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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 18:49:45 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 18:49:45 +0000 |
commit | 2c3c1048746a4622d8c89a29670120dc8fab93c4 (patch) | |
tree | 848558de17fb3008cdf4d861b01ac7781903ce39 /sound/soc/codecs/ad1980.c | |
parent | Initial commit. (diff) | |
download | linux-2c3c1048746a4622d8c89a29670120dc8fab93c4.tar.xz linux-2c3c1048746a4622d8c89a29670120dc8fab93c4.zip |
Adding upstream version 6.1.76.upstream/6.1.76
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'sound/soc/codecs/ad1980.c')
-rw-r--r-- | sound/soc/codecs/ad1980.c | 325 |
1 files changed, 325 insertions, 0 deletions
diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c new file mode 100644 index 000000000..5e777d7fd --- /dev/null +++ b/sound/soc/codecs/ad1980.c @@ -0,0 +1,325 @@ +// SPDX-License-Identifier: GPL-2.0-or-later +/* + * ad1980.c -- ALSA Soc AD1980 codec support + * + * Copyright: Analog Devices Inc. + * Author: Roy Huang <roy.huang@analog.com> + * Cliff Cai <cliff.cai@analog.com> + */ + +/* + * 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; + } +} + +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); + } +} + +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_component *component, int try_warm) +{ + struct snd_ac97 *ac97 = snd_soc_component_get_drvdata(component); + 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_component_write(component, AC97_AD_SERIAL_CFG, 0x9900); + + } while (retry_cnt++ < 10); + + dev_err(component->dev, "Failed to reset: AC97 link error\n"); + + return -EIO; +} + +static int ad1980_soc_probe(struct snd_soc_component *component) +{ + struct snd_ac97 *ac97; + struct regmap *regmap; + int ret; + u16 vendor_id2; + u16 ext_status; + + ac97 = snd_soc_new_ac97_component(component, 0, 0); + if (IS_ERR(ac97)) { + ret = PTR_ERR(ac97); + dev_err(component->dev, "Failed to register AC97 component: %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_component_init_regmap(component, regmap); + snd_soc_component_set_drvdata(component, ac97); + + ret = ad1980_reset(component, 0); + if (ret < 0) + goto reset_err; + + vendor_id2 = snd_soc_component_read(component, AC97_VENDOR_ID2); + if (vendor_id2 == 0x5374) { + dev_warn(component->dev, + "Found AD1981 - only 2/2 IN/OUT Channels supported\n"); + } + + /* unmute captures and playbacks volume */ + snd_soc_component_write(component, AC97_MASTER, 0x0000); + snd_soc_component_write(component, AC97_PCM, 0x0000); + snd_soc_component_write(component, AC97_REC_GAIN, 0x0000); + snd_soc_component_write(component, AC97_CENTER_LFE_MASTER, 0x0000); + snd_soc_component_write(component, AC97_SURROUND_MASTER, 0x0000); + + /*power on LFE/CENTER/Surround DACs*/ + ext_status = snd_soc_component_read(component, AC97_EXTENDED_STATUS); + snd_soc_component_write(component, AC97_EXTENDED_STATUS, ext_status&~0x3800); + + return 0; + +reset_err: + snd_soc_component_exit_regmap(component); +err_free_ac97: + snd_soc_free_ac97_component(ac97); + return ret; +} + +static void ad1980_soc_remove(struct snd_soc_component *component) +{ + struct snd_ac97 *ac97 = snd_soc_component_get_drvdata(component); + + snd_soc_component_exit_regmap(component); + snd_soc_free_ac97_component(ac97); +} + +static const struct snd_soc_component_driver soc_component_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), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, +}; + +static int ad1980_probe(struct platform_device *pdev) +{ + return devm_snd_soc_register_component(&pdev->dev, + &soc_component_dev_ad1980, &ad1980_dai, 1); +} + +static struct platform_driver ad1980_codec_driver = { + .driver = { + .name = "ad1980", + }, + + .probe = ad1980_probe, +}; + +module_platform_driver(ad1980_codec_driver); + +MODULE_DESCRIPTION("ASoC ad1980 driver (Obsolete)"); +MODULE_AUTHOR("Roy Huang, Cliff Cai"); +MODULE_LICENSE("GPL"); |