/* * ad1980.c -- ALSA Soc AD1980 codec support * * Copyright: Analog Device Inc. * Author: Roy Huang * Cliff Cai * * 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 #include #include #include #include #include #include #include #include #include #include #include "ad1980.h" static unsigned int ac97_read(struct snd_soc_codec *codec, unsigned int reg); static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int val); /* * AD1980 register cache */ static const u16 ad1980_reg[] = { 0x0090, 0x8000, 0x8000, 0x8000, /* 0 - 6 */ 0x0000, 0x0000, 0x8008, 0x8008, /* 8 - e */ 0x8808, 0x8808, 0x0000, 0x8808, /* 10 - 16 */ 0x8808, 0x0000, 0x8000, 0x0000, /* 18 - 1e */ 0x0000, 0x0000, 0x0000, 0x0000, /* 20 - 26 */ 0x03c7, 0x0000, 0xbb80, 0xbb80, /* 28 - 2e */ 0xbb80, 0xbb80, 0x0000, 0x8080, /* 30 - 36 */ 0x8080, 0x2000, 0x0000, 0x0000, /* 38 - 3e */ 0x0000, 0x0000, 0x0000, 0x0000, /* reserved */ 0x0000, 0x0000, 0x0000, 0x0000, /* reserved */ 0x0000, 0x0000, 0x0000, 0x0000, /* reserved */ 0x0000, 0x0000, 0x0000, 0x0000, /* reserved */ 0x8080, 0x0000, 0x0000, 0x0000, /* 60 - 66 */ 0x0000, 0x0000, 0x0000, 0x0000, /* reserved */ 0x0000, 0x0000, 0x1001, 0x0000, /* 70 - 76 */ 0x0000, 0x0000, 0x4144, 0x5370 /* 78 - 7e */ }; static const char *ad1980_rec_sel[] = {"Mic", "CD", "NC", "AUX", "Line", "Stereo Mix", "Mono Mix", "Phone"}; static const struct soc_enum ad1980_cap_src = SOC_ENUM_DOUBLE(AC97_REC_SEL, 8, 0, 7, ad1980_rec_sel); static const struct snd_kcontrol_new ad1980_snd_ac97_controls[] = { SOC_DOUBLE("Master Playback Volume", AC97_MASTER, 8, 0, 31, 1), SOC_SINGLE("Master Playback Switch", AC97_MASTER, 15, 1, 1), SOC_DOUBLE("Headphone Playback Volume", AC97_HEADPHONE, 8, 0, 31, 1), SOC_SINGLE("Headphone Playback Switch", AC97_HEADPHONE, 15, 1, 1), SOC_DOUBLE("PCM Playback Volume", AC97_PCM, 8, 0, 31, 1), SOC_SINGLE("PCM Playback Switch", AC97_PCM, 15, 1, 1), SOC_DOUBLE("PCM Capture Volume", AC97_REC_GAIN, 8, 0, 31, 0), SOC_SINGLE("PCM Capture Switch", AC97_REC_GAIN, 15, 1, 1), SOC_SINGLE("Mono Playback Volume", AC97_MASTER_MONO, 0, 31, 1), SOC_SINGLE("Mono Playback Switch", AC97_MASTER_MONO, 15, 1, 1), SOC_SINGLE("Phone Capture Volume", AC97_PHONE, 0, 31, 1), SOC_SINGLE("Phone Capture Switch", AC97_PHONE, 15, 1, 1), SOC_SINGLE("Mic Volume", AC97_MIC, 0, 31, 1), SOC_SINGLE("Mic Switch", AC97_MIC, 15, 1, 1), SOC_SINGLE("Stereo Mic Switch", AC97_AD_MISC, 6, 1, 0), SOC_DOUBLE("Line HP Swap Switch", AC97_AD_MISC, 10, 5, 1, 0), SOC_DOUBLE("Surround Playback Volume", AC97_SURROUND_MASTER, 8, 0, 31, 1), SOC_DOUBLE("Surround Playback Switch", AC97_SURROUND_MASTER, 15, 7, 1, 1), SOC_DOUBLE("Center/LFE Playback Volume", AC97_CENTER_LFE_MASTER, 8, 0, 31, 1), SOC_DOUBLE("Center/LFE Playback Switch", AC97_CENTER_LFE_MASTER, 15, 7, 1, 1), SOC_ENUM("Capture Source", ad1980_cap_src), SOC_SINGLE("Mic Boost Switch", AC97_MIC, 6, 1, 0), }; static unsigned int ac97_read(struct snd_soc_codec *codec, unsigned int reg) { u16 *cache = codec->reg_cache; switch (reg) { case AC97_RESET: case AC97_INT_PAGING: case AC97_POWERDOWN: case AC97_EXTENDED_STATUS: case AC97_VENDOR_ID1: case AC97_VENDOR_ID2: return soc_ac97_ops.read(codec->ac97, reg); default: reg = reg >> 1; if (reg >= ARRAY_SIZE(ad1980_reg)) return -EINVAL; return cache[reg]; } } static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int val) { u16 *cache = codec->reg_cache; soc_ac97_ops.write(codec->ac97, reg, val); reg = reg >> 1; if (reg < ARRAY_SIZE(ad1980_reg)) cache[reg] = val; return 0; } struct snd_soc_dai ad1980_dai = { .name = "AC97", .ac97_control = 1, .playback = { .stream_name = "Playback", .channels_min = 2, .channels_max = 6, .rates = SNDRV_PCM_RATE_48000, .formats = SND_SOC_STD_AC97_FMTS, }, .capture = { .stream_name = "Capture", .