diff options
Diffstat (limited to 'sound/soc/codecs')
73 files changed, 6966 insertions, 1480 deletions
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 59d8efaa17e9..1e1613a438dd 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -29,6 +29,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_ALC5632 if I2C select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC select SND_SOC_CS42L51 if I2C + select SND_SOC_CS42L52 if I2C select SND_SOC_CS42L73 if I2C select SND_SOC_CS4270 if I2C select SND_SOC_CS4271 if SND_SOC_I2C_AND_SPI @@ -37,11 +38,15 @@ config SND_SOC_ALL_CODECS select SND_SOC_DFBMCS320 select SND_SOC_JZ4740_CODEC select SND_SOC_LM4857 if I2C + select SND_SOC_LM49453 if I2C select SND_SOC_MAX98088 if I2C select SND_SOC_MAX98095 if I2C select SND_SOC_MAX9850 if I2C select SND_SOC_MAX9768 if I2C select SND_SOC_MAX9877 if I2C + select SND_SOC_MC13783 if MFD_MC13XXX + select SND_SOC_ML26124 if I2C + select SND_SOC_OMAP_HDMI_CODEC if OMAP4_DSS_HDMI select SND_SOC_PCM3008 select SND_SOC_RT5631 if I2C select SND_SOC_SGTL5000 if I2C @@ -181,6 +186,9 @@ config SND_SOC_CQ0093VC config SND_SOC_CS42L51 tristate +config SND_SOC_CS42L52 + tristate + config SND_SOC_CS42L73 tristate @@ -217,6 +225,9 @@ config SND_SOC_DFBMCS320 config SND_SOC_DMIC tristate +config SND_SOC_LM49453 + tristate + config SND_SOC_MAX98088 tristate @@ -226,6 +237,9 @@ config SND_SOC_MAX98095 config SND_SOC_MAX9850 tristate +config SND_SOC_OMAP_HDMI_CODEC + tristate + config SND_SOC_PCM3008 tristate @@ -435,5 +449,11 @@ config SND_SOC_MAX9768 config SND_SOC_MAX9877 tristate +config SND_SOC_MC13783 + tristate + +config SND_SOC_ML26124 + tristate + config SND_SOC_TPA6130A2 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 6662eb0cdcc0..fc27fec39487 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -15,6 +15,7 @@ snd-soc-ak4642-objs := ak4642.o snd-soc-ak4671-objs := ak4671.o snd-soc-cq93vc-objs := cq93vc.o snd-soc-cs42l51-objs := cs42l51.o +snd-soc-cs42l52-objs := cs42l52.o snd-soc-cs42l73-objs := cs42l73.o snd-soc-cs4270-objs := cs4270.o snd-soc-cs4271-objs := cs4271.o @@ -25,10 +26,14 @@ snd-soc-dmic-objs := dmic.o snd-soc-jz4740-codec-objs := jz4740.o snd-soc-l3-objs := l3.o snd-soc-lm4857-objs := lm4857.o +snd-soc-lm49453-objs := lm49453.o snd-soc-max9768-objs := max9768.o snd-soc-max98088-objs := max98088.o snd-soc-max98095-objs := max98095.o snd-soc-max9850-objs := max9850.o +snd-soc-mc13783-objs := mc13783.o +snd-soc-ml26124-objs := ml26124.o +snd-soc-omap-hdmi-codec-objs := omap-hdmi.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-rt5631-objs := rt5631.o snd-soc-sgtl5000-objs := sgtl5000.o @@ -121,6 +126,7 @@ obj-$(CONFIG_SND_SOC_ALC5623) += snd-soc-alc5623.o obj-$(CONFIG_SND_SOC_ALC5632) += snd-soc-alc5632.o obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o +obj-$(CONFIG_SND_SOC_CS42L52) += snd-soc-cs42l52.o obj-$(CONFIG_SND_SOC_CS42L73) += snd-soc-cs42l73.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_CS4271) += snd-soc-cs4271.o @@ -128,13 +134,17 @@ obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o obj-$(CONFIG_SND_SOC_DFBMCS320) += snd-soc-dfbmcs320.o obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o +obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o obj-$(CONFIG_SND_SOC_LM4857) += snd-soc-lm4857.o -obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o +obj-$(CONFIG_SND_SOC_LM49453) += snd-soc-lm49453.o obj-$(CONFIG_SND_SOC_MAX9768) += snd-soc-max9768.o obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o obj-$(CONFIG_SND_SOC_MAX98095) += snd-soc-max98095.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o +obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o +obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o +obj-$(CONFIG_SND_SOC_OMAP_HDMI_CODEC) += snd-soc-omap-hdmi-codec.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index 1bbad4c16d28..2023c749f232 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -26,13 +26,11 @@ static int ac97_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; int reg = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? AC97_PCM_FRONT_DAC_RATE : AC97_PCM_LR_ADC_RATE; - return snd_ac97_set_rate(codec->ac97, reg, runtime->rate); + return snd_ac97_set_rate(codec->ac97, reg, substream->runtime->rate); } #define STD_AC97_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c index 12e3b4118557..c67b50d8b317 100644 --- a/sound/soc/codecs/ad1836.c +++ b/sound/soc/codecs/ad1836.c @@ -162,9 +162,7 @@ static int ad1836_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { int word_len = 0; - - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; /* bit size */ switch (params_format(params)) { diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index a4a6bef2c0bb..13e62be4f990 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -245,9 +245,7 @@ static int ad193x_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { int word_len = 0, master_rate = 0; - - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct ad193x_priv *ad193x = snd_soc_codec_get_drvdata(codec); /* bit size */ diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index 78e9ce48bb99..3d50fc8646b6 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -258,8 +258,7 @@ static int adau1701_set_playback_pcm_format(struct snd_soc_codec *codec, static int adau1701_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; snd_pcm_format_t format; unsigned int val; diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c index ceb96ecf5588..31d4483245d0 100644 --- a/sound/soc/codecs/ak4104.c +++ b/sound/soc/codecs/ak4104.c @@ -88,8 +88,7 @@ static int ak4104_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; int val = 0; /* set the IEC958 bits: consumer mode, no copyright bit */ diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index 838ae8b22b50..618fdc30f73e 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -262,8 +262,7 @@ static int ak4535_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct ak4535_priv *ak4535 = snd_soc_codec_get_drvdata(codec); u8 mode2 = snd_soc_read(codec, AK4535_MODE2) & ~(0x3 << 5); int rate = params_rate(params), fs = 256; diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c index c4d165a4bddf..543a12f471be 100644 --- a/sound/soc/codecs/ak4641.c +++ b/sound/soc/codecs/ak4641.c @@ -296,8 +296,7 @@ static int ak4641_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct ak4641_priv *ak4641 = snd_soc_codec_get_drvdata(codec); int rate = params_rate(params), fs = 256; u8 mode2; @@ -517,67 +516,24 @@ static int ak4641_resume(struct snd_soc_codec *codec) static int ak4641_probe(struct snd_soc_codec *codec) { - struct ak4641_platform_data *pdata = codec->dev->platform_data; int ret; - - if (pdata) { - if (gpio_is_valid(pdata->gpio_power)) { - ret = gpio_request_one(pdata->gpio_power, - GPIOF_OUT_INIT_LOW, "ak4641 power"); - if (ret) - goto err_out; - } - if (gpio_is_valid(pdata->gpio_npdn)) { - ret = gpio_request_one(pdata->gpio_npdn, - GPIOF_OUT_INIT_LOW, "ak4641 npdn"); - if (ret) - goto err_gpio; - - udelay(1); /* > 150 ns */ - gpio_set_value(pdata->gpio_npdn, 1); - } - } - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - goto err_register; + return ret; } /* power on device */ ak4641_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; - -err_register: - if (pdata) { - if (gpio_is_valid(pdata->gpio_power)) - gpio_set_value(pdata->gpio_power, 0); - if (gpio_is_valid(pdata->gpio_npdn)) - gpio_free(pdata->gpio_npdn); - } -err_gpio: - if (pdata && gpio_is_valid(pdata->gpio_power)) - gpio_free(pdata->gpio_power); -err_out: - return ret; } static int ak4641_remove(struct snd_soc_codec *codec) { - struct ak4641_platform_data *pdata = codec->dev->platform_data; - ak4641_set_bias_level(codec, SND_SOC_BIAS_OFF); - if (pdata) { - if (gpio_is_valid(pdata->gpio_power)) { - gpio_set_value(pdata->gpio_power, 0); - gpio_free(pdata->gpio_power); - } - if (gpio_is_valid(pdata->gpio_npdn)) - gpio_free(pdata->gpio_npdn); - } return 0; } @@ -604,6 +560,7 @@ static struct snd_soc_codec_driver soc_codec_dev_ak4641 = { static int __devinit ak4641_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { + struct ak4641_platform_data *pdata = i2c->dev.platform_data; struct ak4641_priv *ak4641; int ret; @@ -612,16 +569,62 @@ static int __devinit ak4641_i2c_probe(struct i2c_client *i2c, if (!ak4641) return -ENOMEM; + if (pdata) { + if (gpio_is_valid(pdata->gpio_power)) { + ret = gpio_request_one(pdata->gpio_power, + GPIOF_OUT_INIT_LOW, "ak4641 power"); + if (ret) + goto err_out; + } + if (gpio_is_valid(pdata->gpio_npdn)) { + ret = gpio_request_one(pdata->gpio_npdn, + GPIOF_OUT_INIT_LOW, "ak4641 npdn"); + if (ret) + goto err_gpio; + + udelay(1); /* > 150 ns */ + gpio_set_value(pdata->gpio_npdn, 1); + } + } + i2c_set_clientdata(i2c, ak4641); ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_ak4641, ak4641_dai, ARRAY_SIZE(ak4641_dai)); + if (ret != 0) + goto err_gpio2; + + return 0; + +err_gpio2: + if (pdata) { + if (gpio_is_valid(pdata->gpio_power)) + gpio_set_value(pdata->gpio_power, 0); + if (gpio_is_valid(pdata->gpio_npdn)) + gpio_free(pdata->gpio_npdn); + } +err_gpio: + if (pdata && gpio_is_valid(pdata->gpio_power)) + gpio_free(pdata->gpio_power); +err_out: return ret; } static int __devexit ak4641_i2c_remove(struct i2c_client *i2c) { + struct ak4641_platform_data *pdata = i2c->dev.platform_data; + snd_soc_unregister_codec(&i2c->dev); + + if (pdata) { + if (gpio_is_valid(pdata->gpio_power)) { + gpio_set_value(pdata->gpio_power, 0); + gpio_free(pdata->gpio_power); + } + if (gpio_is_valid(pdata->gpio_npdn)) + gpio_free(pdata->gpio_npdn); + } + return 0; } @@ -641,23 +644,7 @@ static struct i2c_driver ak4641_i2c_driver = { .id_table = ak4641_i2c_id, }; -static int __init ak4641_modinit(void) -{ - int ret; - - ret = i2c_add_driver(&ak4641_i2c_driver); - if (ret != 0) - pr_err("Failed to register AK4641 I2C driver: %d\n", ret); - - return ret; -} -module_init(ak4641_modinit); - -static void __exit ak4641_exit(void) -{ - i2c_del_driver(&ak4641_i2c_driver); -} -module_exit(ak4641_exit); +module_i2c_driver(ak4641_i2c_driver); MODULE_DESCRIPTION("SoC AK4641 driver"); MODULE_AUTHOR("Harald Welte <laforge@gnufiish.org>"); diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c index d47b62ddb210..1960478ce6bb 100644 --- a/sound/soc/codecs/alc5623.c +++ b/sound/soc/codecs/alc5623.c @@ -705,8 +705,7 @@ static int alc5623_set_dai_fmt(struct snd_soc_dai *codec_dai, static int alc5623_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec); int coeff, rate; u16 iface; @@ -1084,25 +1083,7 @@ static struct i2c_driver alc5623_i2c_driver = { .id_table = alc5623_i2c_table, }; -static int __init alc5623_modinit(void) -{ - int ret; - - ret = i2c_add_driver(&alc5623_i2c_driver); - if (ret != 0) { - printk(KERN_ERR "%s: can't add i2c driver", __func__); - return ret; - } - - return ret; -} -module_init(alc5623_modinit); - -static void __exit alc5623_modexit(void) -{ - i2c_del_driver(&alc5623_i2c_driver); -} -module_exit(alc5623_modexit); +module_i2c_driver(alc5623_i2c_driver); MODULE_DESCRIPTION("ASoC alc5621/2/3 driver"); MODULE_AUTHOR("Arnaud Patard <arnaud.patard@rtp-net.org>"); diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index e2111e0ccad7..7dd02420b36d 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -861,8 +861,7 @@ static int alc5632_set_dai_fmt(struct snd_soc_dai *codec_dai, static int alc5632_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; int coeff, rate; u16 iface; @@ -1131,7 +1130,7 @@ static __devinit int alc5632_i2c_probe(struct i2c_client *client, i2c_set_clientdata(client, alc5632); - alc5632->regmap = regmap_init_i2c(client, &alc5632_regmap); + alc5632->regmap = devm_regmap_init_i2c(client, &alc5632_regmap); if (IS_ERR(alc5632->regmap)) { ret = PTR_ERR(alc5632->regmap); dev_err(&client->dev, "regmap_init() failed: %d\n", ret); @@ -1143,7 +1142,6 @@ static __devinit int alc5632_i2c_probe(struct i2c_client *client, if (ret1 != 0 || ret2 != 0) { dev_err(&client->dev, "Failed to read chip ID: ret1=%d, ret2=%d\n", ret1, ret2); - regmap_exit(alc5632->regmap); return -EIO; } @@ -1152,14 +1150,12 @@ static __devinit int alc5632_i2c_probe(struct i2c_client *client, if ((vid1 != 0x10EC) || (vid2 != id->driver_data)) { dev_err(&client->dev, "Device is not a ALC5632: VID1=0x%x, VID2=0x%x\n", vid1, vid2); - regmap_exit(alc5632->regmap); return -EINVAL; } ret = alc5632_reset(alc5632->regmap); if (ret < 0) { dev_err(&client->dev, "Failed to issue reset\n"); - regmap_exit(alc5632->regmap); return ret; } @@ -1177,7 +1173,6 @@ static __devinit int alc5632_i2c_probe(struct i2c_client *client, if (ret < 0) { dev_err(&client->dev, "Failed to register codec: %d\n", ret); - regmap_exit(alc5632->regmap); return ret; } @@ -1186,9 +1181,7 @@ static __devinit int alc5632_i2c_probe(struct i2c_client *client, static __devexit int alc5632_i2c_remove(struct i2c_client *client) { - struct alc5632_priv *alc5632 = i2c_get_clientdata(client); snd_soc_unregister_codec(&client->dev); - regmap_exit(alc5632->regmap); return 0; } @@ -1209,25 +1202,7 @@ static struct i2c_driver alc5632_i2c_driver = { .id_table = alc5632_i2c_table, }; -static int __init alc5632_modinit(void) -{ - int ret; - - ret = i2c_add_driver(&alc5632_i2c_driver); - if (ret != 0) { - printk(KERN_ERR "%s: can't add i2c driver", __func__); - return ret; - } - - return ret; -} -module_init(alc5632_modinit); - -static void __exit alc5632_modexit(void) -{ - i2c_del_driver(&alc5632_i2c_driver); -} -module_exit(alc5632_modexit); +module_i2c_driver(alc5632_i2c_driver); MODULE_DESCRIPTION("ASoC ALC5632 driver"); MODULE_AUTHOR("Leon Romanovsky <leon@leon.nu>"); diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 1d672f528662..047917f0b8ae 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -307,8 +307,7 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct cs4270_private *cs4270 = snd_soc_codec_get_drvdata(codec); int ret; unsigned int i; @@ -600,10 +599,12 @@ static int cs4270_soc_suspend(struct snd_soc_codec *codec) static int cs4270_soc_resume(struct snd_soc_codec *codec) { struct cs4270_private *cs4270 = snd_soc_codec_get_drvdata(codec); - int reg; + int reg, ret; - regulator_bulk_enable(ARRAY_SIZE(cs4270->supplies), - cs4270->supplies); + ret = regulator_bulk_enable(ARRAY_SIZE(cs4270->supplies), + cs4270->supplies); + if (ret != 0) + return ret; /* In case the device was put to hard reset during sleep, we need to * wait 500ns here before any I2C communication. */ diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index bf7141280a74..9eb01d7d58a3 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -318,8 +318,7 @@ static int cs4271_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); int i, ret; unsigned int ratio, val; diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index a8bf588e8740..091d0193f507 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -141,15 +141,15 @@ static const struct soc_enum cs42l51_chan_mix = static const struct snd_kcontrol_new cs42l51_snd_controls[] = { SOC_DOUBLE_R_SX_TLV("PCM Playback Volume", CS42L51_PCMA_VOL, CS42L51_PCMB_VOL, - 7, 0xffffff99, 0x18, adc_pcm_tlv), + 6, 0x19, 0x7F, adc_pcm_tlv), SOC_DOUBLE_R("PCM Playback Switch", CS42L51_PCMA_VOL, CS42L51_PCMB_VOL, 7, 1, 1), SOC_DOUBLE_R_SX_TLV("Analog Playback Volume", CS42L51_AOUTA_VOL, CS42L51_AOUTB_VOL, - 8, 0xffffff19, 0x18, aout_tlv), + 0, 0x34, 0xE4, aout_tlv), SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume", CS42L51_ADCA_VOL, CS42L51_ADCB_VOL, - 7, 0xffffff99, 0x18, adc_pcm_tlv), + 6, 0x19, 0x7F, adc_pcm_tlv), SOC_DOUBLE_R("ADC Mixer Switch", CS42L51_ADCA_VOL, CS42L51_ADCB_VOL, 7, 1, 1), SOC_SINGLE("Playback Deemphasis Switch", CS42L51_DAC_CTL, 3, 1, 0), @@ -356,8 +356,7 @@ static int cs42l51_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct cs42l51_private *cs42l51 = snd_soc_codec_get_drvdata(codec); int ret; unsigned int i; diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c new file mode 100644 index 000000000000..a7109413aef1 --- /dev/null +++ b/sound/soc/codecs/cs42l52.c @@ -0,0 +1,1295 @@ +/* + * cs42l52.c -- CS42L52 ALSA SoC audio driver + * + * Copyright 2012 CirrusLogic, Inc. + * + * Author: Georgi Vlaev <joe@nucleusys.com> + * Author: Brian Austin <brian.austin@cirrus.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/version.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/input.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <linux/workqueue.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <sound/cs42l52.h> +#include "cs42l52.h" + +struct sp_config { + u8 spc, format, spfs; + u32 srate; +}; + +struct cs42l52_private { + struct regmap *regmap; + struct snd_soc_codec *codec; + struct device *dev; + struct sp_config config; + struct cs42l52_platform_data pdata; + u32 sysclk; + u8 mclksel; + u32 mclk; + u8 flags; +#if defined(CONFIG_INPUT) || defined(CONFIG_INPUT_MODULE) + struct input_dev *beep; + struct work_struct beep_work; + int beep_rate; +#endif +}; + +static const struct reg_default cs42l52_reg_defaults[] = { + { CS42L52_PWRCTL1, 0x9F }, /* r02 PWRCTL 1 */ + { CS42L52_PWRCTL2, 0x07 }, /* r03 PWRCTL 2 */ + { CS42L52_PWRCTL3, 0xFF }, /* r04 PWRCTL 3 */ + { CS42L52_CLK_CTL, 0xA0 }, /* r05 Clocking Ctl */ + { CS42L52_IFACE_CTL1, 0x00 }, /* r06 Interface Ctl 1 */ + { CS42L52_ADC_PGA_A, 0x80 }, /* r08 Input A Select */ + { CS42L52_ADC_PGA_B, 0x80 }, /* r09 Input B Select */ + { CS42L52_ANALOG_HPF_CTL, 0xA5 }, /* r0A Analog HPF Ctl */ + { CS42L52_ADC_HPF_FREQ, 0x00 }, /* r0B ADC HPF Corner Freq */ + { CS42L52_ADC_MISC_CTL, 0x00 }, /* r0C Misc. ADC Ctl */ + { CS42L52_PB_CTL1, 0x60 }, /* r0D Playback Ctl 1 */ + { CS42L52_MISC_CTL, 0x02 }, /* r0E Misc. Ctl */ + { CS42L52_PB_CTL2, 0x00 }, /* r0F Playback Ctl 2 */ + { CS42L52_MICA_CTL, 0x00 }, /* r10 MICA Amp Ctl */ + { CS42L52_MICB_CTL, 0x00 }, /* r11 MICB Amp Ctl */ + { CS42L52_PGAA_CTL, 0x00 }, /* r12 PGAA Vol, Misc. */ + { CS42L52_PGAB_CTL, 0x00 }, /* r13 PGAB Vol, Misc. */ + { CS42L52_PASSTHRUA_VOL, 0x00 }, /* r14 Bypass A Vol */ + { CS42L52_PASSTHRUB_VOL, 0x00 }, /* r15 Bypass B Vol */ + { CS42L52_ADCA_VOL, 0x00 }, /* r16 ADCA Volume */ + { CS42L52_ADCB_VOL, 0x00 }, /* r17 ADCB Volume */ + { CS42L52_ADCA_MIXER_VOL, 0x80 }, /* r18 ADCA Mixer Volume */ + { CS42L52_ADCB_MIXER_VOL, 0x80 }, /* r19 ADCB Mixer Volume */ + { CS42L52_PCMA_MIXER_VOL, 0x00 }, /* r1A PCMA Mixer Volume */ + { CS42L52_PCMB_MIXER_VOL, 0x00 }, /* r1B PCMB Mixer Volume */ + { CS42L52_BEEP_FREQ, 0x00 }, /* r1C Beep Freq on Time */ + { CS42L52_BEEP_VOL, 0x00 }, /* r1D Beep Volume off Time */ + { CS42L52_BEEP_TONE_CTL, 0x00 }, /* r1E Beep Tone Cfg. */ + { CS42L52_TONE_CTL, 0x00 }, /* r1F Tone Ctl */ + { CS42L52_MASTERA_VOL, 0x88 }, /* r20 Master A Volume */ + { CS42L52_MASTERB_VOL, 0x00 }, /* r21 Master B Volume */ + { CS42L52_HPA_VOL, 0x00 }, /* r22 Headphone A Volume */ + { CS42L52_HPB_VOL, 0x00 }, /* r23 Headphone B Volume */ + { CS42L52_SPKA_VOL, 0x00 }, /* r24 Speaker A Volume */ + { CS42L52_SPKB_VOL, 0x00 }, /* r25 Speaker B Volume */ + { CS42L52_ADC_PCM_MIXER, 0x00 }, /* r26 Channel Mixer and Swap */ + { CS42L52_LIMITER_CTL1, 0x00 }, /* r27 Limit Ctl 1 Thresholds */ + { CS42L52_LIMITER_CTL2, 0x7F }, /* r28 Limit Ctl 2 Release Rate */ + { CS42L52_LIMITER_AT_RATE, 0xC0 }, /* r29 Limiter Attack Rate */ + { CS42L52_ALC_CTL, 0x00 }, /* r2A ALC Ctl 1 Attack Rate */ + { CS42L52_ALC_RATE, 0x3F }, /* r2B ALC Release Rate */ + { CS42L52_ALC_THRESHOLD, 0x3f }, /* r2C ALC Thresholds */ + { CS42L52_NOISE_GATE_CTL, 0x00 }, /* r2D Noise Gate Ctl */ + { CS42L52_CLK_STATUS, 0x00 }, /* r2E Overflow and Clock Status */ + { CS42L52_BATT_COMPEN, 0x00 }, /* r2F battery Compensation */ + { CS42L52_BATT_LEVEL, 0x00 }, /* r30 VP Battery Level */ + { CS42L52_SPK_STATUS, 0x00 }, /* r31 Speaker Status */ + { CS42L52_TEM_CTL, 0x3B }, /* r32 Temp Ctl */ + { CS42L52_THE_FOLDBACK, 0x00 }, /* r33 Foldback */ +}; + +static bool cs42l52_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS42L52_CHIP: + case CS42L52_PWRCTL1: + case CS42L52_PWRCTL2: + case CS42L52_PWRCTL3: + case CS42L52_CLK_CTL: + case CS42L52_IFACE_CTL1: + case CS42L52_IFACE_CTL2: + case CS42L52_ADC_PGA_A: + case CS42L52_ADC_PGA_B: + case CS42L52_ANALOG_HPF_CTL: + case CS42L52_ADC_HPF_FREQ: + case CS42L52_ADC_MISC_CTL: + case CS42L52_PB_CTL1: + case CS42L52_MISC_CTL: + case CS42L52_PB_CTL2: + case CS42L52_MICA_CTL: + case CS42L52_MICB_CTL: + case CS42L52_PGAA_CTL: + case CS42L52_PGAB_CTL: + case CS42L52_PASSTHRUA_VOL: + case CS42L52_PASSTHRUB_VOL: + case CS42L52_ADCA_VOL: + case CS42L52_ADCB_VOL: + case CS42L52_ADCA_MIXER_VOL: + case CS42L52_ADCB_MIXER_VOL: + case CS42L52_PCMA_MIXER_VOL: + case CS42L52_PCMB_MIXER_VOL: + case CS42L52_BEEP_FREQ: + case CS42L52_BEEP_VOL: + case CS42L52_BEEP_TONE_CTL: + case CS42L52_TONE_CTL: + case CS42L52_MASTERA_VOL: + case CS42L52_MASTERB_VOL: + case CS42L52_HPA_VOL: + case CS42L52_HPB_VOL: + case CS42L52_SPKA_VOL: + case CS42L52_SPKB_VOL: + case CS42L52_ADC_PCM_MIXER: + case CS42L52_LIMITER_CTL1: + case CS42L52_LIMITER_CTL2: + case CS42L52_LIMITER_AT_RATE: + case CS42L52_ALC_CTL: + case CS42L52_ALC_RATE: + case CS42L52_ALC_THRESHOLD: + case CS42L52_NOISE_GATE_CTL: + case CS42L52_CLK_STATUS: + case CS42L52_BATT_COMPEN: + case CS42L52_BATT_LEVEL: + case CS42L52_SPK_STATUS: + case CS42L52_TEM_CTL: + case CS42L52_THE_FOLDBACK: + case CS42L52_CHARGE_PUMP: + return true; + default: + return false; + } +} + +static bool cs42l52_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS42L52_IFACE_CTL2: + case CS42L52_CLK_STATUS: + case CS42L52_BATT_LEVEL: + case CS42L52_SPK_STATUS: + case CS42L52_CHARGE_PUMP: + return 1; + default: + return 0; + } +} + +static DECLARE_TLV_DB_SCALE(hl_tlv, -10200, 50, 0); + +static DECLARE_TLV_DB_SCALE(hpd_tlv, -9600, 50, 1); + +static DECLARE_TLV_DB_SCALE(ipd_tlv, -9600, 100, 0); + +static DECLARE_TLV_DB_SCALE(mic_tlv, 1600, 100, 0); + +static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0); + +static const unsigned int limiter_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 2, TLV_DB_SCALE_ITEM(-3000, 600, 0), + 3, 7, TLV_DB_SCALE_ITEM(-1200, 300, 0), +}; + +static const char * const cs42l52_adca_text[] = { + "Input1A", "Input2A", "Input3A", "Input4A", "PGA Input Left"}; + +static const char * const cs42l52_adcb_text[] = { + "Input1B", "Input2B", "Input3B", "Input4B", "PGA Input Right"}; + +static const struct soc_enum adca_enum = + SOC_ENUM_SINGLE(CS42L52_ADC_PGA_A, 5, + ARRAY_SIZE(cs42l52_adca_text), cs42l52_adca_text); + +static const struct soc_enum adcb_enum = + SOC_ENUM_SINGLE(CS42L52_ADC_PGA_B, 5, + ARRAY_SIZE(cs42l52_adcb_text), cs42l52_adcb_text); + +static const struct snd_kcontrol_new adca_mux = + SOC_DAPM_ENUM("Left ADC Input Capture Mux", adca_enum); + +static const struct snd_kcontrol_new adcb_mux = + SOC_DAPM_ENUM("Right ADC Input Capture Mux", adcb_enum); + +static const char * const mic_bias_level_text[] = { + "0.5 +VA", "0.6 +VA", "0.7 +VA", + "0.8 +VA", "0.83 +VA", "0.91 +VA" +}; + +static const struct soc_enum mic_bias_level_enum = + SOC_ENUM_SINGLE(CS42L52_IFACE_CTL1, 0, + ARRAY_SIZE(mic_bias_level_text), mic_bias_level_text); + +static const char * const cs42l52_mic_text[] = { "Single", "Differential" }; + +static const struct soc_enum mica_enum = + SOC_ENUM_SINGLE(CS42L52_MICA_CTL, 5, + ARRAY_SIZE(cs42l52_mic_text), cs42l52_mic_text); + +static const struct soc_enum micb_enum = + SOC_ENUM_SINGLE(CS42L52_MICB_CTL, 5, + ARRAY_SIZE(cs42l52_mic_text), cs42l52_mic_text); + +static const struct snd_kcontrol_new mica_mux = + SOC_DAPM_ENUM("Left Mic Input Capture Mux", mica_enum); + +static const struct snd_kcontrol_new micb_mux = + SOC_DAPM_ENUM("Right Mic Input Capture Mux", micb_enum); + +static const char * const digital_output_mux_text[] = {"ADC", "DSP"}; + +static const struct soc_enum digital_output_mux_enum = + SOC_ENUM_SINGLE(CS42L52_ADC_MISC_CTL, 6, + ARRAY_SIZE(digital_output_mux_text), + digital_output_mux_text); + +static const struct snd_kcontrol_new digital_output_mux = + SOC_DAPM_ENUM("Digital Output Mux", digital_output_mux_enum); + +static const char * const hp_gain_num_text[] = { + "0.3959", "0.4571", "0.5111", "0.6047", + "0.7099", "0.8399", "1.000", "1.1430" +}; + +static const struct soc_enum hp_gain_enum = + SOC_ENUM_SINGLE(CS42L52_PB_CTL1, 4, + ARRAY_SIZE(hp_gain_num_text), hp_gain_num_text); + +static const char * const beep_pitch_text[] = { + "C4", "C5", "D5", "E5", "F5", "G5", "A5", "B5", + "C6", "D6", "E6", "F6", "G6", "A6", "B6", "C7" +}; + +static const struct soc_enum beep_pitch_enum = + SOC_ENUM_SINGLE(CS42L52_BEEP_FREQ, 4, + ARRAY_SIZE(beep_pitch_text), beep_pitch_text); + +static const char * const beep_ontime_text[] = { + "86 ms", "430 ms", "780 ms", "1.20 s", "1.50 s", + "1.80 s", "2.20 s", "2.50 s", "2.80 s", "3.20 s", + "3.50 s", "3.80 s", "4.20 s", "4.50 s", "4.80 s", "5.20 s" +}; + +static const struct soc_enum beep_ontime_enum = + SOC_ENUM_SINGLE(CS42L52_BEEP_FREQ, 0, + ARRAY_SIZE(beep_ontime_text), beep_ontime_text); + +static const char * const beep_offtime_text[] = { + "1.23 s", "2.58 s", "3.90 s", "5.20 s", + "6.60 s", "8.05 s", "9.35 s", "10.80 s" +}; + +static const struct soc_enum beep_offtime_enum = + SOC_ENUM_SINGLE(CS42L52_BEEP_VOL, 5, + ARRAY_SIZE(beep_offtime_text), beep_offtime_text); + +static const char * const beep_config_text[] = { + "Off", "Single", "Multiple", "Continuous" +}; + +static const struct soc_enum beep_config_enum = + SOC_ENUM_SINGLE(CS42L52_BEEP_TONE_CTL, 6, + ARRAY_SIZE(beep_config_text), beep_config_text); + +static const char * const beep_bass_text[] = { + "50 Hz", "100 Hz", "200 Hz", "250 Hz" +}; + +static const struct soc_enum beep_bass_enum = + SOC_ENUM_SINGLE(CS42L52_BEEP_TONE_CTL, 1, + ARRAY_SIZE(beep_bass_text), beep_bass_text); + +static const char * const beep_treble_text[] = { + "5 kHz", "7 kHz", "10 kHz", " 15 kHz" +}; + +static const struct soc_enum beep_treble_enum = + SOC_ENUM_SINGLE(CS42L52_BEEP_TONE_CTL, 3, + ARRAY_SIZE(beep_treble_text), beep_treble_text); + +static const char * const ng_threshold_text[] = { + "-34dB", "-37dB", "-40dB", "-43dB", + "-46dB", "-52dB", "-58dB", "-64dB" +}; + +static const struct soc_enum ng_threshold_enum = + SOC_ENUM_SINGLE(CS42L52_NOISE_GATE_CTL, 2, + ARRAY_SIZE(ng_threshold_text), ng_threshold_text); + +static const char * const cs42l52_ng_delay_text[] = { + "50ms", "100ms", "150ms", "200ms"}; + +static const struct soc_enum ng_delay_enum = + SOC_ENUM_SINGLE(CS42L52_NOISE_GATE_CTL, 0, + ARRAY_SIZE(cs42l52_ng_delay_text), cs42l52_ng_delay_text); + +static const char * const cs42l52_ng_type_text[] = { + "Apply Specific", "Apply All" +}; + +static const struct soc_enum ng_type_enum = + SOC_ENUM_SINGLE(CS42L52_NOISE_GATE_CTL, 6, + ARRAY_SIZE(cs42l52_ng_type_text), cs42l52_ng_type_text); + +static const char * const left_swap_text[] = { + "Left", "LR 2", "Right"}; + +static const char * const right_swap_text[] = { + "Right", "LR 2", "Left"}; + +static const unsigned int swap_values[] = { 0, 1, 3 }; + +static const struct soc_enum adca_swap_enum = + SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 2, 1, + ARRAY_SIZE(left_swap_text), + left_swap_text, + swap_values); + +static const struct snd_kcontrol_new adca_mixer = + SOC_DAPM_ENUM("Route", adca_swap_enum); + +static const struct soc_enum pcma_swap_enum = + SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 6, 1, + ARRAY_SIZE(left_swap_text), + left_swap_text, + swap_values); + +static const struct snd_kcontrol_new pcma_mixer = + SOC_DAPM_ENUM("Route", pcma_swap_enum); + +static const struct soc_enum adcb_swap_enum = + SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 0, 1, + ARRAY_SIZE(right_swap_text), + right_swap_text, + swap_values); + +static const struct snd_kcontrol_new adcb_mixer = + SOC_DAPM_ENUM("Route", adcb_swap_enum); + +static const struct soc_enum pcmb_swap_enum = + SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 4, 1, + ARRAY_SIZE(right_swap_text), + right_swap_text, + swap_values); + +static const struct snd_kcontrol_new pcmb_mixer = + SOC_DAPM_ENUM("Route", pcmb_swap_enum); + + +static const struct snd_kcontrol_new passthrul_ctl = + SOC_DAPM_SINGLE("Switch", CS42L52_MISC_CTL, 6, 1, 0); + +static const struct snd_kcontrol_new passthrur_ctl = + SOC_DAPM_SINGLE("Switch", CS42L52_MISC_CTL, 7, 1, 0); + +static const struct snd_kcontrol_new spkl_ctl = + SOC_DAPM_SINGLE("Switch", CS42L52_PWRCTL3, 0, 1, 1); + +static const struct snd_kcontrol_new spkr_ctl = + SOC_DAPM_SINGLE("Switch", CS42L52_PWRCTL3, 2, 1, 1); + +static const struct snd_kcontrol_new hpl_ctl = + SOC_DAPM_SINGLE("Switch", CS42L52_PWRCTL3, 4, 1, 1); + +static const struct snd_kcontrol_new hpr_ctl = + SOC_DAPM_SINGLE("Switch", CS42L52_PWRCTL3, 6, 1, 1); + +static const struct snd_kcontrol_new cs42l52_snd_controls[] = { + + SOC_DOUBLE_R_SX_TLV("Master Volume", CS42L52_MASTERA_VOL, + CS42L52_MASTERB_VOL, 0, 0x34, 0xE4, hl_tlv), + + SOC_DOUBLE_R_SX_TLV("Headphone Volume", CS42L52_HPA_VOL, + CS42L52_HPB_VOL, 0, 0x34, 0xCC, hpd_tlv), + + SOC_ENUM("Headphone Analog Gain", hp_gain_enum), + + SOC_DOUBLE_R_SX_TLV("Speaker Volume", CS42L52_SPKA_VOL, + CS42L52_SPKB_VOL, 7, 0x1, 0xff, hl_tlv), + + SOC_DOUBLE_R_SX_TLV("Bypass Volume", CS42L52_PASSTHRUA_VOL, + CS42L52_PASSTHRUB_VOL, 6, 0x18, 0x90, pga_tlv), + + SOC_DOUBLE("Bypass Mute", CS42L52_MISC_CTL, 4, 5, 1, 0), + + SOC_DOUBLE_R_TLV("MIC Gain Volume", CS42L52_MICA_CTL, + CS42L52_MICB_CTL, 0, 0x10, 0, mic_tlv), + + SOC_ENUM("MIC Bias Level", mic_bias_level_enum), + + SOC_DOUBLE_R_SX_TLV("ADC Volume", CS42L52_ADCA_VOL, + CS42L52_ADCB_VOL, 7, 0x80, 0xA0, ipd_tlv), + SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume", + CS42L52_ADCA_MIXER_VOL, CS42L52_ADCB_MIXER_VOL, + 6, 0x7f, 0x19, ipd_tlv), + + SOC_DOUBLE("ADC Switch", CS42L52_ADC_MISC_CTL, 0, 1, 1, 0), + + SOC_DOUBLE_R("ADC Mixer Switch", CS42L52_ADCA_MIXER_VOL, + CS42L52_ADCB_MIXER_VOL, 7, 1, 1), + + SOC_DOUBLE_R_SX_TLV("PGA Volume", CS42L52_PGAA_CTL, + CS42L52_PGAB_CTL, 0, 0x28, 0x30, pga_tlv), + + SOC_DOUBLE_R_SX_TLV("PCM Mixer Volume", + CS42L52_PCMA_MIXER_VOL, CS42L52_PCMB_MIXER_VOL, + 6, 0x7f, 0x19, hl_tlv), + SOC_DOUBLE_R("PCM Mixer Switch", + CS42L52_PCMA_MIXER_VOL, CS42L52_PCMB_MIXER_VOL, 7, 1, 1), + + SOC_ENUM("Beep Config", beep_config_enum), + SOC_ENUM("Beep Pitch", beep_pitch_enum), + SOC_ENUM("Beep on Time", beep_ontime_enum), + SOC_ENUM("Beep off Time", beep_offtime_enum), + SOC_SINGLE_TLV("Beep Volume", CS42L52_BEEP_VOL, 0, 0x1f, 0x07, hl_tlv), + SOC_SINGLE("Beep Mixer Switch", CS42L52_BEEP_TONE_CTL, 5, 1, 1), + SOC_ENUM("Beep Treble Corner Freq", beep_treble_enum), + SOC_ENUM("Beep Bass Corner Freq", beep_bass_enum), + + SOC_SINGLE("Tone Control Switch", CS42L52_BEEP_TONE_CTL, 0, 1, 1), + SOC_SINGLE_TLV("Treble Gain