/* * ak4642.c -- AK4642/AK4643 ALSA Soc Audio driver * * Copyright (C) 2009 Renesas Solutions Corp. * Kuninori Morimoto * * Based on wm8731.c by Richard Purdie * Based on ak4535.c by Richard Purdie * Based on wm8753.c by Liam Girdwood * * 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. */ /* ** CAUTION ** * * This is very simple driver. * It can use headphone output / stereo input only * * AK4642 is not tested. * AK4643 is tested. */ #include #include #include #include #include #include #include #include #include #include #include #include #include #include "ak4642.h" #define AK4642_VERSION "0.0.1" #define PW_MGMT1 0x00 #define PW_MGMT2 0x01 #define SG_SL1 0x02 #define SG_SL2 0x03 #define MD_CTL1 0x04 #define MD_CTL2 0x05 #define TIMER 0x06 #define ALC_CTL1 0x07 #define ALC_CTL2 0x08 #define L_IVC 0x09 #define L_DVC 0x0a #define ALC_CTL3 0x0b #define R_IVC 0x0c #define R_DVC 0x0d #define MD_CTL3 0x0e #define MD_CTL4 0x0f #define PW_MGMT3 0x10 #define DF_S 0x11 #define FIL3_0 0x12 #define FIL3_1 0x13 #define FIL3_2 0x14 #define FIL3_3 0x15 #define EQ_0 0x16 #define EQ_1 0x17 #define EQ_2 0x18 #define EQ_3 0x19 #define EQ_4 0x1a #define EQ_5 0x1b #define FIL1_0 0x1c #define FIL1_1 0x1d #define FIL1_2 0x1e #define FIL1_3 0x1f #define PW_MGMT4 0x20 #define MD_CTL5 0x21 #define LO_MS 0x22 #define HP_MS 0x23 #define SPK_MS 0x24 #define AK4642_CACHEREGNUM 0x25 /* PW_MGMT2 */ #define HPMTN (1 << 6) #define PMHPL (1 << 5) #define PMHPR (1 << 4) #define MS (1 << 3) /* master/slave select */ #define MCKO (1 << 1) #define PMPLL (1 << 0) #define PMHP_MASK (PMHPL | PMHPR) #define PMHP PMHP_MASK /* MD_CTL1 */ #define PLL3 (1 << 7) #define PLL2 (1 << 6) #define PLL1 (1 << 5) #define PLL0 (1 << 4) #define PLL_MASK (PLL3 | PLL2 | PLL1 | PLL0) #define BCKO_MASK (1 << 3) #define BCKO_64 BCKO_MASK /* MD_CTL2 */ #define FS0 (1 << 0) #define FS1 (1 << 1) #define FS2 (1 << 2) #define FS3 (1 << 5) #define FS_MASK (FS0 | FS1 | FS2 | FS3) struct snd_soc_codec_device soc_codec_dev_ak4642; /* codec private data */ struct ak4642_priv { struct snd_soc_codec codec; }; static struct snd_soc_codec *ak4642_codec; /* * ak4642 register cache */ static const u16 ak4642_reg[AK4642_CACHEREGNUM] = { 0x0000, 0x0000, 0x0001, 0x0000, 0x0002, 0x0000, 0x0000, 0x0000, 0x00e1, 0x00e1, 0x0018, 0x0000, 0x00e1, 0x0018, 0x0011, 0x0008, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, }; /* * read ak4642 register cache */ static inline unsigned int ak4642_read_reg_cache(struct snd_soc_codec *codec, unsigned int reg) { u16 *cache = codec->reg_cache; if (reg >= AK4642_CACHEREGNUM) return -1; return cache[reg]; } /* * write ak4642 register cache */ static inline void ak4642_write_reg_cache(struct snd_soc_codec *codec, u16 reg, unsigned int value) { u16 *cache = codec->reg_cache; if (reg >= AK4642_CACHEREGNUM) return; cache[reg] = value; } /* * write to the AK4642 register space */ static int ak4642_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { u8 data[2]; /* data is * D15..D8 AK4642 register offset * D7...D0 register data */ data[0] = reg & 0xff; data[1] = value & 0xff; if (codec->hw_write(codec->control_data, data, 2) == 2) { ak4642_write_reg_cache(codec, reg, value); return 0; } else return -EIO; } static int ak4642_sync(struct snd_soc_codec *codec) { u16 *cache = codec->reg_cache; int i, r = 0; for (i = 0; i < AK4642_CACHEREGNUM; i++) r |= ak4642_write(codec, i, cache[i]); return r; }; static int ak4642_dai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; struct snd_soc_codec *codec = dai->codec; if (is_play) { /* * start headphone output * * PLL, Master Mode * Audio I/F Format :MSB justified (ADC & DAC) * Digital Volume: -8dB * Bass Boost Level : Middle * * This operation came from example code of * "ASAHI KASEI AK4642" (japanese) manual p97. */ ak4642_write(codec, 0x0f, 0x09); ak4642_write(codec, 0x0e, 0x19); ak4642_write(codec, 0x09, 0x91); ak4642_write(codec, 0x0c, 0x91); ak4642_write(codec, 0x0a, 0x28); ak4642_write(codec, 0x0d, 0x28); ak4642_write(codec, 0x00, 0x64); snd_soc_update_bits(codec, PW_MGMT2, PMHP_MASK, PMHP); snd_soc_update_bits(codec, PW_MGMT2, HPMTN, HPMTN); } else { /* * start stereo input * * PLL Master Mode * Audio I/F Format:MSB justified (ADC & DAC) * Pre MIC AMP:+20dB * MIC Power On * ALC setting:Refer to Table 35 * ALC bit=“1” * * This operation came from example code of * "ASAHI KASEI AK4642" (japanese) manual p94. */ ak4642_write(codec, 0x02, 0x05); ak4642_write(codec, 0x06, 0x3c); ak4642_write(codec, 0x08, 0xe1); ak4642_write(codec, 0x0b, 0x00); ak4642_write(codec, 0x07, 0x21); ak4642_write(codec, 0x00, 0x41); ak4642_write(codec, 0x10, 0x01); } return 0; } static void ak4642_dai_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; struct snd_soc_codec *codec = dai->codec; if (is_play) { /* stop headphone output */ snd_soc_update_bits(codec, PW_MGMT2, HPMTN, 0); snd_soc_update_bits(codec, PW_MGMT2, PMHP_MASK, 0); ak4642_write(codec, 0x00, 0x40); ak4642_write(codec, 0x0e, 0x11); ak4642_write(codec, 0x0f, 0x08); } else { /* stop stereo input */ ak4642_write(codec, 0x00, 0x40); ak4642_write(codec, 0x10, 0x00); ak4642_write(codec, 0x07, 0x01); } } static int ak4642_dai_set_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { struct snd_soc_codec *codec = codec_dai->codec; u8 pll; switch (freq) { case 11289600: pll = PLL2; break; case 12288000: pll = PLL2 | PLL0; break; case 12000000: pll = PLL2 | PLL1; break; case 24000000: pll = PLL2 | PLL1 | PLL0; break; case 13500000: pll = PLL3 | PLL2; break; case 27000000: pll = PLL3 | PLL2 | PLL0; break; default: return -EINVAL; } snd_soc_update_bits(codec, MD_CTL1, PLL_MASK, pll); return 0; } static int ak4642_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct snd_soc_codec *codec = dai->codec; u8 data; u8 bcko; data = MCKO | PMPLL; /* use MCKO */ bcko = 0; /* set master/slave audio interface */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: data |= MS; bcko = BCKO_64; break; case SND_SOC_DAIFMT_CBS_CFS: break; default: return -EINVAL; } snd_soc_update_bits(codec, PW_MGMT2, MS, data); snd_soc_update_bits(codec, MD_CTL1, BCKO_MASK, bcko); return 0; } static int ak4642_dai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; u8 rate; switch (params_rate(params)) { case 7350: rate = FS2; break; case 8000: rate = 0; break; case 11025: rate = FS2 | FS0; break; case 12000: rate = FS0; break; case 14700: rate = FS2 | FS1; break; case 16000: rate = FS1; break; case 22050: rate = FS2 | FS1 | FS0; break; case 24000: rate = FS1 | FS0; break; case 29400: rate = FS3 | FS2 | FS1; break; case 32000: rate = FS3 | FS1; break; case 44100: rate = FS3 | FS2 | FS1 | FS0; break; case 48000: rate = FS3 | FS1 | FS0; break; default: return -EINVAL; break; } snd_soc_update_bits(codec, MD_CTL2, FS_MASK, rate); return 0; } static struct snd_soc_dai_ops ak4642_dai_ops = { .startup = ak4642_dai_startup, .shutdown = ak4642_dai_shutdown, .set_sysclk = ak4642_dai_set_sysclk, .set_fmt = ak4642_dai_set_fmt, .hw_params = ak4642_dai_hw_params, }; struct snd_soc_dai ak4642_dai = { .name = "AK4642", .playback = { .stream_name = "Playback", .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_48000, .formats = SNDRV_PCM_FMTBIT_S16_LE }, .capture = { .stream_name = "Capture", .