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_48000, .formats = SND_SOC_STD_AC97_FMTS, }, }; EXPORT_SYMBOL_GPL(ad1980_dai); static int ad1980_reset(struct snd_soc_codec *codec, int try_warm) { u16 retry_cnt = 0; retry: if (try_warm && soc_ac97_ops.warm_reset) { soc_ac97_ops.warm_reset(codec->ac97); if (ac97_read(codec, AC97_RESET) == 0x0090) return 1; } soc_ac97_ops.reset(codec->ac97); /* Set bit 16slot in register 74h, then every slot will has only 16 * bits. This command is sent out in 20bit mode, in which case the * first nibble of data is eaten by the addr. (Tag is always 16 bit)*/ ac97_write(codec, AC97_AD_SERIAL_CFG, 0x9900); if (ac97_read(codec, AC97_RESET) != 0x0090) goto err; return 0; err: while (retry_cnt++ < 10) goto retry; printk(KERN_ERR "AD1980 AC97 reset failed\n"); return -EIO; } static int ad1980_soc_probe(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec; int ret = 0; u16 vendor_id2; u16 ext_status; printk(KERN_INFO "AD1980 SoC Audio Codec\n"); socdev->card->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); if (socdev->card->codec == NULL) return -ENOMEM; codec = socdev->card->codec; mutex_init(&codec->mutex); codec->reg_cache = kzalloc(sizeof(u16) * ARRAY_SIZE(ad1980_reg), GFP_KERNEL); if (codec->reg_cache == NULL) { ret = -ENOMEM; goto cache_err; } memcpy(codec->reg_cache, ad1980_reg, sizeof(u16) * \ ARRAY_SIZE(ad1980_reg)); codec->reg_cache_size = sizeof(u16) * ARRAY_SIZE(ad1980_reg); codec->reg_cache_step = 2; codec->name = "AD1980"; codec->owner = THIS_MODULE; codec->dai = &ad1980_dai; codec->num_dai = 1; codec->write = ac97_write; codec->read = ac97_read; INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); if (ret < 0) { printk(KERN_ERR "ad1980: failed to register AC97 codec\n"); goto codec_err; } /* register pcms */ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); if (ret < 0) goto pcm_err; ret = ad1980_reset(codec, 0); if (ret < 0) { printk(KERN_ERR "Failed to reset AD1980: AC97 link error\n"); goto reset_err; } /* Read out vendor ID to make sure it is ad1980 */ if (ac97_read(codec, AC97_VENDOR_ID1) != 0x4144) goto reset_err; vendor_id2 = ac97_read(codec, AC97_VENDOR_ID2); if (vendor_id2 != 0x5370) { if (vendor_id2 != 0x5374) goto reset_err; else printk(KERN_WARNING "ad1980: " "Found AD1981 - only 2/2 IN/OUT Channels " "supported\n"); } /* unmute captures and playbacks volume */ ac97_write(codec, AC97_MASTER, 0x0000); ac97_write(codec, AC97_PCM, 0x0000); ac97_write(codec, AC97_REC_GAIN, 0x0000); ac97_write(codec, AC97_CENTER_LFE_MASTER, 0x0000); ac97_write(codec, AC97_SURROUND_MASTER, 0x0000); /*power on LFE/CENTER/Surround DACs*/ ext_status = ac97_read(codec, AC97_EXTENDED_STATUS); ac97_write(codec, AC97_EXTENDED_STATUS, ext_status&~0x3800); snd_soc_add_controls(codec, ad1980_snd_ac97_controls, ARRAY_SIZE(ad1980_snd_ac97_controls)); return 0; reset_err: snd_soc_free_pcms(socdev); pcm_err: snd_soc_free_ac97_codec(codec); codec_err: kfree(codec->reg_cache); cache_err: kfree(socdev->card->codec); socdev->card->codec = NULL; return ret; } static int ad1980_soc_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->card->codec; if (codec == NULL) return 0; snd_soc_dapm_free(socdev); snd_soc_free_pcms(socdev); snd_soc_free_ac97_codec(codec); kfree(codec->reg_cache); kfree(codec); return 0; } struct snd_soc_codec_device soc_codec_dev_ad1980 = { .probe = ad1980_soc_probe, .remove = ad1980_soc_remove, }; EXPORT_SYMBOL_GPL(soc_codec_dev_ad1980); MODULE_DESCRIPTION("ASoC ad1980 driver"); MODULE_AUTHOR("Roy Huang, Cliff Cai"); MODULE_LICENSE("GPL");