Volume", + CS42L52_TONE_CTL, 4, 15, 1, hl_tlv), + SOC_SINGLE_TLV("Bass Gain Volume", + CS42L52_TONE_CTL, 0, 15, 1, hl_tlv), + + /* Limiter */ + SOC_SINGLE_TLV("Limiter Max Threshold Volume", + CS42L52_LIMITER_CTL1, 5, 7, 0, limiter_tlv), + SOC_SINGLE_TLV("Limiter Cushion Threshold Volume", + CS42L52_LIMITER_CTL1, 2, 7, 0, limiter_tlv), + SOC_SINGLE_TLV("Limiter Release Rate Volume", + CS42L52_LIMITER_CTL2, 0, 63, 0, limiter_tlv), + SOC_SINGLE_TLV("Limiter Attack Rate Volume", + CS42L52_LIMITER_AT_RATE, 0, 63, 0, limiter_tlv), + + SOC_SINGLE("Limiter SR Switch", CS42L52_LIMITER_CTL1, 1, 1, 0), + SOC_SINGLE("Limiter ZC Switch", CS42L52_LIMITER_CTL1, 0, 1, 0), + SOC_SINGLE("Limiter Switch", CS42L52_LIMITER_CTL2, 7, 1, 0), + + /* ALC */ + SOC_SINGLE_TLV("ALC Attack Rate Volume", CS42L52_ALC_CTL, + 0, 63, 0, limiter_tlv), + SOC_SINGLE_TLV("ALC Release Rate Volume", CS42L52_ALC_RATE, + 0, 63, 0, limiter_tlv), + SOC_SINGLE_TLV("ALC Max Threshold Volume", CS42L52_ALC_THRESHOLD, + 5, 7, 0, limiter_tlv), + SOC_SINGLE_TLV("ALC Min Threshold Volume", CS42L52_ALC_THRESHOLD, + 2, 7, 0, limiter_tlv), + + SOC_DOUBLE_R("ALC SR Capture Switch", CS42L52_PGAA_CTL, + CS42L52_PGAB_CTL, 7, 1, 1), + SOC_DOUBLE_R("ALC ZC Capture Switch", CS42L52_PGAA_CTL, + CS42L52_PGAB_CTL, 6, 1, 1), + SOC_DOUBLE("ALC Capture Switch", CS42L52_ALC_CTL, 6, 7, 1, 0), + + /* Noise gate */ + SOC_ENUM("NG Type Switch", ng_type_enum), + SOC_SINGLE("NG Enable Switch", CS42L52_NOISE_GATE_CTL, 6, 1, 0), + SOC_SINGLE("NG Boost Switch", CS42L52_NOISE_GATE_CTL, 5, 1, 1), + SOC_ENUM("NG Threshold", ng_threshold_enum), + SOC_ENUM("NG Delay", ng_delay_enum), + + SOC_DOUBLE("HPF Switch", CS42L52_ANALOG_HPF_CTL, 5, 7, 1, 0), + + SOC_DOUBLE("Analog SR Switch", CS42L52_ANALOG_HPF_CTL, 1, 3, 1, 1), + SOC_DOUBLE("Analog ZC Switch", CS42L52_ANALOG_HPF_CTL, 0, 2, 1, 1), + SOC_SINGLE("Digital SR Switch", CS42L52_MISC_CTL, 1, 1, 0), + SOC_SINGLE("Digital ZC Switch", CS42L52_MISC_CTL, 0, 1, 0), + SOC_SINGLE("Deemphasis Switch", CS42L52_MISC_CTL, 2, 1, 0), + + SOC_SINGLE("Batt Compensation Switch", CS42L52_BATT_COMPEN, 7, 1, 0), + SOC_SINGLE("Batt VP Monitor Switch", CS42L52_BATT_COMPEN, 6, 1, 0), + SOC_SINGLE("Batt VP ref", CS42L52_BATT_COMPEN, 0, 0x0f, 0), + + SOC_SINGLE("PGA AIN1L Switch", CS42L52_ADC_PGA_A, 0, 1, 0), + SOC_SINGLE("PGA AIN1R Switch", CS42L52_ADC_PGA_B, 0, 1, 0), + SOC_SINGLE("PGA AIN2L Switch", CS42L52_ADC_PGA_A, 1, 1, 0), + SOC_SINGLE("PGA AIN2R Switch", CS42L52_ADC_PGA_B, 1, 1, 0), + + SOC_SINGLE("PGA AIN3L Switch", CS42L52_ADC_PGA_A, 2, 1, 0), + SOC_SINGLE("PGA AIN3R Switch", CS42L52_ADC_PGA_B, 2, 1, 0), + + SOC_SINGLE("PGA AIN4L Switch", CS42L52_ADC_PGA_A, 3, 1, 0), + SOC_SINGLE("PGA AIN4R Switch", CS42L52_ADC_PGA_B, 3, 1, 0), + + SOC_SINGLE("PGA MICA Switch", CS42L52_ADC_PGA_A, 4, 1, 0), + SOC_SINGLE("PGA MICB Switch", CS42L52_ADC_PGA_B, 4, 1, 0), + +}; + +static const struct snd_soc_dapm_widget cs42l52_dapm_widgets[] = { + + SND_SOC_DAPM_INPUT("AIN1L"), + SND_SOC_DAPM_INPUT("AIN1R"), + SND_SOC_DAPM_INPUT("AIN2L"), + SND_SOC_DAPM_INPUT("AIN2R"), + SND_SOC_DAPM_INPUT("AIN3L"), + SND_SOC_DAPM_INPUT("AIN3R"), + SND_SOC_DAPM_INPUT("AIN4L"), + SND_SOC_DAPM_INPUT("AIN4R"), + SND_SOC_DAPM_INPUT("MICA"), + SND_SOC_DAPM_INPUT("MICB"), + SND_SOC_DAPM_SIGGEN("Beep"), + + SND_SOC_DAPM_AIF_OUT("AIFOUTL", NULL, 0, + SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIFOUTR", NULL, 0, + SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_MUX("MICA Mux", SND_SOC_NOPM, 0, 0, &mica_mux), + SND_SOC_DAPM_MUX("MICB Mux", SND_SOC_NOPM, 0, 0, &micb_mux), + + SND_SOC_DAPM_ADC("ADC Left", NULL, CS42L52_PWRCTL1, 1, 1), + SND_SOC_DAPM_ADC("ADC Right", NULL, CS42L52_PWRCTL1, 2, 1), + SND_SOC_DAPM_PGA("PGA Left", CS42L52_PWRCTL1, 3, 1, NULL, 0), + SND_SOC_DAPM_PGA("PGA Right", CS42L52_PWRCTL1, 4, 1, NULL, 0), + + SND_SOC_DAPM_MUX("ADC Left Mux", SND_SOC_NOPM, 0, 0, &adca_mux), + SND_SOC_DAPM_MUX("ADC Right Mux", SND_SOC_NOPM, 0, 0, &adcb_mux), + + SND_SOC_DAPM_MUX("ADC Left Swap", SND_SOC_NOPM, + 0, 0, &adca_mixer), + SND_SOC_DAPM_MUX("ADC Right Swap", SND_SOC_NOPM, + 0, 0, &adcb_mixer), + + SND_SOC_DAPM_MUX("Output Mux", SND_SOC_NOPM, + 0, 0, &digital_output_mux), + + SND_SOC_DAPM_PGA("PGA MICA", CS42L52_PWRCTL2, 1, 1, NULL, 0), + SND_SOC_DAPM_PGA("PGA MICB", CS42L52_PWRCTL2, 2, 1, NULL, 0), + + SND_SOC_DAPM_SUPPLY("Mic Bias", CS42L52_PWRCTL2, 0, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Charge Pump", CS42L52_PWRCTL1, 7, 1, NULL, 0), + + SND_SOC_DAPM_AIF_IN("AIFINL", NULL, 0, + SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIFINR", NULL, 0, + SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_DAC("DAC Left", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DAC Right", NULL, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_SWITCH("Bypass Left", CS42L52_MISC_CTL, + 6, 0, &passthrul_ctl), + SND_SOC_DAPM_SWITCH("Bypass Right", CS42L52_MISC_CTL, + 7, 0, &passthrur_ctl), + + SND_SOC_DAPM_MUX("PCM Left Swap", SND_SOC_NOPM, + 0, 0, &pcma_mixer), + SND_SOC_DAPM_MUX("PCM Right Swap", SND_SOC_NOPM, + 0, 0, &pcmb_mixer), + + SND_SOC_DAPM_SWITCH("HP Left Amp", SND_SOC_NOPM, 0, 0, &hpl_ctl), + SND_SOC_DAPM_SWITCH("HP Right Amp", SND_SOC_NOPM, 0, 0, &hpr_ctl), + + SND_SOC_DAPM_SWITCH("SPK Left Amp", SND_SOC_NOPM, 0, 0, &spkl_ctl), + SND_SOC_DAPM_SWITCH("SPK Right Amp", SND_SOC_NOPM, 0, 0, &spkr_ctl), + + SND_SOC_DAPM_OUTPUT("HPOUTA"), + SND_SOC_DAPM_OUTPUT("HPOUTB"), + SND_SOC_DAPM_OUTPUT("SPKOUTA"), + SND_SOC_DAPM_OUTPUT("SPKOUTB"), + +}; + +static const struct snd_soc_dapm_route cs42l52_audio_map[] = { + + {"Capture", NULL, "AIFOUTL"}, + {"Capture", NULL, "AIFOUTL"}, + + {"AIFOUTL", NULL, "Output Mux"}, + {"AIFOUTR", NULL, "Output Mux"}, + + {"Output Mux", "ADC", "ADC Left"}, + {"Output Mux", "ADC", "ADC Right"}, + + {"ADC Left", NULL, "Charge Pump"}, + {"ADC Right", NULL, "Charge Pump"}, + + {"Charge Pump", NULL, "ADC Left Mux"}, + {"Charge Pump", NULL, "ADC Right Mux"}, + + {"ADC Left Mux", "Input1A", "AIN1L"}, + {"ADC Right Mux", "Input1B", "AIN1R"}, + {"ADC Left Mux", "Input2A", "AIN2L"}, + {"ADC Right Mux", "Input2B", "AIN2R"}, + {"ADC Left Mux", "Input3A", "AIN3L"}, + {"ADC Right Mux", "Input3B", "AIN3R"}, + {"ADC Left Mux", "Input4A", "AIN4L"}, + {"ADC Right Mux", "Input4B", "AIN4R"}, + {"ADC Left Mux", "PGA Input Left", "PGA Left"}, + {"ADC Right Mux", "PGA Input Right" , "PGA Right"}, + + {"PGA Left", "Switch", "AIN1L"}, + {"PGA Right", "Switch", "AIN1R"}, + {"PGA Left", "Switch", "AIN2L"}, + {"PGA Right", "Switch", "AIN2R"}, + {"PGA Left", "Switch", "AIN3L"}, + {"PGA Right", "Switch", "AIN3R"}, + {"PGA Left", "Switch", "AIN4L"}, + {"PGA Right", "Switch", "AIN4R"}, + + {"PGA Left", "Switch", "PGA MICA"}, + {"PGA MICA", NULL, "MICA"}, + + {"PGA Right", "Switch", "PGA MICB"}, + {"PGA MICB", NULL, "MICB"}, + + {"HPOUTA", NULL, "HP Left Amp"}, + {"HPOUTB", NULL, "HP Right Amp"}, + {"HP Left Amp", NULL, "Bypass Left"}, + {"HP Right Amp", NULL, "Bypass Right"}, + {"Bypass Left", "Switch", "PGA Left"}, + {"Bypass Right", "Switch", "PGA Right"}, + {"HP Left Amp", "Switch", "DAC Left"}, + {"HP Right Amp", "Switch", "DAC Right"}, + + {"SPKOUTA", NULL, "SPK Left Amp"}, + {"SPKOUTB", NULL, "SPK Right Amp"}, + + {"SPK Left Amp", NULL, "Beep"}, + {"SPK Right Amp", NULL, "Beep"}, + {"SPK Left Amp", "Switch", "Playback"}, + {"SPK Right Amp", "Switch", "Playback"}, + + {"DAC Left", NULL, "Beep"}, + {"DAC Right", NULL, "Beep"}, + {"DAC Left", NULL, "Playback"}, + {"DAC Right", NULL, "Playback"}, + + {"Output Mux", "DSP", "Playback"}, + {"Output Mux", "DSP", "Playback"}, + + {"AIFINL", NULL, "Playback"}, + {"AIFINR", NULL, "Playback"}, + +}; + +struct cs42l52_clk_para { + u32 mclk; + u32 rate; + u8 speed; + u8 group; + u8 videoclk; + u8 ratio; + u8 mclkdiv2; +}; + +static const struct cs42l52_clk_para clk_map_table[] = { + /*8k*/ + {12288000, 8000, CLK_QS_MODE, CLK_32K, CLK_NO_27M, CLK_R_128, 0}, + {18432000, 8000, CLK_QS_MODE, CLK_32K, CLK_NO_27M, CLK_R_128, 0}, + {12000000, 8000, CLK_QS_MODE, CLK_32K, CLK_NO_27M, CLK_R_125, 0}, + {24000000, 8000, CLK_QS_MODE, CLK_32K, CLK_NO_27M, CLK_R_125, 1}, + {27000000, 8000, CLK_QS_MODE, CLK_32K, CLK_27M_MCLK, CLK_R_125, 0}, + + /*11.025k*/ + {11289600, 11025, CLK_QS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0}, + {16934400, 11025, CLK_QS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0}, + + /*16k*/ + {12288000, 16000, CLK_HS_MODE, CLK_32K, CLK_NO_27M, CLK_R_128, 0}, + {18432000, 16000, CLK_HS_MODE, CLK_32K, CLK_NO_27M, CLK_R_128, 0}, + {12000000, 16000, CLK_HS_MODE, CLK_32K, CLK_NO_27M, CLK_R_125, 0}, + {24000000, 16000, CLK_HS_MODE, CLK_32K, CLK_NO_27M, CLK_R_125, 1}, + {27000000, 16000, CLK_HS_MODE, CLK_32K, CLK_27M_MCLK, CLK_R_125, 1}, + + /*22.05k*/ + {11289600, 22050, CLK_HS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0}, + {16934400, 22050, CLK_HS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0}, + + /* 32k */ + {12288000, 32000, CLK_SS_MODE, CLK_32K, CLK_NO_27M, CLK_R_128, 0}, + {18432000, 32000, CLK_SS_MODE, CLK_32K, CLK_NO_27M, CLK_R_128, 0}, + {12000000, 32000, CLK_SS_MODE, CLK_32K, CLK_NO_27M, CLK_R_125, 0}, + {24000000, 32000, CLK_SS_MODE, CLK_32K, CLK_NO_27M, CLK_R_125, 1}, + {27000000, 32000, CLK_SS_MODE, CLK_32K, CLK_27M_MCLK, CLK_R_125, 0}, + + /* 44.1k */ + {11289600, 44100, CLK_SS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0}, + {16934400, 44100, CLK_SS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0}, + + /* 48k */ + {12288000, 48000, CLK_SS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0}, + {18432000, 48000, CLK_SS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0}, + {12000000, 48000, CLK_SS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_125, 0}, + {24000000, 48000, CLK_SS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_125, 1}, + {27000000, 48000, CLK_SS_MODE, CLK_NO_32K, CLK_27M_MCLK, CLK_R_125, 1}, + + /* 88.2k */ + {11289600, 88200, CLK_DS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0}, + {16934400, 88200, CLK_DS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0}, + + /* 96k */ + {12288000, 96000, CLK_DS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0}, + {18432000, 96000, CLK_DS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0}, + {12000000, 96000, CLK_DS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_125, 0}, + {24000000, 96000, CLK_DS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_125, 1}, +}; + +static int cs42l52_get_clk(int mclk, int rate) +{ + int i, ret = 0; + u_int mclk1, mclk2 = 0; + + for (i = 0; i < ARRAY_SIZE(clk_map_table); i++) { + if (clk_map_table[i].rate == rate) { + mclk1 = clk_map_table[i].mclk; + if (abs(mclk - mclk1) < abs(mclk - mclk2)) { + mclk2 = mclk1; + ret = i; + } + } + } + if (ret > ARRAY_SIZE(clk_map_table)) + return -EINVAL; + return ret; +} + +static int cs42l52_set_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec); + + if ((freq >= CS42L52_MIN_CLK) && (freq <= CS42L52_MAX_CLK)) { + cs42l52->sysclk = freq; + } else { + dev_err(codec->dev, "Invalid freq paramter\n"); + return -EINVAL; + } + return 0; +} + +static int cs42l52_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec); + int ret = 0; + u8 iface = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + iface = CS42L52_IFACE_CTL1_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFS: + iface = CS42L52_IFACE_CTL1_SLAVE; + break; + default: + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface |= CS42L52_IFACE_CTL1_ADC_FMT_I2S | + CS42L52_IFACE_CTL1_DAC_FMT_I2S; + break; + case SND_SOC_DAIFMT_RIGHT_J: + iface |= CS42L52_IFACE_CTL1_DAC_FMT_RIGHT_J; + break; + case SND_SOC_DAIFMT_LEFT_J: + iface |= CS42L52_IFACE_CTL1_ADC_FMT_LEFT_J | + CS42L52_IFACE_CTL1_DAC_FMT_LEFT_J; + break; + case SND_SOC_DAIFMT_DSP_A: + iface |= CS42L52_IFACE_CTL1_DSP_MODE_EN; + break; + case SND_SOC_DAIFMT_DSP_B: + break; + default: + return -EINVAL; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + iface |= CS42L52_IFACE_CTL1_INV_SCLK; + break; + case SND_SOC_DAIFMT_IB_NF: + iface |= CS42L52_IFACE_CTL1_INV_SCLK; + break; + case SND_SOC_DAIFMT_NB_IF: + break; + default: + ret = -EINVAL; + } + cs42l52->config.format = iface; + snd_soc_write(codec, CS42L52_IFACE_CTL1, cs42l52->config.format); + + return 0; +} + +static int cs42l52_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + + if (mute) + snd_soc_update_bits(codec, CS42L52_PB_CTL1, + CS42L52_PB_CTL1_MUTE_MASK, + CS42L52_PB_CTL1_MUTE); + else + snd_soc_update_bits(codec, CS42L52_PB_CTL1, + CS42L52_PB_CTL1_MUTE_MASK, + CS42L52_PB_CTL1_UNMUTE); + + return 0; +} + +static int cs42l52_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec); + u32 clk = 0; + int index; + + index = cs42l52_get_clk(cs42l52->sysclk, params_rate(params)); + if (index >= 0) { + cs42l52->sysclk = clk_map_table[index].mclk; + + clk |= (clk_map_table[index].speed << CLK_SPEED_SHIFT) | + (clk_map_table[index].group << CLK_32K_SR_SHIFT) | + (clk_map_table[index].videoclk << CLK_27M_MCLK_SHIFT) | + (clk_map_table[index].ratio << CLK_RATIO_SHIFT) | + clk_map_table[index].mclkdiv2; + + snd_soc_write(codec, CS42L52_CLK_CTL, clk); + } else { + dev_err(codec->dev, "can't get correct mclk\n"); + return -EINVAL; + } + + return 0; +} + +static int cs42l52_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec); + + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + snd_soc_update_bits(codec, CS42L52_PWRCTL1, + CS42L52_PWRCTL1_PDN_CODEC, 0); + break; + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + regcache_cache_only(cs42l52->regmap, false); + regcache_sync(cs42l52->regmap); + } + snd_soc_write(codec, CS42L52_PWRCTL1, CS42L52_PWRCTL1_PDN_ALL); + break; + case SND_SOC_BIAS_OFF: + snd_soc_write(codec, CS42L52_PWRCTL1, CS42L52_PWRCTL1_PDN_ALL); + regcache_cache_only(cs42l52->regmap, true); + break; + } + codec->dapm.bias_level = level; + + return 0; +} + +#define CS42L52_RATES (SNDRV_PCM_RATE_8000_96000) + +#define CS42L52_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE | \ + SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_U18_3LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_U20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_U24_LE) + +static struct snd_soc_dai_ops cs42l52_ops = { + .hw_params = cs42l52_pcm_hw_params, + .digital_mute = cs42l52_digital_mute, + .set_fmt = cs42l52_set_fmt, + .set_sysclk = cs42l52_set_sysclk, +}; + +static struct snd_soc_dai_driver cs42l52_dai = { + .name = "cs42l52", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = CS42L52_RATES, + .formats = CS42L52_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = CS42L52_RATES, + .formats = CS42L52_FORMATS, + }, + .ops = &cs42l52_ops, +}; + +static int cs42l52_suspend(struct snd_soc_codec *codec) +{ + cs42l52_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int cs42l52_resume(struct snd_soc_codec *codec) +{ + cs42l52_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} + +#if defined(CONFIG_INPUT) || defined(CONFIG_INPUT_MODULE) +static int beep_rates[] = { + 261, 522, 585, 667, 706, 774, 889, 1000, + 1043, 1200, 1333, 1412, 1600, 1714, 2000, 2182 +}; + +static void cs42l52_beep_work(struct work_struct *work) +{ + struct cs42l52_private *cs42l52 = + container_of(work, struct cs42l52_private, beep_work); + struct snd_soc_codec *codec = cs42l52->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + int i; + int val = 0; + int best = 0; + + if (cs42l52->beep_rate) { + for (i = 0; i < ARRAY_SIZE(beep_rates); i++) { + if (abs(cs42l52->beep_rate - beep_rates[i]) < + abs(cs42l52->beep_rate - beep_rates[best])) + best = i; + } + + dev_dbg(codec->dev, "Set beep rate %dHz for requested %dHz\n", + beep_rates[best], cs42l52->beep_rate); + + val = (best << CS42L52_BEEP_RATE_SHIFT); + + snd_soc_dapm_enable_pin(dapm, "Beep"); + } else { + dev_dbg(codec->dev, "Disabling beep\n"); + snd_soc_dapm_disable_pin(dapm, "Beep"); + } + + snd_soc_update_bits(codec, CS42L52_BEEP_FREQ, + CS42L52_BEEP_RATE_MASK, val); + + snd_soc_dapm_sync(dapm); +} + +/* For usability define a way of injecting beep events for the device - + * many systems will not have a keyboard. + */ +static int cs42l52_beep_event(struct input_dev *dev, unsigned int type, + unsigned int code, int hz) +{ + struct snd_soc_codec *codec = input_get_drvdata(dev); + struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec); + + dev_dbg(codec->dev, "Beep event %x %x\n", code, hz); + + switch (code) { + case SND_BELL: + if (hz) + hz = 261; + case SND_TONE: + break; + default: + return -1; + } + + /* Kick the beep from a workqueue */ + cs42l52->beep_rate = hz; + schedule_work(&cs42l52->beep_work); + return 0; +} + +static ssize_t cs42l52_beep_set(struct device *dev, + struct device_attribute *attr, + const char *buf, size_t count) +{ + struct cs42l52_private *cs42l52 = dev_get_drvdata(dev); + long int time; + int ret; + + ret = kstrtol(buf, 10, &time); + if (ret != 0) + return ret; + + input_event(cs42l52->beep, EV_SND, SND_TONE, time); + + return count; +} + +static DEVICE_ATTR(beep, 0200, NULL, cs42l52_beep_set); + +static void cs42l52_init_beep(struct snd_soc_codec *codec) +{ + struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec); + int ret; + + cs42l52->beep = input_allocate_device(); + if (!cs42l52->beep) { + dev_err(codec->dev, "Failed to allocate beep device\n"); + return; + } + + INIT_WORK(&cs42l52->beep_work, cs42l52_beep_work); + cs42l52->beep_rate = 0; + + cs42l52->beep->name = "CS42L52 Beep Generator"; + cs42l52->beep->phys = dev_name(codec->dev); + cs42l52->beep->id.bustype = BUS_I2C; + + cs42l52->beep->evbit[0] = BIT_MASK(EV_SND); + cs42l52->beep->sndbit[0] = BIT_MASK(SND_BELL) | BIT_MASK(SND_TONE); + cs42l52->beep->event = cs42l52_beep_event; + cs42l52->beep->dev.parent = codec->dev; + input_set_drvdata(cs42l52->beep, codec); + + ret = input_register_device(cs42l52->beep); + if (ret != 0) { + input_free_device(cs42l52->beep); + cs42l52->beep = NULL; + dev_err(codec->dev, "Failed to register beep device\n"); + } + + ret = device_create_file(codec->dev, &dev_attr_beep); + if (ret != 0) { + dev_err(codec->dev, "Failed to create keyclick file: %d\n", + ret); + } +} + +static void cs42l52_free_beep(struct snd_soc_codec *codec) +{ + struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec); + + device_remove_file(codec->dev, &dev_attr_beep); + input_unregister_device(cs42l52->beep); + cancel_work_sync(&cs42l52->beep_work); + cs42l52->beep = NULL; + + snd_soc_update_bits(codec, CS42L52_BEEP_TONE_CTL, + CS42L52_BEEP_EN_MASK, 0); +} +#else +static void cs42l52_init_beep(struct snd_soc_codec *codec) +{ +} + +static void cs42l52_free_beep(struct snd_soc_codec *codec) +{ +} +#endif + +static int cs42l52_probe(struct snd_soc_codec *codec) +{ + struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec); + int ret; + + codec->control_data = cs42l52->regmap; + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + regcache_cache_only(cs42l52->regmap, true); + + cs42l52_init_beep(codec); + + cs42l52_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + cs42l52->sysclk = CS42L52_DEFAULT_CLK; + cs42l52->config.format = CS42L52_DEFAULT_FORMAT; + + /* Set Platform MICx CFG */ + snd_soc_update_bits(codec, CS42L52_MICA_CTL, + CS42L52_MIC_CTL_TYPE_MASK, + cs42l52->pdata.mica_cfg << + CS42L52_MIC_CTL_TYPE_SHIFT); + + snd_soc_update_bits(codec, CS42L52_MICB_CTL, + CS42L52_MIC_CTL_TYPE_MASK, + cs42l52->pdata.micb_cfg << + CS42L52_MIC_CTL_TYPE_SHIFT); + + /* if Single Ended, Get Mic_Select */ + if (cs42l52->pdata.mica_cfg) + snd_soc_update_bits(codec, CS42L52_MICA_CTL, + CS42L52_MIC_CTL_MIC_SEL_MASK, + cs42l52->pdata.mica_sel << + CS42L52_MIC_CTL_MIC_SEL_SHIFT); + if (cs42l52->pdata.micb_cfg) + snd_soc_update_bits(codec, CS42L52_MICB_CTL, + CS42L52_MIC_CTL_MIC_SEL_MASK, + cs42l52->pdata.micb_sel << + CS42L52_MIC_CTL_MIC_SEL_SHIFT); + + /* Set Platform Charge Pump Freq */ + snd_soc_update_bits(codec, CS42L52_CHARGE_PUMP, + CS42L52_CHARGE_PUMP_MASK, + cs42l52->pdata.chgfreq << + CS42L52_CHARGE_PUMP_SHIFT); + + /* Set Platform Bias Level */ + snd_soc_update_bits(codec, CS42L52_IFACE_CTL2, + CS42L52_IFACE_CTL2_BIAS_LVL, + cs42l52->pdata.micbias_lvl); + + return ret; +} + +static int cs42l52_remove(struct snd_soc_codec *codec) +{ + cs42l52_free_beep(codec); + cs42l52_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_cs42l52 = { + .probe = cs42l52_probe, + .remove = cs42l52_remove, + .suspend = cs42l52_suspend, + .resume = cs42l52_resume, + .set_bias_level = cs42l52_set_bias_level, + + .dapm_widgets = cs42l52_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs42l52_dapm_widgets), + .dapm_routes = cs42l52_audio_map, + .num_dapm_routes = ARRAY_SIZE(cs42l52_audio_map), + + .controls = cs42l52_snd_controls, + .num_controls = ARRAY_SIZE(cs42l52_snd_controls), +}; + +/* Current and threshold powerup sequence Pg37 */ +static const struct reg_default cs42l52_threshold_patch[] = { + + { 0x00, 0x99 }, + { 0x3E, 0xBA }, + { 0x47, 0x80 }, + { 0x32, 0xBB }, + { 0x32, 0x3B }, + { 0x00, 0x00 }, + +}; + +static struct regmap_config cs42l52_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = CS42L52_MAX_REGISTER, + .reg_defaults = cs42l52_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(cs42l52_reg_defaults), + .readable_reg = cs42l52_readable_register, + .volatile_reg = cs42l52_volatile_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int cs42l52_i2c_probe(struct i2c_client *i2c_client, + const struct i2c_device_id *id) +{ + struct cs42l52_private *cs42l52; + int ret; + unsigned int devid = 0; + unsigned int reg; + + cs42l52 = devm_kzalloc(&i2c_client->dev, sizeof(struct cs42l52_private), + GFP_KERNEL); + if (cs42l52 == NULL) + return -ENOMEM; + cs42l52->dev = &i2c_client->dev; + + cs42l52->regmap = regmap_init_i2c(i2c_client, &cs42l52_regmap); + if (IS_ERR(cs42l52->regmap)) { + ret = PTR_ERR(cs42l52->regmap); + dev_err(&i2c_client->dev, "regmap_init() failed: %d\n", ret); + goto err; + } + + i2c_set_clientdata(i2c_client, cs42l52); + + if (dev_get_platdata(&i2c_client->dev)) + memcpy(&cs42l52->pdata, dev_get_platdata(&i2c_client->dev), + sizeof(cs42l52->pdata)); + + ret = regmap_register_patch(cs42l52->regmap, cs42l52_threshold_patch, + ARRAY_SIZE(cs42l52_threshold_patch)); + if (ret != 0) + dev_warn(cs42l52->dev, "Failed to apply regmap patch: %d\n", + ret); + + ret = regmap_read(cs42l52->regmap, CS42L52_CHIP, ®); + devid = reg & CS42L52_CHIP_ID_MASK; + if (devid != CS42L52_CHIP_ID) { + ret = -ENODEV; + dev_err(&i2c_client->dev, + "CS42L52 Device ID (%X). Expected %X\n", + devid, CS42L52_CHIP_ID); + goto err_regmap; + } + + regcache_cache_only(cs42l52->regmap, true); + + ret = snd_soc_register_codec(&i2c_client->dev, + &soc_codec_dev_cs42l52, &cs42l52_dai, 1); + if (ret < 0) + goto err_regmap; + return 0; + +err_regmap: + regmap_exit(cs42l52->regmap); + +err: + return ret; +} + +static int cs42l52_i2c_remove(struct i2c_client *client) +{ + struct cs42l52_private *cs42l52 = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + regmap_exit(cs42l52->regmap); + + return 0; +} + +static const struct i2c_device_id cs42l52_id[] = { + { "cs42l52", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, cs42l52_id); + +static struct i2c_driver cs42l52_i2c_driver = { + .driver = { + .name = "cs42l52", + .owner = THIS_MODULE, + }, + .id_table = cs42l52_id, + .probe = cs42l52_i2c_probe, + .remove = __devexit_p(cs42l52_i2c_remove), +}; + +module_i2c_driver(cs42l52_i2c_driver); + +MODULE_DESCRIPTION("ASoC CS42L52 driver"); +MODULE_AUTHOR("Georgi Vlaev, Nucleus Systems Ltd, <joe@nucleusys.com>"); +MODULE_AUTHOR("Brian Austin, Cirrus Logic Inc, <brian.austin@cirrus.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs42l52.h b/sound/soc/codecs/cs42l52.h new file mode 100644 index 000000000000..60985c059071 --- /dev/null +++ b/sound/soc/codecs/cs42l52.h @@ -0,0 +1,274 @@ +/* + * cs42l52.h -- CS42L52 ALSA SoC audio driver + * + * Copyright 2012 CirrusLogic, Inc. + * + * Author: Georgi Vlaev <joe@nucleusys.com> + * Author: Brian Austin <brian.austin@cirrus.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + */ + +#ifndef __CS42L52_H__ +#define __CS42L52_H__ + +#define CS42L52_NAME "CS42L52" +#define CS42L52_DEFAULT_CLK 12000000 +#define CS42L52_MIN_CLK 11000000 +#define CS42L52_MAX_CLK 27000000 +#define CS42L52_DEFAULT_FORMAT SNDRV_PCM_FMTBIT_S16_LE +#define CS42L52_DEFAULT_MAX_CHANS 2 +#define CS42L52_SYSCLK 1 + +#define CS42L52_CHIP_SWICTH (1 << 17) +#define CS42L52_ALL_IN_ONE (1 << 16) +#define CS42L52_CHIP_ONE 0x00 +#define CS42L52_CHIP_TWO 0x01 +#define CS42L52_CHIP_THR 0x02 +#define CS42L52_CHIP_MASK 0x0f + +#define CS42L52_FIX_BITS_CTL 0x00 +#define CS42L52_CHIP 0x01 +#define CS42L52_CHIP_ID 0xE0 +#define CS42L52_CHIP_ID_MASK 0xF8 +#define CS42L52_CHIP_REV_A0 0x00 +#define CS42L52_CHIP_REV_A1 0x01 +#define CS42L52_CHIP_REV_B0 0x02 +#define CS42L52_CHIP_REV_MASK 0x03 + +#define CS42L52_PWRCTL1 0x02 +#define CS42L52_PWRCTL1_PDN_ALL 0x9F +#define CS42L52_PWRCTL1_PDN_CHRG 0x80 +#define CS42L52_PWRCTL1_PDN_PGAB 0x10 +#define CS42L52_PWRCTL1_PDN_PGAA 0x08 +#define CS42L52_PWRCTL1_PDN_ADCB 0x04 +#define CS42L52_PWRCTL1_PDN_ADCA 0x02 +#define CS42L52_PWRCTL1_PDN_CODEC 0x01 + +#define CS42L52_PWRCTL2 0x03 +#define CS42L52_PWRCTL2_OVRDB (1 << 4) +#define CS42L52_PWRCTL2_OVRDA (1 << 3) +#define CS42L52_PWRCTL2_PDN_MICB (1 << 2) +#define CS42L52_PWRCTL2_PDN_MICB_SHIFT 2 +#define CS42L52_PWRCTL2_PDN_MICA (1 << 1) +#define CS42L52_PWRCTL2_PDN_MICA_SHIFT 1 +#define CS42L52_PWRCTL2_PDN_MICBIAS (1 << 0) +#define CS42L52_PWRCTL2_PDN_MICBIAS_SHIFT 0 + +#define CS42L52_PWRCTL3 0x04 +#define CS42L52_PWRCTL3_HPB_PDN_SHIFT 6 +#define CS42L52_PWRCTL3_HPB_ON_LOW 0x00 +#define CS42L52_PWRCTL3_HPB_ON_HIGH 0x01 +#define CS42L52_PWRCTL3_HPB_ALWAYS_ON 0x02 +#define CS42L52_PWRCTL3_HPB_ALWAYS_OFF 0x03 +#define CS42L52_PWRCTL3_HPA_PDN_SHIFT 4 +#define CS42L52_PWRCTL3_HPA_ON_LOW 0x00 +#define CS42L52_PWRCTL3_HPA_ON_HIGH 0x01 +#define CS42L52_PWRCTL3_HPA_ALWAYS_ON 0x02 +#define CS42L52_PWRCTL3_HPA_ALWAYS_OFF 0x03 +#define CS42L52_PWRCTL3_SPKB_PDN_SHIFT 2 +#define CS42L52_PWRCTL3_SPKB_ON_LOW 0x00 +#define CS42L52_PWRCTL3_SPKB_ON_HIGH 0x01 +#define CS42L52_PWRCTL3_SPKB_ALWAYS_ON 0x02 +#define CS42L52_PWRCTL3_PDN_SPKB (1 << 2) +#define CS42L52_PWRCTL3_PDN_SPKA (1 << 0) +#define CS42L52_PWRCTL3_SPKA_PDN_SHIFT 0 +#define CS42L52_PWRCTL3_SPKA_ON_LOW 0x00 +#define CS42L52_PWRCTL3_SPKA_ON_HIGH 0x01 +#define CS42L52_PWRCTL3_SPKA_ALWAYS_ON 0x02 + +#define CS42L52_DEFAULT_OUTPUT_STATE 0x05 +#define CS42L52_PWRCTL3_CONF_MASK 0x03 + +#define CS42L52_CLK_CTL 0x05 +#define CLK_AUTODECT_ENABLE (1 << 7) +#define CLK_SPEED_SHIFT 5 +#define CLK_DS_MODE 0x00 +#define CLK_SS_MODE 0x01 +#define CLK_HS_MODE 0x02 +#define CLK_QS_MODE 0x03 +#define CLK_32K_SR_SHIFT 4 +#define CLK_32K 0x01 +#define CLK_NO_32K 0x00 +#define CLK_27M_MCLK_SHIFT 3 +#define CLK_27M_MCLK 0x01 +#define CLK_NO_27M 0x00 +#define CLK_RATIO_SHIFT 1 +#define CLK_R_128 0x00 +#define CLK_R_125 0x01 +#define CLK_R_132 0x02 +#define CLK_R_136 0x03 + +#define CS42L52_IFACE_CTL1 0x06 +#define CS42L52_IFACE_CTL1_MASTER (1 << 7) +#define CS42L52_IFACE_CTL1_SLAVE (0 << 7) +#define CS42L52_IFACE_CTL1_INV_SCLK (1 << 6) +#define CS42L52_IFACE_CTL1_ADC_FMT_I2S (1 << 5) +#define CS42L52_IFACE_CTL1_ADC_FMT_LEFT_J (0 << 5) +#define CS42L52_IFACE_CTL1_DSP_MODE_EN (1 << 4) +#define CS42L52_IFACE_CTL1_DAC_FMT_LEFT_J (0 << 2) +#define CS42L52_IFACE_CTL1_DAC_FMT_I2S (1 << 2) +#define CS42L52_IFACE_CTL1_DAC_FMT_RIGHT_J (2 << 2) +#define CS42L52_IFACE_CTL1_WL_32BIT (0x00) +#define CS42L52_IFACE_CTL1_WL_24BIT (0x01) +#define CS42L52_IFACE_CTL1_WL_20BIT (0x02) +#define CS42L52_IFACE_CTL1_WL_16BIT (0x03) +#define CS42L52_IFACE_CTL1_WL_MASK 0xFFFF + +#define CS42L52_IFACE_CTL2 0x07 +#define CS42L52_IFACE_CTL2_SC_MC_EQ (1 << 6) +#define CS42L52_IFACE_CTL2_LOOPBACK (1 << 5) +#define CS42L52_IFACE_CTL2_S_MODE_OUTPUT_EN (0 << 4) +#define CS42L52_IFACE_CTL2_S_MODE_OUTPUT_HIZ (1 << 4) +#define CS42L52_IFACE_CTL2_HP_SW_INV (1 << 3) +#define CS42L52_IFACE_CTL2_BIAS_LVL 0x07 + +#define CS42L52_ADC_PGA_A 0x08 +#define CS42L52_ADC_PGA_B 0x09 +#define CS42L52_ADC_SEL_SHIFT 5 +#define CS42L52_ADC_SEL_AIN1 0x00 +#define CS42L52_ADC_SEL_AIN2 0x01 +#define CS42L52_ADC_SEL_AIN3 0x02 +#define CS42L52_ADC_SEL_AIN4 0x03 +#define CS42L52_ADC_SEL_PGA 0x04 + +#define CS42L52_ANALOG_HPF_CTL 0x0A +#define CS42L52_HPF_CTL_ANLGSFTB (1 << 3) +#define CS42L52_HPF_CTL_ANLGSFTA (1 << 0) + +#define CS42L52_ADC_HPF_FREQ 0x0B +#define CS42L52_ADC_MISC_CTL 0x0C +#define CS42L52_ADC_MISC_CTL_SOURCE_DSP (1 << 6) + +#define CS42L52_PB_CTL1 0x0D +#define CS42L52_PB_CTL1_HP_GAIN_SHIFT 5 +#define CS42L52_PB_CTL1_HP_GAIN_03959 0x00 +#define CS42L52_PB_CTL1_HP_GAIN_04571 0x01 +#define CS42L52_PB_CTL1_HP_GAIN_05111 0x02 +#define CS42L52_PB_CTL1_HP_GAIN_06047 0x03 +#define CS42L52_PB_CTL1_HP_GAIN_07099 0x04 +#define CS42L52_PB_CTL1_HP_GAIN_08399 0x05 +#define CS42L52_PB_CTL1_HP_GAIN_10000 0x06 +#define CS42L52_PB_CTL1_HP_GAIN_11430 0x07 +#define CS42L52_PB_CTL1_INV_PCMB (1 << 3) +#define CS42L52_PB_CTL1_INV_PCMA (1 << 2) +#define CS42L52_PB_CTL1_MSTB_MUTE (1 << 1) +#define CS42L52_PB_CTL1_MSTA_MUTE (1 << 0) +#define CS42L52_PB_CTL1_MUTE_MASK 0xFFFD +#define CS42L52_PB_CTL1_MUTE 3 +#define CS42L52_PB_CTL1_UNMUTE 0 + +#define CS42L52_MISC_CTL 0x0E +#define CS42L52_MISC_CTL_DEEMPH (1 << 2) +#define CS42L52_MISC_CTL_DIGSFT (1 << 1) +#define CS42L52_MISC_CTL_DIGZC (1 << 0) + +#define CS42L52_PB_CTL2 0x0F +#define CS42L52_PB_CTL2_HPB_MUTE (1 << 7) +#define CS42L52_PB_CTL2_HPA_MUTE (1 << 6) +#define CS42L52_PB_CTL2_SPKB_MUTE (1 << 5) +#define CS42L52_PB_CTL2_SPKA_MUTE (1 << 4) +#define CS42L52_PB_CTL2_SPK_SWAP (1 << 2) +#define CS42L52_PB_CTL2_SPK_MONO (1 << 1) +#define CS42L52_PB_CTL2_SPK_MUTE50 (1 << 0) + +#define CS42L52_MICA_CTL 0x10 +#define CS42L52_MICB_CTL 0x11 +#define CS42L52_MIC_CTL_MIC_SEL_MASK 0xBF +#define CS42L52_MIC_CTL_MIC_SEL_SHIFT 6 +#define CS42L52_MIC_CTL_TYPE_MASK 0xDF +#define CS42L52_MIC_CTL_TYPE_SHIFT 5 + + +#define CS42L52_PGAA_CTL 0x12 +#define CS42L52_PGAB_CTL 0x13 +#define CS42L52_PGAX_CTL_VOL_12DB 24 +#define CS42L52_PGAX_CTL_VOL_6DB 12 /*step size 0.5db*/ + +#define CS42L52_PASSTHRUA_VOL 0x14 +#define CS42L52_PASSTHRUB_VOL 0x15 + +#define CS42L52_ADCA_VOL 0x16 +#define CS42L52_ADCB_VOL 0x17 +#define CS42L52_ADCX_VOL_24DB 24 /*step size 1db*/ +#define CS42L52_ADCX_VOL_12DB 12 +#define CS42L52_ADCX_VOL_6DB 6 + +#define CS42L52_ADCA_MIXER_VOL 0x18 +#define CS42L52_ADCB_MIXER_VOL 0x19 +#define CS42L52_ADC_MIXER_VOL_12DB 0x18 + +#define CS42L52_PCMA_MIXER_VOL 0x1A +#define CS42L52_PCMB_MIXER_VOL 0x1B + +#define CS42L52_BEEP_FREQ 0x1C +#define CS42L52_BEEP_VOL 0x1D +#define CS42L52_BEEP_TONE_CTL 0x1E +#define CS42L52_BEEP_RATE_SHIFT 4 +#define CS42L52_BEEP_RATE_MASK 0x0F + +#define CS42L52_TONE_CTL 