Merge commit 'linus/master' into merge-linus

Conflicts:

	arch/x86/kvm/i8254.c

1 files changed, 308 insertions, 0 deletions

diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c
new file mode 100644
index 000000000000..1397b8e06c0b
--- /dev/null
+++ b/sound/soc/codecs/ad1980.c
@@ -0,0 +1,308 @@
+/*
+ * 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/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");