; +}; + +/* + * AIC23 register cache + */ +static const u16 aic23_reg[AIC23_CACHEREGNUM] = { + 0x0097, 0x0097, 0x00F9, 0x00F9, /* 0 */ + 0x001A, 0x0004, 0x0007, 0x0001, /* 4 */ + 0x0020, 0x0000, 0x0000, 0x0000, /* 8 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 12 */ +}; + +/* + * read aic23 register cache + */ +static inline unsigned int aic23_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + if (reg >= AIC23_CACHEREGNUM) + return -1; + return cache[reg]; +} + +/* + * write aic23 register cache + */ +static inline void aic23_write_reg_cache(struct snd_soc_codec *codec, + u8 reg, u16 value) +{ + u16 *cache = codec->reg_cache; + if (reg >= AIC23_CACHEREGNUM) + return; + cache[reg] = value; +} + +/* + * write to the aic23 register space + */ +static int aic23_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + + u8 data; + + /* TLV320AIC23 has 7 bit address and 9 bits of data + * so we need to switch one data bit into reg and rest + * of data into val + */ + + if ((reg < 0 || reg > 9) && (reg != 15)) { + printk(KERN_WARNING "Invalid register R%d\n", reg); + return -1; + } + + data = (reg << 1) | (value >> 8 & 0x01); + + aic23_write_reg_cache(codec, reg, value); + + if (codec->hw_write(codec->control_data, (u8) data, + (u8) (value & 0xff)) == 0) + return 0; + + printk(KERN_ERR "I2C: cannot write %03x to register R%d\n", value, reg); + + return -EIO; +} + +static const struct snd_kcontrol_new aic23_snd_controls[] = { + SOC_DOUBLE_R("PCM Playback Volume", LEFT_CHANNEL_VOLUME_ADDR, + RIGHT_CHANNEL_VOLUME_ADDR, 0, 0x7f, 0), + SOC_SINGLE("PCM Playback Switch", DIGITAL_AUDIO_CONTROL_ADDR, 3, 0x01, + 1), + SOC_DOUBLE_R("Line Input Mute", LEFT_LINE_VOLUME_ADDR, + RIGHT_LINE_VOLUME_ADDR, 7, 0x01, 0), + SOC_DOUBLE_R("Line Input Volume", LEFT_LINE_VOLUME_ADDR, + RIGHT_LINE_VOLUME_ADDR, 0, 0x1f, 0), + SOC_SINGLE("Mic Mute", ANALOG_AUDIO_CONTROL_ADDR, 1, 0x01, 0), + SOC_SINGLE("Mic Booster Switch", ANALOG_AUDIO_CONTROL_ADDR, 0, 0x01, 0), + +}; + +/* add non dapm controls */ +static int aic23_add_controls(struct snd_soc_codec *codec) +{ + + int err, i; + + for (i = 0; i < ARRAY_SIZE(aic23_snd_controls); i++) { + err = snd_ctl_add(codec->card, + snd_soc_cnew(&aic23_snd_controls[i], + codec, NULL)); + if (err < 0) + return err; + } + + return 0; + +} +static const char *aic23_rec_src[] = { "Line", "Mic" }; + +static const struct soc_enum aic23_enum[] = { + SOC_ENUM_SINGLE(ANALOG_AUDIO_CONTROL_ADDR, 2, 2, aic23_rec_src), +}; + +static const struct snd_kcontrol_new aic23_rec_src_mux_controls = +SOC_DAPM_ENUM("Route", aic23_enum[0]); + +/* PGA Mixer controls for Line and Mic switch */ +static const struct snd_kcontrol_new aic23_pga_mixer_controls[] = { + SOC_DAPM_SINGLE("Line Switch", POWER_DOWN_CONTROL_ADDR, 0, 1, 1), + SOC_DAPM_SINGLE("Mic Switch", POWER_DOWN_CONTROL_ADDR, 1, 1, 1), +}; + +static const struct snd_soc_dapm_widget aic23_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC", "Playback", POWER_DOWN_CONTROL_ADDR, 3, 1), + SND_SOC_DAPM_PGA("OUT", POWER_DOWN_CONTROL_ADDR, 4, 1, NULL, 0), + + SND_SOC_DAPM_ADC("ADC", "Capture", POWER_DOWN_CONTROL_ADDR, 2, 1), + SND_SOC_DAPM_MIXER("PGA Mixer", SND_SOC_NOPM, 0, 0, + &aic23_pga_mixer_controls[0], + ARRAY_SIZE(aic23_pga_mixer_controls)), + SND_SOC_DAPM_MUX("Capture Source", SND_SOC_NOPM, 0, 0, + &aic23_rec_src_mux_controls), + + SND_SOC_DAPM_OUTPUT("LHPOUT"), + SND_SOC_DAPM_OUTPUT("RHPOUT"), + SND_SOC_DAPM_OUTPUT("LOUT"), + SND_SOC_DAPM_OUTPUT("ROUT"), + SND_SOC_DAPM_INPUT("LLINEIN"), + SND_SOC_DAPM_INPUT("RLINEIN"), + SND_SOC_DAPM_INPUT("MICIN"), +}; + +static