0x1F +#define CS42L52_BEEP_EN_MASK 0x3F + +#define CS42L52_MASTERA_VOL 0x20 +#define CS42L52_MASTERB_VOL 0x21 + +#define CS42L52_HPA_VOL 0x22 +#define CS42L52_HPB_VOL 0x23 +#define CS42L52_DEFAULT_HP_VOL 0xF0 + +#define CS42L52_SPKA_VOL 0x24 +#define CS42L52_SPKB_VOL 0x25 +#define CS42L52_DEFAULT_SPK_VOL 0xF0 + +#define CS42L52_ADC_PCM_MIXER 0x26 + +#define CS42L52_LIMITER_CTL1 0x27 +#define CS42L52_LIMITER_CTL2 0x28 +#define CS42L52_LIMITER_AT_RATE 0x29 + +#define CS42L52_ALC_CTL 0x2A +#define CS42L52_ALC_CTL_ALCB_ENABLE_SHIFT 7 +#define CS42L52_ALC_CTL_ALCA_ENABLE_SHIFT 6 +#define CS42L52_ALC_CTL_FASTEST_ATTACK 0 + +#define CS42L52_ALC_RATE 0x2B +#define CS42L52_ALC_SLOWEST_RELEASE 0x3F + +#define CS42L52_ALC_THRESHOLD 0x2C +#define CS42L52_ALC_MAX_RATE_SHIFT 5 +#define CS42L52_ALC_MIN_RATE_SHIFT 2 +#define CS42L52_ALC_RATE_0DB 0 +#define CS42L52_ALC_RATE_3DB 1 +#define CS42L52_ALC_RATE_6DB 2 + +#define CS42L52_NOISE_GATE_CTL 0x2D +#define CS42L52_NG_ENABLE_SHIFT 6 +#define CS42L52_NG_THRESHOLD_SHIFT 2 +#define CS42L52_NG_MIN_70DB 2 +#define CS42L52_NG_DELAY_SHIFT 0 +#define CS42L52_NG_DELAY_100MS 1 + +#define CS42L52_CLK_STATUS 0x2E +#define CS42L52_BATT_COMPEN 0x2F + +#define CS42L52_BATT_LEVEL 0x30 +#define CS42L52_SPK_STATUS 0x31 +#define CS42L52_SPK_STATUS_PIN_SHIFT 3 +#define CS42L52_SPK_STATUS_PIN_HIGH 1 + +#define CS42L52_TEM_CTL 0x32 +#define CS42L52_TEM_CTL_SET 0x80 +#define CS42L52_THE_FOLDBACK 0x33 +#define CS42L52_CHARGE_PUMP 0x34 +#define CS42L52_CHARGE_PUMP_MASK 0xF0 +#define CS42L52_CHARGE_PUMP_SHIFT 4 +#define CS42L52_FIX_BITS1 0x3E +#define CS42L52_FIX_BITS2 0x47 + +#define CS42L52_MAX_REGISTER 0x34 + +#endif diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 3686417f5ea5..e0d45fdaa750 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -43,9 +43,6 @@ struct cs42l73_private { }; static const struct reg_default cs42l73_reg_defaults[] = { - { 1, 0x42 }, /* r01 - Device ID A&B */ - { 2, 0xA7 }, /* r02 - Device ID C&D */ - { 3, 0x30 }, /* r03 - Device ID E */ { 6, 0xF1 }, /* r06 - Power Ctl 1 */ { 7, 0xDF }, /* r07 - Power Ctl 2 */ { 8, 0x3F }, /* r08 - Power Ctl 3 */ @@ -402,37 +399,37 @@ static const struct snd_kcontrol_new ear_amp_ctl = static const struct snd_kcontrol_new cs42l73_snd_controls[] = { SOC_DOUBLE_R_SX_TLV("Headphone Analog Playback Volume", - CS42L73_HPAAVOL, CS42L73_HPBAVOL, 7, - 0xffffffC1, 0x0C, hpaloa_tlv), + CS42L73_HPAAVOL, CS42L73_HPBAVOL, 0, + 0x41, 0x4B, hpaloa_tlv), SOC_DOUBLE_R_SX_TLV("LineOut Analog Playback Volume", CS42L73_LOAAVOL, - CS42L73_LOBAVOL, 7, 0xffffffC1, 0x0C, hpaloa_tlv), + CS42L73_LOBAVOL, 0, 0x41, 0x4B, hpaloa_tlv), SOC_DOUBLE_R_SX_TLV("Input PGA Analog Volume", CS42L73_MICAPREPGAAVOL, - CS42L73_MICBPREPGABVOL, 5, 0xffffff35, - 0x34, micpga_tlv), + CS42L73_MICBPREPGABVOL, 5, 0x34, + 0x24, micpga_tlv), SOC_DOUBLE_R("MIC Preamp Switch", CS42L73_MICAPREPGAAVOL, CS42L73_MICBPREPGABVOL, 6, 1, 1), SOC_DOUBLE_R_SX_TLV("Input Path Digital Volume", CS42L73_IPADVOL, - CS42L73_IPBDVOL, 7, 0xffffffA0, 0xA0, ipd_tlv), + CS42L73_IPBDVOL, 0, 0xA0, 0x6C, ipd_tlv), SOC_DOUBLE_R_SX_TLV("HL Digital Playback Volume", - CS42L73_HLADVOL, CS42L73_HLBDVOL, 7, 0xffffffE5, - 0xE4, hl_tlv), + CS42L73_HLADVOL, CS42L73_HLBDVOL, + 0, 0x34, 0xE4, hl_tlv), SOC_SINGLE_TLV("ADC A Boost Volume", CS42L73_ADCIPC, 2, 0x01, 1, adc_boost_tlv), SOC_SINGLE_TLV("ADC B Boost Volume", - CS42L73_ADCIPC, 6, 0x01, 1, adc_boost_tlv), + CS42L73_ADCIPC, 6, 0x01, 1, adc_boost_tlv), - SOC_SINGLE_TLV("Speakerphone Digital Playback Volume", - CS42L73_SPKDVOL, 0, 0xE4, 1, hl_tlv), + SOC_SINGLE_SX_TLV("Speakerphone Digital Volume", + CS42L73_SPKDVOL, 0, 0x34, 0xE4, hl_tlv), - SOC_SINGLE_TLV("Ear Speaker Digital Playback Volume", - CS42L73_ESLDVOL, 0, 0xE4, 1, hl_tlv), + SOC_SINGLE_SX_TLV("Ear Speaker Digital Volume", + CS42L73_ESLDVOL, 0, 0x34, 0xE4, hl_tlv), SOC_DOUBLE_R("Headphone Analog Playback Switch", CS42L73_HPAAVOL, CS42L73_HPBAVOL, 7, 1, 1), @@ -599,17 +596,17 @@ static const struct snd_soc_dapm_widget cs42l73_dapm_widgets[] = { SND_SOC_DAPM_INPUT("MIC2"), SND_SOC_DAPM_SUPPLY("MIC2 Bias", CS42L73_PWRCTL2, 7, 1, NULL, 0), - SND_SOC_DAPM_AIF_OUT("XSPOUTL", "XSP Capture", 0, + SND_SOC_DAPM_AIF_OUT("XSPOUTL", NULL, 0, CS42L73_PWRCTL2, 1, 1), - SND_SOC_DAPM_AIF_OUT("XSPOUTR", "XSP Capture", 0, + SND_SOC_DAPM_AIF_OUT("XSPOUTR", NULL, 0, CS42L73_PWRCTL2, 1, 1), - SND_SOC_DAPM_AIF_OUT("ASPOUTL", "ASP Capture", 0, + SND_SOC_DAPM_AIF_OUT("ASPOUTL", NULL, 0, CS42L73_PWRCTL2, 3, 1), - SND_SOC_DAPM_AIF_OUT("ASPOUTR", "ASP Capture", 0, + SND_SOC_DAPM_AIF_OUT("ASPOUTR", NULL, 0, CS42L73_PWRCTL2, 3, 1), - SND_SOC_DAPM_AIF_OUT("VSPOUTL", "VSP Capture", 0, + SND_SOC_DAPM_AIF_OUT("VSPOUTL", NULL, 0, CS42L73_PWRCTL2, 4, 1), - SND_SOC_DAPM_AIF_OUT("VSPOUTR", "VSP Capture", 0, + SND_SOC_DAPM_AIF_OUT("VSPOUTR", NULL, 0, CS42L73_PWRCTL2, 4, 1), SND_SOC_DAPM_PGA("PGA Left", SND_SOC_NOPM, 0, 0, NULL, 0), @@ -638,21 +635,21 @@ static const struct snd_soc_dapm_widget cs42l73_dapm_widgets[] = { SND_SOC_DAPM_MIXER("VSPL Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_MIXER("VSPR Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_AIF_IN("XSPINL", "XSP Playback", 0, + SND_SOC_DAPM_AIF_IN("XSPINL", NULL, 0, CS42L73_PWRCTL2, 0, 1), - SND_SOC_DAPM_AIF_IN("XSPINR", "XSP Playback", 0, + SND_SOC_DAPM_AIF_IN("XSPINR", NULL, 0, CS42L73_PWRCTL2, 0, 1), - SND_SOC_DAPM_AIF_IN("XSPINM", "XSP Playback", 0, + SND_SOC_DAPM_AIF_IN("XSPINM", NULL, 0, CS42L73_PWRCTL2, 0, 1), - SND_SOC_DAPM_AIF_IN("ASPINL", "ASP Playback", 0, + SND_SOC_DAPM_AIF_IN("ASPINL", NULL, 0, CS42L73_PWRCTL2, 2, 1), - SND_SOC_DAPM_AIF_IN("ASPINR", "ASP Playback", 0, + SND_SOC_DAPM_AIF_IN("ASPINR", NULL, 0, CS42L73_PWRCTL2, 2, 1), - SND_SOC_DAPM_AIF_IN("ASPINM", "ASP Playback", 0, + SND_SOC_DAPM_AIF_IN("ASPINM", NULL, 0, CS42L73_PWRCTL2, 2, 1), - SND_SOC_DAPM_AIF_IN("VSPIN", "VSP Playback", 0, + SND_SOC_DAPM_AIF_IN("VSPIN", NULL, 0, CS42L73_PWRCTL2, 4, 1), SND_SOC_DAPM_MIXER("HL Left Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), @@ -776,6 +773,14 @@ static const struct snd_soc_dapm_route cs42l73_audio_map[] = { {"HL Left Mixer", NULL, "VSPIN"}, {"HL Right Mixer", NULL, "VSPIN"}, + {"ASPINL", NULL, "ASP Playback"}, + {"ASPINM", NULL, "ASP Playback"}, + {"ASPINR", NULL, "ASP Playback"}, + {"XSPINL", NULL, "XSP Playback"}, + {"XSPINM", NULL, "XSP Playback"}, + {"XSPINR", NULL, "XSP Playback"}, + {"VSPIN", NULL, "VSP Playback"}, + /* Capture Paths */ {"MIC1", NULL, "MIC1 Bias"}, {"PGA Left Mux", "Mic 1", "MIC1"}, @@ -822,6 +827,13 @@ static const struct snd_soc_dapm_route cs42l73_audio_map[] = { {"VSPOUTL", NULL, "VSPL Output Mixer"}, {"VSPOUTR", NULL, "VSPR Output Mixer"}, + + {"ASP Capture", NULL, "ASPOUTL"}, + {"ASP Capture", NULL, "ASPOUTR"}, + {"XSP Capture", NULL, "XSPOUTL"}, + {"XSP Capture", NULL, "XSPOUTR"}, + {"VSP Capture", NULL, "VSPOUTL"}, + {"VSP Capture", NULL, "VSPOUTR"}, }; struct cs42l73_mclk_div { @@ -1091,8 +1103,7 @@ static int cs42l73_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec); int id = dai->id; int mclk_coeff; @@ -1429,25 +1440,7 @@ static struct i2c_driver cs42l73_i2c_driver = { }; -static int __init cs42l73_modinit(void) -{ - int ret; - ret = i2c_add_driver(&cs42l73_i2c_driver); - if (ret != 0) { - pr_err("Failed to register CS42L73 I2C driver: %d\n", ret); - return ret; - } - return 0; -} - -module_init(cs42l73_modinit); - -static void __exit cs42l73_exit(void) -{ - i2c_del_driver(&cs42l73_i2c_driver); -} - -module_exit(cs42l73_exit); +module_i2c_driver(cs42l73_i2c_driver); MODULE_DESCRIPTION("ASoC CS42L73 driver"); MODULE_AUTHOR("Georgi Vlaev, Nucleus Systems Ltd, <joe@nucleusys.com>"); diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 7843711729bc..af5db7080519 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -17,6 +17,7 @@ #include <linux/delay.h> #include <linux/i2c.h> +#include <linux/spi/spi.h> #include <linux/regmap.h> #include <linux/slab.h> #include <linux/module.h> @@ -27,6 +28,7 @@ #include <sound/tlv.h> /* DA7210 register space */ +#define DA7210_PAGE_CONTROL 0x00 #define DA7210_CONTROL 0x01 #define DA7210_STATUS 0x02 #define DA7210_STARTUP1 0x03 @@ -146,6 +148,7 @@ #define DA7210_DAI_EN (1 << 7) /*PLL_DIV3 bit fields */ +#define DA7210_PLL_DIV_L_MASK (0xF << 0) #define DA7210_MCLK_RANGE_10_20_MHZ (1 << 4) #define DA7210_PLL_BYP (1 << 6) @@ -162,12 +165,16 @@ #define DA7210_PLL_FS_48000 (0xB << 0) #define DA7210_PLL_FS_88200 (0xE << 0) #define DA7210_PLL_FS_96000 (0xF << 0) +#define DA7210_MCLK_DET_EN (0x1 << 5) +#define DA7210_MCLK_SRM_EN (0x1 << 6) #define DA7210_PLL_EN (0x1 << 7) /* SOFTMUTE bit fields */ #define DA7210_RAMP_EN (1 << 6) /* CONTROL bit fields */ +#define DA7210_REG_EN (1 << 0) +#define DA7210_BIAS_EN (1 << 2) #define DA7210_NOISE_SUP_EN (1 << 3) /* IN_GAIN bit fields */ @@ -206,6 +213,47 @@ #define DA7210_OUT2_OUTMIX_L (1 << 6) #define DA7210_OUT2_EN (1 << 7) +struct pll_div { + int fref; + int fout; + u8 div1; + u8 div2; + u8 div3; + u8 mode; /* 0 = slave, 1 = master */ +}; + +/* PLL dividers table */ +static const struct pll_div da7210_pll_div[] = { + /* for MASTER mode, fs = 44.1Khz */ + { 12000000, 2822400, 0xE8, 0x6C, 0x2, 1}, /* MCLK=12Mhz */ + { 13000000, 2822400, 0xDF, 0x28, 0xC, 1}, /* MCLK=13Mhz */ + { 13500000, 2822400, 0xDB, 0x0A, 0xD, 1}, /* MCLK=13.5Mhz */ + { 14400000, 2822400, 0xD4, 0x5A, 0x2, 1}, /* MCLK=14.4Mhz */ + { 19200000, 2822400, 0xBB, 0x43, 0x9, 1}, /* MCLK=19.2Mhz */ + { 19680000, 2822400, 0xB9, 0x6D, 0xA, 1}, /* MCLK=19.68Mhz */ + { 19800000, 2822400, 0xB8, 0xFB, 0xB, 1}, /* MCLK=19.8Mhz */ + /* for MASTER mode, fs = 48Khz */ + { 12000000, 3072000, 0xF3, 0x12, 0x7, 1}, /* MCLK=12Mhz */ + { 13000000, 3072000, 0xE8, 0xFD, 0x5, 1}, /* MCLK=13Mhz */ + { 13500000, 3072000, 0xE4, 0x82, 0x3, 1}, /* MCLK=13.5Mhz */ + { 14400000, 3072000, 0xDD, 0x3A, 0x0, 1}, /* MCLK=14.4Mhz */ + { 19200000, 3072000, 0xC1, 0xEB, 0x8, 1}, /* MCLK=19.2Mhz */ + { 19680000, 3072000, 0xBF, 0xEC, 0x0, 1}, /* MCLK=19.68Mhz */ + { 19800000, 3072000, 0xBF, 0x70, 0x0, 1}, /* MCLK=19.8Mhz */ + /* for SLAVE mode with SRM */ + { 12000000, 2822400, 0xED, 0xBF, 0x5, 0}, /* MCLK=12Mhz */ + { 13000000, 2822400, 0xE4, 0x13, 0x0, 0}, /* MCLK=13Mhz */ + { 13500000, 2822400, 0xDF, 0xC6, 0x8, 0}, /* MCLK=13.5Mhz */ + { 14400000, 2822400, 0xD8, 0xCA, 0x1, 0}, /* MCLK=14.4Mhz */ + { 19200000, 2822400, 0xBE, 0x97, 0x9, 0}, /* MCLK=19.2Mhz */ + { 19680000, 2822400, 0xBC, 0xAC, 0xD, 0}, /* MCLK=19.68Mhz */ + { 19800000, 2822400, 0xBC, 0x35, 0xE, 0}, /* MCLK=19.8Mhz */ +}; + +enum clk_src { + DA7210_CLKSRC_MCLK +}; + #define DA7210_VERSION "0.0.1" /* @@ -628,9 +676,12 @@ static const struct snd_soc_dapm_route da7210_audio_map[] = { /* Codec private data */ struct da7210_priv { struct regmap *regmap; + unsigned int mclk_rate; + int master; }; static struct reg_default da7210_reg_defaults[] = { + { 0x00, 0x00 }, { 0x01, 0x11 }, { 0x03, 0x00 }, { 0x04, 0x00 }, @@ -713,10 +764,10 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; + struct da7210_priv *da7210 = snd_soc_codec_get_drvdata(codec); u32 dai_cfg1; - u32 fs, bypass; + u32 fs, sysclk; /* set DAI source to Left and Right ADC */ snd_soc_write(codec, DA7210_DAI_SRC_SEL, @@ -749,43 +800,43 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, switch (params_rate(params)) { case 8000: fs = DA7210_PLL_FS_8000; - bypass = DA7210_PLL_BYP; + sysclk = 3072000; break; case 11025: fs = DA7210_PLL_FS_11025; - bypass = 0; + sysclk = 2822400; break; case 12000: fs = DA7210_PLL_FS_12000; - bypass = DA7210_PLL_BYP; + sysclk = 3072000; break; case 16000: fs = DA7210_PLL_FS_16000; - bypass = DA7210_PLL_BYP; + sysclk = 3072000; break; case 22050: fs = DA7210_PLL_FS_22050; - bypass = 0; + sysclk = 2822400; break; case 32000: fs = DA7210_PLL_FS_32000; - bypass = DA7210_PLL_BYP; + sysclk = 3072000; break; case 44100: fs = DA7210_PLL_FS_44100; - bypass = 0; + sysclk = 2822400; break; case 48000: fs = DA7210_PLL_FS_48000; - bypass = DA7210_PLL_BYP; + sysclk = 3072000; break; case 88200: fs = DA7210_PLL_FS_88200; - bypass = 0; + sysclk = 2822400; break; case 96000: fs = DA7210_PLL_FS_96000; - bypass = DA7210_PLL_BYP; + sysclk = 3072000; break; default: return -EINVAL; @@ -795,8 +846,26 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, snd_soc_update_bits(codec, DA7210_STARTUP1, DA7210_SC_MST_EN, 0); snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_FS_MASK, fs); - snd_soc_update_bits(codec, DA7210_PLL_DIV3, DA7210_PLL_BYP, bypass); + if (da7210->mclk_rate && (da7210->mclk_rate != sysclk)) { + /* PLL mode, disable PLL bypass */ + snd_soc_update_bits(codec, DA7210_PLL_DIV3, DA7210_PLL_BYP, 0); + + if (!da7210->master) { + /* PLL slave mode, also enable SRM */ + snd_soc_update_bits(codec, DA7210_PLL, + (DA7210_MCLK_SRM_EN | + DA7210_MCLK_DET_EN), + (DA7210_MCLK_SRM_EN | + DA7210_MCLK_DET_EN)); + } + } else { + /* PLL bypass mode, enable PLL bypass and Auto Detection */ + snd_soc_update_bits(codec, DA7210_PLL, DA7210_MCLK_DET_EN, + DA7210_MCLK_DET_EN); + snd_soc_update_bits(codec, DA7210_PLL_DIV3, DA7210_PLL_BYP, + DA7210_PLL_BYP); + } /* Enable active mode */ snd_soc_update_bits(codec, DA7210_STARTUP1, DA7210_SC_MST_EN, DA7210_SC_MST_EN); @@ -810,17 +879,24 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) { struct snd_soc_codec *codec = codec_dai->codec; + struct da7210_priv *da7210 = snd_soc_codec_get_drvdata(codec); u32 dai_cfg1; u32 dai_cfg3; dai_cfg1 = 0x7f & snd_soc_read(codec, DA7210_DAI_CFG1); dai_cfg3 = 0xfc & snd_soc_read(codec, DA7210_DAI_CFG3); + if ((snd_soc_read(codec, DA7210_PLL) & DA7210_PLL_EN) && + (!(snd_soc_read(codec, DA7210_PLL_DIV3) & DA7210_PLL_BYP))) + return -EINVAL; + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: + da7210->master = 1; dai_cfg1 |= DA7210_DAI_MODE_MASTER; break; case SND_SOC_DAIFMT_CBS_CFS: + da7210->master = 0; dai_cfg1 |= DA7210_DAI_MODE_SLAVE; break; default: @@ -872,10 +948,101 @@ static int da7210_mute(struct snd_soc_dai *dai, int mute) #define DA7210_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) +static int da7210_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct da7210_priv *da7210 = snd_soc_codec_get_drvdata(codec); + + switch (clk_id) { + case DA7210_CLKSRC_MCLK: + switch (freq) { + case 12000000: + case 13000000: + case 13500000: + case 14400000: + case 19200000: + case 19680000: + case 19800000: + da7210->mclk_rate = freq; + return 0; + default: + dev_err(codec_dai->dev, "Unsupported MCLK value %d\n", + freq); + return -EINVAL; + } + break; + default: + dev_err(codec_dai->dev, "Unknown clock source %d\n", clk_id); + return -EINVAL; + } +} + +/** + * da7210_set_dai_pll :Configure the codec PLL + * @param codec_dai : pointer to codec DAI + * @param pll_id : da7210 has only one pll, so pll_id is always zero + * @param fref : MCLK frequency, should be < 20MHz + * @param fout : FsDM value, Refer page 44 & 45 of datasheet + * @return int : Zero for success, negative error code for error + * + * Note: Supported PLL input frequencies are 12MHz, 13MHz, 13.5MHz, 14.4MHz, + * 19.2MHz, 19.6MHz and 19.8MHz + */ +static int da7210_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int fref, unsigned int fout) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct da7210_priv *da7210 = snd_soc_codec_get_drvdata(codec); + + u8 pll_div1, pll_div2, pll_div3, cnt; + + /* In slave mode, there is only one set of divisors */ + if (!da7210->master) + fout = 2822400; + + /* Search pll div array for correct divisors */ + for (cnt = 0; cnt < ARRAY_SIZE(da7210_pll_div); cnt++) { + /* check fref, mode and fout */ + if ((fref == da7210_pll_div[cnt].fref) && + (da7210->master == da7210_pll_div[cnt].mode) && + (fout == da7210_pll_div[cnt].fout)) { + /* all match, pick up divisors */ + pll_div1 = da7210_pll_div[cnt].div1; + pll_div2 = da7210_pll_div[cnt].div2; + pll_div3 = da7210_pll_div[cnt].div3; + break; + } + } + if (cnt >= ARRAY_SIZE(da7210_pll_div)) + goto err; + + /* Disable active mode */ + snd_soc_update_bits(codec, DA7210_STARTUP1, DA7210_SC_MST_EN, 0); + /* Write PLL dividers */ + snd_soc_write(codec, DA7210_PLL_DIV1, pll_div1); + snd_soc_write(codec, DA7210_PLL_DIV2, pll_div2); + snd_soc_update_bits(codec, DA7210_PLL_DIV3, + DA7210_PLL_DIV_L_MASK, pll_div3); + + /* Enable PLL */ + snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_EN, DA7210_PLL_EN); + + /* Enable active mode */ + snd_soc_update_bits(codec, DA7210_STARTUP1, DA7210_SC_MST_EN, + DA7210_SC_MST_EN); + return 0; +err: + dev_err(codec_dai->dev, "Unsupported PLL input frequency %d\n", fref); + return -EINVAL; +} + /* DAI operations */ static const struct snd_soc_dai_ops da7210_dai_ops = { .hw_params = da7210_hw_params, .set_fmt = da7210_set_dai_fmt, + .set_sysclk = da7210_set_dai_sysclk, + .set_pll = da7210_set_dai_pll, .digital_mute = da7210_mute, }; @@ -915,24 +1082,11 @@ static int da7210_probe(struct snd_soc_codec *codec) return ret; } - /* FIXME - * - * This driver use fixed value here - * And below settings expects MCLK = 12.288MHz - * - * When you select different MCLK, please check... - * DA7210_PLL_DIV1 val - * DA7210_PLL_DIV2 val - * DA7210_PLL_DIV3 val - * DA7210_PLL_DIV3 :: DA7210_MCLK_RANGExxx - */ + da7210->mclk_rate = 0; /* This will be set from set_sysclk() */ + da7210->master = 0; /* This will be set from set_fmt() */ - /* - * make sure that DA7210 use bypass mode before start up - */ - snd_soc_write(codec, DA7210_STARTUP1, 0); - snd_soc_write(codec, DA7210_PLL_DIV3, - DA7210_MCLK_RANGE_10_20_MHZ | DA7210_PLL_BYP); + /* Enable internal regulator & bias current */ + snd_soc_write(codec, DA7210_CONTROL, DA7210_REG_EN | DA7210_BIAS_EN); /* * ADC settings @@ -1007,34 +1161,13 @@ static int da7210_probe(struct snd_soc_codec *codec) /* Enable Aux2 */ snd_soc_write(codec, DA7210_AUX2, DA7210_AUX2_EN); + /* Set PLL Master clock range 10-20 MHz, enable PLL bypass */ + snd_soc_write(codec, DA7210_PLL_DIV3, DA7210_MCLK_RANGE_10_20_MHZ | + DA7210_PLL_BYP); + /* Diable PLL and bypass it */ snd_soc_write(codec, DA7210_PLL, DA7210_PLL_FS_48000); - /* - * If 48kHz sound came, it use bypass mode, - * and when it is 44.1kHz, it use PLL. - * - * This time, this driver sets PLL always ON - * and controls bypass/PLL mode by switching - * DA7210_PLL_DIV3 :: DA7210_PLL_BYP bit. - * see da7210_hw_params - */ - snd_soc_write(codec, DA7210_PLL_DIV1, 0xE5); /* MCLK = 12.288MHz */ - snd_soc_write(codec, DA7210_PLL_DIV2, 0x99); - snd_soc_write(codec, DA7210_PLL_DIV3, 0x0A | - DA7210_MCLK_RANGE_10_20_MHZ | DA7210_PLL_BYP); - snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_EN, DA7210_PLL_EN); - - /* As suggested by Dialog */ - /* unlock */ - regmap_write(da7210->regmap, DA7210_A_HID_UNLOCK, 0x8B); - regmap_write(da7210->regmap, DA7210_A_TEST_UNLOCK, 0xB4); - regmap_write(da7210->regmap, DA7210_A_PLL1, 0x01); - regmap_write(da7210->regmap, DA7210_A_CP_MODE, 0x7C); - /* re-lock */ - regmap_write(da7210->regmap, DA7210_A_HID_UNLOCK, 0x00); - regmap_write(da7210->regmap, DA7210_A_TEST_UNLOCK, 0x00); - /* Activate all enabled subsystem */ snd_soc_write(codec, DA7210_STARTUP1, DA7210_SC_MST_EN); @@ -1055,7 +1188,26 @@ static struct snd_soc_codec_driver soc_codec_dev_da7210 = { .num_dapm_routes = ARRAY_SIZE(da7210_audio_map), }; -static struct regmap_config da7210_regmap = { +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + +static struct reg_default da7210_regmap_i2c_patch[] = { + + /* System controller master disable */ + { DA7210_STARTUP1, 0x00 }, + /* Set PLL Master clock range 10-20 MHz */ + { DA7210_PLL_DIV3, DA7210_MCLK_RANGE_10_20_MHZ }, + + /* to unlock */ + { DA7210_A_HID_UNLOCK, 0x8B}, + { DA7210_A_TEST_UNLOCK, 0xB4}, + { DA7210_A_PLL1, 0x01}, + { DA7210_A_CP_MODE, 0x7C}, + /* to re-lock */ + { DA7210_A_HID_UNLOCK, 0x00}, + { DA7210_A_TEST_UNLOCK, 0x00}, +}; + +static const struct regmap_config da7210_regmap_config_i2c = { .reg_bits = 8, .val_bits = 8, @@ -1066,7 +1218,6 @@ static struct regmap_config da7210_regmap = { .cache_type = REGCACHE_RBTREE, }; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) static int __devinit da7210_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -1080,13 +1231,18 @@ static int __devinit da7210_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, da7210); - da7210->regmap = regmap_init_i2c(i2c, &da7210_regmap); + da7210->regmap = regmap_init_i2c(i2c, &da7210_regmap_config_i2c); if (IS_ERR(da7210->regmap)) { ret = PTR_ERR(da7210->regmap); dev_err(&i2c->dev, "regmap_init() failed: %d\n", ret); return ret; } + ret = regmap_register_patch(da7210->regmap, da7210_regmap_i2c_patch, + ARRAY_SIZE(da7210_regmap_i2c_patch)); + if (ret != 0) + dev_warn(&i2c->dev, "Failed to apply regmap patch: %d\n", ret); + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_da7210, &da7210_dai, 1); if (ret < 0) { @@ -1119,7 +1275,7 @@ MODULE_DEVICE_TABLE(i2c, da7210_i2c_id); /* I2C codec control layer */ static struct i2c_driver da7210_i2c_driver = { .driver = { - .name = "da7210-codec", + .name = "da7210", .owner = THIS_MODULE, }, .probe = da7210_i2c_probe, @@ -1128,12 +1284,112 @@ static struct i2c_driver da7210_i2c_driver = { }; #endif +#if defined(CONFIG_SPI_MASTER) + +static struct reg_default da7210_regmap_spi_patch[] = { + /* Dummy read to give two pulses over nCS for SPI */ + { DA7210_AUX2, 0x00 }, + { DA7210_AUX2, 0x00 }, + + /* System controller master disable */ + { DA7210_STARTUP1, 0x00 }, + /* Set PLL Master clock range 10-20 MHz */ + { DA7210_PLL_DIV3, DA7210_MCLK_RANGE_10_20_MHZ }, + + /* to set PAGE1 of SPI register space */ + { DA7210_PAGE_CONTROL, 0x80 }, + /* to unlock */ + { DA7210_A_HID_UNLOCK, 0x8B}, + { DA7210_A_TEST_UNLOCK, 0xB4}, + { DA7210_A_PLL1, 0x01}, + { DA7210_A_CP_MODE, 0x7C}, + /* to re-lock */ + { DA7210_A_HID_UNLOCK, 0x00}, + { DA7210_A_TEST_UNLOCK, 0x00}, + /* to set back PAGE0 of SPI register space */ + { DA7210_PAGE_CONTROL, 0x00 }, +}; + +static const struct regmap_config da7210_regmap_config_spi = { + .reg_bits = 8, + .val_bits = 8, + .read_flag_mask = 0x01, + .write_flag_mask = 0x00, + + .reg_defaults = da7210_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(da7210_reg_defaults), + .volatile_reg = da7210_volatile_register, + .readable_reg = da7210_readable_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int __devinit da7210_spi_probe(struct spi_device *spi) +{ + struct da7210_priv *da7210; + int ret; + + da7210 = devm_kzalloc(&spi->dev, sizeof(struct da7210_priv), + GFP_KERNEL); + if (!da7210) + return -ENOMEM; + + spi_set_drvdata(spi, da7210); + da7210->regmap = devm_regmap_init_spi(spi, &da7210_regmap_config_spi); + if (IS_ERR(da7210->regmap)) { + ret = PTR_ERR(da7210->regmap); + dev_err(&spi->dev, "Failed to register regmap: %d\n", ret); + return ret; + } + + ret = regmap_register_patch(da7210->regmap, da7210_regmap_spi_patch, + ARRAY_SIZE(da7210_regmap_spi_patch)); + if (ret != 0) + dev_warn(&spi->dev, "Failed to apply regmap patch: %d\n", ret); + + ret = snd_soc_register_codec(&spi->dev, + &soc_codec_dev_da7210, &da7210_dai, 1); + if (ret < 0) + goto err_regmap; + + return ret; + +err_regmap: + regmap_exit(da7210->regmap); + + return ret; +} + +static int __devexit da7210_spi_remove(struct spi_device *spi) +{ + struct da7210_priv *da7210 = spi_get_drvdata(spi); + snd_soc_unregister_codec(&spi->dev); + regmap_exit(da7210->regmap); + return 0; +} + +static struct spi_driver da7210_spi_driver = { + .driver = { + .name = "da7210", + .owner = THIS_MODULE, + }, + .probe = da7210_spi_probe, + .remove = __devexit_p(da7210_spi_remove) +}; +#endif + static int __init da7210_modinit(void) { int ret = 0; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) ret = i2c_add_driver(&da7210_i2c_driver); #endif +#if defined(CONFIG_SPI_MASTER) + ret = spi_register_driver(&da7210_spi_driver); + if (ret) { + printk(KERN_ERR "Failed to register da7210 SPI driver: %d\n", + ret); + } +#endif return ret; } module_init(da7210_modinit); @@ -1143,6 +1399,9 @@ static void __exit da7210_exit(void) #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) i2c_del_driver(&da7210_i2c_driver); #endif +#if defined(CONFIG_SPI_MASTER) + spi_unregister_driver(&da7210_spi_driver); +#endif } module_exit(da7210_exit); diff --git a/sound/soc/codecs/jz4740.c b/sound/soc/codecs/jz4740.c index 4624e752a188..85d9cabe6d55 100644 --- a/sound/soc/codecs/jz4740.c +++ b/sound/soc/codecs/jz4740.c @@ -164,8 +164,7 @@ static int jz4740_codec_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { uint32_t val; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec =rtd->codec; + struct snd_soc_codec *codec = dai->codec; switch (params_rate(params)) { case 8000: diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c new file mode 100644 index 000000000000..802b9f176b16 --- /dev/null +++ b/sound/soc/codecs/lm49453.c @@ -0,0 +1,1550 @@ +/* + * lm49453.c - LM49453 ALSA Soc Audio driver + * + * Copyright (c) 2012 Texas Instruments, Inc + * + * 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; version 2 of the License. + * + * Initially based on sound/soc/codecs/wm8350.c + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/version.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> +#include <sound/jack.h> +#include <sound/initval.h> +#include <asm/div64.h> +#include "lm49453.h" + +static struct reg_default lm49453_reg_defs[] = { + { 0, 0x00 }, + { 1, 0x00 }, + { 2, 0x00 }, + { 3, 0x00 }, + { 4, 0x00 }, + { 5, 0x00 }, + { 6, 0x00 }, + { 7, 0x00 }, + { 8, 0x00 }, + { 9, 0x00 }, + { 10, 0x00 }, + { 11, 0x00 }, + { 12, 0x00 }, + { 13, 0x00 }, + { 14, 0x00 }, + { 15, 0x00 }, + { 16, 0x00 }, + { 17, 0x00 }, + { 18, 0x00 }, + { 19, 0x00 }, + { 20, 0x00 }, + { 21, 0x00 }, + { 22, 0x00 }, + { 23, 0x00 }, + { 32, 0x00 }, + { 33, 0x00 }, + { 35, 0x00 }, + { 36, 0x00 }, + { 37, 0x00 }, + { 46, 0x00 }, + { 48, 0x00 }, + { 49, 0x00 }, + { 51, 0x00 }, + { 56, 0x00 }, + { 58, 0x00 }, + { 59, 0x00 }, + { 60, 0x00 }, + { 61, 0x00 }, + { 62, 0x00 }, + { 63, 0x00 }, + { 64, 0x00 }, + { 65, 0x00 }, + { 66, 0x00 }, + { 67, 0x00 }, + { 68, 0x00 }, + { 69, 0x00 }, + { 70, 0x00 }, + { 71, 0x00 }, + { 72, 0x00 }, + { 73, 0x00 }, + { 74, 0x00 }, + { 75, 0x00 }, + { 76, 0x00 }, + { 77, 0x00 }, + { 78, 0x00 }, + { 79, 0x00 }, + { 80, 0x00 }, + { 81, 0x00 }, + { 82, 0x00 }, + { 83, 0x00 }, + { 85, 0x00 }, + { 85, 0x00 }, + { 86, 0x00 }, + { 87, 0x00 }, + { 88, 0x00 }, + { 89, 0x00 }, + { 90, 0x00 }, + { 91, 0x00 }, + { 92, 0x00 }, + { 93, 0x00 }, + { 94, 0x00 }, + { 95, 0x00 }, + { 96, 0x01 }, + { 97, 0x00 }, + { 98, 0x00 }, + { 99, 0x00 }, + { 100, 0x00 }, + { 101, 0x00 }, + { 102, 0x00 }, + { 103, 0x01 }, + { 105, 0x01 }, + { 106, 0x00 }, + { 107, 0x01 }, + { 107, 0x00 }, + { 108, 0x00 }, + { 109, 0x00 }, + { 110, 0x00 }, + { 111, 0x02 }, + { 112, 0x02 }, + { 113, 0x00 }, + { 121, 0x80 }, + { 122, 0xBB }, + { 123, 0x80 }, + { 124, 0xBB }, + { 128, 0x00 }, + { 130, 0x00 }, + { 131, 0x00 }, + { 132, 0x00 }, + { 133, 0x0A }, + { 134, 0x0A }, + { 135, 0x0A }, + { 136, 0x0F }, + { 137, 0x00 }, + { 138, 0x73 }, + { 139, 0x33 }, + { 140, 0x73 }, + { 141, 0x33 }, + { 142, 0x73 }, + { 143, 0x33 }, + { 144, 0x73 }, + { 145, 0x33 }, + { 146, 0x73 }, + { 147, 0x33 }, + { 148, 0x73 }, + { 149, 0x33 }, + { 150, 0x73 }, + { 151, 0x33 }, + { 152, 0x00 }, + { 153, 0x00 }, + { 154, 0x00 }, + { 155, 0x00 }, + { 176, 0x00 }, + { 177, 0x00 }, + { 178, 0x00 }, + { 179, 0x00 }, + { 180, 0x00 }, + { 181, 0x00 }, + { 182, 0x00 }, + { 183, 0x00 }, + { 184, 0x00 }, + { 185, 0x00 }, + { 186, 0x00 }, + { 189, 0x00 }, + { 188, 0x00 }, + { 194, 0x00 }, + { 195, 0x00 }, + { 196, 0x00 }, + { 197, 0x00 }, + { 200, 0x00 }, + { 201, 0x00 }, + { 202, 0x00 }, + { 203, 0x00 }, + { 204, 0x00 }, + { 205, 0x00 }, + { 208, 0x00 }, + { 209, 0x00 }, + { 210, 0x00 }, + { 211, 0x00 }, + { 213, 0x00 }, + { 214, 0x00 }, + { 215, 0x00 }, + { 216, 0x00 }, + { 217, 0x00 }, + { 218, 0x00 }, + { 219, 0x00 }, + { 221, 0x00 }, + { 222, 0x00 }, + { 224, 0x00 }, + { 225, 0x00 }, + { 226, 0x00 }, + { 227, 0x00 }, + { 228, 0x00 }, + { 229, 0x00 }, + { 230, 0x13 }, + { 231, 0x00 }, + { 232, 0x80 }, + { 233, 0x0C }, + { 234, 0xDD }, + { 235, 0x00 }, + { 236, 0x04 }, + { 237, 0x00 }, + { 238, 0x00 }, + { 239, 0x00 }, + { 240, 0x00 }, + { 241, 0x00 }, + { 242, 0x00 }, + { 243, 0x00 }, + { 244, 0x00 }, + { 245, 0x00 }, + { 248, 0x00 }, + { 249, 0x00 }, + { 254, 0x00 }, + { 255, 0x00 }, +}; + +/* codec private data */ +struct lm49453_priv { + struct regmap *regmap; + int fs_rate; +}; + +/* capture path controls */ + +static const char *lm49453_mic2mode_text[] = {"Single Ended", "Differential"}; + +static const SOC_ENUM_SINGLE_DECL(lm49453_mic2mode_enum, LM49453_P0_MICR_REG, 5, + lm49453_mic2mode_text); + +static const char *lm49453_dmic_cfg_text[] = {"DMICDAT1", "DMICDAT2"}; + +static const SOC_ENUM_SINGLE_DECL(lm49453_dmic12_cfg_enum, + LM49453_P0_DIGITAL_MIC1_CONFIG_REG, + 7, lm49453_dmic_cfg_text); + +static const SOC_ENUM_SINGLE_DECL(lm49453_dmic34_cfg_enum, + LM49453_P0_DIGITAL_MIC2_CONFIG_REG, + 7, lm49453_dmic_cfg_text); + +/* MUX Controls */ +static const char *lm49453_adcl_mux_text[] = { "MIC1", "Aux_L" }; + +static const char *lm49453_adcr_mux_text[] = { "MIC2", "Aux_R" }; + +static const struct soc_enum lm49453_adcl_enum = + SOC_ENUM_SINGLE(LM49453_P0_ANALOG_MIXER_ADC_REG, 0, + ARRAY_SIZE(lm49453_adcl_mux_text), + lm49453_adcl_mux_text); + +static const struct soc_enum lm49453_adcr_enum = + SOC_ENUM_SINGLE(LM49453_P0_ANALOG_MIXER_ADC_REG, 1, + ARRAY_SIZE(lm49453_adcr_mux_text), + lm49453_adcr_mux_text); + +static const struct snd_kcontrol_new lm49453_adcl_mux_control = + SOC_DAPM_ENUM("ADC Left Mux", lm49453_adcl_enum); + +static const struct snd_kcontrol_new lm49453_adcr_mux_control = + SOC_DAPM_ENUM("ADC Right Mux", lm49453_adcr_enum); + +static const struct snd_kcontrol_new lm49453_headset_left_mixer[] = { +SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACHPL1_REG, 0, 1, 0), +SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACHPL1_REG, 1, 1, 0), +SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACHPL1_REG, 2, 1, 0), +SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACHPL1_REG, 3, 1, 0), +SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACHPL1_REG, 4, 1, 0), +SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACHPL1_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACHPL1_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACHPL1_REG, 7, 1, 0), +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACHPL2_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACHPL2_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACHPL2_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACHPL2_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACHPL2_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACHPL2_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACHPL2_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACHPL2_REG, 