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author | David S. Miller <davem@davemloft.net> | 2008-10-11 12:39:35 -0700 |
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committer | David S. Miller <davem@davemloft.net> | 2008-10-11 12:39:35 -0700 |
commit | 56c5d900dbb8e042bfad035d18433476931d8f93 (patch) | |
tree | 00b793965beeef10db03e0ff021d2d965c410759 /sound/soc/codecs/ad1980.c | |
parent | 4dd95b63ae25c5cad6986829b5e8788e9faa0330 (diff) | |
parent | ead9d23d803ea3a73766c3cb27bf7563ac8d7266 (diff) | |
download | op-kernel-dev-56c5d900dbb8e042bfad035d18433476931d8f93.zip op-kernel-dev-56c5d900dbb8e042bfad035d18433476931d8f93.tar.gz |
Merge branch 'master' of master.kernel.org:/pub/scm/linux/kernel/git/torvalds/linux-2.6
Conflicts:
sound/core/memalloc.c
Diffstat (limited to 'sound/soc/codecs/ad1980.c')
-rw-r--r-- | sound/soc/codecs/ad1980.c | 309 |
1 files changed, 309 insertions, 0 deletions
diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c new file mode 100644 index 0000000..4e09c1f --- /dev/null +++ b/sound/soc/codecs/ad1980.c @@ -0,0 +1,309 @@ +/* + * ad1980.c -- ALSA Soc AD1980 codec support + * + * Copyright: Analog Device Inc. + * Author: Roy Huang <roy.huang@analog.com> + * Cliff Cai <cliff.cai@analog.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. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/version.h> +#include <linux/kernel.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/ac97_codec.h> +#include <sound/initval.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#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_ENUM("Capture Source", ad1980_cap_src), + +SOC_SINGLE("Mic Boost Switch", AC97_MIC, 6, 1, 0), +}; + +/* add non dapm controls */ +static int ad1980_add_controls(struct snd_soc_codec *codec) +{ + int err, i; + + for (i = 0; i < ARRAY_SIZE(ad1980_snd_ac97_controls); i++) { + err = snd_ctl_add(codec->card, snd_soc_cnew( + &ad1980_snd_ac97_controls[i], codec, NULL)); + if (err < 0) + return err; + } + return 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", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, }, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, }, +}; +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; + + printk(KERN_INFO "AD1980 SoC Audio Codec\n"); + + socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (socdev->codec == NULL) + return -ENOMEM; + codec = socdev->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 "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"); + } + + ac97_write(codec, AC97_MASTER, 0x0000); /* unmute line out volume */ + ac97_write(codec, AC97_PCM, 0x0000); /* unmute PCM out volume */ + ac97_write(codec, AC97_REC_GAIN, 0x0000);/* unmute record volume */ + + ad1980_add_controls(codec); + ret = snd_soc_register_card(socdev); + if (ret < 0) { + printk(KERN_ERR "ad1980: failed to register card\n"); + goto reset_err; + } + + 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->codec); + socdev->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->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"); |