const struct snd_soc_dapm_route intercon[] = { + {"OUT", NULL, "DAC"}, + + {"LHPOUT", NULL, "OUT"}, + {"RHPOUT", NULL, "OUT"}, + + {"LOUT", NULL, "OUT"}, + {"ROUT", NULL, "OUT"}, + + {"Capture Source", "Line", "LLINEIN"}, + {"Capture Source", "Line", "RLINEIN"}, + + {"Capture Source", "Mic", "MICIN"}, + + {"PGA Mixer", "Line Switch", "Capture Source"}, + {"PGA Mixer", "Mic Switch", "Capture Source"}, + + {"ADC", NULL, "PGA Mixer"}, + +}; + +/* aic23 related */ +static const struct aic23_samplerate_reg_info + rate_reg_info[NUMBER_SAMPLE_RATES_SUPPORTED] = { + {4000, 0x06, 1}, /* 4000 */ + {8000, 0x06, 0}, /* 8000 */ + {16000, 0x0C, 1}, /* 16000 */ + {22050, 0x11, 1}, /* 22050 */ + {24000, 0x00, 1}, /* 24000 */ + {32000, 0x0C, 0}, /* 32000 */ + {44100, 0x11, 0}, /* 44100 */ + {48000, 0x00, 0}, /* 48000 */ + {88200, 0x1F, 0}, /* 88200 */ + {96000, 0x0E, 0}, /* 96000 */ +}; + +static int aic23_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, aic23_dapm_widgets, + ARRAY_SIZE(aic23_dapm_widgets)); + + /* set up audio path interconnects */ + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +static int aic23_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_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + u16 iface_reg, data; + u8 count = 0; + + iface_reg = + aic23_read_reg_cache(codec, + DIGITAL_AUDIO_FORMAT_ADDR) & ~(0x03 << 2); + + /* Search for the right sample rate */ + /* Verify what happens if the rate is not supported + * now it goes to 96Khz */ + while ((rate_reg_info[count].sample_rate != params_rate(params)) && + (count < ARRAY_SIZE(rate_reg_info))) { + count++; + } + + data = (rate_reg_info[count].divider << CLKIN_SHIFT) | + (rate_reg_info[count].control << BOSR_SHIFT) | USB_CLK_ON; + + aic23_write(codec, SAMPLE_RATE_CONTROL_ADDR, data); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + iface_reg |= (0x01 << 2); + break; + case SNDRV_PCM_FORMAT_S24_LE: + iface_reg |= (0x02 << 2); + break; + case SNDRV_PCM_FORMAT_S32_LE: + iface_reg |= (0x03 << 2); + break; + } + aic23_write(codec, DIGITAL_AUDIO_FORMAT_ADDR, iface_reg); + + return 0; +} + +static int aic23_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u16 reg; + + reg = + aic23_read_reg_cache(codec, + DIGITAL_AUDIO_CONTROL_ADDR) & ~DACM_MUTE; + + if (mute) + aic23_write(codec, DIGITAL_AUDIO_CONTROL_ADDR, reg | DACM_MUTE); + + else + aic23_write(codec, DIGITAL_AUDIO_CONTROL_ADDR, reg); + + return 0; +} + +static int aic23_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct aic23_priv *aic23 = codec->private_data; + + aic23->sysclk = freq; + return 0; +} + +static int aic23_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct aic23_priv *aic23 = codec->private_data; + u16 iface_reg; + + iface_reg = + aic23_read_reg_cache(codec, DIGITAL_AUDIO_FORMAT_ADDR) & (~0x03); + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + aic23->master = 1; + iface_reg |= MS_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFS: + aic23->master = 0; + break; + default: + return -EINVAL; + + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface_reg |= FOR_I2S; + break; + case SND_SOC_DAIFMT_DSP_A: + iface_reg |= FOR_DSP; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + iface_reg |= FOR_LJUST; + break; + default: + return -EINVAL; + + } + + aic23_write(codec, DIGITAL_AUDIO_FORMAT_ADDR, iface_reg); + + return 0; +} + +static