7, 1, 0), +SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 0, 0, 0), +}; + +static const struct snd_kcontrol_new lm49453_headset_right_mixer[] = { +SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACHPR1_REG, 0, 1, 0), +SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACHPR1_REG, 1, 1, 0), +SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACHPR1_REG, 2, 1, 0), +SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACHPR1_REG, 3, 1, 0), +SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACHPR1_REG, 4, 1, 0), +SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACHPR1_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACHPR1_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACHPR1_REG, 7, 1, 0), +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACHPR2_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACHPR2_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACHPR2_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACHPR2_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACHPR2_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACHPR2_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACHPR2_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACHPR2_REG, 7, 1, 0), +SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 1, 0, 0), +}; + +static const struct snd_kcontrol_new lm49453_speaker_left_mixer[] = { +SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACLSL1_REG, 0, 1, 0), +SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACLSL1_REG, 1, 1, 0), +SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACLSL1_REG, 2, 1, 0), +SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACLSL1_REG, 3, 1, 0), +SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACLSL1_REG, 4, 1, 0), +SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACLSL1_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACLSL1_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACLSL1_REG, 7, 1, 0), +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACLSL2_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACLSL2_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACLSL2_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACLSL2_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACLSL2_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACLSL2_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACLSL2_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACLSL2_REG, 7, 1, 0), +SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 2, 0, 0), +}; + +static const struct snd_kcontrol_new lm49453_speaker_right_mixer[] = { +SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACLSR1_REG, 0, 1, 0), +SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACLSR1_REG, 1, 1, 0), +SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACLSR1_REG, 2, 1, 0), +SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACLSR1_REG, 3, 1, 0), +SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACLSR1_REG, 4, 1, 0), +SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACLSR1_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACLSR1_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACLSR1_REG, 7, 1, 0), +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACLSR2_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACLSR2_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACLSR2_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACLSR2_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACLSR2_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACLSR2_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACLSR2_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACLSR2_REG, 7, 1, 0), +SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 3, 0, 0), +}; + +static const struct snd_kcontrol_new lm49453_haptic_left_mixer[] = { +SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACHAL1_REG, 0, 1, 0), +SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACHAL1_REG, 1, 1, 0), +SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACHAL1_REG, 2, 1, 0), +SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACHAL1_REG, 3, 1, 0), +SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACHAL1_REG, 4, 1, 0), +SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACHAL1_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACHAL1_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACHAL1_REG, 7, 1, 0), +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACHAL2_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACHAL2_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACHAL2_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACHAL2_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACHAL2_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACHAL2_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACHAL2_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACHAL2_REG, 7, 1, 0), +SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 4, 0, 0), +}; + +static const struct snd_kcontrol_new lm49453_haptic_right_mixer[] = { +SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACHAR1_REG, 0, 1, 0), +SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACHAR1_REG, 1, 1, 0), +SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACHAR1_REG, 2, 1, 0), +SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACHAR1_REG, 3, 1, 0), +SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACHAR1_REG, 4, 1, 0), +SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACHAR1_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACHAR1_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACHAR1_REG, 7, 1, 0), +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACHAR2_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACHAR2_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACHAR2_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACHAR2_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACHAR2_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACHAR2_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACHAR2_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACHAR2_REG, 7, 1, 0), +SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 5, 0, 0), +}; + +static const struct snd_kcontrol_new lm49453_lineout_left_mixer[] = { +SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACLOL1_REG, 0, 1, 0), +SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACLOL1_REG, 1, 1, 0), +SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACLOL1_REG, 2, 1, 0), +SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACLOL1_REG, 3, 1, 0), +SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACLOL1_REG, 4, 1, 0), +SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACLOL1_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACLOL1_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACLOL1_REG, 7, 1, 0), +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACLOL2_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACLOL2_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACLOL2_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACLOL2_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACLOL2_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACLOL2_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACLOL2_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACLOL2_REG, 7, 1, 0), +SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 6, 0, 0), +}; + +static const struct snd_kcontrol_new lm49453_lineout_right_mixer[] = { +SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACLOR1_REG, 0, 1, 0), +SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACLOR1_REG, 1, 1, 0), +SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACLOR1_REG, 2, 1, 0), +SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACLOR1_REG, 3, 1, 0), +SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACLOR1_REG, 4, 1, 0), +SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACLOR1_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACLOR1_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACLOR1_REG, 7, 1, 0), +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACLOR2_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACLOR2_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACLOR2_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACLOR2_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACLOR2_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACLOR2_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACLOR2_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACLOR2_REG, 7, 1, 0), +SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 7, 0, 0), +}; + +static const struct snd_kcontrol_new lm49453_port1_tx1_mixer[] = { +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX1_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX1_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX1_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX1_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX1_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX1_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_PORT1_TX1_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_PORT1_TX1_REG, 7, 1, 0), +}; + +static const struct snd_kcontrol_new lm49453_port1_tx2_mixer[] = { +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX2_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX2_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX2_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX2_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX2_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX2_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_PORT1_TX2_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_PORT1_TX2_REG, 7, 1, 0), +}; + +static const struct snd_kcontrol_new lm49453_port1_tx3_mixer[] = { +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX3_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX3_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX3_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX3_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX3_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX3_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_PORT1_TX3_REG, 6, 1, 0), +}; + +static const struct snd_kcontrol_new lm49453_port1_tx4_mixer[] = { +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX4_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX4_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX4_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX4_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX4_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX4_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_PORT1_TX4_REG, 6, 1, 0), +}; + +static const struct snd_kcontrol_new lm49453_port1_tx5_mixer[] = { +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX5_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX5_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX5_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX5_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX5_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX5_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_PORT1_TX5_REG, 6, 1, 0), +}; + +static const struct snd_kcontrol_new lm49453_port1_tx6_mixer[] = { +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX6_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX6_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX6_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX6_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX6_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX6_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_PORT1_TX6_REG, 6, 1, 0), +}; + +static const struct snd_kcontrol_new lm49453_port1_tx7_mixer[] = { +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX7_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX7_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX7_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX7_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX7_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX7_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_PORT1_TX7_REG, 6, 1, 0), +}; + +static const struct snd_kcontrol_new lm49453_port1_tx8_mixer[] = { +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX8_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX8_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX8_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX8_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX8_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX8_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_PORT1_TX8_REG, 6, 1, 0), +}; + +static const struct snd_kcontrol_new lm49453_port2_tx1_mixer[] = { +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT2_TX1_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT2_TX1_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT2_TX1_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT2_TX1_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT2_TX1_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT2_TX1_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_PORT2_TX1_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_PORT2_TX1_REG, 7, 1, 0), +}; + +static const struct snd_kcontrol_new lm49453_port2_tx2_mixer[] = { +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT2_TX2_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT2_TX2_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT2_TX2_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT2_TX2_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT2_TX2_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT2_TX2_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_PORT2_TX2_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_PORT2_TX2_REG, 7, 1, 0), +}; + +/* TLV Declarations */ +static const DECLARE_TLV_DB_SCALE(digital_tlv, -7650, 150, 1); +static const DECLARE_TLV_DB_SCALE(port_tlv, 0, 600, 0); + +static const struct snd_kcontrol_new lm49453_sidetone_mixer_controls[] = { +/* Sidetone supports mono only */ +SOC_DAPM_SINGLE_TLV("Sidetone ADCL Volume", LM49453_P0_STN_VOL_ADCL_REG, + 0, 0x3F, 0, digital_tlv), +SOC_DAPM_SINGLE_TLV("Sidetone ADCR Volume", LM49453_P0_STN_VOL_ADCR_REG, + 0, 0x3F, 0, digital_tlv), +SOC_DAPM_SINGLE_TLV("Sidetone DMIC1L Volume", LM49453_P0_STN_VOL_DMIC1L_REG, + 0, 0x3F, 0, digital_tlv), +SOC_DAPM_SINGLE_TLV("Sidetone DMIC1R Volume", LM49453_P0_STN_VOL_DMIC1R_REG, + 0, 0x3F, 0, digital_tlv), +SOC_DAPM_SINGLE_TLV("Sidetone DMIC2L Volume", LM49453_P0_STN_VOL_DMIC2L_REG, + 0, 0x3F, 0, digital_tlv), +SOC_DAPM_SINGLE_TLV("Sidetone DMIC2R Volume", LM49453_P0_STN_VOL_DMIC2R_REG, + 0, 0x3F, 0, digital_tlv), +}; + +static const struct snd_kcontrol_new lm49453_snd_controls[] = { + /* mic1 and mic2 supports mono only */ + SOC_SINGLE_TLV("Mic1 Volume", LM49453_P0_ADC_LEVELL_REG, 0, 6, + 0, digital_tlv), + SOC_SINGLE_TLV("Mic2 Volume", LM49453_P0_ADC_LEVELR_REG, 0, 6, + 0, digital_tlv), + + SOC_DOUBLE_R_TLV("DMIC1 Volume", LM49453_P0_DMIC1_LEVELL_REG, + LM49453_P0_DMIC1_LEVELR_REG, 0, 6, 0, digital_tlv), + SOC_DOUBLE_R_TLV("DMIC2 Volume", LM49453_P0_DMIC2_LEVELL_REG, + LM49453_P0_DMIC2_LEVELR_REG, 0, 6, 0, digital_tlv), + + SOC_DAPM_ENUM("Mic2Mode", lm49453_mic2mode_enum), + SOC_DAPM_ENUM("DMIC12 SRC", lm49453_dmic12_cfg_enum), + SOC_DAPM_ENUM("DMIC34 SRC", lm49453_dmic34_cfg_enum), + + /* Capture path filter enable */ + SOC_SINGLE("DMIC1 HPFilter Switch", LM49453_P0_ADC_FX_ENABLES_REG, + 0, 1, 0), + SOC_SINGLE("DMIC2 HPFilter Switch", LM49453_P0_ADC_FX_ENABLES_REG, + 1, 1, 0), + SOC_SINGLE("ADC HPFilter Switch", LM49453_P0_ADC_FX_ENABLES_REG, + 2, 1, 0), + + SOC_DOUBLE_R_TLV("DAC HP Volume", LM49453_P0_DAC_HP_LEVELL_REG, + LM49453_P0_DAC_HP_LEVELR_REG, 0, 6, 0, digital_tlv), + SOC_DOUBLE_R_TLV("DAC LO Volume", LM49453_P0_DAC_LO_LEVELL_REG, + LM49453_P0_DAC_LO_LEVELR_REG, 0, 6, 0, digital_tlv), + SOC_DOUBLE_R_TLV("DAC LS Volume", LM49453_P0_DAC_LS_LEVELL_REG, + LM49453_P0_DAC_LS_LEVELR_REG, 0, 6, 0, digital_tlv), + SOC_DOUBLE_R_TLV("DAC HA Volume", LM49453_P0_DAC_HA_LEVELL_REG, + LM49453_P0_DAC_HA_LEVELR_REG, 0, 6, 0, digital_tlv), + + SOC_SINGLE_TLV("EP Volume", LM49453_P0_DAC_LS_LEVELL_REG, + 0, 6, 0, digital_tlv), + + SOC_SINGLE_TLV("PORT1_1_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL1_REG, + 0, 3, 0, port_tlv), + SOC_SINGLE_TLV("PORT1_2_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL1_REG, + 2, 3, 0, port_tlv), + SOC_SINGLE_TLV("PORT1_3_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL1_REG, + 4, 3, 0, port_tlv), + SOC_SINGLE_TLV("PORT1_4_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL1_REG, + 6, 3, 0, port_tlv), + SOC_SINGLE_TLV("PORT1_5_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL2_REG, + 0, 3, 0, port_tlv), + SOC_SINGLE_TLV("PORT1_6_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL2_REG, + 2, 3, 0, port_tlv), + SOC_SINGLE_TLV("PORT1_7_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL2_REG, + 4, 3, 0, port_tlv), + SOC_SINGLE_TLV("PORT1_8_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL2_REG, + 6, 3, 0, port_tlv), + + SOC_SINGLE_TLV("PORT2_1_RX_LVL Volume", LM49453_P0_PORT2_RX_LVL_REG, + 0, 3, 0, port_tlv), + SOC_SINGLE_TLV("PORT2_2_RX_LVL Volume", LM49453_P0_PORT2_RX_LVL_REG, + 2, 3, 0, port_tlv), + + SOC_SINGLE("Port1 Playback Switch", LM49453_P0_AUDIO_PORT1_BASIC_REG, + 1, 1, 0), + SOC_SINGLE("Port2 Playback Switch", LM49453_P0_AUDIO_PORT2_BASIC_REG, + 1, 1, 0), + SOC_SINGLE("Port1 Capture Switch", LM49453_P0_AUDIO_PORT1_BASIC_REG, + 2, 1, 0), + SOC_SINGLE("Port2 Capture Switch", LM49453_P0_AUDIO_PORT2_BASIC_REG, + 2, 1, 0) + +}; + +/* DAPM widgets */ +static const struct snd_soc_dapm_widget lm49453_dapm_widgets[] = { + + /* All end points HP,EP, LS, Lineout and Haptic */ + SND_SOC_DAPM_OUTPUT("HPOUTL"), + SND_SOC_DAPM_OUTPUT("HPOUTR"), + SND_SOC_DAPM_OUTPUT("EPOUT"), + SND_SOC_DAPM_OUTPUT("LSOUTL"), + SND_SOC_DAPM_OUTPUT("LSOUTR"), + SND_SOC_DAPM_OUTPUT("LOOUTR"), + SND_SOC_DAPM_OUTPUT("LOOUTL"), + SND_SOC_DAPM_OUTPUT("HAOUTL"), + SND_SOC_DAPM_OUTPUT("HAOUTR"), + + SND_SOC_DAPM_INPUT("AMIC1"), + SND_SOC_DAPM_INPUT("AMIC2"), + SND_SOC_DAPM_INPUT("DMIC1DAT"), + SND_SOC_DAPM_INPUT("DMIC2DAT"), + SND_SOC_DAPM_INPUT("AUXL"), + SND_SOC_DAPM_INPUT("AUXR"), + + SND_SOC_DAPM_PGA("PORT1_1_RX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PORT1_2_RX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PORT1_3_RX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PORT1_4_RX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PORT1_5_RX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PORT1_6_RX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PORT1_7_RX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PORT1_8_RX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PORT2_1_RX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PORT2_2_RX", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("AMIC1Bias", LM49453_P0_MICL_REG, 6, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("AMIC2Bias", LM49453_P0_MICR_REG, 6, 0, NULL, 0), + + /* playback path driver enables */ + SND_SOC_DAPM_OUT_DRV("Headset Switch", + LM49453_P0_PMC_SETUP_REG, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("Earpiece Switch", + LM49453_P0_EP_REG, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("Speaker Left Switch", + LM49453_P0_DIS_PKVL_FB_REG, 0, 1, NULL, 0), + SND_SOC_DAPM_OUT_DRV("Speaker Right Switch", + LM49453_P0_DIS_PKVL_FB_REG, 1, 1, NULL, 0), + SND_SOC_DAPM_OUT_DRV("Haptic Left Switch", + LM49453_P0_DIS_PKVL_FB_REG, 2, 1, NULL, 0), + SND_SOC_DAPM_OUT_DRV("Haptic Right Switch", + LM49453_P0_DIS_PKVL_FB_REG, 3, 1, NULL, 0), + + /* DAC */ + SND_SOC_DAPM_DAC("HPL DAC", "Headset", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("HPR DAC", "Headset", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("LSL DAC", "Speaker", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("LSR DAC", "Speaker", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("HAL DAC", "Haptic", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("HAR DAC", "Haptic", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("LOL DAC", "Lineout", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("LOR DAC", "Lineout", SND_SOC_NOPM, 0, 0), + + + SND_SOC_DAPM_PGA("AUXL Input", + LM49453_P0_ANALOG_MIXER_ADC_REG, 2, 0, NULL, 0), + SND_SOC_DAPM_PGA("AUXR Input", + LM49453_P0_ANALOG_MIXER_ADC_REG, 3, 0, NULL, 0), + + SND_SOC_DAPM_PGA("Sidetone", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* ADC */ + SND_SOC_DAPM_ADC("DMIC1 Left", "Capture", SND_SOC_NOPM, 1, 0), + SND_SOC_DAPM_ADC("DMIC1 Right", "Capture", SND_SOC_NOPM, 1, 0), + SND_SOC_DAPM_ADC("DMIC2 Left", "Capture", SND_SOC_NOPM, 1, 0), + SND_SOC_DAPM_ADC("DMIC2 Right", "Capture", SND_SOC_NOPM, 1, 0), + + SND_SOC_DAPM_ADC("ADC Left", "Capture", SND_SOC_NOPM, 1, 0), + SND_SOC_DAPM_ADC("ADC Right", "Capture", SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_MUX("ADCL Mux", SND_SOC_NOPM, 0, 0, + &lm49453_adcl_mux_control), + SND_SOC_DAPM_MUX("ADCR Mux", SND_SOC_NOPM, 0, 0, + &lm49453_adcr_mux_control), + + SND_SOC_DAPM_MUX("Mic1 Input", + SND_SOC_NOPM, 0, 0, &lm49453_adcl_mux_control), + + SND_SOC_DAPM_MUX("Mic2 Input", + SND_SOC_NOPM, 0, 0, &lm49453_adcr_mux_control), + + /* AIF */ + SND_SOC_DAPM_AIF_IN("PORT1_SDI", NULL, 0, + LM49453_P0_PULL_CONFIG1_REG, 2, 0), + SND_SOC_DAPM_AIF_IN("PORT2_SDI", NULL, 0, + LM49453_P0_PULL_CONFIG1_REG, 6, 0), + + SND_SOC_DAPM_AIF_OUT("PORT1_SDO", NULL, 0, + LM49453_P0_PULL_CONFIG1_REG, 3, 0), + SND_SOC_DAPM_AIF_OUT("PORT2_SDO", NULL, 0, + LM49453_P0_PULL_CONFIG1_REG, 7, 0), + + /* Port1 TX controls */ + SND_SOC_DAPM_OUT_DRV("P1_1_TX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("P1_2_TX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("P1_3_TX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("P1_4_TX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("P1_5_TX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("P1_6_TX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("P1_7_TX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("P1_8_TX", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Port2 TX controls */ + SND_SOC_DAPM_OUT_DRV("P2_1_TX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("P2_2_TX", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Sidetone Mixer */ + SND_SOC_DAPM_MIXER("Sidetone Mixer", SND_SOC_NOPM, 0, 0, + lm49453_sidetone_mixer_controls, + ARRAY_SIZE(lm49453_sidetone_mixer_controls)), + + /* DAC MIXERS */ + SND_SOC_DAPM_MIXER("HPL Mixer", SND_SOC_NOPM, 0, 0, + lm49453_headset_left_mixer, + ARRAY_SIZE(lm49453_headset_left_mixer)), + SND_SOC_DAPM_MIXER("HPR Mixer", SND_SOC_NOPM, 0, 0, + lm49453_headset_right_mixer, + ARRAY_SIZE(lm49453_headset_right_mixer)), + SND_SOC_DAPM_MIXER("LOL Mixer", SND_SOC_NOPM, 0, 0, + lm49453_lineout_left_mixer, + ARRAY_SIZE(lm49453_lineout_left_mixer)), + SND_SOC_DAPM_MIXER("LOR Mixer", SND_SOC_NOPM, 0, 0, + lm49453_lineout_right_mixer, + ARRAY_SIZE(lm49453_lineout_right_mixer)), + SND_SOC_DAPM_MIXER("LSL Mixer", SND_SOC_NOPM, 0, 0, + lm49453_speaker_left_mixer, + ARRAY_SIZE(lm49453_speaker_left_mixer)), + SND_SOC_DAPM_MIXER("LSR Mixer", SND_SOC_NOPM, 0, 0, + lm49453_speaker_right_mixer, + ARRAY_SIZE(lm49453_speaker_right_mixer)), + SND_SOC_DAPM_MIXER("HAL Mixer", SND_SOC_NOPM, 0, 0, + lm49453_haptic_left_mixer, + ARRAY_SIZE(lm49453_haptic_left_mixer)), + SND_SOC_DAPM_MIXER("HAR Mixer", SND_SOC_NOPM, 0, 0, + lm49453_haptic_right_mixer, + ARRAY_SIZE(lm49453_haptic_right_mixer)), + + /* Capture Mixer */ + SND_SOC_DAPM_MIXER("Port1_1 Mixer", SND_SOC_NOPM, 0, 0, + lm49453_port1_tx1_mixer, + ARRAY_SIZE(lm49453_port1_tx1_mixer)), + SND_SOC_DAPM_MIXER("Port1_2 Mixer", SND_SOC_NOPM, 0, 0, + lm49453_port1_tx2_mixer, + ARRAY_SIZE(lm49453_port1_tx2_mixer)), + SND_SOC_DAPM_MIXER("Port1_3 Mixer", SND_SOC_NOPM, 0, 0, + lm49453_port1_tx3_mixer, + ARRAY_SIZE(lm49453_port1_tx3_mixer)), + SND_SOC_DAPM_MIXER("Port1_4 Mixer", SND_SOC_NOPM, 0, 0, + lm49453_port1_tx4_mixer, + ARRAY_SIZE(lm49453_port1_tx4_mixer)), + SND_SOC_DAPM_MIXER("Port1_5 Mixer", SND_SOC_NOPM, 0, 0, + lm49453_port1_tx5_mixer, + ARRAY_SIZE(lm49453_port1_tx5_mixer)), + SND_SOC_DAPM_MIXER("Port1_6 Mixer", SND_SOC_NOPM, 0, 0, + lm49453_port1_tx6_mixer, + ARRAY_SIZE(lm49453_port1_tx6_mixer)), + SND_SOC_DAPM_MIXER("Port1_7 Mixer", SND_SOC_NOPM, 0, 0, + lm49453_port1_tx7_mixer, + ARRAY_SIZE(lm49453_port1_tx7_mixer)), + SND_SOC_DAPM_MIXER("Port1_8 Mixer", SND_SOC_NOPM, 0, 0, + lm49453_port1_tx8_mixer, + ARRAY_SIZE(lm49453_port1_tx8_mixer)), + + SND_SOC_DAPM_MIXER("Port2_1 Mixer", SND_SOC_NOPM, 0, 0, + lm49453_port2_tx1_mixer, + ARRAY_SIZE(lm49453_port2_tx1_mixer)), + SND_SOC_DAPM_MIXER("Port2_2 Mixer", SND_SOC_NOPM, 0, 0, + lm49453_port2_tx2_mixer, + ARRAY_SIZE(lm49453_port2_tx2_mixer)), +}; + +static const struct snd_soc_dapm_route lm49453_audio_map[] = { + /* Port SDI mapping */ + { "PORT1_1_RX", "Port1 Playback Switch", "PORT1_SDI" }, + { "PORT1_2_RX", "Port1 Playback Switch", "PORT1_SDI" }, + { "PORT1_3_RX", "Port1 Playback Switch", "PORT1_SDI" }, + { "PORT1_4_RX", "Port1 Playback Switch", "PORT1_SDI" }, + { "PORT1_5_RX", "Port1 Playback Switch", "PORT1_SDI" }, + { "PORT1_6_RX", "Port1 Playback Switch", "PORT1_SDI" }, + { "PORT1_7_RX", "Port1 Playback Switch", "PORT1_SDI" }, + { "PORT1_8_RX", "Port1 Playback Switch", "PORT1_SDI" }, + + { "PORT2_1_RX", "Port2 Playback Switch", "PORT2_SDI" }, + { "PORT2_2_RX", "Port2 Playback Switch", "PORT2_SDI" }, + + /* HP mapping */ + { "HPL Mixer", "Port1_1 Switch", "PORT1_1_RX" }, + { "HPL Mixer", "Port1_2 Switch", "PORT1_2_RX" }, + { "HPL Mixer", "Port1_3 Switch", "PORT1_3_RX" }, + { "HPL Mixer", "Port1_4 Switch", "PORT1_4_RX" }, + { "HPL Mixer", "Port1_5 Switch", "PORT1_5_RX" }, + { "HPL Mixer", "Port1_6 Switch", "PORT1_6_RX" }, + { "HPL Mixer", "Port1_7 Switch", "PORT1_7_RX" }, + { "HPL Mixer", "Port1_8 Switch", "PORT1_8_RX" }, + + { "HPL Mixer", "Port2_1 Switch", "PORT2_1_RX" }, + { "HPL Mixer", "Port2_2 Switch", "PORT2_2_RX" }, + + { "HPL Mixer", "ADCL Switch", "ADC Left" }, + { "HPL Mixer", "ADCR Switch", "ADC Right" }, + { "HPL Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "HPL Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "HPL Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "HPL Mixer", "DMIC2R Switch", "DMIC2 Right" }, + { "HPL Mixer", "Sidetone Switch", "Sidetone" }, + + { "HPL DAC", NULL, "HPL Mixer" }, + + { "HPR Mixer", "Port1_1 Switch", "PORT1_1_RX" }, + { "HPR Mixer", "Port1_2 Switch", "PORT1_2_RX" }, + { "HPR Mixer", "Port1_3 Switch", "PORT1_3_RX" }, + { "HPR Mixer", "Port1_4 Switch", "PORT1_4_RX" }, + { "HPR Mixer", "Port1_5 Switch", "PORT1_5_RX" }, + { "HPR Mixer", "Port1_6 Switch", "PORT1_6_RX" }, + { "HPR Mixer", "Port1_7 Switch", "PORT1_7_RX" }, + { "HPR Mixer", "Port1_8 Switch", "PORT1_8_RX" }, + + /* Port 2 */ + { "HPR Mixer", "Port2_1 Switch", "PORT2_1_RX" }, + { "HPR Mixer", "Port2_2 Switch", "PORT2_2_RX" }, + + { "HPR Mixer", "ADCL Switch", "ADC Left" }, + { "HPR Mixer", "ADCR Switch", "ADC Right" }, + { "HPR Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "HPR Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "HPR Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "HPR Mixer", "DMIC2L Switch", "DMIC2 Right" }, + { "HPR Mixer", "Sidetone Switch", "Sidetone" }, + + { "HPR DAC", NULL, "HPR Mixer" }, + + { "HPOUTL", "Headset Switch", "HPL DAC"}, + { "HPOUTR", "Headset Switch", "HPR DAC"}, + + /* EP map */ + { "EPOUT", "Earpiece Switch", "HPL DAC" }, + + /* Speaker map */ + { "LSL Mixer", "Port1_1 Switch", "PORT1_1_RX" }, + { "LSL Mixer", "Port1_2 Switch", "PORT1_2_RX" }, + { "LSL Mixer", "Port1_3 Switch", "PORT1_3_RX" }, + { "LSL Mixer", "Port1_4 Switch", "PORT1_4_RX" }, + { "LSL Mixer", "Port1_5 Switch", "PORT1_5_RX" }, + { "LSL Mixer", "Port1_6 Switch", "PORT1_6_RX" }, + { "LSL Mixer", "Port1_7 Switch", "PORT1_7_RX" }, + { "LSL Mixer", "Port1_8 Switch", "PORT1_8_RX" }, + + /* Port 2 */ + { "LSL Mixer", "Port2_1 Switch", "PORT2_1_RX" }, + { "LSL Mixer", "Port2_2 Switch", "PORT2_2_RX" }, + + { "LSL Mixer", "ADCL Switch", "ADC Left" }, + { "LSL Mixer", "ADCR Switch", "ADC Right" }, + { "LSL Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "LSL Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "LSL Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "LSL Mixer", "DMIC2R Switch", "DMIC2 Right" }, + { "LSL Mixer", "Sidetone Switch", "Sidetone" }, + + { "LSL DAC", NULL, "LSL Mixer" }, + + { "LSR Mixer", "Port1_1 Switch", "PORT1_1_RX" }, + { "LSR Mixer", "Port1_2 Switch", "PORT1_2_RX" }, + { "LSR Mixer", "Port1_3 Switch", "PORT1_3_RX" }, + { "LSR Mixer", "Port1_4 Switch", "PORT1_4_RX" }, + { "LSR Mixer", "Port1_5 Switch", "PORT1_5_RX" }, + { "LSR Mixer", "Port1_6 Switch", "PORT1_6_RX" }, + { "LSR Mixer", "Port1_7 Switch", "PORT1_7_RX" }, + { "LSR Mixer", "Port1_8 Switch", "PORT1_8_RX" }, + + /* Port 2 */ + { "LSR Mixer", "Port2_1 Switch", "PORT2_1_RX" }, + { "LSR Mixer", "Port2_2 Switch", "PORT2_2_RX" }, + + { "LSR Mixer", "ADCL Switch", "ADC Left" }, + { "LSR Mixer", "ADCR Switch", "ADC Right" }, + { "LSR Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "LSR Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "LSR Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "LSR Mixer", "DMIC2R Switch", "DMIC2 Right" }, + { "LSR Mixer", "Sidetone Switch", "Sidetone" }, + + { "LSR DAC", NULL, "LSR Mixer" }, + + { "LSOUTL", "Speaker Left Switch", "LSL DAC"}, + { "LSOUTR", "Speaker Left Switch", "LSR DAC"}, + + /* Haptic map */ + { "HAL Mixer", "Port1_1 Switch", "PORT1_1_RX" }, + { "HAL Mixer", "Port1_2 Switch", "PORT1_2_RX" }, + { "HAL Mixer", "Port1_3 Switch", "PORT1_3_RX" }, + { "HAL Mixer", "Port1_4 Switch", "PORT1_4_RX" }, + { "HAL Mixer", "Port1_5 Switch", "PORT1_5_RX" }, + { "HAL Mixer", "Port1_6 Switch", "PORT1_6_RX" }, + { "HAL Mixer", "Port1_7 Switch", "PORT1_7_RX" }, + { "HAL Mixer", "Port1_8 Switch", "PORT1_8_RX" }, + + /* Port 2 */ + { "HAL Mixer", "Port2_1 Switch", "PORT2_1_RX" }, + { "HAL Mixer", "Port2_2 Switch", "PORT2_2_RX" }, + + { "HAL Mixer", "ADCL Switch", "ADC Left" }, + { "HAL Mixer", "ADCR Switch", "ADC Right" }, + { "HAL Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "HAL Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "HAL Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "HAL Mixer", "DMIC2R Switch", "DMIC2 Right" }, + { "HAL Mixer", "Sidetone Switch", "Sidetone" }, + + { "HAL DAC", NULL, "HAL Mixer" }, + + { "HAR Mixer", "Port1_1 Switch", "PORT1_1_RX" }, + { "HAR Mixer", "Port1_2 Switch", "PORT1_2_RX" }, + { "HAR Mixer", "Port1_3 Switch", "PORT1_3_RX" }, + { "HAR