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_48000, .formats = SNDRV_PCM_FMTBIT_S16_LE }, .ops = &ak4642_dai_ops, .symmetric_rates = 1, }; EXPORT_SYMBOL_GPL(ak4642_dai); static int ak4642_resume(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->card->codec; ak4642_sync(codec); return 0; } /* * initialise the AK4642 driver * register the mixer and dsp interfaces with the kernel */ static int ak4642_init(struct ak4642_priv *ak4642) { struct snd_soc_codec *codec = &ak4642->codec; int ret = 0; if (ak4642_codec) { dev_err(codec->dev, "Another ak4642 is registered\n"); return -EINVAL; } mutex_init(&codec->mutex); INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); codec->private_data = ak4642; codec->name = "AK4642"; codec->owner = THIS_MODULE; codec->read = ak4642_read_reg_cache; codec->write = ak4642_write; codec->dai = &ak4642_dai; codec->num_dai = 1; codec->hw_write = (hw_write_t)i2c_master_send; codec->reg_cache_size = ARRAY_SIZE(ak4642_reg); codec->reg_cache = kmemdup(ak4642_reg, sizeof(ak4642_reg), GFP_KERNEL); if (!codec->reg_cache) return -ENOMEM; ak4642_dai.dev = codec->dev; ak4642_codec = codec; ret = snd_soc_register_codec(codec); if (ret) { dev_err(codec->dev, "Failed to register codec: %d\n", ret); goto reg_cache_err; } ret = snd_soc_register_dai(&ak4642_dai); if (ret) { dev_err(codec->dev, "Failed to register DAI: %d\n", ret); snd_soc_unregister_codec(codec); goto reg_cache_err; } return ret; reg_cache_err: kfree(codec->reg_cache); codec->reg_cache = NULL; return ret; } #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) static int ak4642_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct ak4642_priv *ak4642; struct snd_soc_codec *codec; int ret; ak4642 = kzalloc(sizeof(struct ak4642_priv), GFP_KERNEL); if (!ak4642) return -ENOMEM; codec = &ak4642->codec; codec->dev = &i2c->dev; i2c_set_clientdata(i2c, ak4642); codec->control_data = i2c; ret = ak4642_init(ak4642); if (ret < 0) printk(KERN_ERR "failed to initialise AK4642\n"); return ret; } static int ak4642_i2c_remove(struct i2c_client *client) { struct ak4642_priv *ak4642 = i2c_get_clientdata(client); snd_soc_unregister_dai(&ak4642_dai); snd_soc_unregister_codec(&ak4642->codec); kfree(ak4642->codec.reg_cache); kfree(ak4642); ak4642_codec = NULL; return 0; } static const struct i2c_device_id ak4642_i2c_id[] = { { "ak4642", 0 }, { "ak4643", 0 }, { } }; MODULE_DEVICE_TABLE(i2c, ak4642_i2c_id); static struct i2c_driver ak4642_i2c_driver = { .driver = { .name = "AK4642 I2C Codec", .owner = THIS_MODULE, }, .probe = ak4642_i2c_probe, .remove = ak4642_i2c_remove, .id_table = ak4642_i2c_id, }; #endif static int ak4642_probe(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); int ret; if (!ak4642_codec) { dev_err(&pdev->dev, "Codec device not registered\n"); return -ENODEV; } socdev->card->codec = ak4642_codec; /* register pcms */ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); if (ret < 0) { printk(KERN_ERR "ak4642: failed to create pcms\n"); goto pcm_err; } dev_info(&pdev->dev, "AK4642 Audio Codec %s", AK4642_VERSION); return ret; pcm_err: return ret; } /* power down chip */ static int ak4642_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); return 0; } struct snd_soc_codec_device soc_codec_dev_ak4642 = { .probe = ak4642_probe, .remove = ak4642_remove, .resume = ak4642_resume, }; EXPORT_SYMBOL_GPL(soc_codec_dev_ak4642); static int __init ak4642_modinit(void) { int ret = 0; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) ret = i2c_add_driver(&ak4642_i2c_driver); #endif return ret; } module_init(ak4642_modinit); static void __exit ak4642_exit(void) { #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) i2c_del_driver(&ak4642_i2c_driver); #endif } module_exit(ak4642_exit); MODULE_DESCRIPTION("Soc AK4642 driver"); MODULE_AUTHOR("Kuninori Morimoto "); MODULE_LICENSE("GPL");