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Release 4.7 sound/soc/codecs/ad1980.c

Directory: sound/soc/codecs
/*
 * 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.
 */

/*
 * WARNING:
 *
 * Because Analog Devices Inc. discontinued the ad1980 sound chip since
 * Sep. 2009, this ad1980 driver is not maintained, tested and supported
 * by ADI now.
 */

#include <linux/init.h>
#include <linux/slab.h>
#include <linux/module.h>
#include <linux/kernel.h>
#include <linux/device.h>
#include <linux/regmap.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/ac97_codec.h>
#include <sound/initval.h>
#include <sound/soc.h>


static const struct reg_default ad1980_reg_defaults[] = {
	{ 0x02, 0x8000 },
	{ 0x04, 0x8000 },
	{ 0x06, 0x8000 },
	{ 0x0c, 0x8008 },
	{ 0x0e, 0x8008 },
	{ 0x10, 0x8808 },
	{ 0x12, 0x8808 },
	{ 0x16, 0x8808 },
	{ 0x18, 0x8808 },
	{ 0x1a, 0x0000 },
	{ 0x1c, 0x8000 },
	{ 0x20, 0x0000 },
	{ 0x28, 0x03c7 },
	{ 0x2c, 0xbb80 },
	{ 0x2e, 0xbb80 },
	{ 0x30, 0xbb80 },
	{ 0x32, 0xbb80 },
	{ 0x36, 0x8080 },
	{ 0x38, 0x8080 },
	{ 0x3a, 0x2000 },
	{ 0x60, 0x0000 },
	{ 0x62, 0x0000 },
	{ 0x72, 0x0000 },
	{ 0x74, 0x1001 },
	{ 0x76, 0x0000 },
};


static bool ad1980_readable_reg(struct device *dev, unsigned int reg) { switch (reg) { case AC97_RESET ... AC97_MASTER_MONO: case AC97_PHONE ... AC97_CD: case AC97_AUX ... AC97_GENERAL_PURPOSE: case AC97_POWERDOWN ... AC97_PCM_LR_ADC_RATE: case AC97_SPDIF: case AC97_CODEC_CLASS_REV: case AC97_PCI_SVID: case AC97_AD_CODEC_CFG: case AC97_AD_JACK_SPDIF: case AC97_AD_SERIAL_CFG: case AC97_VENDOR_ID1: case AC97_VENDOR_ID2: return true; default: return false; } }

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static bool ad1980_writeable_reg(struct device *dev, unsigned int reg) { switch (reg) { case AC97_VENDOR_ID1: case AC97_VENDOR_ID2: return false; default: return ad1980_readable_reg(dev, reg); } }

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static const struct regmap_config ad1980_regmap_config = { .reg_bits = 16, .reg_stride = 2, .val_bits = 16, .max_register = 0x7e, .cache_type = REGCACHE_RBTREE, .volatile_reg = regmap_ac97_default_volatile, .readable_reg = ad1980_readable_reg, .writeable_reg = ad1980_writeable_reg, .reg_defaults = ad1980_reg_defaults, .num_reg_defaults = ARRAY_SIZE(ad1980_reg_defaults), }; static const char *ad1980_rec_sel[] = {"Mic", "CD", "NC", "AUX", "Line", "Stereo Mix", "Mono Mix", "Phone"}; static SOC_ENUM_DOUBLE_DECL(ad1980_cap_src, AC97_REC_SEL, 8, 0, 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 const struct snd_soc_dapm_widget ad1980_dapm_widgets[] = { SND_SOC_DAPM_INPUT("MIC1"), SND_SOC_DAPM_INPUT("MIC2"), SND_SOC_DAPM_INPUT("CD_L"), SND_SOC_DAPM_INPUT("CD_R"), SND_SOC_DAPM_INPUT("AUX_L"), SND_SOC_DAPM_INPUT("AUX_R"), SND_SOC_DAPM_INPUT("LINE_IN_L"), SND_SOC_DAPM_INPUT("LINE_IN_R"), SND_SOC_DAPM_OUTPUT("LFE_OUT"), SND_SOC_DAPM_OUTPUT("CENTER_OUT"), SND_SOC_DAPM_OUTPUT("LINE_OUT_L"), SND_SOC_DAPM_OUTPUT("LINE_OUT_R"), SND_SOC_DAPM_OUTPUT("MONO_OUT"), SND_SOC_DAPM_OUTPUT("HP_OUT_L"), SND_SOC_DAPM_OUTPUT("HP_OUT_R"), }; static const struct snd_soc_dapm_route ad1980_dapm_routes[] = { { "Capture", NULL, "MIC1" }, { "Capture", NULL, "MIC2" }, { "Capture", NULL, "CD_L" }, { "Capture", NULL, "CD_R" }, { "Capture", NULL, "AUX_L" }, { "Capture", NULL, "AUX_R" }, { "Capture", NULL, "LINE_IN_L" }, { "Capture", NULL, "LINE_IN_R" }, { "LFE_OUT", NULL, "Playback" }, { "CENTER_OUT", NULL, "Playback" }, { "LINE_OUT_L", NULL, "Playback" }, { "LINE_OUT_R", NULL, "Playback" }, { "MONO_OUT", NULL, "Playback" }, { "HP_OUT_L", NULL, "Playback" }, { "HP_OUT_R", NULL, "Playback" }, }; static struct snd_soc_dai_driver ad1980_dai = { .name = "ad1980-hifi", .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, }, }; #define AD1980_VENDOR_ID 0x41445300 #define AD1980_VENDOR_MASK 0xffffff00
static int ad1980_reset(struct snd_soc_codec *codec, int try_warm) { struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); unsigned int retry_cnt = 0; int ret; do { ret = snd_ac97_reset(ac97, true, AD1980_VENDOR_ID, AD1980_VENDOR_MASK); if (ret >= 0) return 0; /* * 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) */ snd_soc_write(codec, AC97_AD_SERIAL_CFG, 0x9900); } while (retry_cnt++ < 10); dev_err(codec->dev, "Failed to reset: AC97 link error\n"); return -EIO; }