Mixer", "Port1_4 Switch", "PORT1_4_RX" }, + { "HAR Mixer", "Port1_5 Switch", "PORT1_5_RX" }, + { "HAR Mixer", "Port1_6 Switch", "PORT1_6_RX" }, + { "HAR Mixer", "Port1_7 Switch", "PORT1_7_RX" }, + { "HAR Mixer", "Port1_8 Switch", "PORT1_8_RX" }, + + /* Port 2 */ + { "HAR Mixer", "Port2_1 Switch", "PORT2_1_RX" }, + { "HAR Mixer", "Port2_2 Switch", "PORT2_2_RX" }, + + { "HAR Mixer", "ADCL Switch", "ADC Left" }, + { "HAR Mixer", "ADCR Switch", "ADC Right" }, + { "HAR Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "HAR Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "HAR Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "HAR Mixer", "DMIC2R Switch", "DMIC2 Right" }, + { "HAR Mixer", "Sideton Switch", "Sidetone" }, + + { "HAR DAC", NULL, "HAR Mixer" }, + + { "HAOUTL", "Haptic Left Switch", "HAL DAC" }, + { "HAOUTR", "Haptic Right Switch", "HAR DAC" }, + + /* Lineout map */ + { "LOL Mixer", "Port1_1 Switch", "PORT1_1_RX" }, + { "LOL Mixer", "Port1_2 Switch", "PORT1_2_RX" }, + { "LOL Mixer", "Port1_3 Switch", "PORT1_3_RX" }, + { "LOL Mixer", "Port1_4 Switch", "PORT1_4_RX" }, + { "LOL Mixer", "Port1_5 Switch", "PORT1_5_RX" }, + { "LOL Mixer", "Port1_6 Switch", "PORT1_6_RX" }, + { "LOL Mixer", "Port1_7 Switch", "PORT1_7_RX" }, + { "LOL Mixer", "Port1_8 Switch", "PORT1_8_RX" }, + + /* Port 2 */ + { "LOL Mixer", "Port2_1 Switch", "PORT2_1_RX" }, + { "LOL Mixer", "Port2_2 Switch", "PORT2_2_RX" }, + + { "LOL Mixer", "ADCL Switch", "ADC Left" }, + { "LOL Mixer", "ADCR Switch", "ADC Right" }, + { "LOL Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "LOL Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "LOL Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "LOL Mixer", "DMIC2R Switch", "DMIC2 Right" }, + { "LOL Mixer", "Sidetone Switch", "Sidetone" }, + + { "LOL DAC", NULL, "LOL Mixer" }, + + { "LOR Mixer", "Port1_1 Switch", "PORT1_1_RX" }, + { "LOR Mixer", "Port1_2 Switch", "PORT1_2_RX" }, + { "LOR Mixer", "Port1_3 Switch", "PORT1_3_RX" }, + { "LOR Mixer", "Port1_4 Switch", "PORT1_4_RX" }, + { "LOR Mixer", "Port1_5 Switch", "PORT1_5_RX" }, + { "LOR Mixer", "Port1_6 Switch", "PORT1_6_RX" }, + { "LOR Mixer", "Port1_7 Switch", "PORT1_7_RX" }, + { "LOR Mixer", "Port1_8 Switch", "PORT1_8_RX" }, + + /* Port 2 */ + { "LOR Mixer", "Port2_1 Switch", "PORT2_1_RX" }, + { "LOR Mixer", "Port2_2 Switch", "PORT2_2_RX" }, + + { "LOR Mixer", "ADCL Switch", "ADC Left" }, + { "LOR Mixer", "ADCR Switch", "ADC Right" }, + { "LOR Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "LOR Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "LOR Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "LOR Mixer", "DMIC2R Switch", "DMIC2 Right" }, + { "LOR Mixer", "Sidetone Switch", "Sidetone" }, + + { "LOR DAC", NULL, "LOR Mixer" }, + + { "LOOUTL", NULL, "LOL DAC" }, + { "LOOUTR", NULL, "LOR DAC" }, + + /* TX map */ + /* Port1 mappings */ + { "Port1_1 Mixer", "ADCL Switch", "ADC Left" }, + { "Port1_1 Mixer", "ADCR Switch", "ADC Right" }, + { "Port1_1 Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "Port1_1 Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "Port1_1 Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "Port1_1 Mixer", "DMIC2R Switch", "DMIC2 Right" }, + + { "Port1_2 Mixer", "ADCL Switch", "ADC Left" }, + { "Port1_2 Mixer", "ADCR Switch", "ADC Right" }, + { "Port1_2 Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "Port1_2 Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "Port1_2 Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "Port1_2 Mixer", "DMIC2R Switch", "DMIC2 Right" }, + + { "Port1_3 Mixer", "ADCL Switch", "ADC Left" }, + { "Port1_3 Mixer", "ADCR Switch", "ADC Right" }, + { "Port1_3 Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "Port1_3 Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "Port1_3 Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "Port1_3 Mixer", "DMIC2R Switch", "DMIC2 Right" }, + + { "Port1_4 Mixer", "ADCL Switch", "ADC Left" }, + { "Port1_4 Mixer", "ADCR Switch", "ADC Right" }, + { "Port1_4 Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "Port1_4 Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "Port1_4 Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "Port1_4 Mixer", "DMIC2R Switch", "DMIC2 Right" }, + + { "Port1_5 Mixer", "ADCL Switch", "ADC Left" }, + { "Port1_5 Mixer", "ADCR Switch", "ADC Right" }, + { "Port1_5 Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "Port1_5 Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "Port1_5 Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "Port1_5 Mixer", "DMIC2R Switch", "DMIC2 Right" }, + + { "Port1_6 Mixer", "ADCL Switch", "ADC Left" }, + { "Port1_6 Mixer", "ADCR Switch", "ADC Right" }, + { "Port1_6 Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "Port1_6 Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "Port1_6 Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "Port1_6 Mixer", "DMIC2R Switch", "DMIC2 Right" }, + + { "Port1_7 Mixer", "ADCL Switch", "ADC Left" }, + { "Port1_7 Mixer", "ADCR Switch", "ADC Right" }, + { "Port1_7 Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "Port1_7 Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "Port1_7 Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "Port1_7 Mixer", "DMIC2R Switch", "DMIC2 Right" }, + + { "Port1_8 Mixer", "ADCL Switch", "ADC Left" }, + { "Port1_8 Mixer", "ADCR Switch", "ADC Right" }, + { "Port1_8 Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "Port1_8 Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "Port1_8 Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "Port1_8 Mixer", "DMIC2R Switch", "DMIC2 Right" }, + + { "Port2_1 Mixer", "ADCL Switch", "ADC Left" }, + { "Port2_1 Mixer", "ADCR Switch", "ADC Right" }, + { "Port2_1 Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "Port2_1 Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "Port2_1 Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "Port2_1 Mixer", "DMIC2R Switch", "DMIC2 Right" }, + + { "Port2_2 Mixer", "ADCL Switch", "ADC Left" }, + { "Port2_2 Mixer", "ADCR Switch", "ADC Right" }, + { "Port2_2 Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "Port2_2 Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "Port2_2 Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "Port2_2 Mixer", "DMIC2R Switch", "DMIC2 Right" }, + + { "P1_1_TX", NULL, "Port1_1 Mixer" }, + { "P1_2_TX", NULL, "Port1_2 Mixer" }, + { "P1_3_TX", NULL, "Port1_3 Mixer" }, + { "P1_4_TX", NULL, "Port1_4 Mixer" }, + { "P1_5_TX", NULL, "Port1_5 Mixer" }, + { "P1_6_TX", NULL, "Port1_6 Mixer" }, + { "P1_7_TX", NULL, "Port1_7 Mixer" }, + { "P1_8_TX", NULL, "Port1_8 Mixer" }, + + { "P2_1_TX", NULL, "Port2_1 Mixer" }, + { "P2_2_TX", NULL, "Port2_2 Mixer" }, + + { "PORT1_SDO", "Port1 Capture Switch", "P1_1_TX"}, + { "PORT1_SDO", "Port1 Capture Switch", "P1_2_TX"}, + { "PORT1_SDO", "Port1 Capture Switch", "P1_3_TX"}, + { "PORT1_SDO", "Port1 Capture Switch", "P1_4_TX"}, + { "PORT1_SDO", "Port1 Capture Switch", "P1_5_TX"}, + { "PORT1_SDO", "Port1 Capture Switch", "P1_6_TX"}, + { "PORT1_SDO", "Port1 Capture Switch", "P1_7_TX"}, + { "PORT1_SDO", "Port1 Capture Switch", "P1_8_TX"}, + + { "PORT2_SDO", "Port2 Capture Switch", "P2_1_TX"}, + { "PORT2_SDO", "Port2 Capture Switch", "P2_2_TX"}, + + { "Mic1 Input", NULL, "AMIC1" }, + { "Mic2 Input", NULL, "AMIC2" }, + + { "AUXL Input", NULL, "AUXL" }, + { "AUXR Input", NULL, "AUXR" }, + + /* AUX connections */ + { "ADCL Mux", "Aux_L", "AUXL Input" }, + { "ADCL Mux", "MIC1", "Mic1 Input" }, + + { "ADCR Mux", "Aux_R", "AUXR Input" }, + { "ADCR Mux", "MIC2", "Mic2 Input" }, + + /* ADC connection */ + { "ADC Left", NULL, "ADCL Mux"}, + { "ADC Right", NULL, "ADCR Mux"}, + + { "DMIC1 Left", NULL, "DMIC1DAT"}, + { "DMIC1 Right", NULL, "DMIC1DAT"}, + { "DMIC2 Left", NULL, "DMIC2DAT"}, + { "DMIC2 Right", NULL, "DMIC2DAT"}, + + /* Sidetone map */ + { "Sidetone Mixer", NULL, "ADC Left" }, + { "Sidetone Mixer", NULL, "ADC Right" }, + { "Sidetone Mixer", NULL, "DMIC1 Left" }, + { "Sidetone Mixer", NULL, "DMIC1 Right" }, + { "Sidetone Mixer", NULL, "DMIC2 Left" }, + { "Sidetone Mixer", NULL, "DMIC2 Right" }, + + { "Sidetone", "Sidetone Switch", "Sidetone Mixer" }, +}; + +static int lm49453_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct lm49453_priv *lm49453 = snd_soc_codec_get_drvdata(codec); + u16 clk_div = 0; + + lm49453->fs_rate = params_rate(params); + + /* Setting DAC clock dividers based on substream sample rate. */ + switch (lm49453->fs_rate) { + case 8000: + case 16000: + case 32000: + case 24000: + case 48000: + clk_div = 256; + break; + case 11025: + case 22050: + case 44100: + clk_div = 216; + break; + case 96000: + clk_div = 127; + break; + default: + return -EINVAL; + } + + snd_soc_write(codec, LM49453_P0_ADC_CLK_DIV_REG, clk_div); + snd_soc_write(codec, LM49453_P0_DAC_HP_CLK_DIV_REG, clk_div); + + return 0; +} + +static int lm49453_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + + u16 aif_val; + int mode = 0; + int clk_phase = 0; + int clk_shift = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + aif_val = 0; + break; + case SND_SOC_DAIFMT_CBS_CFM: + aif_val = LM49453_AUDIO_PORT1_BASIC_SYNC_MS; + break; + case SND_SOC_DAIFMT_CBM_CFS: + aif_val = LM49453_AUDIO_PORT1_BASIC_CLK_MS; + break; + case SND_SOC_DAIFMT_CBM_CFM: + aif_val = LM49453_AUDIO_PORT1_BASIC_CLK_MS | + LM49453_AUDIO_PORT1_BASIC_SYNC_MS; + break; + default: + return -EINVAL; + } + + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_DSP_A: + mode = 1; + clk_phase = (1 << 5); + clk_shift = 1; + break; + case SND_SOC_DAIFMT_DSP_B: + mode = 1; + clk_phase = (1 << 5); + clk_shift = 0; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, LM49453_P0_AUDIO_PORT1_BASIC_REG, + LM49453_AUDIO_PORT1_BASIC_FMT_MASK|BIT(1)|BIT(5), + (aif_val | mode | clk_phase)); + + snd_soc_write(codec, LM49453_P0_AUDIO_PORT1_RX_MSB_REG, clk_shift); + + return 0; +} + +static int lm49453_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + u16 pll_clk = 0; + + switch (freq) { + case 12288000: + case 26000000: + case 19200000: + /* pll clk slection */ + pll_clk = 0; + break; + case 48000: + case 32576: + /* fll clk slection */ + pll_clk = BIT(4); + return 0; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, LM49453_P0_PMC_SETUP_REG, BIT(4), pll_clk); + + return 0; +} + +static int lm49453_hp_mute(struct snd_soc_dai *dai, int mute) +{ + snd_soc_update_bits(dai->codec, LM49453_P0_DAC_DSP_REG, BIT(1)|BIT(0), + (mute ? (BIT(1)|BIT(0)) : 0)); + return 0; +} + +static int lm49453_lo_mute(struct snd_soc_dai *dai, int mute) +{ + snd_soc_update_bits(dai->codec, LM49453_P0_DAC_DSP_REG, BIT(3)|BIT(2), + (mute ? (BIT(3)|BIT(2)) : 0)); + return 0; +} + +static int lm49453_ls_mute(struct snd_soc_dai *dai, int mute) +{ + snd_soc_update_bits(dai->codec, LM49453_P0_DAC_DSP_REG, BIT(5)|BIT(4), + (mute ? (BIT(5)|BIT(4)) : 0)); + return 0; +} + +static int lm49453_ep_mute(struct snd_soc_dai *dai, int mute) +{ + snd_soc_update_bits(dai->codec, LM49453_P0_DAC_DSP_REG, BIT(4), + (mute ? BIT(4) : 0)); + return 0; +} + +static int lm49453_ha_mute(struct snd_soc_dai *dai, int mute) +{ + snd_soc_update_bits(dai->codec, LM49453_P0_DAC_DSP_REG, BIT(7)|BIT(6), + (mute ? (BIT(7)|BIT(6)) : 0)); + return 0; +} + +static int lm49453_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct lm49453_priv *lm49453 = snd_soc_codec_get_drvdata(codec); + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + break; + + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) + regcache_sync(lm49453->regmap); + + snd_soc_update_bits(codec, LM49453_P0_PMC_SETUP_REG, + LM49453_PMC_SETUP_CHIP_EN, LM49453_CHIP_EN); + break; + + case SND_SOC_BIAS_OFF: + snd_soc_update_bits(codec, LM49453_P0_PMC_SETUP_REG, + LM49453_PMC_SETUP_CHIP_EN, 0); + break; + } + + codec->dapm.bias_level = level; + + return 0; +} + +/* Formates supported by LM49453 driver. */ +#define LM49453_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops lm49453_headset_dai_ops = { + .hw_params = lm49453_hw_params, + .set_sysclk = lm49453_set_dai_sysclk, + .set_fmt = lm49453_set_dai_fmt, + .digital_mute = lm49453_hp_mute, +}; + +static struct snd_soc_dai_ops lm49453_speaker_dai_ops = { + .hw_params = lm49453_hw_params, + .set_sysclk = lm49453_set_dai_sysclk, + .set_fmt = lm49453_set_dai_fmt, + .digital_mute = lm49453_ls_mute, +}; + +static struct snd_soc_dai_ops lm49453_haptic_dai_ops = { + .hw_params = lm49453_hw_params, + .set_sysclk = lm49453_set_dai_sysclk, + .set_fmt = lm49453_set_dai_fmt, + .digital_mute = lm49453_ha_mute, +}; + +static struct snd_soc_dai_ops lm49453_ep_dai_ops = { + .hw_params = lm49453_hw_params, + .set_sysclk = lm49453_set_dai_sysclk, + .set_fmt = lm49453_set_dai_fmt, + .digital_mute = lm49453_ep_mute, +}; + +static struct snd_soc_dai_ops lm49453_lineout_dai_ops = { + .hw_params = lm49453_hw_params, + .set_sysclk = lm49453_set_dai_sysclk, + .set_fmt = lm49453_set_dai_fmt, + .digital_mute = lm49453_lo_mute, +}; + +/* LM49453 dai structure. */ +static const struct snd_soc_dai_driver lm49453_dai[] = { + { + .name = "LM49453 Headset", + .playback = { + .stream_name = "Headset", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = LM49453_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 5, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = LM49453_FORMATS, + }, + .ops = &lm49453_headset_dai_ops, + .symmetric_rates = 1, + }, + { + .name = "LM49453 Speaker", + .playback = { + .stream_name = "Speaker", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = LM49453_FORMATS, + }, + .ops = &lm49453_speaker_dai_ops, + }, + { + .name = "LM49453 Haptic", + .playback = { + .stream_name = "Haptic", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = LM49453_FORMATS, + }, + .ops = &lm49453_haptic_dai_ops, + }, + { + .name = "LM49453 Earpiece", + .playback = { + .stream_name = "Earpiece", + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = LM49453_FORMATS, + }, + .ops = &lm49453_ep_dai_ops, + }, + { + .name = "LM49453 line out", + .playback = { + .stream_name = "Lineout", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = LM49453_FORMATS, + }, + .ops = &lm49453_lineout_dai_ops, + }, +}; + +static int lm49453_suspend(struct snd_soc_codec *codec) +{ + lm49453_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int lm49453_resume(struct snd_soc_codec *codec) +{ + lm49453_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + return 0; +} + +static int lm49453_probe(struct snd_soc_codec *codec) +{ + struct lm49453_priv *lm49453 = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + codec->control_data = lm49453->regmap; + + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + return 0; +} + +/* power down chip */ +static int lm49453_remove(struct snd_soc_codec *codec) +{ + lm49453_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_lm49453 = { + .probe = lm49453_probe, + .remove = lm49453_remove, + .suspend = lm49453_suspend, + .resume = lm49453_resume, + .set_bias_level = lm49453_set_bias_level, + .controls = lm49453_snd_controls, + .num_controls = ARRAY_SIZE(lm49453_snd_controls), + .dapm_widgets = lm49453_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(lm49453_dapm_widgets), + .dapm_routes = lm49453_audio_map, + .num_dapm_routes = ARRAY_SIZE(lm49453_audio_map), + .idle_bias_off = true, +}; + +static const struct regmap_config lm49453_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = LM49453_MAX_REGISTER, + .reg_defaults = lm49453_reg_defs, + .num_reg_defaults = ARRAY_SIZE(lm49453_reg_defs), + .cache_type = REGCACHE_RBTREE, +}; + +static __devinit int lm49453_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct lm49453_priv *lm49453; + int ret = 0; + + lm49453 = devm_kzalloc(&i2c->dev, sizeof(struct lm49453_priv), + GFP_KERNEL); + + if (lm49453 == NULL) + return -ENOMEM; + + i2c_set_clientdata(i2c, lm49453); + + lm49453->regmap = regmap_init_i2c(i2c, &lm49453_regmap_config); + if (IS_ERR(lm49453->regmap)) { + ret = PTR_ERR(lm49453->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + ret = snd_soc_register_codec(&i2c->dev, + &soc_codec_dev_lm49453, + lm49453_dai, ARRAY_SIZE(lm49453_dai)); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to register codec: %d\n", ret); + regmap_exit(lm49453->regmap); + return ret; + } + + return ret; +} + +static int __devexit lm49453_i2c_remove(struct i2c_client *client) +{ + struct lm49453_priv *lm49453 = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + regmap_exit(lm49453->regmap); + return 0; +} + +static const struct i2c_device_id lm49453_i2c_id[] = { + { "lm49453", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, lm49453_i2c_id); + +static struct i2c_driver lm49453_i2c_driver = { + .driver = { + .name = "lm49453", + .owner = THIS_MODULE, + }, + .probe = lm49453_i2c_probe, + .remove = __devexit_p(lm49453_i2c_remove), + .id_table = lm49453_i2c_id, +}; + +module_i2c_driver(lm49453_i2c_driver); + +MODULE_DESCRIPTION("ASoC LM49453 driver"); +MODULE_AUTHOR("M R Swami Reddy <MR.Swami.Reddy@ti.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/lm49453.h b/sound/soc/codecs/lm49453.h new file mode 100644 index 000000000000..a63cfa5c0883 --- /dev/null +++ b/sound/soc/codecs/lm49453.h @@ -0,0 +1,380 @@ +/* + * lm49453.h - LM49453 ALSA Soc Audio drive + * + * Copyright (c) 2012 Texas Instruments, Inc + * + * 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; version 2 of the License. + * + */ + +#ifndef _LM49453_H +#define _LM49453_H + +#include <linux/bitops.h> + +/* LM49453_P0 register space for page0 */ +#define LM49453_P0_PMC_SETUP_REG 0x00 +#define LM49453_P0_PLL_CLK_SEL1_REG 0x01 +#define LM49453_P0_PLL_CLK_SEL2_REG 0x02 +#define LM49453_P0_PMC_CLK_DIV_REG 0x03 +#define LM49453_P0_HSDET_CLK_DIV_REG 0x04 +#define LM49453_P0_DMIC_CLK_DIV_REG 0x05 +#define LM49453_P0_ADC_CLK_DIV_REG 0x06 +#define LM49453_P0_DAC_OT_CLK_DIV_REG 0x07 +#define LM49453_P0_PLL_HF_M_REG 0x08 +#define LM49453_P0_PLL_LF_M_REG 0x09 +#define LM49453_P0_PLL_NL_REG 0x0A +#define LM49453_P0_PLL_N_MODL_REG 0x0B +#define LM49453_P0_PLL_N_MODH_REG 0x0C +#define LM49453_P0_PLL_P1_REG 0x0D +#define LM49453_P0_PLL_P2_REG 0x0E +#define LM49453_P0_FLL_REF_FREQL_REG 0x0F +#define LM49453_P0_FLL_REF_FREQH_REG 0x10 +#define LM49453_P0_VCO_TARGETLL_REG 0x11 +#define LM49453_P0_VCO_TARGETLH_REG 0x12 +#define LM49453_P0_VCO_TARGETHL_REG 0x13 +#define LM49453_P0_VCO_TARGETHH_REG 0x14 +#define LM49453_P0_PLL_CONFIG_REG 0x15 +#define LM49453_P0_DAC_CLK_SEL_REG 0x16 +#define LM49453_P0_DAC_HP_CLK_DIV_REG 0x17 + +/* Analog Mixer Input Stages */ +#define LM49453_P0_MICL_REG 0x20 +#define LM49453_P0_MICR_REG 0x21 +#define LM49453_P0_EP_REG 0x24 +#define LM49453_P0_DIS_PKVL_FB_REG 0x25 + +/* Analog Mixer Output Stages */ +#define LM49453_P0_ANALOG_MIXER_ADC_REG 0x2E + +/*ADC or DAC */ +#define LM49453_P0_ADC_DSP_REG 0x30 +#define LM49453_P0_DAC_DSP_REG 0x31 + +/* EFFECTS ENABLES */ +#define LM49453_P0_ADC_FX_ENABLES_REG 0x33 + +/* GPIO */ +#define LM49453_P0_GPIO1_REG 0x38 +#define LM49453_P0_GPIO2_REG 0x39 +#define LM49453_P0_GPIO3_REG 0x3A +#define LM49453_P0_HAP_CTL_REG 0x3B +#define LM49453_P0_HAP_FREQ_PROG_LEFTL_REG 0x3C +#define LM49453_P0_HAP_FREQ_PROG_LEFTH_REG 0x3D +#define LM49453_P0_HAP_FREQ_PROG_RIGHTL_REG 0x3E +#define LM49453_P0_HAP_FREQ_PROG_RIGHTH_REG 0x3F + +/* DIGITAL MIXER */ +#define LM49453_P0_DMIX_CLK_SEL_REG 0x40 +#define LM49453_P0_PORT1_RX_LVL1_REG 0x41 +#define LM49453_P0_PORT1_RX_LVL2_REG 0x42 +#define LM49453_P0_PORT2_RX_LVL_REG 0x43 +#define LM49453_P0_PORT1_TX1_REG 0x44 +#define LM49453_P0_PORT1_TX2_REG 0x45 +#define LM49453_P0_PORT1_TX3_REG 0x46 +#define LM49453_P0_PORT1_TX4_REG 0x47 +#define LM49453_P0_PORT1_TX5_REG 0x48 +#define LM49453_P0_PORT1_TX6_REG 0x49 +#define LM49453_P0_PORT1_TX7_REG 0x4A +#define LM49453_P0_PORT1_TX8_REG 0x4B +#define LM49453_P0_PORT2_TX1_REG 0x4C +#define LM49453_P0_PORT2_TX2_REG 0x4D +#define LM49453_P0_STN_SEL_REG 0x4F +#define LM49453_P0_DACHPL1_REG 0x50 +#define LM49453_P0_DACHPL2_REG 0x51 +#define LM49453_P0_DACHPR1_REG 0x52 +#define LM49453_P0_DACHPR2_REG 0x53 +#define LM49453_P0_DACLOL1_REG 0x54 +#define LM49453_P0_DACLOL2_REG 0x55 +#define LM49453_P0_DACLOR1_REG 0x56 +#define LM49453_P0_DACLOR2_REG 0x57 +#define LM49453_P0_DACLSL1_REG 0x58 +#define LM49453_P0_DACLSL2_REG 0x59 +#define LM49453_P0_DACLSR1_REG 0x5A +#define LM49453_P0_DACLSR2_REG 0x5B +#define LM49453_P0_DACHAL1_REG 0x5C +#define LM49453_P0_DACHAL2_REG 0x5D +#define LM49453_P0_DACHAR1_REG 0x5E +#define LM49453_P0_DACHAR2_REG 0x5F + +/* AUDIO PORT 1 (TDM) */ +#define LM49453_P0_AUDIO_PORT1_BASIC_REG 0x60 +#define LM49453_P0_AUDIO_PORT1_CLK_GEN1_REG 0x61 +#define LM49453_P0_AUDIO_PORT1_CLK_GEN2_REG 0x62 +#define LM49453_P0_AUDIO_PORT1_CLK_GEN3_REG 0x63 +#define LM49453_P0_AUDIO_PORT1_SYNC_RATE_REG 0x64 +#define LM49453_P0_AUDIO_PORT1_SYNC_SDO_SETUP_REG 0x65 +#define LM49453_P0_AUDIO_PORT1_DATA_WIDTH_REG 0x66 +#define LM49453_P0_AUDIO_PORT1_RX_MSB_REG 0x67 +#define LM49453_P0_AUDIO_PORT1_TX_MSB_REG 0x68 +#define LM49453_P0_AUDIO_PORT1_TDM_CHANNELS_REG 0x69 + +/* AUDIO PORT 2 */ +#define LM49453_P0_AUDIO_PORT2_BASIC_REG 0x6A +#define LM49453_P0_AUDIO_PORT2_CLK_GEN1_REG 0x6B +#define LM49453_P0_AUDIO_PORT2_CLK_GEN2_REG 0x6C +#define LM49453_P0_AUDIO_PORT2_SYNC_GEN_REG 0x6D +#define LM49453_P0_AUDIO_PORT2_DATA_WIDTH_REG 0x6E +#define LM49453_P0_AUDIO_PORT2_RX_MODE_REG 0x6F +#define LM49453_P0_AUDIO_PORT2_TX_MODE_REG 0x70 + +/* SAMPLE RATE */ +#define LM49453_P0_PORT1_SR_LSB_REG 0x79 +#define LM49453_P0_PORT1_SR_MSB_REG 0x7A +#define LM49453_P0_PORT2_SR_LSB_REG 0x7B +#define LM49453_P0_PORT2_SR_MSB_REG 0x7C + +/* EFFECTS - HPFs */ +#define LM49453_P0_HPF_REG 0x80 + +/* EFFECTS ADC ALC */ +#define LM49453_P0_ADC_ALC1_REG 0x82 +#define LM49453_P0_ADC_ALC2_REG 0x83 +#define LM49453_P0_ADC_ALC3_REG 0x84 +#define LM49453_P0_ADC_ALC4_REG 0x85 +#define LM49453_P0_ADC_ALC5_REG 0x86 +#define LM49453_P0_ADC_ALC6_REG 0x87 +#define LM49453_P0_ADC_ALC7_REG 0x88 +#define LM49453_P0_ADC_ALC8_REG 0x89 +#define LM49453_P0_DMIC1_LEVELL_REG 0x8A +#define LM49453_P0_DMIC1_LEVELR_REG 0x8B +#define LM49453_P0_DMIC2_LEVELL_REG 0x8C +#define LM49453_P0_DMIC2_LEVELR_REG 0x8D +#define LM49453_P0_ADC_LEVELL_REG 0x8E +#define LM49453_P0_ADC_LEVELR_REG 0x8F +#define LM49453_P0_DAC_HP_LEVELL_REG 0x90 +#define LM49453_P0_DAC_HP_LEVELR_REG 0x91 +#define LM49453_P0_DAC_LO_LEVELL_REG 0x92 +#define LM49453_P0_DAC_LO_LEVELR_REG 0x93 +#define LM49453_P0_DAC_LS_LEVELL_REG 0x94 +#define LM49453_P0_DAC_LS_LEVELR_REG 0x95 +#define LM49453_P0_DAC_HA_LEVELL_REG 0x96 +#define LM49453_P0_DAC_HA_LEVELR_REG 0x97 +#define LM49453_P0_SOFT_MUTE_REG 0x98 +#define LM49453_P0_DMIC_MUTE_CFG_REG 0x99 +#define LM49453_P0_ADC_MUTE_CFG_REG 0x9A +#define LM49453_P0_DAC_MUTE_CFG_REG 0x9B + +/*DIGITAL MIC1 */ +#define LM49453_P0_DIGITAL_MIC1_CONFIG_REG 0xB0 +#define LM49453_P0_DIGITAL_MIC1_DATA_DELAYL_REG 0xB1 +#define LM49453_P0_DIGITAL_MIC1_DATA_DELAYR_REG 0xB2 + +/*DIGITAL MIC2 */ +#define LM49453_P0_DIGITAL_MIC2_CONFIG_REG 0xB3 +#define LM49453_P0_DIGITAL_MIC2_DATA_DELAYL_REG 0xB4 +#define LM49453_P0_DIGITAL_MIC2_DATA_DELAYR_REG 0xB5 + +/* ADC DECIMATOR */ +#define LM49453_P0_ADC_DECIMATOR_REG 0xB6 + +/* DAC CONFIGURE */ +#define LM49453_P0_DAC_CONFIG_REG 0xB7 + +/* SIDETONE */ +#define LM49453_P0_STN_VOL_ADCL_REG 0xB8 +#define LM49453_P0_STN_VOL_ADCR_REG 0xB9 +#define LM49453_P0_STN_VOL_DMIC1L_REG 0xBA +#define LM49453_P0_STN_VOL_DMIC1R_REG 0xBB +#define LM49453_P0_STN_VOL_DMIC2L_REG 0xBC +#define LM49453_P0_STN_VOL_DMIC2R_REG 0xBD + +/* ADC/DAC CLIPPING MONITORS (Read Only/Write to Clear) */ +#define LM49453_P0_ADC_DEC_CLIP_REG 0xC2 +#define LM49453_P0_ADC_HPF_CLIP_REG 0xC3 +#define LM49453_P0_ADC_LVL_CLIP_REG 0xC4 +#define LM49453_P0_DAC_LVL_CLIP_REG 0xC5 + +/* ADC ALC EFFECT MONITORS (Read Only) */ +#define LM49453_P0_ADC_LVLMONL_REG 0xC8 +#define LM49453_P0_ADC_LVLMONR_REG 0xC9 +#define LM49453_P0_ADC_ALCMONL_REG 0xCA +#define LM49453_P0_ADC_ALCMONR_REG 0xCB +#define LM49453_P0_ADC_MUTED_REG 0xCC +#define LM49453_P0_DAC_MUTED_REG 0xCD + +/* HEADSET DETECT */ +#define LM49453_P0_HSD_PPB_LONG_CNT_LIMITL_REG 0xD0 +#define LM49453_P0_HSD_PPB_LONG_CNT_LIMITR_REG 0xD1 +#define LM49453_P0_HSD_PIN3_4_EX_LOOP_CNT_LIMITL_REG 0xD2 +#define LM49453_P0_HSD_PIN3_4_EX_LOOP_CNT_LIMITH_REG 0xD3 +#define LM49453_P0_HSD_TIMEOUT1_REG 0xD4 +#define LM49453_P0_HSD_TIMEOUT2_REG 0xD5 +#define LM49453_P0_HSD_TIMEOUT3_REG 0xD6 +#define LM49453_P0_HSD_PIN3_4_CFG_REG 0xD7 +#define LM49453_P0_HSD_IRQ1_REG 0xD8 +#define LM49453_P0_HSD_IRQ2_REG 0xD9 +#define LM49453_P0_HSD_IRQ3_REG 0xDA +#define LM49453_P0_HSD_IRQ4_REG 0xDB +#define LM49453_P0_HSD_IRQ_MASK1_REG 0xDC +#define LM49453_P0_HSD_IRQ_MASK2_REG 0xDD +#define LM49453_P0_HSD_IRQ_MASK3_REG 0xDE +#define LM49453_P0_HSD_R_HPLL_REG 0xE0 +#define LM49453_P0_HSD_R_HPLH_REG 0xE1 +#define LM49453_P0_HSD_R_HPLU_REG 0xE2 +#define LM49453_P0_HSD_R_HPRL_REG 0xE3 +#define LM49453_P0_HSD_R_HPRH_REG 0xE4 +#define LM49453_P0_HSD_R_HPRU_REG 0xE5 +#define LM49453_P0_HSD_VEL_L_FINALL_REG 0xE6 +#define LM49453_P0_HSD_VEL_L_FINALH_REG 0xE7 +#define LM49453_P0_HSD_VEL_L_FINALU_REG 0xE8 +#define LM49453_P0_HSD_RO_FINALL_REG 0xE9 +#define LM49453_P0_HSD_RO_FINALH_REG 0xEA +#define LM49453_P0_HSD_RO_FINALU_REG 0xEB +#define LM49453_P0_HSD_VMIC_BIAS_FINALL_REG 0xEC +#define LM49453_P0_HSD_VMIC_BIAS_FINALH_REG 0xED +#define LM49453_P0_HSD_VMIC_BIAS_FINALU_REG 0xEE +#define LM49453_P0_HSD_PIN_CONFIG_REG 0xEF +#define LM49453_P0_HSD_PLUG_DETECT_BB_IRQ_STATUS1_REG 0xF1 +#define LM49453_P0_HSD_PLUG_DETECT_BB_IRQ_STATUS2_REG 0xF2 +#define LM49453_P0_HSD_PLUG_DETECT_BB_IRQ_STATUS3_REG 0xF3 +#define LM49453_P0_HSD_PLUG_DETECT_BB_IRQ_STATEL_REG 0xF4 +#define LM49453_P0_HSD_PLUG_DETECT_BB_IRQ_STATEH_REG 0xF5 + +/* I/O PULLDOWN CONFIG */ +#define LM49453_P0_PULL_CONFIG1_REG 0xF8 +#define LM49453_P0_PULL_CONFIG2_REG 0xF9 +#define LM49453_P0_PULL_CONFIG3_REG 0xFA + +/* RESET */ +#define LM49453_P0_RESET_REG 0xFE + +/* PAGE */ +#define LM49453_PAGE_REG 0xFF + +#define LM49453_MAX_REGISTER (0xFF+1) + +/* LM49453_P0_PMC_SETUP_REG (0x00h) */ +#define LM49453_PMC_SETUP_CHIP_EN (BIT(1)|BIT(0)) +#define LM49453_PMC_SETUP_PLL_EN BIT(2) +#define LM49453_PMC_SETUP_PLL_P2_EN BIT(3) +#define LM49453_PMC_SETUP_PLL_FLL BIT(4) +#define LM49453_PMC_SETUP_MCLK_OVER BIT(5) +#define LM49453_PMC_SETUP_RTC_CLK_OVER BIT(6) +#define LM49453_PMC_SETUP_CHIP_ACTIVE BIT(7) + +/* Chip Enable bits */ +#define LM49453_CHIP_EN_SHUTDOWN 0x00 +#define LM49453_CHIP_EN 0x01 +#define LM49453_CHIP_EN_HSD_DETECT 0x02 +#define LM49453_CHIP_EN_INVALID_HSD 0x03 + +/* LM49453_P0_PLL_CLK_SEL1_REG (0x01h) */ +#define LM49453_CLK_SEL1_MCLK_SEL 0x11 +#define LM49453_CLK_SEL1_RTC_SEL 0x11 +#define LM49453_CLK_SEL1_PORT1_SEL 0x10 +#define LM49453_CLK_SEL1_PORT2_SEL 0x11 + +/* LM49453_P0_PLL_CLK_SEL2_REG (0x02h) */ +#define LM49453_CLK_SEL2_ADC_CLK_SEL 0x38 + +/* LM49453_P0_FLL_REF_FREQL_REG (0x0F) */ +#define LM49453_FLL_REF_FREQ_VAL 0x8ca0001 + +/* LM49453_P0_VCO_TARGETLL_REG (0x11) */ +#define LM49453_VCO_TARGET_VAL 0x8ca0001 + +/* LM49453_P0_ADC_DSP_REG (0x30h) */ +#define LM49453_ADC_DSP_ADC_MUTEL BIT(0) +#define LM49453_ADC_DSP_ADC_MUTER BIT(1) +#define LM49453_ADC_DSP_DMIC1_MUTEL BIT(2) +#define LM49453_ADC_DSP_DMIC1_MUTER BIT(3) +#define LM49453_ADC_DSP_DMIC2_MUTEL BIT(4) +#define LM49453_ADC_DSP_DMIC2_MUTER BIT(5) +#define LM49453_ADC_DSP_MUTE_ALL 0x3F + +/* LM49453_P0_DAC_DSP_REG (0x31h) */ +#define LM49453_DAC_DSP_MUTE_ALL 0xFF + +/* LM49453_P0_AUDIO_PORT1_BASIC_REG (0x60h) */ +#define LM49453_AUDIO_PORT1_BASIC_FMT_MASK (BIT(4)|BIT(3)) +#define LM49453_AUDIO_PORT1_BASIC_CLK_MS BIT(3) +#define LM49453_AUDIO_PORT1_BASIC_SYNC_MS BIT(4) + +/* LM49453_P0_RESET_REG (0xFEh) */ +#define LM49453_RESET_REG_RST BIT(0) + +/* Page select register bits (0xFF) */ +#define LM49453_PAGE0_SELECT 0x0 +#define LM49453_PAGE1_SELECT 0x1 + +/* LM49453_P0_HSD_PIN3_4_CFG_REG (Jack Pin config - 0xD7) */ +#define LM49453_JACK_DISABLE 0x00 +#define LM49453_JACK_CONFIG1 0x01 +#define LM49453_JACK_CONFIG2 0x02 +#define LM49453_JACK_CONFIG3 0x03 +#define LM49453_JACK_CONFIG4 0x04 +#define LM49453_JACK_CONFIG5 0x05 + +/* Page 1 REGISTERS */ + +/* SIDETONE */ +#define LM49453_P1_SIDETONE_SA0L_REG 0x80 +#define LM49453_P1_SIDETONE_SA0H_REG 0x81 +#define LM49453_P1_SIDETONE_SAB0U_REG 0x82 +#define LM49453_P1_SIDETONE_SB0L_REG 0x83 +#define LM49453_P1_SIDETONE_SB0H_REG 0x84 +#define LM49453_P1_SIDETONE_SH0L_REG 0x85 +#define LM49453_P1_SIDETONE_SH0H_REG 0x86 +#define LM49453_P1_SIDETONE_SH0U_REG 0x87 +#define LM49453_P1_SIDETONE_SA1L_REG 0x88 +#define LM49453_P1_SIDETONE_SA1H_REG 0x89 +#define LM49453_P1_SIDETONE_SAB1U_REG 0x8A +#define LM49453_P1_SIDETONE_SB1L_REG 0x8B +#define LM49453_P1_SIDETONE_SB1H_REG 0x8C +#define LM49453_P1_SIDETONE_SH1L_REG 0x8D +#define LM49453_P1_SIDETONE_SH1H_REG 0x8E +#define LM49453_P1_SIDETONE_SH1U_REG 0x8F +#define LM49453_P1_SIDETONE_SA2L_REG 0x90 +#define LM49453_P1_SIDETONE_SA2H_REG 0x91 +#define LM49453_P1_SIDETONE_SAB2U_REG 0x92 +#define LM49453_P1_SIDETONE_SB2L_REG 0x93 +#define LM49453_P1_SIDETONE_SB2H_REG 0x94 +#define LM49453_P1_SIDETONE_SH2L_REG 0x95 +#define LM49453_P1_SIDETONE_SH2H_REG 0x96 +#define LM49453_P1_SIDETONE_SH2U_REG 0x97 +#define LM49453_P1_SIDETONE_SA3L_REG 0x98 +#define LM49453_P1_SIDETONE_SA3H_REG 0x99 +#define LM49453_P1_SIDETONE_SAB3U_REG 0x9A +#define LM49453_P1_SIDETONE_SB3L_REG 0x9B +#define LM49453_P1_SIDETONE_SB3H_REG 0x9C +#define LM49453_P1_SIDETONE_SH3L_REG 0x9D +#define LM49453_P1_SIDETONE_SH3H_REG 0x9E +#define LM49453_P1_SIDETONE_SH3U_REG 0x9F +#define LM49453_P1_SIDETONE_SA4L_REG 0xA0 +#define LM49453_P1_SIDETONE_SA4H_REG 0xA1 +#define LM49453_P1_SIDETONE_SAB4U_REG 0xA2 +#define LM49453_P1_SIDETONE_SB4L_REG 0xA3 +#define LM49453_P1_SIDETONE_SB4H_REG 0xA4 +#define LM49453_P1_SIDETONE_SH4L_REG 0xA5 +#define LM49453_P1_SIDETONE_SH4H_REG 0xA6 +#define LM49453_P1_SIDETONE_SH4U_REG 0xA7 +#define LM49453_P1_SIDETONE_SA5L_REG 0xA8 +#define LM49453_P1_SIDETONE_SA5H_REG 0xA9 +#define LM49453_P1_SIDETONE_SAB5U_REG 0xAA +#define LM49453_P1_SIDETONE_SB5L_REG 0xAB +#define LM49453_P1_SIDETONE_SB5H_REG 0xAC +#define LM49453_P1_SIDETONE_SH5L_REG 0xAD +#define LM49453_P1_SIDETONE_SH5H_REG 0xAE +#define LM49453_P1_SIDETONE_SH5U_REG 0xAF + +/* CHARGE PUMP CONFIG */ +#define LM49453_P1_CP_CONFIG1_REG 0xB0 +#define LM49453_P1_CP_CONFIG2_REG 0xB1 +#define LM49453_P1_CP_CONFIG3_REG 0xB2 +#define LM49453_P1_CP_CONFIG4_REG 0xB3 +#define LM49453_P1_CP_LA_VTH1L_REG 0xB4 +#define LM49453_P1_CP_LA_VTH1M_REG 0xB5 +#define LM49453_P1_CP_LA_VTH2L_REG 0xB6 +#define LM49453_P1_CP_LA_VTH2M_REG 0xB7 +#define LM49453_P1_CP_LA_VTH3L_REG 0xB8 +#define LM49453_P1_CP_LA_VTH3H_REG 0xB9 +#define LM49453_P1_CP_CLK_DIV_REG 0xBA + +/* DAC */ +#define LM49453_P1_DAC_CHOP_REG 0xC0 + +#define LM49453_CLK_SRC_MCLK 