int aic23_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + u16 reg; + + switch (level) { + case SND_SOC_BIAS_ON: + reg = aic23_read_reg_cache(codec, POWER_DOWN_CONTROL_ADDR); + aic23_write(codec, POWER_DOWN_CONTROL_ADDR, + reg & (~DEVICE_POWER_OFF)); + /* Activate interface */ + reg = aic23_read_reg_cache(codec, DIGITAL_INTERFACE_ACT_ADDR); + aic23_write(codec, DIGITAL_INTERFACE_ACT_ADDR, reg | ACT_ON); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + reg = aic23_read_reg_cache(codec, POWER_DOWN_CONTROL_ADDR); + aic23_write(codec, POWER_DOWN_CONTROL_ADDR, + reg | DEVICE_POWER_OFF); + /* Deactivate interface */ + reg = aic23_read_reg_cache(codec, DIGITAL_INTERFACE_ACT_ADDR); + aic23_write(codec, DIGITAL_INTERFACE_ACT_ADDR, reg & (~ACT_ON)); + + break; + case SND_SOC_BIAS_OFF: + break; + } + + codec->bias_level = level; + + return 0; +} + +#define AIC23_RATES SNDRV_PCM_RATE_8000_96000 +#define AIC23_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) + +struct snd_soc_dai aic23_dai = { + .name = "aic23", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = AIC23_RATES, + .formats = AIC23_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = AIC23_RATES, + .formats = AIC23_FORMATS,}, + .ops = { + .hw_params = aic23_hw_params, + }, + .dai_ops = { + .digital_mute = aic23_mute, + .set_sysclk = aic23_set_dai_sysclk, + .set_fmt = aic23_set_dai_fmt, + } +}; +EXPORT_SYMBOL_GPL(aic23_dai); + +static int aic23_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + aic23_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int aic23_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + int i; + u8 data[2]; + u8 *cache = codec->reg_cache; + + /* Sync reg_cache with the hardware */ + for (i = 0; i < ARRAY_SIZE(aic23_reg); i++) { + data[0] = i; + data[1] = cache[i]; + codec->hw_write(codec->control_data, data, 2); + } + + aic23_set_bias_level(codec, codec->suspend_bias_level); + + return 0; +} + +/* + * initialise the AIC23 driver + * register the mixer and dsp interfaces with the kernel + */ +static int aic23_init(struct snd_soc_device *socdev) +{ + struct snd_soc_codec *codec = socdev->codec; + int ret = 0; + u16 reg; + + codec->name = "aic23"; + codec->owner = THIS_MODULE; + codec->read = aic23_read_reg_cache; + codec->write = aic23_write; + codec->set_bias_level = aic23_set_bias_level; + codec->dai = &aic23_dai; + codec->num_dai = 1; + codec->reg_cache_size = ARRAY_SIZE(aic23_reg); + codec->reg_cache = kmemdup(aic23_reg, sizeof(aic23_reg), GFP_KERNEL); + if (codec->reg_cache == NULL) + return -ENOMEM; + + /* Reset codec */ + aic23_write(codec, RESET_CONTROL_ADDR, 0); + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "aic23: failed to create pcms\n"); + goto pcm_err; + } + + aic23_write(codec, DIGITAL_AUDIO_CONTROL_ADDR, DEEMP_44K); + + /* Unmute input */ + reg = aic23_read_reg_cache(codec, LEFT_LINE_VOLUME_ADDR); + aic23_write(codec, LEFT_LINE_VOLUME_ADDR, + (reg & (~LIM_MUTED)) | (LRS_ENABLED)); + reg = aic23_read_reg_cache(codec, RIGHT_LINE_VOLUME_ADDR); + aic23_write(codec, RIGHT_LINE_VOLUME_ADDR, + (reg & (~LIM_MUTED)) | LRS_ENABLED); + reg = aic23_read_reg_cache(codec, ANALOG_AUDIO_CONTROL_ADDR); + aic23_write(codec, ANALOG_AUDIO_CONTROL_ADDR, + (reg | DAC_SELECTED) & (~MICM_MUTED) & (~BYPASS_ON)); + + /* Default output volume */ + aic23_write(codec, LEFT_CHANNEL_VOLUME_ADDR, + DEFAULT_OUTPUT_VOLUME & OUTPUT_VOLUME_MASK); + aic23_write(codec, RIGHT_CHANNEL_VOLUME_ADDR, + DEFAULT_OUTPUT_VOLUME & OUTPUT_VOLUME_MASK); + + /* off, with power on */ + aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + aic23_add_controls(codec); + aic23_add_widgets(codec); + ret = snd_soc_register_card(socdev); + if (ret < 0) { + printk(KERN_ERR "aic23: failed to register card\n"); + goto card_err; + } + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + kfree(codec->reg_cache); + return ret; +} +static struct snd_soc_device *aic23_socdev; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +/* + * If the i2c layer weren't so broken, we could pass this kind of data + * around + */ +static int aic23_codec_probe(struct i2c_client *i2c, + const struct i2c_device_id *i2c_id) +{ + struct snd_soc_device *socdev = aic23_socdev; + struct snd_soc_codec *codec = socdev->codec; + int ret; + + if (!i2c_check_functionality(i2c->adapter, I2C_FUNC_SMBUS_BYTE_DATA)) + return -EINVAL; + + + i2c_set_clientdata(i2c, codec); + codec->control_data = i2c; + + ret = aic23_init(socdev); + if (ret < 0) { + printk(KERN_ERR "aic23: failed to initialise AIC23\n"); + goto err; + } + return ret; + +err: + kfree(codec); + kfree(i2c); + return ret; +} +static int __exit aic23_i2c_remove(struct i2c_client *i2c) +{ + + put_device(&i2c->dev); + return 0; +} + +static const struct i2c_device_id tlvaic23b_id[] = { + {"tlvaic23b", 0}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, tlvaic23b_id); + +static struct i2c_driver aic23b_i2c_driver = { + .driver = { + .name = "tlvaic23b", + }, + .probe = aic23_codec_probe, + .remove = __exit_p(aic23_i2c_remove), + .id_table = tlvaic23b_id, +}; + +#endif + +static int aic23_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + struct aic23_priv *aic23; + int ret = 0; + + printk(KERN_INFO "AIC23 Audio Codec %s\n", AIC23_VERSION); + + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) + return -ENOMEM; + + aic23 = kzalloc(sizeof(struct aic23_priv), GFP_KERNEL); + if (aic23 == NULL) { + kfree(codec); + return -ENOMEM; + } + + codec->private_data = aic23; + socdev->codec = codec; + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + aic23_socdev = socdev; +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + codec->hw_write = (hw_write_t) i2c_smbus_write_byte_data; + codec->hw_read = NULL; + ret = i2c_add_driver(&aic23b_i2c_driver); + if (ret != 0) + printk(KERN_ERR "can't add i2c driver"); +#else + /* Add other interfaces here */ +#endif + return ret; +} + +static int aic23_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + /* power down chip */ + if (codec->control_data) + aic23_set_bias_level(codec, SND_SOC_BIAS_OFF); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&aic23b_i2c_driver); +#endif + kfree(codec->private_data); + kfree(codec->reg_cache); + kfree(codec); + + return 0; +} +struct snd_soc_codec_device soc_codec_dev_aic23 = { + .probe = aic23_probe, + .remove = aic23_remove, + .suspend = aic23_suspend, + .resume = aic23_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_aic23); + +MODULE_DESCRIPTION("ASoC TLV320AIC23 codec driver"); +MODULE_AUTHOR("Arun KS"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tlv320aic23.h b/sound/soc/codecs/tlv320aic23.h new file mode 100644 index 0000000..9c3808c --- /dev/null +++ b/sound/soc/codecs/tlv320aic23.h @@ -0,0 +1,143 @@ +/* + * ALSA SoC TLV320AIC23 codec driver + * + * Author: Arun KS, arunks@mistralsolutions.com + * Copyright: (C) 2008 Mistral Solutions Pvt Ltd + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _AIC23_H +#define _AIC23_H + +#define AIC23_CACHEREGNUM 16 +/* Codec TLV320AIC23 */ +#define LEFT_LINE_VOLUME_ADDR 0x00 +#define RIGHT_LINE_VOLUME_ADDR 0x01 +#define LEFT_CHANNEL_VOLUME_ADDR 0x02 +#define RIGHT_CHANNEL_VOLUME_ADDR 0x03 +#define ANALOG_AUDIO_CONTROL_ADDR 0x04 +#define DIGITAL_AUDIO_CONTROL_ADDR 0x05 +#define POWER_DOWN_CONTROL_ADDR 0x06 +#define DIGITAL_AUDIO_FORMAT_ADDR 0x07 +#define SAMPLE_RATE_CONTROL_ADDR 0x08 +#define DIGITAL_INTERFACE_ACT_ADDR 0x09 +#define RESET_CONTROL_ADDR 0x0F + +/* Left (right) line input volume control register */ +#define LRS_ENABLED 0x0100 +#define LIM_MUTED 0x0080 +#define LIV_DEFAULT 0x0017 +#define LIV_MAX 0x001f +#define LIV_MIN 0x0000 + +/* Left (right) channel headphone volume control register */ +#define LZC_ON 0x0080 +#define LHV_DEFAULT 0x0079 +#define LHV_MAX 0x007f +#define LHV_MIN 0x0000 + +/* Analog audio path control register */ +#define STA_REG(x) ((x)<<6) +#define STE_ENABLED 0x0020 +#define DAC_SELECTED 0x0010 +#define BYPASS_ON 0x0008 +#define INSEL_MIC 0x0004 +#define MICM_MUTED 0x0002 +#define MICB_20DB 0x0001 + +/* Digital audio path control register */ +#define DACM_MUTE 0x0008 +#define DEEMP_32K 0x0002 +#define DEEMP_44K 0x0004 +#define DEEMP_48K 0x0006 +#define ADCHP_ON 0x0001 + +/* Power control down register */ +#define DEVICE_POWER_OFF 0x0080 +#define CLK_OFF 0x0040 +#define OSC_OFF 0x0020 +#define OUT_OFF 0x0010 +#define DAC_OFF 0x0008 +#define ADC_OFF 0x0004 +#define MIC_OFF 0x0002 +#define LINE_OFF 0x0001 + +/* Digital audio interface register */ +#define MS_MASTER 0x0040 +#define LRSWAP_ON 0x0020 +#define LRP_ON 0x0010 +#define IWL_16 0x0000 +#define IWL_20 0x0004 +#define IWL_24 0x0008 +#define IWL_32 0x000C +#define FOR_I2S 0x0002 +#define FOR_DSP 0x0003 +#define FOR_LJUST 0x0001 + +/* Sample rate control register */ +#define CLKOUT_HALF 0x0080 +#define CLKIN_HALF 0x0040 +#define BOSR_384fs 0x0002 /* BOSR_272fs when in USB mode */ +#define USB_CLK_ON 0x0001 +#define SR_MASK 0xf +#define CLKOUT_SHIFT 7 +#define CLKIN_SHIFT 6 +#define SR_SHIFT 2 +#define BOSR_SHIFT 1 + +/* Digital interface register */ +#define ACT_ON 0x0001 + +/* Define to set the AIC23 as the master w.r.t McBSP */ +#define AIC23_MASTER + +#define NUMBER_SAMPLE_RATES_SUPPORTED 10 + +/* + * AUDIO related MACROS + */ +#ifndef DEFAULT_BITPERSAMPLE +#define DEFAULT_BITPERSAMPLE 16 +#endif + +#define DEFAULT_SAMPLE_RATE 44100 +#define CODEC_CLOCK 12000000 +#define AUDIO_MCBSP OMAP_MCBSP1 + +#define DEFAULT_OUTPUT_VOLUME 0x70 +#define DEFAULT_INPUT_VOLUME 0x10 /* 0 ==> mute line in */ + +#define OUTPUT_VOLUME_MIN LHV_MIN +#define OUTPUT_VOLUME_MAX LHV_MAX +#define OUTPUT_VOLUME_RANGE (OUTPUT_VOLUME_MAX - OUTPUT_VOLUME_MIN) +#define OUTPUT_VOLUME_MASK OUTPUT_VOLUME_MAX + +#define INPUT_VOLUME_MIN LIV_MIN +#define INPUT_VOLUME_MAX LIV_MAX +#define INPUT_VOLUME_RANGE (INPUT_VOLUME_MAX - INPUT_VOLUME_MIN) +#define INPUT_VOLUME_MASK INPUT_VOLUME_MAX + +#define SIDETONE_MASK 0x1c0 +#define SIDETONE_0 0x100 +#define SIDETONE_6 0x000 +#define SIDETONE_9 0x040 +#define SIDETONE_12 0x080 +#define SIDETONE_18 0x0c0 + +#define DEFAULT_ANALOG_AUDIO_CONTROL (DAC_SELECTED | STE_ENABLED | \ + INSEL_MIC | MICB_20DB) + +struct aic23_samplerate_reg_info { + u32 sample_rate; + u8 control; /* SR3, SR2, SR1, SR0 and BOSR */ + u8 divider /* if 0 CLKIN = MCLK, if 1 CLKIN = MCLK/2 */ +}; + +extern struct snd_soc_dai aic23_dai; +extern struct snd_soc_codec_device soc_codec_dev_aic23; + +#endif /* _AIC23_H */ +