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static int ad1980_soc_probe(struct snd_soc_codec *codec) { struct snd_ac97 *ac97; struct regmap *regmap; int ret; u16 vendor_id2; u16 ext_status; ac97 = snd_soc_new_ac97_codec(codec, 0, 0); if (IS_ERR(ac97)) { ret = PTR_ERR(ac97); dev_err(codec->dev, "Failed to register AC97 codec: %d\n", ret); return ret; } regmap = regmap_init_ac97(ac97, &ad1980_regmap_config); if (IS_ERR(regmap)) { ret = PTR_ERR(regmap); goto err_free_ac97; } snd_soc_codec_init_regmap(codec, regmap); snd_soc_codec_set_drvdata(codec, ac97); ret = ad1980_reset(codec, 0); if (ret < 0) goto reset_err; vendor_id2 = snd_soc_read(codec, AC97_VENDOR_ID2); if (vendor_id2 == 0x5374) { dev_warn(codec->dev, "Found AD1981 - only 2/2 IN/OUT Channels supported\n"); } /* unmute captures and playbacks volume */ snd_soc_write(codec, AC97_MASTER, 0x0000); snd_soc_write(codec, AC97_PCM, 0x0000); snd_soc_write(codec, AC97_REC_GAIN, 0x0000); snd_soc_write(codec, AC97_CENTER_LFE_MASTER, 0x0000); snd_soc_write(codec, AC97_SURROUND_MASTER, 0x0000); /*power on LFE/CENTER/Surround DACs*/ ext_status = snd_soc_read(codec, AC97_EXTENDED_STATUS); snd_soc_write(codec, AC97_EXTENDED_STATUS, ext_status&~0x3800); return 0; reset_err: snd_soc_codec_exit_regmap(codec); err_free_ac97: snd_soc_free_ac97_codec(ac97); return ret; }

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axel linaxel lin10.41%110.00%
Total246100.00%10100.00%


static int ad1980_soc_remove(struct snd_soc_codec *codec) { struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); snd_soc_codec_exit_regmap(codec); snd_soc_free_ac97_codec(ac97); return 0; }

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static struct snd_soc_codec_driver soc_codec_dev_ad1980 = { .probe = ad1980_soc_probe, .remove = ad1980_soc_remove, .controls = ad1980_snd_ac97_controls, .num_controls = ARRAY_SIZE(ad1980_snd_ac97_controls), .dapm_widgets = ad1980_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(ad1980_dapm_widgets), .dapm_routes = ad1980_dapm_routes, .num_dapm_routes = ARRAY_SIZE(ad1980_dapm_routes), };
static int ad1980_probe(struct platform_device *pdev) { return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_ad1980, &ad1980_dai, 1); }

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static int ad1980_remove(struct platform_device *pdev) { snd_soc_unregister_codec(&pdev->dev); return 0; }

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static struct platform_driver ad1980_codec_driver = { .driver = { .name = "ad1980", }, .probe = ad1980_probe, .remove = ad1980_remove, }; module_platform_driver(ad1980_codec_driver); MODULE_DESCRIPTION("ASoC ad1980 driver (Obsolete)"); MODULE_AUTHOR("Roy Huang, Cliff Cai"); MODULE_LICENSE("GPL");

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lars-peter clausenlars-peter clausen43728.62%840.00%
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liam girdwoodliam girdwood895.83%15.00%
takashi iwaitakashi iwai40.26%15.00%
tejun heotejun heo30.20%15.00%
sonic zhangsonic zhang20.13%15.00%
mike frysingermike frysinger10.07%15.00%
axel linaxel lin10.07%15.00%
Total1527100.00%20100.00%
Directory: sound/soc/codecs
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