1 +#endif diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index 0bb511a0388d..35179e2c23c9 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -24,6 +24,7 @@ #include <linux/slab.h> #include <asm/div64.h> #include <sound/max98095.h> +#include <sound/jack.h> #include "max98095.h" enum max98095_type { @@ -51,6 +52,8 @@ struct max98095_priv { u8 lin_state; unsigned int mic1pre; unsigned int mic2pre; + struct snd_soc_jack *headphone_jack; + struct snd_soc_jack *mic_jack; }; static const u8 max98095_reg_def[M98095_REG_CNT] = { @@ -2173,9 +2176,125 @@ static void max98095_handle_pdata(struct snd_soc_codec *codec) max98095_handle_bq_pdata(codec); } +static irqreturn_t max98095_report_jack(int irq, void *data) +{ + struct snd_soc_codec *codec = data; + struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); + unsigned int value; + int hp_report = 0; + int mic_report = 0; + + /* Read the Jack Status Register */ + value = snd_soc_read(codec, M98095_007_JACK_AUTO_STS); + + /* If ddone is not set, then detection isn't finished yet */ + if ((value & M98095_DDONE) == 0) + return IRQ_NONE; + + /* if hp, check its bit, and if set, clear it */ + if ((value & M98095_HP_IN || value & M98095_LO_IN) && + max98095->headphone_jack) + hp_report |= SND_JACK_HEADPHONE; + + /* if mic, check its bit, and if set, clear it */ + if ((value & M98095_MIC_IN) && max98095->mic_jack) + mic_report |= SND_JACK_MICROPHONE; + + if (max98095->headphone_jack == max98095->mic_jack) { + snd_soc_jack_report(max98095->headphone_jack, + hp_report | mic_report, + SND_JACK_HEADSET); + } else { + if (max98095->headphone_jack) + snd_soc_jack_report(max98095->headphone_jack, + hp_report, SND_JACK_HEADPHONE); + if (max98095->mic_jack) + snd_soc_jack_report(max98095->mic_jack, + mic_report, SND_JACK_MICROPHONE); + } + + return IRQ_HANDLED; +} + +int max98095_jack_detect_enable(struct snd_soc_codec *codec) +{ + struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); + int ret = 0; + int detect_enable = M98095_JDEN; + unsigned int slew = M98095_DEFAULT_SLEW_DELAY; + + if (max98095->pdata->jack_detect_pin5en) + detect_enable |= M98095_PIN5EN; + + if (max98095->pdata->jack_detect_delay) + slew = max98095->pdata->jack_detect_delay; + + ret = snd_soc_write(codec, M98095_08E_JACK_DC_SLEW, slew); + if (ret < 0) { + dev_err(codec->dev, "Failed to cfg auto detect %d\n", ret); + return ret; + } + + /* configure auto detection to be enabled */ + ret = snd_soc_write(codec, M98095_089_JACK_DET_AUTO, detect_enable); + if (ret < 0) { + dev_err(codec->dev, "Failed to cfg auto detect %d\n", ret); + return ret; + } + + return ret; +} + +int max98095_jack_detect_disable(struct snd_soc_codec *codec) +{ + int ret = 0; + + /* configure auto detection to be disabled */ + ret = snd_soc_write(codec, M98095_089_JACK_DET_AUTO, 0x0); + if (ret < 0) { + dev_err(codec->dev, "Failed to cfg auto detect %d\n", ret); + return ret; + } + + return ret; +} + +int max98095_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *hp_jack, struct snd_soc_jack *mic_jack) +{ + struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); + struct i2c_client *client = to_i2c_client(codec->dev); + int ret = 0; + + max98095->headphone_jack = hp_jack; + max98095->mic_jack = mic_jack; + + /* only progress if we have at least 1 jack pointer */ + if (!hp_jack && !mic_jack) + return -EINVAL; + + max98095_jack_detect_enable(codec); + + /* enable interrupts for headphone jack detection */ + ret = snd_soc_update_bits(codec, M98095_013_JACK_INT_EN, + M98095_IDDONE, M98095_IDDONE); + if (ret < 0) { + dev_err(codec->dev, "Failed to cfg jack irqs %d\n", ret); + return ret; + } + + max98095_report_jack(client->irq, codec); + return 0; +} + #ifdef CONFIG_PM static int max98095_suspend(struct snd_soc_codec *codec) { + struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); + + if (max98095->headphone_jack || max98095->mic_jack) + max98095_jack_detect_disable(codec); + max98095_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; @@ -2183,8 +2302,16 @@ static int max98095_suspend(struct snd_soc_codec *codec) static int max98095_resume(struct snd_soc_codec *codec) { + struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); + struct i2c_client *client = to_i2c_client(codec->dev); + max98095_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + if (max98095->headphone_jack || max98095->mic_jack) { + max98095_jack_detect_enable(codec); + max98095_report_jack(client->irq, codec); + } + return 0; } #else @@ -2227,6 +2354,7 @@ static int max98095_probe(struct snd_soc_codec *codec) { struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); struct max98095_cdata *cdata; + struct i2c_client *client; int ret = 0; ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); @@ -2238,6 +2366,8 @@ static int max98095_probe(struct snd_soc_codec *codec) /* reset the codec, the DSP core, and disable all interrupts */ max98095_reset(codec); + client = to_i2c_client(codec->dev); + /* initialize private data */ max98095->sysclk = (unsigned)-1; @@ -2266,11 +2396,23 @@ static int max98095_probe(struct snd_soc_codec *codec) max98095->mic1pre = 0; max98095->mic2pre = 0; + if (client->irq) { + /* register an audio interrupt */ + ret = request_threaded_irq(client->irq, NULL, + max98095_report_jack, + IRQF_TRIGGER_FALLING | IRQF_TRIGGER_RISING, + "max98095", codec); + if (ret) { + dev_err(codec->dev, "Failed to request IRQ: %d\n", ret); + goto err_access; + } + } + ret = snd_soc_read(codec, M98095_0FF_REV_ID); if (ret < 0) { dev_err(codec->dev, "Failure reading hardware revision: %d\n", ret); - goto err_access; + goto err_irq; } dev_info(codec->dev, "Hardware revision: %c\n", ret - 0x40 + 'A'); @@ -2306,14 +2448,28 @@ static int max98095_probe(struct snd_soc_codec *codec) max98095_add_widgets(codec); + return 0; + +err_irq: + if (client->irq) + free_irq(client->irq, codec); err_access: return ret; } static int max98095_remove(struct snd_soc_codec *codec) { + struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); + struct i2c_client *client = to_i2c_client(codec->dev); + max98095_set_bias_level(codec, SND_SOC_BIAS_OFF); + if (max98095->headphone_jack || max98095->mic_jack) + max98095_jack_detect_disable(codec); + + if (client->irq) + free_irq(client->irq, codec); + return 0; } diff --git a/sound/soc/codecs/max98095.h b/sound/soc/codecs/max98095.h index 891584a0eb03..2ebbe4e894bf 100644 --- a/sound/soc/codecs/max98095.h +++ b/sound/soc/codecs/max98095.h @@ -175,11 +175,23 @@ /* MAX98095 Registers Bit Fields */ +/* M98095_007_JACK_AUTO_STS */ + #define M98095_MIC_IN (1<<3) + #define M98095_LO_IN (1<<5) + #define M98095_HP_IN (1<<6) + #define M98095_DDONE (1<<7) + /* M98095_00F_HOST_CFG */ #define M98095_SEG (1<<0) #define M98095_XTEN (1<<1) #define M98095_MDLLEN (1<<2) +/* M98095_013_JACK_INT_EN */ + #define M98095_IMIC_IN (1<<3) + #define M98095_ILO_IN (1<<5) + #define M98095_IHP_IN (1<<6) + #define M98095_IDDONE (1<<7) + /* M98095_027_DAI1_CLKMODE, M98095_031_DAI2_CLKMODE, M98095_03B_DAI3_CLKMODE */ #define M98095_CLKMODE_MASK 0xFF @@ -255,6 +267,10 @@ #define M98095_EQ2EN (1<<1) #define M98095_EQ1EN (1<<0) +/* M98095_089_JACK_DET_AUTO */ + #define M98095_PIN5EN (1<<2) + #define M98095_JDEN (1<<7) + /* M98095_090_PWR_EN_IN */ #define M98095_INEN (1<<7) #define M98095_MB2EN (1<<3) @@ -296,4 +312,10 @@ #define M98095_174_DAI1_BQ_BASE 0x74 #define M98095_17E_DAI2_BQ_BASE 0x7E +/* Default Delay used in Slew Rate Calculation for Jack detection */ +#define M98095_DEFAULT_SLEW_DELAY 0x18 + +extern int max98095_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *hp_jack, struct snd_soc_jack *mic_jack); + #endif diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c new file mode 100644 index 000000000000..6276e352125f --- /dev/null +++ b/sound/soc/codecs/mc13783.c @@ -0,0 +1,786 @@ +/* + * Copyright 2008 Juergen Beisert, kernel@pengutronix.de + * Copyright 2009 Sascha Hauer, s.hauer@pengutronix.de + * Copyright 2012 Philippe Retornaz, philippe.retornaz@epfl.ch + * + * Initial development of this code was funded by + * Phytec Messtechnik GmbH, http://www.phytec.de + * + * 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. + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, + * MA 02110-1301, USA. + */ +#include <linux/module.h> +#include <linux/device.h> +#include <linux/mfd/mc13xxx.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/control.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/initval.h> +#include <sound/soc-dapm.h> + +#include "mc13783.h" + +#define MC13783_AUDIO_RX0 36 +#define MC13783_AUDIO_RX1 37 +#define MC13783_AUDIO_TX 38 +#define MC13783_SSI_NETWORK 39 +#define MC13783_AUDIO_CODEC 40 +#define MC13783_AUDIO_DAC 41 + +#define AUDIO_RX0_ALSPEN (1 << 5) +#define AUDIO_RX0_ALSPSEL (1 << 7) +#define AUDIO_RX0_ADDCDC (1 << 21) +#define AUDIO_RX0_ADDSTDC (1 << 22) +#define AUDIO_RX0_ADDRXIN (1 << 23) + +#define AUDIO_RX1_PGARXEN (1 << 0); +#define AUDIO_RX1_PGASTEN (1 << 5) +#define AUDIO_RX1_ARXINEN (1 << 10) + +#define AUDIO_TX_AMC1REN (1 << 5) +#define AUDIO_TX_AMC1LEN (1 << 7) +#define AUDIO_TX_AMC2EN (1 << 9) +#define AUDIO_TX_ATXINEN (1 << 11) +#define AUDIO_TX_RXINREC (1 << 13) + +#define SSI_NETWORK_CDCTXRXSLOT(x) (((x) & 0x3) << 2) +#define SSI_NETWORK_CDCTXSECSLOT(x) (((x) & 0x3) << 4) +#define SSI_NETWORK_CDCRXSECSLOT(x) (((x) & 0x3) << 6) +#define SSI_NETWORK_CDCRXSECGAIN(x) (((x) & 0x3) << 8) +#define SSI_NETWORK_CDCSUMGAIN(x) (1 << 10) +#define SSI_NETWORK_CDCFSDLY(x) (1 << 11) +#define SSI_NETWORK_DAC_SLOTS_8 (1 << 12) +#define SSI_NETWORK_DAC_SLOTS_4 (2 << 12) +#define SSI_NETWORK_DAC_SLOTS_2 (3 << 12) +#define SSI_NETWORK_DAC_SLOT_MASK (3 << 12) +#define SSI_NETWORK_DAC_RXSLOT_0_1 (0 << 14) +#define SSI_NETWORK_DAC_RXSLOT_2_3 (1 << 14) +#define SSI_NETWORK_DAC_RXSLOT_4_5 (2 << 14) +#define SSI_NETWORK_DAC_RXSLOT_6_7 (3 << 14) +#define SSI_NETWORK_DAC_RXSLOT_MASK (3 << 14) +#define SSI_NETWORK_STDCRXSECSLOT(x) (((x) & 0x3) << 16) +#define SSI_NETWORK_STDCRXSECGAIN(x) (((x) & 0x3) << 18) +#define SSI_NETWORK_STDCSUMGAIN (1 << 20) + +/* + * MC13783_AUDIO_CODEC and MC13783_AUDIO_DAC mostly share the same + * register layout + */ +#define AUDIO_SSI_SEL (1 << 0) +#define AUDIO_CLK_SEL (1 << 1) +#define AUDIO_CSM (1 << 2) +#define AUDIO_BCL_INV (1 << 3) +#define AUDIO_CFS_INV (1 << 4) +#define AUDIO_CFS(x) (((x) & 0x3) << 5) +#define AUDIO_CLK(x) (((x) & 0x7) << 7) +#define AUDIO_C_EN (1 << 11) +#define AUDIO_C_CLK_EN (1 << 12) +#define AUDIO_C_RESET (1 << 15) + +#define AUDIO_CODEC_CDCFS8K16K (1 << 10) +#define AUDIO_DAC_CFS_DLY_B (1 << 10) + +struct mc13783_priv { + struct snd_soc_codec codec; + struct mc13xxx *mc13xxx; + + enum mc13783_ssi_port adc_ssi_port; + enum mc13783_ssi_port dac_ssi_port; +}; + +static unsigned int mc13783_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec); + unsigned int value = 0; + + mc13xxx_lock(priv->mc13xxx); + + mc13xxx_reg_read(priv->mc13xxx, reg, &value); + + mc13xxx_unlock(priv->mc13xxx); + + return value; +} + +static int mc13783_write(struct snd_soc_codec *codec, + unsigned int reg, unsigned int value) +{ + struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec); + int ret; + + mc13xxx_lock(priv->mc13xxx); + + ret = mc13xxx_reg_write(priv->mc13xxx, reg, value); + + mc13xxx_unlock(priv->mc13xxx); + + return ret; +} + +/* Mapping between sample rates and register value */ +static unsigned int mc13783_rates[] = { + 8000, 11025, 12000, 16000, + 22050, 24000, 32000, 44100, + 48000, 64000, 96000 +}; + +static int mc13783_pcm_hw_params_dac(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + unsigned int rate = params_rate(params); + int i; + + for (i = 0; i < ARRAY_SIZE(mc13783_rates); i++) { + if (rate == mc13783_rates[i]) { + snd_soc_update_bits(codec, MC13783_AUDIO_DAC, + 0xf << 17, i << 17); + return 0; + } + } + + return -EINVAL; +} + +static int mc13783_pcm_hw_params_codec(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + unsigned int rate = params_rate(params); + unsigned int val; + + switch (rate) { + case 8000: + val = 0; + break; + case 16000: + val = AUDIO_CODEC_CDCFS8K16K; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, MC13783_AUDIO_CODEC, AUDIO_CODEC_CDCFS8K16K, + val); + + return 0; +} + +static int mc13783_pcm_hw_params_sync(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + return mc13783_pcm_hw_params_dac(substream, params, dai); + else + return mc13783_pcm_hw_params_codec(substream, params, dai); +} + +static int mc13783_set_fmt(struct snd_soc_dai *dai, unsigned int fmt, + unsigned int reg) +{ + struct snd_soc_codec *codec = dai->codec; + unsigned int val = 0; + unsigned int mask = AUDIO_CFS(3) | AUDIO_BCL_INV | AUDIO_CFS_INV | + AUDIO_CSM | AUDIO_C_CLK_EN | AUDIO_C_RESET; + + + /* DAI mode */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + val |= AUDIO_CFS(2); + break; + case SND_SOC_DAIFMT_DSP_A: + val |= AUDIO_CFS(1); + break; + default: + return -EINVAL; + } + + /* DAI clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + val |= AUDIO_BCL_INV; + break; + case SND_SOC_DAIFMT_NB_IF: + val |= AUDIO_BCL_INV | AUDIO_CFS_INV; + break; + case SND_SOC_DAIFMT_IB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + val |= AUDIO_CFS_INV; + break; + } + + /* DAI clock master masks */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + val |= AUDIO_C_CLK_EN; + break; + case SND_SOC_DAIFMT_CBS_CFS: + val |= AUDIO_CSM; + break; + case SND_SOC_DAIFMT_CBM_CFS: + case SND_SOC_DAIFMT_CBS_CFM: + return -EINVAL; + } + + val |= AUDIO_C_RESET; + + snd_soc_update_bits(codec, reg, mask, val); + + return 0; +} + +static int mc13783_set_fmt_async(struct snd_soc_dai *dai, unsigned int fmt) +{ + if (dai->id == MC13783_ID_STEREO_DAC) + return mc13783_set_fmt(dai, fmt, MC13783_AUDIO_DAC); + else + return mc13783_set_fmt(dai, fmt, MC13783_AUDIO_CODEC); +} + +static int mc13783_set_fmt_sync(struct snd_soc_dai *dai, unsigned int fmt) +{ + int ret; + + ret = mc13783_set_fmt(dai, fmt, MC13783_AUDIO_DAC); + if (ret) + return ret; + + /* + * In synchronous mode force the voice codec into slave mode + * so that the clock / framesync from the stereo DAC is used + */ + fmt &= ~SND_SOC_DAIFMT_MASTER_MASK; + fmt |= SND_SOC_DAIFMT_CBS_CFS; + ret = mc13783_set_fmt(dai, fmt, MC13783_AUDIO_CODEC); + + return ret; +} + +static int mc13783_sysclk[] = { + 13000000, + 15360000, + 16800000, + -1, + 26000000, + -1, /* 12000000, invalid for voice codec */ + -1, /* 3686400, invalid for voice codec */ + 33600000, +}; + +static int mc13783_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir, + unsigned int reg) +{ + struct snd_soc_codec *codec = dai->codec; + int clk; + unsigned int val = 0; + unsigned int mask = AUDIO_CLK(0x7) | AUDIO_CLK_SEL; + + for (clk = 0; clk < ARRAY_SIZE(mc13783_sysclk); clk++) { + if (mc13783_sysclk[clk] < 0) + continue; + if (mc13783_sysclk[clk] == freq) + break; + } + + if (clk == ARRAY_SIZE(mc13783_sysclk)) + return -EINVAL; + + if (clk_id == MC13783_CLK_CLIB) + val |= AUDIO_CLK_SEL; + + val |= AUDIO_CLK(clk); + + snd_soc_update_bits(codec, reg, mask, val); + + return 0; +} + +static int mc13783_set_sysclk_dac(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + return mc13783_set_sysclk(dai, clk_id, freq, dir, MC13783_AUDIO_DAC); +} + +static int mc13783_set_sysclk_codec(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + return mc13783_set_sysclk(dai, clk_id, freq, dir, MC13783_AUDIO_CODEC); +} + +static int mc13783_set_sysclk_sync(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + int ret; + + ret = mc13783_set_sysclk(dai, clk_id, freq, dir, MC13783_AUDIO_DAC); + if (ret) + return ret; + + return mc13783_set_sysclk(dai, clk_id, freq, dir, MC13783_AUDIO_CODEC); +} + +static int mc13783_set_tdm_slot_dac(struct snd_soc_dai *dai, + unsigned int tx_mask, unsigned int rx_mask, int slots, + int slot_width) +{ + struct snd_soc_codec *codec = dai->codec; + unsigned int val = 0; + unsigned int mask = SSI_NETWORK_DAC_SLOT_MASK | + SSI_NETWORK_DAC_RXSLOT_MASK; + + switch (slots) { + case 2: + val |= SSI_NETWORK_DAC_SLOTS_2; + break; + case 4: + val |= SSI_NETWORK_DAC_SLOTS_4; + break; + case 8: + val |= SSI_NETWORK_DAC_SLOTS_8; + break; + default: + return -EINVAL; + } + + switch (rx_mask) { + case 0xfffffffc: + val |= SSI_NETWORK_DAC_RXSLOT_0_1; + break; + case 0xfffffff3: + val |= SSI_NETWORK_DAC_RXSLOT_2_3; + break; + case 0xffffffcf: + val |= SSI_NETWORK_DAC_RXSLOT_4_5; + break; + case 0xffffff3f: + val |= SSI_NETWORK_DAC_RXSLOT_6_7; + break; + default: + return -EINVAL; + }; + + snd_soc_update_bits(codec, MC13783_SSI_NETWORK, mask, val); + + return 0; +} + +static int mc13783_set_tdm_slot_codec(struct snd_soc_dai *dai, + unsigned int tx_mask, unsigned int rx_mask, int slots, + int slot_width) +{ + struct snd_soc_codec *codec = dai->codec; + unsigned int val = 0; + unsigned int mask = 0x3f; + + if (slots != 4) + return -EINVAL; + + if (tx_mask != 0xfffffffc) + return -EINVAL; + + val |= (0x00 << 2); /* primary timeslot RX/TX(?) is 0 */ + val |= (0x01 << 4); /* secondary timeslot TX is 1 */ + + snd_soc_update_bits(codec, MC13783_SSI_NETWORK, mask, val); + + return 0; +} + +static int mc13783_set_tdm_slot_sync(struct snd_soc_dai *dai, + unsigned int tx_mask, unsigned int rx_mask, int slots, + int slot_width) +{ + int ret; + + ret = mc13783_set_tdm_slot_dac(dai, tx_mask, rx_mask, slots, + slot_width); + if (ret) + return ret; + + ret = mc13783_set_tdm_slot_codec(dai, tx_mask, rx_mask, slots, + slot_width); + + return ret; +} + +static const struct snd_kcontrol_new mc1l_amp_ctl = + SOC_DAPM_SINGLE("Switch", 38, 7, 1, 0); + +static const struct snd_kcontrol_new mc1r_amp_ctl = + SOC_DAPM_SINGLE("Switch", 38, 5, 1, 0); + +static const struct snd_kcontrol_new mc2_amp_ctl = + SOC_DAPM_SINGLE("Switch", 38, 9, 1, 0); + +static const struct snd_kcontrol_new atx_amp_ctl = + SOC_DAPM_SINGLE("Switch", 38, 11, 1, 0); + + +/* Virtual mux. The chip does the input selection automatically + * as soon as we enable one input. */ +static const char * const adcl_enum_text[] = { + "MC1L", "RXINL", +}; + +static const struct soc_enum adcl_enum = + SOC_ENUM_SINGLE(0, 0, ARRAY_SIZE(adcl_enum_text), adcl_enum_text); + +static const struct snd_kcontrol_new left_input_mux = + SOC_DAPM_ENUM_VIRT("Route", adcl_enum); + +static const char * const adcr_enum_text[] = { + "MC1R", "MC2", "RXINR", "TXIN", +}; + +static const struct soc_enum adcr_enum = + SOC_ENUM_SINGLE(0, 0, ARRAY_SIZE(adcr_enum_text), adcr_enum_text); + +static const struct snd_kcontrol_new right_input_mux = + SOC_DAPM_ENUM_VIRT("Route", adcr_enum); + +static const struct snd_kcontrol_new samp_ctl = + SOC_DAPM_SINGLE("Switch", 36, 3, 1, 0); + +static const struct snd_kcontrol_new lamp_ctl = + SOC_DAPM_SINGLE("Switch", 36, 5, 1, 0); + +static const struct snd_kcontrol_new hlamp_ctl = + SOC_DAPM_SINGLE("Switch", 36, 10, 1, 0); + +static const struct snd_kcontrol_new hramp_ctl = + SOC_DAPM_SINGLE("Switch", 36, 9, 1, 0); + +static const struct snd_kcontrol_new llamp_ctl = + SOC_DAPM_SINGLE("Switch", 36, 16, 1, 0); + +static const struct snd_kcontrol_new lramp_ctl = + SOC_DAPM_SINGLE("Switch", 36, 15, 1, 0); + +static const struct snd_soc_dapm_widget mc13783_dapm_widgets[] = { +/* Input */ + SND_SOC_DAPM_INPUT("MC1LIN"), + SND_SOC_DAPM_INPUT("MC1RIN"), + SND_SOC_DAPM_INPUT("MC2IN"), + SND_SOC_DAPM_INPUT("RXINR"), + SND_SOC_DAPM_INPUT("RXINL"), + SND_SOC_DAPM_INPUT("TXIN"), + + SND_SOC_DAPM_SUPPLY("MC1 Bias", 38, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MC2 Bias", 38, 1, 0, NULL, 0), + + SND_SOC_DAPM_SWITCH("MC1L Amp", 38, 7, 0, &mc1l_amp_ctl), + SND_SOC_DAPM_SWITCH("MC1R Amp", 38, 5, 0, &mc1r_amp_ctl), + SND_SOC_DAPM_SWITCH("MC2 Amp", 38, 9, 0, &mc2_amp_ctl), + SND_SOC_DAPM_SWITCH("TXIN Amp", 38, 11, 0, &atx_amp_ctl), + + SND_SOC_DAPM_VIRT_MUX("PGA Left Input Mux", SND_SOC_NOPM, 0, 0, + &left_input_mux), + SND_SOC_DAPM_VIRT_MUX("PGA Right Input Mux", SND_SOC_NOPM, 0, 0, + &right_input_mux), + + SND_SOC_DAPM_PGA("PGA Left Input", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PGA Right Input", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_ADC("ADC", "Capture", 40, 11, 0), + SND_SOC_DAPM_SUPPLY("ADC_Reset", 40, 15, 0, NULL, 0), + +/* Output */ + SND_SOC_DAPM_SUPPLY("DAC_E", 41, 11, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC_Reset", 41, 15, 0, NULL, 0), + SND_SOC_DAPM_OUTPUT("RXOUTL"), + SND_SOC_DAPM_OUTPUT("RXOUTR"), + SND_SOC_DAPM_OUTPUT("HSL"), + SND_SOC_DAPM_OUTPUT("HSR"), + SND_SOC_DAPM_OUTPUT("LSP"), + SND_SOC_DAPM_OUTPUT("SP"), + + SND_SOC_DAPM_SWITCH("Speaker Amp", 36, 3, 0, &samp_ctl), + SND_SOC_DAPM_SWITCH("Loudspeaker Amp", SND_SOC_NOPM, 0, 0, &lamp_ctl), + SND_SOC_DAPM_SWITCH("Headset Amp Left", 36, 10, 0, &hlamp_ctl), + SND_SOC_DAPM_SWITCH("Headset Amp Right", 36, 9, 0, &hramp_ctl), + SND_SOC_DAPM_SWITCH("Line out Amp Left", 36, 16, 0, &llamp_ctl), + SND_SOC_DAPM_SWITCH("Line out Amp Right", 36, 15, 0, &lramp_ctl), + SND_SOC_DAPM_DAC("DAC", "Playback", 36, 22, 0), + SND_SOC_DAPM_PGA("DAC PGA", 37, 5, 0, NULL, 0), +}; + +static struct snd_soc_dapm_route mc13783_routes[] = { +/* Input */ + { "MC1L Amp", NULL, "MC1LIN"}, + { "MC1R Amp", NULL, "MC1RIN" }, + { "MC2 Amp", NULL, "MC2IN" }, + { "TXIN Amp", NULL, "TXIN"}, + + { "PGA Left Input Mux", "MC1L", "MC1L Amp" }, + { "PGA Left Input Mux", "RXINL", "RXINL"}, + { "PGA Right Input Mux", "MC1R", "MC1R Amp" }, + { "PGA Right Input Mux", "MC2", "MC2 Amp"}, + { "PGA Right Input Mux", "TXIN", "TXIN Amp"}, + { "PGA Right Input Mux", "RXINR", "RXINR"}, + + { "PGA Left Input", NULL, "PGA Left Input Mux"}, + { "PGA Right Input", NULL, "PGA Right Input Mux"}, + + { "ADC", NULL, "PGA Left Input"}, + { "ADC", NULL, "PGA Right Input"}, + { "ADC", NULL, "ADC_Reset"}, + +/* Output */ + { "HSL", NULL, "Headset Amp Left" }, + { "HSR", NULL, "Headset Amp Right"}, + { "RXOUTL", NULL, "Line out Amp Left"}, + { "RXOUTR", NULL, "Line out Amp Right"}, + { "SP", NULL, "Speaker Amp"}, + { "Speaker Amp", NULL, "DAC PGA"}, + { "LSP", NULL, "DAC PGA"}, + { "Headset Amp Left", NULL, "DAC PGA"}, + { "Headset Amp Right", NULL, "DAC PGA"}, + { "Line out Amp Left", NULL, "DAC PGA"}, + { "Line out Amp Right", NULL, "DAC PGA"}, + { "DAC PGA", NULL, "DAC"}, + { "DAC", NULL, "DAC_E"}, +}; + +static const char * const mc13783_3d_mixer[] = {"Stereo", "Phase Mix", + "Mono", "Mono Mix"}; + +static const struct soc_enum mc13783_enum_3d_mixer = + SOC_ENUM_SINGLE(MC13783_AUDIO_RX1, 16, ARRAY_SIZE(mc13783_3d_mixer), + mc13783_3d_mixer); + +static struct snd_kcontrol_new mc13783_control_list[] = { + SOC_SINGLE("Loudspeaker enable", MC13783_AUDIO_RX0, 5, 1, 0), + SOC_SINGLE("PCM Playback Volume", MC13783_AUDIO_RX1, 6, 15, 0), + SOC_DOUBLE("PCM Capture Volume", MC13783_AUDIO_TX, 19, 14, 31, 0), + SOC_ENUM("3D Control", mc13783_enum_3d_mixer), +}; + +static int mc13783_probe(struct snd_soc_codec *codec) +{ + struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec); + + mc13xxx_lock(priv->mc13xxx); + + /* these are the reset values */ + mc13xxx_reg_write(priv->mc13xxx, MC13783_AUDIO_RX0, 0x25893); + mc13xxx_reg_write(priv->mc13xxx, MC13783_AUDIO_RX1, 0x00d35A); + mc13xxx_reg_write(priv->mc13xxx, MC13783_AUDIO_TX, 0x420000); + mc13xxx_reg_write(priv->mc13xxx, MC13783_SSI_NETWORK, 0x013060); + mc13xxx_reg_write(priv->mc13xxx, MC13783_AUDIO_CODEC, 0x180027); + mc13xxx_reg_write(priv->mc13xxx, MC13783_AUDIO_DAC, 0x0e0004); + + if (priv->adc_ssi_port == MC13783_SSI1_PORT) + mc13xxx_reg_rmw(priv->mc13xxx, MC13783_AUDIO_CODEC, + AUDIO_SSI_SEL, 0); + else + mc13xxx_reg_rmw(priv->mc13xxx, MC13783_AUDIO_CODEC, + 0, AUDIO_SSI_SEL); + + if (priv->dac_ssi_port == MC13783_SSI1_PORT) + mc13xxx_reg_rmw(priv->mc13xxx, MC13783_AUDIO_DAC, + AUDIO_SSI_SEL, 0); + else + mc13xxx_reg_rmw(priv->mc13xxx, MC13783_AUDIO_DAC, + 0, AUDIO_SSI_SEL); + + mc13xxx_unlock(priv->mc13xxx); + + return 0; +} + +static int mc13783_remove(struct snd_soc_codec *codec) +{ + struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec); + + mc13xxx_lock(priv->mc13xxx); + + /* Make sure VAUDIOON is off */ + mc13xxx_reg_rmw(priv->mc13xxx, MC13783_AUDIO_RX0, 0x3, 0); + + mc13xxx_unlock(priv->mc13xxx); + + return 0; +} + +#define MC13783_RATES_RECORD (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000) + +#define MC13783_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_ops mc13783_ops_dac = { + .hw_params = mc13783_pcm_hw_params_dac, + .set_fmt = mc13783_set_fmt_async, + .set_sysclk = mc13783_set_sysclk_dac, + .set_tdm_slot = mc13783_set_tdm_slot_dac, +}; + +static struct snd_soc_dai_ops mc13783_ops_codec = { + .hw_params = mc13783_pcm_hw_params_codec, + .set_fmt = mc13783_set_fmt_async, + .set_sysclk = mc13783_set_sysclk_codec, + .set_tdm_slot = mc13783_set_tdm_slot_codec, +}; + +/* + * The mc13783 has two SSI ports, both of them can be routed either + * to the voice codec or the stereo DAC. When two different SSI ports + * are used for the voice codec and the stereo DAC we can do different + * formats and sysclock settings for playback and capture + * (mc13783-hifi-playback and mc13783-hifi-capture). Using the same port + * forces us to use symmetric rates (mc13783-hifi). + */ +static struct snd_soc_dai_driver mc13783_dai_async[] = { + { + .name = "mc13783-hifi-playback", + .id = MC13783_ID_STEREO_DAC, + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = MC13783_FORMATS, + }, + .ops = &mc13783_ops_dac, + }, { + .name = "mc13783-hifi-capture", + .id = MC13783_ID_STEREO_CODEC, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = MC13783_RATES_RECORD, + .formats = MC13783_FORMATS, + }, + .ops = &mc13783_ops_codec, + }, +}; + +static struct snd_soc_dai_ops mc13783_ops_sync = { + .hw_params = mc13783_pcm_hw_params_sync, + .set_fmt = mc13783_set_fmt_sync, + .set_sysclk = mc13783_set_sysclk_sync, + .set_tdm_slot = mc13783_set_tdm_slot_sync, +}; + +static struct snd_soc_dai_driver mc13783_dai_sync[] = { + { + .name = "mc13783-hifi", + .id = MC13783_ID_SYNC, + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = MC13783_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = MC13783_RATES_RECORD, + .formats = MC13783_FORMATS, + }, + .ops = &mc13783_ops_sync, + .symmetric_rates = 1, + } +}; + +static struct snd_soc_codec_driver soc_codec_dev_mc13783 = { + .probe = mc13783_probe, + .remove = mc13783_remove, + .read = mc13783_read, + .write = mc13783_write, + .controls = mc13783_control_list, + .num_controls = ARRAY_SIZE(mc13783_control_list), + .dapm_widgets = mc13783_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(mc13783_dapm_widgets), + .dapm_routes = mc13783_routes, + .num_dapm_routes = ARRAY_SIZE(mc13783_routes), +}; + +static int mc13783_codec_probe(struct platform_device *pdev) +{ + struct mc13xxx *mc13xxx; + struct mc13783_priv *priv; + struct mc13xxx_codec_platform_data *pdata = pdev->dev.platform_data; + int ret; + + mc13xxx = dev_get_drvdata(pdev->dev.parent); + + + priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); + if (priv == NULL) + return -ENOMEM; + + dev_set_drvdata(&pdev->dev, priv); + priv->mc13xxx = mc13xxx; + if (pdata) { + priv->adc_ssi_port = pdata->adc_ssi_port; + priv->dac_ssi_port = pdata->dac_ssi_port; + } else { + priv->adc_ssi_port = MC13783_SSI1_PORT; + priv->dac_ssi_port = MC13783_SSI2_PORT; + } + + if (priv->adc_ssi_port == priv->dac_ssi_port) + ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_mc13783, + mc13783_dai_sync, ARRAY_SIZE(mc13783_dai_sync)); + else + ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_mc13783, + mc13783_dai_async, ARRAY_SIZE(mc13783_dai_async)); + + if (ret) + goto err_register_codec; + + return 0; + +err_register_codec: + dev_err(&pdev->dev, "register codec failed with %d\n", ret); + + return ret; +} + +static int mc13783_codec_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + + return 0; +} + +static struct platform_driver mc13783_codec_driver = { + .driver = { + .name = "mc13783-codec", + .owner = THIS_MODULE, + }, + .probe = mc13783_codec_probe, + .remove = __devexit_p(mc13783_codec_remove), +}; + +module_platform_driver(mc13783_codec_driver); + +MODULE_DESCRIPTION("ASoC MC13783 driver"); +MODULE_AUTHOR("Sascha Hauer, Pengutronix <s.hauer@pengutronix.de>"); +MODULE_AUTHOR("Philippe Retornaz <philippe.retornaz@epfl.ch>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/mc13783.h b/sound/soc/codecs/mc13783.h new file mode 100644 index 000000000000..3a6d1993a217 --- /dev/null +++ b/sound/soc/codecs/mc13783.h @@ -0,0 +1,28 @@ +/* + * Copyright 2008 Juergen Beisert, kernel@pengutronix.de + * + * 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. + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software Foundation, Inc. + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA. + */ + +#ifndef MC13783_MIXER_H +#define MC13783_MIXER_H + +#define MC13783_CLK_CLIA 1 +#define MC13783_CLK_CLIB 2 + +#define MC13783_ID_STEREO_DAC 1 +#define MC13783_ID_STEREO_CODEC 2 +#define MC13783_ID_SYNC 3 + +#endif /* MC13783_MIXER_H */ diff --git a/sound/soc/codecs/ml26124.c b/sound/soc/codecs/ml26124.c new file mode 100644 index 000000000000..22cb5bf59273 --- /dev/null +++ b/sound/soc/codecs/ml26124.c @@ -0,0 +1,681 @@ +/* + * Copyright (C) 2011 LAPIS Semiconductor Co., Ltd. + * + * 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; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307, USA. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/slab.h> +#include <linux/platform_device.h> +#include <linux/regmap.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include "ml26124.h" + +#define DVOL_CTL_DVMUTE_ON BIT(4) /* Digital volume MUTE On */ +#define DVOL_CTL_DVMUTE_OFF 0 /* Digital volume MUTE Off */ +#define ML26124_SAI_NO_DELAY BIT(1) +#define ML26124_SAI_FRAME_SYNC (BIT(5) | BIT(0)) /* For mono (Telecodec) */ +#define ML26134_CACHESIZE 212 +#define ML26124_VMID BIT(1) +#define ML26124_RATES (SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_32000 |\ + SNDRV_PCM_RATE_48000) +#define ML26124_FORMATS (SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S32_LE) +#define ML26124_NUM_REGISTER ML26134_CACHESIZE + +struct ml26124_priv { + u32 mclk; + u32 rate; + struct regmap *regmap; + int clk_in; + struct snd_pcm_substream *substream; +}; + +struct clk_coeff { + u32 mclk; + u32 rate; + u8 pllnl; + u8 pllnh; + u8 pllml; + u8 pllmh; + u8 plldiv; +}; + +/* ML26124 configuration */ +static const DECLARE_TLV_DB_SCALE(digital_tlv, -7150, 50, 0); + +static const DECLARE_TLV_DB_SCALE(alclvl, -2250, 150, 0); +static const DECLARE_TLV_DB_SCALE(mingain, -1200, 600, 0); +static const DECLARE_TLV_DB_SCALE(maxgain, -675, 600, 0); +static const DECLARE_TLV_DB_SCALE(boost_vol, -1200, 75, 0); +static const DECLARE_TLV_DB_SCALE(ngth, -7650, 150, 0); + +static const char * const ml26124_companding[] = {"16bit PCM", "u-law", + "A-law"}; + +static const struct soc_enum ml26124_adc_companding_enum + = SOC_ENUM_SINGLE(ML26124_SAI_TRANS_CTL, 6, 3, ml26124_companding); + +static const struct soc_enum ml26124_dac_companding_enum + = SOC_ENUM_SINGLE(ML26124_SAI_RCV_CTL, 6, 3, ml26124_companding); + +static const struct snd_kcontrol_new ml26124_snd_controls[] = { + SOC_SINGLE_TLV("Capture Digital Volume", ML26124_RECORD_DIG_VOL, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("Playback Digital Volume", ML26124_PLBAK_DIG_VOL, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("Digital Boost Volume", ML26124_DIGI_BOOST_VOL, 0, + 0x3f, 0, boost_vol), + SOC_SINGLE_TLV("EQ Band0 Volume", ML26124_EQ_GAIN_BRAND0, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("EQ Band1 Volume", ML26124_EQ_GAIN_BRAND1, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("EQ Band2 Volume", ML26124_EQ_GAIN_BRAND2, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("EQ Band3 Volume", ML26124_EQ_GAIN_BRAND3, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("EQ Band4 Volume", ML26124_EQ_GAIN_BRAND4, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("ALC Target Level", ML26124_ALC_TARGET_LEV, 0, + 0xf, 1, alclvl), + SOC_SINGLE_TLV("ALC Min Input Volume", ML26124_ALC_MAXMIN_GAIN, 0, + 7, 0, mingain), + SOC_SINGLE_TLV("ALC Max Input Volume", ML26124_ALC_MAXMIN_GAIN, 4, + 7, 1, maxgain), + SOC_SINGLE_TLV("Playback Limiter Min Input Volume", + ML26124_PL_MAXMIN_GAIN, 0, 7, 0, mingain), + SOC_SINGLE_TLV("Playback Limiter Max Input Volume", + ML26124_PL_MAXMIN_GAIN, 4, 7, 1, maxgain), + SOC_SINGLE_TLV("Playback Boost Volume", ML26124_PLYBAK_BOST_VOL, 0, + 0x3f, 0, boost_vol), + SOC_SINGLE("DC High Pass Filter Switch", ML26124_FILTER_EN, 0, 1, 0), + SOC_SINGLE("Noise High Pass Filter Switch", ML26124_FILTER_EN, 1, 1, 0), + SOC_SINGLE("ZC Switch", ML26124_PW_ZCCMP_PW_MNG, 1, + 1, 0), + SOC_SINGLE("EQ Band0 Switch", ML26124_FILTER_EN, 2, 1, 0), + SOC_SINGLE("EQ Band1 Switch", ML26124_FILTER_EN, 3, 1, 0), + SOC_SINGLE("EQ Band2 Switch", ML26124_FILTER_EN, 4, 1, 0), + SOC_SINGLE("EQ Band3 Switch", ML26124_FILTER_EN, 5, 1, 0), + SOC_SINGLE("EQ Band4 Switch", ML26124_FILTER_EN, 6, 1, 0), + SOC_SINGLE("Play Limiter", ML26124_DVOL_CTL, 0, 1, 0), + SOC_SINGLE("Capture Limiter", ML26124_DVOL_CTL, 1, 1, 0), + SOC_SINGLE("Digital Volume Fade Switch", ML26124_DVOL_CTL, 3, 1, 0), + SOC_SINGLE("Digital Switch", ML26124_DVOL_CTL, 4, 1, 0), + SOC_ENUM("DAC Companding", ml26124_dac_companding_enum), + SOC_ENUM("ADC Companding", ml26124_adc_companding_enum), +}; + +static const struct snd_kcontrol_new ml26124_output_mixer_controls[] = { + SOC_DAPM_SINGLE("DAC Switch", ML26124_SPK_AMP_OUT, 1, 1, 0), + SOC_DAPM_SINGLE("Line in loopback Switch", ML26124_SPK_AMP_OUT, 3, 1, + 0), + SOC_DAPM_SINGLE("PGA Switch", ML26124_SPK_AMP_OUT, 5, 1, 0), +}; + +/* Input mux */ +static const char * const ml26124_input_select[] = {"Analog MIC SingleEnded in", + "Digital MIC in", "Analog MIC Differential in"}; + +static const struct soc_enum ml26124_insel_enum = + SOC_ENUM_SINGLE(ML26124_MIC_IF_CTL, 0, 3, ml26124_input_select); + +static const struct snd_kcontrol_new ml26124_input_mux_controls = + SOC_DAPM_ENUM("Input Select", ml26124_insel_enum); + +static const struct snd_kcontrol_new ml26124_line_control = + SOC_DAPM_SINGLE("Switch", ML26124_PW_LOUT_PW_MNG, 1, 1, 0); + +static const struct snd_soc_dapm_widget ml26124_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("MCLKEN", ML26124_CLK_EN, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLLEN", ML26124_CLK_EN, 1, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLLOE", ML26124_CLK_EN, 2, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS", ML26124_PW_REF_PW_MNG, 2, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Output Mixer", SND_SOC_NOPM, 0, 0, + &ml26124_output_mixer_controls[0], + ARRAY_SIZE(ml26124_output_mixer_controls)), + SND_SOC_DAPM_DAC("DAC", "Playback", ML26124_PW_DAC_PW_MNG, 1, 0), + SND_SOC_DAPM_ADC("ADC", "Capture", ML26124_PW_IN_PW_MNG, 1, 0), + SND_SOC_DAPM_PGA("PGA", ML26124_PW_IN_PW_MNG, 3, 0, NULL, 0), + SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0, + &ml26124_input_mux_controls), + SND_SOC_DAPM_SWITCH("Line Out Enable", SND_SOC_NOPM, 0, 0, + &ml26124_line_control), + SND_SOC_DAPM_INPUT("MDIN"), + SND_SOC_DAPM_INPUT("MIN"), + SND_SOC_DAPM_INPUT("LIN"), + SND_SOC_DAPM_OUTPUT("SPOUT"), + SND_SOC_DAPM_OUTPUT("LOUT"), +}; + +static const struct snd_soc_dapm_route ml26124_intercon[] = { + /* Supply */ + {"DAC", NULL, "MCLKEN"}, + {"ADC", NULL, "MCLKEN"}, + {"DAC", NULL, "PLLEN"}, + {"ADC", NULL, "PLLEN"}, + {"DAC", NULL, "PLLOE"}, + {"ADC", NULL, "PLLOE"}, + + /* output mixer */ + {"Output Mixer", "DAC Switch", "DAC"}, + {"Output Mixer", "Line in loopback Switch", "LIN"}, + + /* outputs */ + {"LOUT", NULL, "Output Mixer"}, + {"SPOUT", NULL, "Output Mixer"}, + {"Line Out Enable", NULL, "LOUT"}, + + /* input */ + {"ADC", NULL, "Input Mux"}, + {"Input Mux", "Analog MIC SingleEnded in", "PGA"}, + {"Input Mux", "Analog MIC Differential in", "PGA"}, + {"PGA", NULL, "MIN"}, +}; + +/* PLLOutputFreq(Hz) = InputMclkFreq(Hz) * PLLM / (PLLN * PLLDIV) */ +static const struct clk_coeff coeff_div[] = { + {12288000, 16000, 0xc, 0x0, 0x20, 0x0, 0x4}, + {12288000, 32000, 0xc, 0x0, 0x20, 0x0, 0x4}, + {12288000, 48000, 0xc, 0x0, 0x30, 0x0, 0x4}, +}; + +static struct reg_default ml26124_reg[] = { + /* CLOCK control Register */ + {0x00, 0x00 }, /* Sampling Rate */ + {0x02, 0x00}, /* PLL NL */ + {0x04, 0x00}, /* PLLNH */ + {0x06, 0x00}, /* PLLML */ + {0x08, 0x00}, /* MLLMH */ + {0x0a, 0x00}, /* PLLDIV */ + {0x0c, 0x00}, /* Clock Enable */ + {0x0e, 0x00}, /* CLK Input/Output Control */ + + /* System Control Register */ + {0x10, 0x00}, /* Software RESET */ + {0x12, 0x00}, /* Record/Playback Run */ + {0x14, 0x00}, /* Mic Input/Output control */ + + /* Power Management Register */ + {0x20, 0x00}, /* Reference Power Management */ + {0x22, 0x00}, /* Input Power Management */ + {0x24, 0x00}, /* DAC Power Management */ + {0x26, 0x00}, /* SP-AMP Power Management */ + {0x28, 0x00}, /* LINEOUT Power Management */ + {0x2a, 0x00}, /* VIDEO Power Management */ + {0x2e, 0x00}, /* AC-CMP Power Management */ + + /* Analog reference Control Register */ + {0x30, 0x04}, /* MICBIAS Voltage Control */ + + /* Input/Output Amplifier Control Register */ + {0x32, 0x10}, /* MIC Input Volume */ + {0x38, 0x00}, /* Mic Boost Volume */ + {0x3a, 0x33}, /* Speaker AMP Volume */ + {0x48, 0x00}, /* AMP Volume Control Function Enable */ + {0x4a, 0x00}, /* Amplifier Volume Fader Control */ + + /* Analog Path Control Register */ + {0x54, 0x00}, /* Speaker AMP Output Control */ + {0x5a, 0x00}, /* Mic IF Control */ + {0xe8, 0x01}, /* Mic Select Control */ + + /* Audio Interface Control Register */ + {0x60, 0x00}, /* SAI-Trans Control */ + {0x62, 0x00}, /* SAI-Receive Control */ + {0x64, 0x00}, /* SAI Mode select */ + + /* DSP Control Register */ + {0x66, 0x01}, /* Filter Func Enable */ + {0x68, 0x00}, /* Volume Control Func Enable */ + {0x6A, 0x00}, /* Mixer & Volume Control*/ + {0x6C, 0xff}, /* Record Digital Volume */ + {0x70, 0xff}, /* Playback Digital Volume */ + {0x72, 0x10}, /* Digital Boost Volume */ + {0x74, 0xe7}, /* EQ gain Band0 */ + {0x76, 0xe7}, /* EQ gain Band1 */ + {0x78, 0xe7}, /* EQ gain Band2 */ + {0x7A, 0xe7}, /* EQ gain Band3 */ + {0x7C, 0xe7}, /* EQ gain Band4 */ + {0x7E, 0x00}, /* HPF2 CutOff*/ + {0x80, 0x00}, /* EQ Band0 Coef0L */ + {0x82, 0x00}, /* EQ Band0 Coef0H */ + {0x84, 0x00}, /* EQ Band0 Coef0L */ + {0x86, 0x00}, /* EQ Band0 Coef0H */ + {0x88, 0x00}, /* EQ Band1 Coef0L */ + {0x8A, 0x00}, /* EQ Band1 Coef0H */ + {0x8C, 0x00}, /* EQ Band1 Coef0L */ + {0x8E, 0x00}, /* EQ Band1 Coef0H */ + {0x90, 0x00}, /* EQ Band2 Coef0L */ + {0x92, 0x00}, /* EQ Band2 Coef0H */ + {0x94, 0x00}, /* EQ Band2 Coef0L */ + {0x96, 0x00}, /* EQ Band2 Coef0H */ + {0x98, 0x00}, /* EQ Band3 Coef0L */ + {0x9A, 0x00}, /* EQ Band3 Coef0H */ + {0x9C, 0x00}, /* EQ Band3 Coef0L */ + {0x9E, 0x00}, /* EQ Band3 Coef0H */ + {0xA0, 0x00}, /* EQ Band4 Coef0L */ + {0xA2, 0x00}, /* EQ Band4 Coef0H */ + {0xA4, 0x00}, /* EQ Band4 Coef0L */ + {0xA6, 0x00}, /* EQ Band4 Coef0H */ + + /* ALC Control Register */ + {0xb0, 0x00}, /* ALC Mode */ + {0xb2, 0x02}, /* ALC Attack Time */ + {0xb4, 0x03}, /* ALC Decay Time */ + {0xb6, 0x00}, /* ALC Hold Time */ + {0xb8, 0x0b}, /* ALC Target Level */ + {0xba, 0x70}, /* ALC Max/Min Gain */ + {0xbc, 0x00}, /* Noise Gate Threshold */ + {0xbe, 0x00}, /* ALC ZeroCross TimeOut */ + + /* Playback Limiter Control Register */ + {0xc0, 0x04}, /* PL Attack Time */ + {0xc2, 0x05}, /* PL Decay Time */ + {0xc4, 0x0d}, /* PL Target Level */ + {0xc6, 0x70}, /* PL Max/Min Gain */ + {0xc8, 0x10}, /* Playback Boost Volume */ + {0xca, 0x00}, /* PL ZeroCross TimeOut */ + + /* Video Amplifier Control Register */ + {0xd0, 0x01}, /* VIDEO AMP Gain Control */ + {0xd2, 0x01}, /* VIDEO AMP Setup 1 */ + {0xd4, 0x01}, /* VIDEO AMP Control2 */ +}; + +/* Get sampling rate value of sampling rate setting register (0x0) */ +static inline int get_srate(int rate) +{ + int srate; + + switch (rate) { + case 16000: + srate = 3; + break; + case 32000: + srate = 6; + break; + case 48000: + srate = 8; + break; + default: + return -EINVAL; + } + return srate; +} + +static inline int get_coeff(int mclk, int rate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(coeff_div); i++) { + if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk) + return i; + } + return -EINVAL; +} + +static int ml26124_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec); + int i = get_coeff(priv->mclk, params_rate(hw_params)); + + priv->substream = substream; + priv->rate = params_rate(hw_params); + + if (priv->clk_in) { + switch (priv->mclk / params_rate(hw_params)) { + case 256: + snd_soc_update_bits(codec, ML26124_CLK_CTL, + BIT(0) | BIT(1), 1); + break; + case 512: + snd_soc_update_bits(codec, ML26124_CLK_CTL, + BIT(0) | BIT(1), 2); + break; + case 1024: + snd_soc_update_bits(codec, ML26124_CLK_CTL, + BIT(0) | BIT(1), 3); + break; + default: + dev_err(codec->dev, "Unsupported MCLKI\n"); + break; + } + } else { + snd_soc_update_bits(codec, ML26124_CLK_CTL, + BIT(0) | BIT(1), 0); + } + + switch (params_rate(hw_params)) { + case 16000: + snd_soc_update_bits(codec, ML26124_SMPLING_RATE, 0xf, + get_srate(params_rate(hw_params))); + snd_soc_update_bits(codec, ML26124_PLLNL, 0xff, + coeff_div[i].pllnl); + snd_soc_update_bits(codec, ML26124_PLLNH, 0x1, + coeff_div[i].pllnh); + snd_soc_update_bits(codec, ML26124_PLLML, 0xff, + coeff_div[i].pllml); + snd_soc_update_bits(codec, ML26124_PLLMH, 0x3f, + coeff_div[i].pllmh); + snd_soc_update_bits(codec, ML26124_PLLDIV, 0x1f, + coeff_div[i].plldiv); + break; + case 32000: + snd_soc_update_bits(codec, ML26124_SMPLING_RATE, 0xf, + get_srate(params_rate(hw_params))); + snd_soc_update_bits(codec, ML26124_PLLNL, 0xff, + coeff_div[i].pllnl); + snd_soc_update_bits(codec, ML26124_PLLNH, 0x1, + coeff_div[i].pllnh); + snd_soc_update_bits(codec, ML26124_PLLML, 0xff, + coeff_div[i].pllml); + snd_soc_update_bits(codec, ML26124_PLLMH, 0x3f, + coeff_div[i].pllmh); + snd_soc_update_bits(codec, ML26124_PLLDIV, 0x1f, + coeff_div[i].plldiv); + break; + case 48000: + snd_soc_update_bits(codec, ML26124_SMPLING_RATE, 0xf, + get_srate(params_rate(hw_params))); + snd_soc_update_bits(codec, ML26124_PLLNL, 0xff, + coeff_div[i].pllnl); + snd_soc_update_bits(codec, ML26124_PLLNH, 0x1, + coeff_div[i].pllnh); + snd_soc_update_bits(codec, ML26124_PLLML, 0xff, + coeff_div[i].pllml); + snd_soc_update_bits(codec, ML26124_PLLMH, 0x3f, + coeff_div[i].pllmh); + snd_soc_update_bits(codec, ML26124_PLLDIV, 0x1f, + coeff_div[i].plldiv); + break; + default: + pr_err("%s:this rate is no support for ml26124\n", __func__); + return -EINVAL; + } + + return 0; +} + +static int ml26124_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec); + + switch (priv->substream->stream) { + case SNDRV_PCM_STREAM_CAPTURE: + snd_soc_update_bits(codec, ML26124_REC_PLYBAK_RUN, BIT(0), 1); + break; + case SNDRV_PCM_STREAM_PLAYBACK: + snd_soc_update_bits(codec, ML26124_REC_PLYBAK_RUN, BIT(1), 2); + break; + } + + if (mute) + snd_soc_update_bits(codec, ML26124_DVOL_CTL, BIT(4), + DVOL_CTL_DVMUTE_ON); + else + snd_soc_update_bits(codec, ML26124_DVOL_CTL, BIT(4), + DVOL_CTL_DVMUTE_OFF); + + return 0; +} + +static int ml26124_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + unsigned char mode; + struct snd_soc_codec *codec = codec_dai->codec; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + mode = 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + mode = 0; + break; + default: + return -EINVAL; + } + snd_soc_update_bits(codec, ML26124_SAI_MODE_SEL, BIT(0), mode); + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + default: + return -EINVAL; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + default: + return -EINVAL; + } + + return 0; +} + +static int ml26124_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec); + + switch (clk_id) { + case ML26124_USE_PLLOUT: + priv->clk_in = ML26124_USE_PLLOUT; + break; + case ML26124_USE_MCLKI: + priv->clk_in = ML26124_USE_MCLKI; + break; + default: + return -EINVAL; + } + + priv->mclk = freq; + + return 0; +} + +static int ml26124_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec); + + switch (level) { + case SND_SOC_BIAS_ON: + snd_soc_update_bits(codec, ML26124_PW_SPAMP_PW_MNG, + ML26124_R26_MASK, ML26124_BLT_PREAMP_ON); + msleep(100); + snd_soc_update_bits(codec, ML26124_PW_SPAMP_PW_MNG, + ML26124_R26_MASK, + ML26124_MICBEN_ON | ML26124_BLT_ALL_ON); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + /* VMID ON */ + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + snd_soc_update_bits(codec, ML26124_PW_REF_PW_MNG, + ML26124_VMID, ML26124_VMID); + msleep(500); + regcache_sync(priv->regmap); + } + break; + case SND_SOC_BIAS_OFF: + /* VMID OFF */ + snd_soc_update_bits(codec, ML26124_PW_REF_PW_MNG, + ML26124_VMID, 0); + break; + } + codec->dapm.bias_level = level; + return 0; +} + +static const struct snd_soc_dai_ops ml26124_dai_ops = { + .hw_params = ml26124_hw_params, + .digital_mute = ml26124_mute, + .set_fmt = ml26124_set_dai_fmt, + .set_sysclk = ml26124_set_dai_sysclk, +}; + +static struct snd_soc_dai_driver ml26124_dai = { + .name = "ml26124-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = ML26124_RATES, + .formats = ML26124_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = ML26124_RATES, + .formats = ML26124_FORMATS,}, + .ops = &ml26124_dai_ops, + .symmetric_rates = 1, +}; + +#ifdef CONFIG_PM +static int ml26124_suspend(struct snd_soc_codec *codec) +{ + ml26124_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int ml26124_resume(struct snd_soc_codec *codec) +{ + ml26124_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} +#else +#define ml26124_suspend NULL +#define ml26124_resume NULL +#endif + +static int ml26124_probe(struct snd_soc_codec *codec) +{ + int ret; + struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec); + codec->control_data = priv->regmap; + + ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + /* Software Reset */ + snd_soc_update_bits(codec, ML26124_SW_RST, 0x01, 1); + snd_soc_update_bits(codec, ML26124_SW_RST, 0x01, 0); + + ml26124_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_ml26124 = { + .probe = ml26124_probe, + .suspend = ml26124_suspend, + .resume = ml26124_resume, + .set_bias_level = ml26124_set_bias_level, + .dapm_widgets = ml26124_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ml26124_dapm_widgets), + .dapm_routes = ml26124_intercon, + .num_dapm_routes = ARRAY_SIZE(ml26124_intercon), + .controls = ml26124_snd_controls, + .num_controls = ARRAY_SIZE(ml26124_snd_controls), +}; + +static const struct regmap_config ml26124_i2c_regmap = { + .val_bits = 8, + .reg_bits = 8, + .max_register = ML26124_NUM_REGISTER, + .reg_defaults = ml26124_reg, + .num_reg_defaults = ARRAY_SIZE(ml26124_reg), + .cache_type = REGCACHE_RBTREE, + .write_flag_mask = 0x01, +}; + +static __devinit int ml26124_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct ml26124_priv *priv; + int ret; + + priv = devm_kzalloc(&i2c->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + i2c_set_clientdata(i2c, priv); + + priv->regmap = regmap_init_i2c(i2c, &ml26124_i2c_regmap); + if (IS_ERR(priv->regmap)) { + ret = PTR_ERR(priv->regmap); + dev_err(&i2c->dev, "regmap_init_i2c() failed: %d\n", ret); + return ret; + } + + return snd_soc_register_codec(&i2c->dev, + &soc_codec_dev_ml26124, &ml26124_dai, 1); +} + +static __devexit int ml26124_i2c_remove(struct i2c_client *client) +{ + struct ml26124_priv *priv = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + regmap_exit(priv->regmap); + return 0; +} + +static const struct i2c_device_id ml26124_i2c_id[] = { + { "ml26124", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ml26124_i2c_id); + +static struct i2c_driver ml26124_i2c_driver = { + .driver = { + .name = "ml26124", + .owner = THIS_MODULE, + }, + .probe = ml26124_i2c_probe, + .remove = __devexit_p(ml26124_i2c_remove), + .id_table = ml26124_i2c_id, +}; + +module_i2c_driver(ml26124_i2c_driver); + +MODULE_AUTHOR("Tomoya MORINAGA <tomoya.rohm@gmail.com>"); +MODULE_DESCRIPTION("LAPIS Semiconductor ML26124 ALSA SoC codec driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ml26124.h b/sound/soc/codecs/ml26124.h new file mode 100644 index 000000000000..5ea0cbb8c46c --- /dev/null +++ b/sound/soc/codecs/ml26124.h @@ -0,0 +1,184 @@ +/* + * Copyright (C) 2011 LAPIS Semiconductor Co., Ltd. + * + * 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; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307, USA. + */ + +#ifndef ML26124_H +#define ML26124_H + +/* Clock Control Register */ +#define ML26124_SMPLING_RATE 0x00 +#define ML26124_PLLNL 0x02 +#define ML26124_PLLNH 0x04 +#define ML26124_PLLML 0x06 +#define ML26124_PLLMH 0x08 +#define ML26124_PLLDIV 0x0a +#define ML26124_CLK_EN 0x0c +#define ML26124_CLK_CTL 0x0e + +/* System Control Register */ +#define ML26124_SW_RST 0x10 +#define ML26124_REC_PLYBAK_RUN 0x12 +#define ML26124_MIC_TIM 0x14 + +/* Power Mnagement Register */ +#define ML26124_PW_REF_PW_MNG 0x20 +#define ML26124_PW_IN_PW_MNG 0x22 +#define ML26124_PW_DAC_PW_MNG 0x24 +#define ML26124_PW_SPAMP_PW_MNG 0x26 +#define ML26124_PW_LOUT_PW_MNG 0x28 +#define ML26124_PW_VOUT_PW_MNG 0x2a +#define ML26124_PW_ZCCMP_PW_MNG 0x2e + +/* Analog Reference Control Register */ +#define ML26124_PW_MICBIAS_VOL 0x30 + +/* Input/Output Amplifier Control Register */ +#define ML26124_PW_MIC_IN_VOL 0x32 +#define ML26124_PW_MIC_BOST_VOL 0x38 +#define ML26124_PW_SPK_AMP_VOL 0x3a +#define ML26124_PW_AMP_VOL_FUNC 0x48 +#define ML26124_PW_AMP_VOL_FADE 0x4a + +/* Analog Path Control Register */ +#define ML26124_SPK_AMP_OUT 0x54 +#define ML26124_MIC_IF_CTL 0x5a +#define ML26124_MIC_SELECT 0xe8 + +/* Audio Interface Control Register */ +#define ML26124_SAI_TRANS_CTL 0x60 +#define ML26124_SAI_RCV_CTL 0x62 +#define ML26124_SAI_MODE_SEL 0x64 + +/* DSP Control Register */ +#define ML26124_FILTER_EN 0x66 +#define ML26124_DVOL_CTL 0x68 +#define ML26124_MIXER_VOL_CTL 0x6a +#define ML26124_RECORD_DIG_VOL 0x6c +#define ML26124_PLBAK_DIG_VOL 0x70 +#define ML26124_DIGI_BOOST_VOL 0x72 +#define ML26124_EQ_GAIN_BRAND0 0x74 +#define ML26124_EQ_GAIN_BRAND1 0x76 +#define ML26124_EQ_GAIN_BRAND2 0x78 +#define ML26124_EQ_GAIN_BRAND3 0x7a +#define ML26124_EQ_GAIN_BRAND4 0x7c +#define ML26124_HPF2_CUTOFF 0x7e +#define ML26124_EQBRAND0_F0L 0x80 +#define ML26124_EQBRAND0_F0H 0x82 +#define ML26124_EQBRAND0_F1L 0x84 +#define ML26124_EQBRAND0_F1H 0x86 +#define ML26124_EQBRAND1_F0L 0x88 +#define ML26124_EQBRAND1_F0H 0x8a +#define ML26124_EQBRAND1_F1L 0x8c +#define ML26124_EQBRAND1_F1H 0x8e +#define ML26124_EQBRAND2_F0L 0x90 +#define ML26124_EQBRAND2_F0H 0x92 +#define ML26124_EQBRAND2_F1L 0x94 +#define ML26124_EQBRAND2_F1H 0x96 +#define ML26124_EQBRAND3_F0L 0x98 +#define ML26124_EQBRAND3_F0H 0x9a +#define ML26124_EQBRAND3_F1L 0x9c +#define ML26124_EQBRAND3_F1H 0x9e +#define ML26124_EQBRAND4_F0L 0xa0 +#define ML26124_EQBRAND4_F0H 0xa2 +#define ML26124_EQBRAND4_F1L 0xa4 +#define ML26124_EQBRAND4_F1H 0xa6 + +/* ALC Control Register */ +#define ML26124_ALC_MODE 0xb0 +#define ML26124_ALC_ATTACK_TIM 0xb2 +#define ML26124_ALC_DECAY_TIM 0xb4 +#define ML26124_ALC_HOLD_TIM 0xb6 +#define ML26124_ALC_TARGET_LEV 0xb8 +#define ML26124_ALC_MAXMIN_GAIN 0xba +#define ML26124_NOIS_GATE_THRSH 0xbc +#define ML26124_ALC_ZERO_TIMOUT 0xbe + +/* Playback Limiter Control Register */ +#define ML26124_PL_ATTACKTIME 0xc0 +#define ML26124_PL_DECAYTIME 0xc2 +#define ML26124_PL_TARGETTIME 0xc4 +#define ML26124_PL_MAXMIN_GAIN 0xc6 +#define ML26124_PLYBAK_BOST_VOL 0xc8 +#define ML26124_PL_0CROSS_TIMOUT 0xca + +/* Video Amplifer Control Register */ +#define ML26124_VIDEO_AMP_GAIN_CTL 0xd0 +#define ML26124_VIDEO_AMP_SETUP1 0xd2 +#define ML26124_VIDEO_AMP_CTL2 0xd4 + +/* Clock select for machine driver */ +#define ML26124_USE_PLL 0 +#define ML26124_USE_MCLKI_256FS 1 +#define ML26124_USE_MCLKI_512FS 2 +#define ML26124_USE_MCLKI_1024FS 3 + +/* Register Mask */ +#define ML26124_R0_MASK 0xf +#define ML26124_R2_MASK 0xff +#define ML26124_R4_MASK 0x1 +#define ML26124_R6_MASK 0xf +#define ML26124_R8_MASK 0x3f +#define ML26124_Ra_MASK 0x1f +#define ML26124_Rc_MASK 0x1f +#define ML26124_Re_MASK 0x7 +#define ML26124_R10_MASK 0x1 +#define ML26124_R12_MASK 0x17 +#define ML26124_R14_MASK 0x3f +#define ML26124_R20_MASK 0x47 +#define ML26124_R22_MASK 0xa +#define ML26124_R24_MASK 0x2 +#define ML26124_R26_MASK 0x1f +#define ML26124_R28_MASK 0x2 +#define ML26124_R2a_MASK 0x2 +#define ML26124_R2e_MASK 0x2 +#define ML26124_R30_MASK 0x7 +#define ML26124_R32_MASK 0x3f +#define ML26124_R38_MASK 0x38 +#define ML26124_R3a_MASK 0x3f +#define ML26124_R48_MASK 0x3 +#define ML26124_R4a_MASK 0x7 +#define ML26124_R54_MASK 0x2a +#define ML26124_R5a_MASK 0x3 +#define ML26124_Re8_MASK 0x3 +#define ML26124_R60_MASK 0xff +#define ML26124_R62_MASK 0xff +#define ML26124_R64_MASK 0x1 +#define ML26124_R66_MASK 0xff +#define ML26124_R68_MASK 0x3b +#define ML26124_R6a_MASK 0xf3 +#define ML26124_R6c_MASK 0xff +#define ML26124_R70_MASK 0xff + +#define ML26124_MCLKEN BIT(0) +#define ML26124_PLLEN BIT(1) +#define ML26124_PLLOE BIT(2) +#define ML26124_MCLKOE BIT(3) + +#define ML26124_BLT_ALL_ON 0x1f +#define ML26124_BLT_PREAMP_ON 0x13 + +#define ML26124_MICBEN_ON BIT(2) + +enum ml26124_regs { + ML26124_MCLK = 0, +}; + +enum ml26124_clk_in { + ML26124_USE_PLLOUT = 0, + ML26124_USE_MCLKI, +}; + +#endif diff --git a/sound/soc/codecs/omap-hdmi.c b/sound/soc/codecs/omap-hdmi.c new file mode 100644 index 000000000000..1bf5c74f5f96 --- /dev/null +++ b/sound/soc/codecs/omap-hdmi.c @@ -0,0 +1,69 @@ +/* + * ALSA SoC codec driver for HDMI audio on OMAP processors. + * Copyright (C) 2012 Texas Instruments Incorporated - http://www.ti.com/ + * Author: Ricardo Neri <ricardo.neri@ti.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ +#include <linux/module.h> +#include <sound/soc.h> + +#define DRV_NAME "hdmi-audio-codec" + +static struct snd_soc_codec_driver omap_hdmi_codec; + +static struct snd_soc_dai_driver omap_hdmi_codec_dai = { + .name = "omap-hdmi-hifi", + .playback = { + .channels_min = 2, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE, + }, +}; + +static __devinit int omap_hdmi_codec_probe(struct platform_device *pdev) +{ + return snd_soc_register_codec(&pdev->dev, &omap_hdmi_codec, + &omap_hdmi_codec_dai, 1); +} + +static __devexit int omap_hdmi_codec_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + return 0; +} + +static struct platform_driver omap_hdmi_codec_driver = { + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + }, + + .probe = omap_hdmi_codec_probe, + .remove = __devexit_p(omap_hdmi_codec_remove), +}; + +module_platform_driver(omap_hdmi_codec_driver); + +MODULE_AUTHOR("Ricardo Neri <ricardo.neri@ti.com>"); +MODULE_DESCRIPTION("ASoC OMAP HDMI codec driver"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:" DRV_NAME); diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c index 20c324c7c349..960d0e93cce9 100644 --- a/sound/soc/codecs/rt5631.c +++ b/sound/soc/codecs/rt5631.c @@ -18,7 +18,7 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> -#include <linux/spi/spi.h> +#include <linux/regmap.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -30,6 +30,7 @@ #include "rt5631.h" struct rt5631_priv { + struct regmap *regmap; int codec_version; int master; int sysclk; @@ -38,33 +39,33 @@ struct rt5631_priv { int dmic_used_flag; }; -static const u16 rt5631_reg[RT5631_VENDOR_ID2 + 1] = { - [RT5631_SPK_OUT_VOL] = 0x8888, - [RT5631_HP_OUT_VOL] = 0x8080, - [RT5631_MONO_AXO_1_2_VOL] = 0xa080, - [RT5631_AUX_IN_VOL] = 0x0808, - [RT5631_ADC_REC_MIXER] = 0xf0f0, - [RT5631_VDAC_DIG_VOL] = 0x0010, - [RT5631_OUTMIXER_L_CTRL] = 0xffc0, - [RT5631_OUTMIXER_R_CTRL] = 0xffc0, - [RT5631_AXO1MIXER_CTRL] = 0x88c0, - [RT5631_AXO2MIXER_CTRL] = 0x88c0, - [RT5631_DIG_MIC_CTRL] = 0x3000, - [RT5631_MONO_INPUT_VOL] = 0x8808, - [RT5631_SPK_MIXER_CTRL] = 0xf8f8, - [RT5631_SPK_MONO_OUT_CTRL] = 0xfc00, - [RT5631_SPK_MONO_HP_OUT_CTRL] = 0x4440, - [RT5631_SDP_CTRL] = 0x8000, - [RT5631_MONO_SDP_CTRL] = 0x8000, - [RT5631_STEREO_AD_DA_CLK_CTRL] = 0x2010, - [RT5631_GEN_PUR_CTRL_REG] = 0x0e00, - [RT5631_INT_ST_IRQ_CTRL_2] = 0x071a, - [RT5631_MISC_CTRL] = 0x2040, - [RT5631_DEPOP_FUN_CTRL_2] = 0x8000, - [RT5631_SOFT_VOL_CTRL] = 0x07e0, - [RT5631_ALC_CTRL_1] = 0x0206, - [RT5631_ALC_CTRL_3] = 0x2000, - [RT5631_PSEUDO_SPATL_CTRL] = 0x0553, +static const struct reg_default rt5631_reg[] = { + { RT5631_SPK_OUT_VOL, 0x8888 }, + { RT5631_HP_OUT_VOL, 0x8080 }, + { RT5631_MONO_AXO_1_2_VOL, 0xa080 }, + { RT5631_AUX_IN_VOL, 0x0808 }, + { RT5631_ADC_REC_MIXER, 0xf0f0 }, + { RT5631_VDAC_DIG_VOL, 0x0010 }, + { RT5631_OUTMIXER_L_CTRL, 0xffc0 }, + { RT5631_OUTMIXER_R_CTRL, 0xffc0 }, + { RT5631_AXO1MIXER_CTRL, 0x88c0 }, + { RT5631_AXO2MIXER_CTRL, 0x88c0 }, + { RT5631_DIG_MIC_CTRL, 0x3000 }, + { RT5631_MONO_INPUT_VOL, 0x8808 }, + { RT5631_SPK_MIXER_CTRL, 0xf8f8 }, + { RT5631_SPK_MONO_OUT_CTRL, 0xfc00 }, + { RT5631_SPK_MONO_HP_OUT_CTRL, 0x4440 }, + { RT5631_SDP_CTRL, 0x8000 }, + { RT5631_MONO_SDP_CTRL, 0x8000 }, + { RT5631_STEREO_AD_DA_CLK_CTRL, 0x2010 }, + { RT5631_GEN_PUR_CTRL_REG, 0x0e00 }, + { RT5631_INT_ST_IRQ_CTRL_2, 0x071a }, + { RT5631_MISC_CTRL, 0x2040 }, + { RT5631_DEPOP_FUN_CTRL_2, 0x8000 }, + { RT5631_SOFT_VOL_CTRL, 0x07e0 }, + { RT5631_ALC_CTRL_1, 0x0206 }, + { RT5631_ALC_CTRL_3, 0x2000 }, + { RT5631_PSEUDO_SPATL_CTRL, 0x0553 }, }; /** @@ -96,8 +97,7 @@ static int rt5631_reset(struct snd_soc_codec *codec) return snd_soc_write(codec, RT5631_RESET, 0); } -static int rt5631_volatile_register(struct snd_soc_codec *codec, - unsigned int reg) +static bool rt5631_volatile_register(struct device *dev, unsigned int reg) { switch (reg) { case RT5631_RESET: @@ -111,8 +111,7 @@ static int rt5631_volatile_register(struct snd_soc_codec *codec, } } -static int rt5631_readable_register(struct snd_soc_codec *codec, - unsigned int reg) +static bool rt5631_readable_register(struct device *dev, unsigned int reg) { switch (reg) { case RT5631_RESET: @@ -1361,8 +1360,7 @@ static int get_coeff(int mclk, int rate, int timesofbclk) static int rt5631_hifi_pcm_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct rt5631_priv *rt5631 = snd_soc_codec_get_drvdata(codec); int timesofbclk = 32, coeff; unsigned int iface = 0; @@ -1544,6 +1542,8 @@ static int rt5631_codec_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, static int rt5631_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct rt5631_priv *rt5631 = snd_soc_codec_get_drvdata(codec); + switch (level) { case SND_SOC_BIAS_ON: case SND_SOC_BIAS_PREPARE: @@ -1561,8 +1561,8 @@ static int rt5631_set_bias_level(struct snd_soc_codec *codec, snd_soc_update_bits(codec, RT5631_PWR_MANAG_ADD3, RT5631_PWR_FAST_VREF_CTRL, RT5631_PWR_FAST_VREF_CTRL); - codec->cache_only = false; - snd_soc_cache_sync(codec); + regcache_cache_only(rt5631->regmap, false); + regcache_sync(rt5631->regmap); } break; @@ -1587,7 +1587,9 @@ static int rt5631_probe(struct snd_soc_codec *codec) unsigned int val; int ret; - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_I2C); + codec->control_data = rt5631->regmap; + + ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; @@ -1698,12 +1700,6 @@ static struct snd_soc_codec_driver soc_codec_dev_rt5631 = { .suspend = rt5631_suspend, .resume = rt5631_resume, .set_bias_level = rt5631_set_bias_level, - .reg_cache_size = RT5631_VENDOR_ID2 + 1, - .reg_word_size = sizeof(u16), - .reg_cache_default = rt5631_reg, - .volatile_register = rt5631_volatile_register, - .readable_register = rt5631_readable_register, - .reg_cache_step = 1, .controls = rt5631_snd_controls, .num_controls = ARRAY_SIZE(rt5631_snd_controls), .dapm_widgets = rt5631_dapm_widgets, @@ -1718,6 +1714,18 @@ static const struct i2c_device_id rt5631_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, rt5631_i2c_id); +static const struct regmap_config rt5631_regmap_config = { + .reg_bits = 8, + .val_bits = 16, + + .readable_reg = rt5631_readable_register, + .volatile_reg = rt5631_volatile_register, + .max_register = RT5631_VENDOR_ID2, + .reg_defaults = rt5631_reg, + .num_reg_defaults = ARRAY_SIZE(rt5631_reg), + .cache_type = REGCACHE_RBTREE, +}; + static int rt5631_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -1731,6 +1739,10 @@ static int rt5631_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, rt5631); + rt5631->regmap = devm_regmap_init_i2c(i2c, &rt5631_regmap_config); + if (IS_ERR(rt5631->regmap)) + return PTR_ERR(rt5631->regmap); + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5631, rt5631_dai, ARRAY_SIZE(rt5631_dai)); return ret; @@ -1752,17 +1764,7 @@ static struct i2c_driver rt5631_i2c_driver = { .id_table = rt5631_i2c_id, }; -static int __init rt5631_modinit(void) -{ - return i2c_add_driver(&rt5631_i2c_driver); -} -module_init(rt5631_modinit); - -static void __exit rt5631_modexit(void) -{ - i2c_del_driver(&rt5631_i2c_driver); -} -module_exit(rt5631_modexit); +module_i2c_driver(rt5631_i2c_driver); MODULE_DESCRIPTION("ASoC RT5631 driver"); MODULE_AUTHOR("flove <flove@realtek.com>"); diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index c395ec370445..8af6a5245b18 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -84,8 +84,8 @@ static struct regulator_consumer_supply ldo_consumer[] = { static struct regulator_init_data ldo_init_data = { .constraints = { - .min_uV = 850000, - .max_uV = 1600000, + .min_uV = 1200000, + .max_uV = 1200000, .valid_modes_mask = REGULATOR_MODE_NORMAL, .valid_ops_mask = REGULATOR_CHANGE_STATUS, }, @@ -197,9 +197,9 @@ static const struct snd_soc_dapm_widget sgtl5000_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("HP_OUT"), SND_SOC_DAPM_OUTPUT("LINE_OUT"), - SND_SOC_DAPM_MICBIAS_E("Mic Bias", SGTL5000_CHIP_MIC_CTRL, 8, 0, - mic_bias_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_SUPPLY("Mic Bias", SGTL5000_CHIP_MIC_CTRL, 8, 0, + mic_bias_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_PGA("HP", SGTL5000_CHIP_ANA_POWER, 4, 0, NULL, 0), SND_SOC_DAPM_PGA("LO", SGTL5000_CHIP_ANA_POWER, 0, 0, NULL, 0), @@ -665,8 +665,7 @@ static int sgtl5000_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); int channels = params_channels(params); int i2s_ctl = 0; @@ -1455,17 +1454,7 @@ static struct i2c_driver sgtl5000_i2c_driver = { .id_table = sgtl5000_id, }; -static int __init sgtl5000_modinit(void) -{ - return i2c_add_driver(&sgtl5000_i2c_driver); -} -module_init(sgtl5000_modinit); - -static void __exit sgtl5000_exit(void) -{ - i2c_del_driver(&sgtl5000_i2c_driver); -} -module_exit(sgtl5000_exit); +module_i2c_driver(sgtl5000_i2c_driver); MODULE_DESCRIPTION("Freescale SGTL5000 ALSA SoC Codec Driver"); MODULE_AUTHOR("Zeng Zhaoming <zengzm.kernel@gmail.com>"); diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index de2b20544ceb..079066fef425 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -33,6 +33,7 @@ #include <linux/pm.h> #include <linux/i2c.h> #include <linux/spi/spi.h> +#include <linux/regmap.h> #include <linux/slab.h> #include <sound/core.h> #include <sound/pcm.h> @@ -43,8 +44,6 @@ #include "ssm2602.h" -#define SSM2602_VERSION "0.1" - enum ssm2602_type { SSM2602, SSM2604, @@ -53,10 +52,12 @@ enum ssm2602_type { /* codec private data */ struct ssm2602_priv { unsigned int sysclk; - enum snd_soc_control_type control_type; + struct snd_pcm_hw_constraint_list *sysclk_constraints; struct snd_pcm_substream *master_substream; struct snd_pcm_substream *slave_substream; + struct regmap *regmap; + enum ssm2602_type type; unsigned int clk_out_pwr; }; @@ -73,7 +74,6 @@ static const u16 ssm2602_reg[SSM2602_CACHEREGNUM] = { 0x0000, 0x0000 }; -#define ssm2602_reset(c) snd_soc_write(c, SSM2602_RESET, 0) /*Appending several "None"s just for OSS mixer use*/ static const char *ssm2602_input_select[] = { @@ -195,6 +195,24 @@ static const struct snd_soc_dapm_route ssm2604_routes[] = { {"ADC", NULL, "Line Input"}, }; +static const unsigned int ssm2602_rates_12288000[] = { + 8000, 32000, 48000, 96000, +}; + +static struct snd_pcm_hw_constraint_list ssm2602_constraints_12288000 = { + .list = ssm2602_rates_12288000, + .count = ARRAY_SIZE(ssm2602_rates_12288000), +}; + +static const unsigned int ssm2602_rates_11289600[] = { + 8000, 44100, 88200, +}; + +static struct snd_pcm_hw_constraint_list ssm2602_constraints_11289600 = { + .list = ssm2602_rates_11289600, + .count = ARRAY_SIZE(ssm2602_rates_11289600), +}; + struct ssm2602_coeff { u32 mclk; u32 rate; @@ -254,11 +272,10 @@ static int ssm2602_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); - u16 iface = snd_soc_read(codec, SSM2602_IFACE) & 0xfff3; int srate = ssm2602_get_coeff(ssm2602->sysclk, params_rate(params)); + unsigned int iface; if (substream == ssm2602->slave_substream) { dev_dbg(codec->dev, "Ignoring hw_params for slave substream\n"); @@ -268,31 +285,34 @@ static int ssm2602_hw_params(struct snd_pcm_substream *substream, if (srate < 0) return srate; - snd_soc_write(codec, SSM2602_SRATE, srate); + regmap_write(ssm2602->regmap, SSM2602_SRATE, srate); /* bit size */ switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: + iface = 0x0; break; case SNDRV_PCM_FORMAT_S20_3LE: - iface |= 0x0004; + iface = 0x4; break; case SNDRV_PCM_FORMAT_S24_LE: - iface |= 0x0008; + iface = 0x8; break; case SNDRV_PCM_FORMAT_S32_LE: - iface |= 0x000c; + iface = 0xc; break; + default: + return -EINVAL; } - snd_soc_write(codec, SSM2602_IFACE, iface); + regmap_update_bits(ssm2602->regmap, SSM2602_IFACE, + IFACE_AUDIO_DATA_LEN, iface); return 0; } static int ssm2602_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); struct snd_pcm_runtime *master_runtime; @@ -322,14 +342,19 @@ static int ssm2602_startup(struct snd_pcm_substream *substream, } else ssm2602->master_substream = substream; + if (ssm2602->sysclk_constraints) { + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + ssm2602->sysclk_constraints); + } + return 0; } static void ssm2602_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); if (ssm2602->master_substream == substream) @@ -341,14 +366,14 @@ static void ssm2602_shutdown(struct snd_pcm_substream *substream, static int ssm2602_mute(struct snd_soc_dai *dai, int mute) { - struct snd_soc_codec *codec = dai->codec; + struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(dai->codec); if (mute) - snd_soc_update_bits(codec, SSM2602_APDIGI, + regmap_update_bits(ssm2602->regmap, SSM2602_APDIGI, APDIGI_ENABLE_DAC_MUTE, APDIGI_ENABLE_DAC_MUTE); else - snd_soc_update_bits(codec, SSM2602_APDIGI, + regmap_update_bits(ssm2602->regmap, SSM2602_APDIGI, APDIGI_ENABLE_DAC_MUTE, 0); return 0; } @@ -364,16 +389,21 @@ static int ssm2602_set_dai_sysclk(struct snd_soc_dai *codec_dai, return -EINVAL; switch (freq) { - case 11289600: - case 12000000: case 12288000: - case 16934400: case 18432000: - ssm2602->sysclk = freq; + ssm2602->sysclk_constraints = &ssm2602_constraints_12288000; + break; + case 11289600: + case 16934400: + ssm2602->sysclk_constraints = &ssm2602_constraints_11289600; + break; + case 12000000: + ssm2602->sysclk_constraints = NULL; break; default: return -EINVAL; } + ssm2602->sysclk = freq; } else { unsigned int mask; @@ -393,7 +423,7 @@ static int ssm2602_set_dai_sysclk(struct snd_soc_dai *codec_dai, else ssm2602->clk_out_pwr &= ~mask; - snd_soc_update_bits(codec, SSM2602_PWR, + regmap_update_bits(ssm2602->regmap, SSM2602_PWR, PWR_CLK_OUT_PDN | PWR_OSC_PDN, ssm2602->clk_out_pwr); } @@ -403,8 +433,8 @@ static int ssm2602_set_dai_sysclk(struct snd_soc_dai *codec_dai, static int ssm2602_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { - struct snd_soc_codec *codec = codec_dai->codec; - u16 iface = 0; + struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec_dai->codec); + unsigned int iface = 0; /* set master/slave audio interface */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { @@ -455,7 +485,7 @@ static int ssm2602_set_dai_fmt(struct snd_soc_dai *codec_dai, } /* set iface */ - snd_soc_write(codec, SSM2602_IFACE, iface); + regmap_write(ssm2602->regmap, SSM2602_IFACE, iface); return 0; } @@ -467,7 +497,7 @@ static int ssm2602_set_bias_level(struct snd_soc_codec *codec, switch (level) { case SND_SOC_BIAS_ON: /* vref/mid on, osc and clkout on if enabled */ - snd_soc_update_bits(codec, SSM2602_PWR, + regmap_update_bits(ssm2602->regmap, SSM2602_PWR, PWR_POWER_OFF | PWR_CLK_OUT_PDN | PWR_OSC_PDN, ssm2602->clk_out_pwr); break; @@ -475,13 +505,13 @@ static int ssm2602_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: /* everything off except vref/vmid, */ - snd_soc_update_bits(codec, SSM2602_PWR, + regmap_update_bits(ssm2602->regmap, SSM2602_PWR, PWR_POWER_OFF | PWR_CLK_OUT_PDN | PWR_OSC_PDN, PWR_CLK_OUT_PDN | PWR_OSC_PDN); break; case SND_SOC_BIAS_OFF: /* everything off */ - snd_soc_update_bits(codec, SSM2602_PWR, + regmap_update_bits(ssm2602->regmap, SSM2602_PWR, PWR_POWER_OFF, PWR_POWER_OFF); break; @@ -540,12 +570,13 @@ static int ssm2602_resume(struct snd_soc_codec *codec) static int ssm2602_probe(struct snd_soc_codec *codec) { + struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); struct snd_soc_dapm_context *dapm = &codec->dapm; int ret; - snd_soc_update_bits(codec, SSM2602_LOUT1V, + regmap_update_bits(ssm2602->regmap, SSM2602_LOUT1V, LOUT1V_LRHP_BOTH, LOUT1V_LRHP_BOTH); - snd_soc_update_bits(codec, SSM2602_ROUT1V, + regmap_update_bits(ssm2602->regmap, SSM2602_ROUT1V, ROUT1V_RLHP_BOTH, ROUT1V_RLHP_BOTH); ret = snd_soc_add_codec_controls(codec, ssm2602_snd_controls, @@ -581,27 +612,26 @@ static int ssm260x_probe(struct snd_soc_codec *codec) struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); int ret; - pr_info("ssm2602 Audio Codec %s", SSM2602_VERSION); - - ret = snd_soc_codec_set_cache_io(codec, 7, 9, ssm2602->control_type); + codec->control_data = ssm2602->regmap; + ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; } - ret = ssm2602_reset(codec); + ret = regmap_write(ssm2602->regmap, SSM2602_RESET, 0); if (ret < 0) { dev_err(codec->dev, "Failed to issue reset: %d\n", ret); return ret; } /* set the update bits */ - snd_soc_update_bits(codec, SSM2602_LINVOL, + regmap_update_bits(ssm2602->regmap, SSM2602_LINVOL, LINVOL_LRIN_BOTH, LINVOL_LRIN_BOTH); - snd_soc_update_bits(codec, SSM2602_RINVOL, + regmap_update_bits(ssm2602->regmap, SSM2602_RINVOL, RINVOL_RLIN_BOTH, RINVOL_RLIN_BOTH); /*select Line in as default input*/ - snd_soc_write(codec, SSM2602_APANA, APANA_SELECT_DAC | + regmap_write(ssm2602->regmap, SSM2602_APANA, APANA_SELECT_DAC | APANA_ENABLE_MIC_BOOST); switch (ssm2602->type) { @@ -634,9 +664,6 @@ static struct snd_soc_codec_driver soc_codec_dev_ssm2602 = { .suspend = ssm2602_suspend, .resume = ssm2602_resume, .set_bias_level = ssm2602_set_bias_level, - .reg_cache_size = ARRAY_SIZE(ssm2602_reg), - .reg_word_size = sizeof(u16), - .reg_cache_default = ssm2602_reg, .controls = ssm260x_snd_controls, .num_controls = ARRAY_SIZE(ssm260x_snd_controls), @@ -646,6 +673,23 @@ static struct snd_soc_codec_driver soc_codec_dev_ssm2602 = { .num_dapm_routes = ARRAY_SIZE(ssm260x_routes), }; +static bool ssm2602_register_volatile(struct device *dev, unsigned int reg) +{ + return reg == SSM2602_RESET; +} + +static const struct regmap_config ssm2602_regmap_config = { + .val_bits = 9, + .reg_bits = 7, + + .max_register = SSM2602_RESET, + .volatile_reg = ssm2602_register_volatile, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults_raw = ssm2602_reg, + .num_reg_defaults_raw = ARRAY_SIZE(ssm2602_reg), +}; + #if defined(CONFIG_SPI_MASTER) static int __devinit ssm2602_spi_probe(struct spi_device *spi) { @@ -658,9 +702,12 @@ static int __devinit ssm2602_spi_probe(struct spi_device *spi) return -ENOMEM; spi_set_drvdata(spi, ssm2602); - ssm2602->control_type = SND_SOC_SPI; ssm2602->type = SSM2602; + ssm2602->regmap = devm_regmap_init_spi(spi, &ssm2602_regmap_config); + if (IS_ERR(ssm2602->regmap)) + return PTR_ERR(ssm2602->regmap); + ret = snd_soc_register_codec(&spi->dev, &soc_codec_dev_ssm2602, &ssm2602_dai, 1); return ret; @@ -701,9 +748,12 @@ static int __devinit ssm2602_i2c_probe(struct i2c_client *i2c, return -ENOMEM; i2c_set_clientdata(i2c, ssm2602); - ssm2602->control_type = SND_SOC_I2C; ssm2602->type = id->driver_data; + ssm2602->regmap = devm_regmap_init_i2c(i2c, &ssm2602_regmap_config); + if (IS_ERR(ssm2602->regmap)) + return PTR_ERR(ssm2602->regmap); + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_ssm2602, &ssm2602_dai, 1); return ret; diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index 7db6fa515028..8d717f4b5a87 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -609,8 +609,7 @@ static int sta32x_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); unsigned int rate; int i, mcs = -1, ir = -1; diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index df1e07ffac32..31762ebdd774 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -34,8 +34,6 @@ #include "tlv320aic23.h" -#define AIC23_VERSION "0.1" - /* * AIC23 register cache */ @@ -325,8 +323,7 @@ static int tlv320aic23_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; u16 iface_reg; int ret; struct aic23 *aic23 = snd_soc_codec_get_drvdata(codec); @@ -371,8 +368,7 @@ static int tlv320aic23_hw_params(struct snd_pcm_substream *substream, static int tlv320aic23_pcm_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; /* set active */ snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x0001); @@ -383,8 +379,7 @@ static int tlv320aic23_pcm_prepare(struct snd_pcm_substream *substream, static void tlv320aic23_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct aic23 *aic23 = snd_soc_codec_get_drvdata(codec); /* deactivate */ @@ -548,8 +543,6 @@ static int tlv320aic23_probe(struct snd_soc_codec *codec) struct aic23 *aic23 = snd_soc_codec_get_drvdata(codec); int ret; - printk(KERN_INFO "AIC23 Audio Codec %s\n", AIC23_VERSION); - ret = snd_soc_codec_set_cache_io(codec, 7, 9, aic23->control_type); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index 802064b5030d..85944e953578 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -126,8 +126,7 @@ static int aic26_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct aic26 *aic26 = snd_soc_codec_get_drvdata(codec); int fsref, divisor, wlen, pval, jval, dval, qval; u16 reg; diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 8d20f6ec20f3..64d2a4fa34b2 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -802,8 +802,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec =rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); int codec_clk = 0, bypass_pll = 0, fsref, last_clk = 0; u8 data, j, r, p, pll_q, pll_p = 1, pll_r = 1, pll_j = 1; @@ -1161,24 +1160,6 @@ static int aic3x_set_bias_level(struct snd_soc_codec *codec, return 0; } -void aic3x_set_headset_detection(struct snd_soc_codec *codec, int detect, - int headset_debounce, int button_debounce) -{ - u8 val; - - val = ((detect & AIC3X_HEADSET_DETECT_MASK) - << AIC3X_HEADSET_DETECT_SHIFT) | - ((headset_debounce & AIC3X_HEADSET_DEBOUNCE_MASK) - << AIC3X_HEADSET_DEBOUNCE_SHIFT) | - ((button_debounce & AIC3X_BUTTON_DEBOUNCE_MASK) - << AIC3X_BUTTON_DEBOUNCE_SHIFT); - - if (detect & AIC3X_HEADSET_DETECT_MASK) - val |= AIC3X_HEADSET_DETECT_ENABLED; - - snd_soc_write(codec, AIC3X_HEADSET_DETECT_CTRL_A, val); -} - #define AIC3X_RATES SNDRV_PCM_RATE_8000_96000 #define AIC3X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 4587ddd0fbf8..0dd41077ab79 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -62,8 +62,10 @@ #define UTHR_FROM_PERIOD_SIZE(samples, playrate, burstrate) \ (((samples)*5000) / (((burstrate)*5000) / ((burstrate) - (playrate)))) -static void dac33_calculate_times(struct snd_pcm_substream *substream); -static int dac33_prepare_chip(struct snd_pcm_substream *substream); +static void dac33_calculate_times(struct snd_pcm_substream *substream, + struct snd_soc_codec *codec); +static int dac33_prepare_chip(struct snd_pcm_substream *substream, + struct snd_soc_codec *codec); enum dac33_state { DAC33_IDLE = 0, @@ -427,8 +429,8 @@ static int dac33_playback_event(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_PRE_PMU: if (likely(dac33->substream)) { - dac33_calculate_times(dac33->substream); - dac33_prepare_chip(dac33->substream); + dac33_calculate_times(dac33->substream, w->codec); + dac33_prepare_chip(dac33->substream, w->codec); } break; case SND_SOC_DAPM_POST_PMD: @@ -799,8 +801,7 @@ static void dac33_oscwait(struct snd_soc_codec *codec) static int dac33_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); /* Stream started, save the substream pointer */ @@ -812,8 +813,7 @@ static int dac33_startup(struct snd_pcm_substream *substream, static void dac33_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); dac33->substream = NULL; @@ -825,8 +825,7 @@ static int dac33_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); /* Check parameters for validity */ @@ -868,10 +867,9 @@ static int dac33_hw_params(struct snd_pcm_substream *substream, * writes happens in different order, than dac33 might end up in unknown state. * Use the known, working sequence of register writes to initialize the dac33. */ -static int dac33_prepare_chip(struct snd_pcm_substream *substream) +static int dac33_prepare_chip(struct snd_pcm_substream *substream, + struct snd_soc_codec *codec) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); unsigned int oscset, ratioset, pwr_ctrl, reg_tmp; u8 aictrl_a, aictrl_b, fifoctrl_a; @@ -1067,10 +1065,9 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream) return 0; } -static void dac33_calculate_times(struct snd_pcm_substream *substream) +static void dac33_calculate_times(struct snd_pcm_substream *substream, + struct snd_soc_codec *codec) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); unsigned int period_size = substream->runtime->period_size; unsigned int rate = substream->runtime->rate; @@ -1128,8 +1125,7 @@ static void dac33_calculate_times(struct snd_pcm_substream *substream) static int dac33_pcm_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); int ret = 0; @@ -1161,8 +1157,7 @@ static snd_pcm_sframes_t dac33_dai_delay( struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); unsigned long long t0, t1, t_now; unsigned int time_delta, uthr; diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 170cf9a8fc79..391fcfc7b63b 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1685,8 +1685,7 @@ static void twl4030_tdm_enable(struct snd_soc_codec *codec, int direction, static int twl4030_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); if (twl4030->master_substream) { @@ -1715,8 +1714,7 @@ static int twl4030_startup(struct snd_pcm_substream *substream, static void twl4030_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); if (twl4030->master_substream == substream) @@ -1740,8 +1738,7 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); u8 mode, old_mode, format, old_format; @@ -1974,8 +1971,7 @@ static void twl4030_voice_enable(struct snd_soc_codec *codec, int direction, static int twl4030_voice_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); u8 mode; @@ -2007,8 +2003,7 @@ static int twl4030_voice_startup(struct snd_pcm_substream *substream, static void twl4030_voice_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; /* Enable voice digital filters */ twl4030_voice_enable(codec, substream->stream, 0); @@ -2017,8 +2012,7 @@ static void twl4030_voice_shutdown(struct snd_pcm_substream *substream, static int twl4030_voice_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); u8 old_mode, mode; diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index dc7509b9d53a..a36e9fcdf184 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -46,17 +46,6 @@ #define TWL6040_OUTHF_0dB 0x03 #define TWL6040_OUTHF_M52dB 0x1D -#define TWL6040_RAMP_NONE 0 -#define TWL6040_RAMP_UP 1 -#define TWL6040_RAMP_DOWN 2 - -#define TWL6040_HSL_VOL_MASK 0x0F -#define TWL6040_HSL_VOL_SHIFT 0 -#define TWL6040_HSR_VOL_MASK 0xF0 -#define TWL6040_HSR_VOL_SHIFT 4 -#define TWL6040_HF_VOL_MASK 0x1F -#define TWL6040_HF_VOL_SHIFT 0 - /* Shadow register used by the driver */ #define TWL6040_REG_SW_SHADOW 0x2F #define TWL6040_CACHEREGNUM (TWL6040_REG_SW_SHADOW + 1) @@ -64,18 +53,6 @@ /* TWL6040_REG_SW_SHADOW (0x2F) fields */ #define TWL6040_EAR_PATH_ENABLE 0x01 -struct twl6040_output { - u16 active; - u16 left_vol; - u16 right_vol; - u16 left_step; - u16 right_step; - unsigned int step_delay; - u16 ramp; - struct delayed_work work; - struct completion ramp_done; -}; - struct twl6040_jack_data { struct snd_soc_jack *jack; struct delayed_work work; @@ -100,8 +77,6 @@ struct twl6040_data { struct snd_soc_codec *codec; struct workqueue_struct *workqueue; struct mutex mutex; - struct twl6040_output headset; - struct twl6040_output handsfree; }; /* @@ -311,318 +286,6 @@ static void twl6040_restore_regs(struct snd_soc_codec *codec) } } -/* - * Ramp HS PGA volume to minimise pops at stream startup and shutdown. - */ -static inline int twl6040_hs_ramp_step(struct snd_soc_codec *codec, - unsigned int left_step, unsigned int right_step) -{ - - struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); - struct twl6040_output *headset = &priv->headset; - int left_complete = 0, right_complete = 0; - u8 reg, val; - - /* left channel */ - left_step = (left_step > 0xF) ? 0xF : left_step; - reg = twl6040_read_reg_cache(codec, TWL6040_REG_HSGAIN); - val = (~reg & TWL6040_HSL_VOL_MASK); - - if (headset->ramp == TWL6040_RAMP_UP) { - /* ramp step up */ - if (val < headset->left_vol) { - if (val + left_step > headset->left_vol) - val = headset->left_vol; - else - val += left_step; - - reg &= ~TWL6040_HSL_VOL_MASK; - twl6040_write(codec, TWL6040_REG_HSGAIN, - (reg | (~val & TWL6040_HSL_VOL_MASK))); - } else { - left_complete = 1; - } - } else if (headset->ramp == TWL6040_RAMP_DOWN) { - /* ramp step down */ - if (val > 0x0) { - if ((int)val - (int)left_step < 0) - val = 0; - else - val -= left_step; - - reg &= ~TWL6040_HSL_VOL_MASK; - twl6040_write(codec, TWL6040_REG_HSGAIN, reg | - (~val & TWL6040_HSL_VOL_MASK)); - } else { - left_complete = 1; - } - } - - /* right channel */ - right_step = (right_step > 0xF) ? 0xF : right_step; - reg = twl6040_read_reg_cache(codec, TWL6040_REG_HSGAIN); - val = (~reg & TWL6040_HSR_VOL_MASK) >> TWL6040_HSR_VOL_SHIFT; - - if (headset->ramp == TWL6040_RAMP_UP) { - /* ramp step up */ - if (val < headset->right_vol) { - if (val + right_step > headset->right_vol) - val = headset->right_vol; - else - val += right_step; - - reg &= ~TWL6040_HSR_VOL_MASK; - twl6040_write(codec, TWL6040_REG_HSGAIN, - (reg | (~val << TWL6040_HSR_VOL_SHIFT))); - } else { - right_complete = 1; - } - } else if (headset->ramp == TWL6040_RAMP_DOWN) { - /* ramp step down */ - if (val > 0x0) { - if ((int)val - (int)right_step < 0) - val = 0; - else - val -= right_step; - - reg &= ~TWL6040_HSR_VOL_MASK; - twl6040_write(codec, TWL6040_REG_HSGAIN, - reg | (~val << TWL6040_HSR_VOL_SHIFT)); - } else { - right_complete = 1; - } - } - - return left_complete & right_complete; -} - -/* - * Ramp HF PGA volume to minimise pops at stream startup and shutdown. - */ -static inline int twl6040_hf_ramp_step(struct snd_soc_codec *codec, - unsigned int left_step, unsigned int right_step) -{ - struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); - struct twl6040_output *handsfree = &priv->handsfree; - int left_complete = 0, right_complete = 0; - u16 reg, val; - - /* left channel */ - left_step = (left_step > 0x1D) ? 0x1D : left_step; - reg = twl6040_read_reg_cache(codec, TWL6040_REG_HFLGAIN); - reg = 0x1D - reg; - val = (reg & TWL6040_HF_VOL_MASK); - if (handsfree->ramp == TWL6040_RAMP_UP) { - /* ramp step up */ - if (val < handsfree->left_vol) { - if (val + left_step > handsfree->left_vol) - val = handsfree->left_vol; - else - val += left_step; - - reg &= ~TWL6040_HF_VOL_MASK; - twl6040_write(codec, TWL6040_REG_HFLGAIN, - reg | (0x1D - val)); - } else { - left_complete = 1; - } - } else if (handsfree->ramp == TWL6040_RAMP_DOWN) { - /* ramp step down */ - if (val > 0) { - if ((int)val - (int)left_step < 0) - val = 0; - else - val -= left_step; - - reg &= ~TWL6040_HF_VOL_MASK; - twl6040_write(codec, TWL6040_REG_HFLGAIN, - reg | (0x1D - val)); - } else { - left_complete = 1; - } - } - - /* right channel */ - right_step = (right_step > 0x1D) ? 0x1D : right_step; - reg = twl6040_read_reg_cache(codec, TWL6040_REG_HFRGAIN); - reg = 0x1D - reg; - val = (reg & TWL6040_HF_VOL_MASK); - if (handsfree->ramp == TWL6040_RAMP_UP) { - /* ramp step up */ - if (val < handsfree->right_vol) { - if (val + right_step > handsfree->right_vol) - val = handsfree->right_vol; - else - val += right_step; - - reg &= ~TWL6040_HF_VOL_MASK; - twl6040_write(codec, TWL6040_REG_HFRGAIN, - reg | (0x1D - val)); - } else { - right_complete = 1; - } - } else if (handsfree->ramp == TWL6040_RAMP_DOWN) { - /* ramp step down */ - if (val > 0) { - if ((int)val - (int)right_step < 0) - val = 0; - else - val -= right_step; - - reg &= ~TWL6040_HF_VOL_MASK; - twl6040_write(codec, TWL6040_REG_HFRGAIN, - reg | (0x1D - val)); - } - } - - return left_complete & right_complete; -} - -/* - * This work ramps both output PGAs at stream start/stop time to - * minimise pop associated with DAPM power switching. - */ -static void twl6040_pga_hs_work(struct work_struct *work) -{ - struct twl6040_data *priv = - container_of(work, struct twl6040_data, headset.work.work); - struct snd_soc_codec *codec = priv->codec; - struct twl6040_output *headset = &priv->headset; - int i, headset_complete; - - /* do we need to ramp at all ? */ - if (headset->ramp == TWL6040_RAMP_NONE) - return; - - /* HS PGA gain range: 0x0 - 0xf (0 - 15) */ - for (i = 0; i < 16; i++) { - headset_complete = twl6040_hs_ramp_step(codec, - headset->left_step, - headset->right_step); - - /* ramp finished ? */ - if (headset_complete) - break; - - schedule_timeout_interruptible( - msecs_to_jiffies(headset->step_delay)); - } - - if (headset->ramp == TWL6040_RAMP_DOWN) { - headset->active = 0; - complete(&headset->ramp_done); - } else { - headset->active = 1; - } - headset->ramp = TWL6040_RAMP_NONE; -} - -static void twl6040_pga_hf_work(struct work_struct *work) -{ - struct twl6040_data *priv = - container_of(work, struct twl6040_data, handsfree.work.work); - struct snd_soc_codec *codec = priv->codec; - struct twl6040_output *handsfree = &priv->handsfree; - int i, handsfree_complete; - - /* do we need to ramp at all ? */ - if (handsfree->ramp == TWL6040_RAMP_NONE) - return; - - /* - * HF PGA gain range: 0x00 - 0x1d (0 - 29) */ - for (i = 0; i < 30; i++) { - handsfree_complete = twl6040_hf_ramp_step(codec, - handsfree->left_step, - handsfree->right_step); - - /* ramp finished ? */ - if (handsfree_complete) - break; - - schedule_timeout_interruptible( - msecs_to_jiffies(handsfree->step_delay)); - } - - - if (handsfree->ramp == TWL6040_RAMP_DOWN) { - handsfree->active = 0; - complete(&handsfree->ramp_done); - } else - handsfree->active = 1; - handsfree->ramp = TWL6040_RAMP_NONE; -} - -static int out_drv_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) -{ - struct snd_soc_codec *codec = w->codec; - struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); - struct twl6040_output *out; - struct delayed_work *work; - - switch (w->shift) { - case 2: /* Headset output driver */ - out = &priv->headset; - work = &out->work; - /* - * Make sure, that we do not mess up variables for already - * executing work. - */ - cancel_delayed_work_sync(work); - - out->left_step = priv->hs_left_step; - out->right_step = priv->hs_right_step; - out->step_delay = 5; /* 5 ms between volume ramp steps */ - break; - case 4: /* Handsfree output driver */ - out = &priv->handsfree; - work = &out->work; - /* - * Make sure, that we do not mess up variables for already - * executing work. - */ - cancel_delayed_work_sync(work); - - out->left_step = priv->hf_left_step; - out->right_step = priv->hf_right_step; - out->step_delay = 5; /* 5 ms between volume ramp steps */ - break; - default: - return -1; - } - - switch (event) { - case SND_SOC_DAPM_POST_PMU: - if (out->active) - break; - - /* don't use volume ramp for power-up */ - out->ramp = TWL6040_RAMP_UP; - out->left_step = out->left_vol; - out->right_step = out->right_vol; - - queue_delayed_work(priv->workqueue, work, msecs_to_jiffies(1)); - break; - - case SND_SOC_DAPM_PRE_PMD: - if (!out->active) - break; - - /* use volume ramp for power-down */ - out->ramp = TWL6040_RAMP_DOWN; - INIT_COMPLETION(out->ramp_done); - - queue_delayed_work(priv->workqueue, work, msecs_to_jiffies(1)); - - wait_for_completion_timeout(&out->ramp_done, - msecs_to_jiffies(2000)); - break; - } - - return 0; -} - /* set headset dac and driver power mode */ static int headset_power_mode(struct snd_soc_codec *codec, int high_perf) { @@ -747,71 +410,6 @@ static irqreturn_t twl6040_audio_handler(int irq, void *data) return IRQ_HANDLED; } -static int twl6040_put_volsw(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - struct twl6040_data *twl6040_priv = snd_soc_codec_get_drvdata(codec); - struct twl6040_output *out = NULL; - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - int ret; - - /* For HS and HF we shadow the values and only actually write - * them out when active in order to ensure the amplifier comes on - * as quietly as possible. */ - switch (mc->reg) { - case TWL6040_REG_HSGAIN: - out = &twl6040_priv->headset; - break; - case TWL6040_REG_HFLGAIN: - out = &twl6040_priv->handsfree; - break; - default: - dev_warn(codec->dev, "%s: Unexpected register: 0x%02x\n", - __func__, mc->reg); - return -EINVAL; - } - - out->left_vol = ucontrol->value.integer.value[0]; - out->right_vol = ucontrol->value.integer.value[1]; - if (!out->active) - return 1; - - ret = snd_soc_put_volsw(kcontrol, ucontrol); - if (ret < 0) - return ret; - - return 1; -} - -static int twl6040_get_volsw(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - struct twl6040_data *twl6040_priv = snd_soc_codec_get_drvdata(codec); - struct twl6040_output *out = &twl6040_priv->headset; - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - - switch (mc->reg) { - case TWL6040_REG_HSGAIN: - out = &twl6040_priv->headset; - break; - case TWL6040_REG_HFLGAIN: - out = &twl6040_priv->handsfree; - break; - default: - dev_warn(codec->dev, "%s: Unexpected register: 0x%02x\n", - __func__, mc->reg); - return -EINVAL; - } - - ucontrol->value.integer.value[0] = out->left_vol; - ucontrol->value.integer.value[1] = out->right_vol; - return 0; -} - static int twl6040_soc_dapm_put_vibra_enum(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -1076,12 +674,10 @@ static const struct snd_kcontrol_new twl6040_snd_controls[] = { TWL6040_REG_LINEGAIN, 0, 3, 7, 0, afm_amp_tlv), /* Playback gains */ - SOC_DOUBLE_EXT_TLV("Headset Playback Volume", - TWL6040_REG_HSGAIN, 0, 4, 0xF, 1, twl6040_get_volsw, - twl6040_put_volsw, hs_tlv), - SOC_DOUBLE_R_EXT_TLV("Handsfree Playback Volume", - TWL6040_REG_HFLGAIN, TWL6040_REG_HFRGAIN, 0, 0x1D, 1, - twl6040_get_volsw, twl6040_put_volsw, hf_tlv), + SOC_DOUBLE_TLV("Headset Playback Volume", + TWL6040_REG_HSGAIN, 0, 4, 0xF, 1, hs_tlv), + SOC_DOUBLE_R_TLV("Handsfree Playback Volume", + TWL6040_REG_HFLGAIN, TWL6040_REG_HFRGAIN, 0, 0x1D, 1, hf_tlv), SOC_SINGLE_TLV("Earphone Playback Volume", TWL6040_REG_EARCTL, 1, 0xF, 1, ep_tlv), @@ -1180,22 +776,14 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { &auxr_switch_control), /* Analog playback drivers */ - SND_SOC_DAPM_OUT_DRV_E("HF Left Driver", - TWL6040_REG_HFLCTL, 4, 0, NULL, 0, - out_drv_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), - SND_SOC_DAPM_OUT_DRV_E("HF Right Driver", - TWL6040_REG_HFRCTL, 4, 0, NULL, 0, - out_drv_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), - SND_SOC_DAPM_OUT_DRV_E("HS Left Driver", - TWL6040_REG_HSLCTL, 2, 0, NULL, 0, - out_drv_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), - SND_SOC_DAPM_OUT_DRV_E("HS Right Driver", - TWL6040_REG_HSRCTL, 2, 0, NULL, 0, - out_drv_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_OUT_DRV("HF Left Driver", + TWL6040_REG_HFLCTL, 4, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("HF Right Driver", + TWL6040_REG_HFRCTL, 4, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("HS Left Driver", + TWL6040_REG_HSLCTL, 2, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("HS Right Driver", + TWL6040_REG_HSRCTL, 2, 0, NULL, 0), SND_SOC_DAPM_OUT_DRV_E("Earphone Driver", TWL6040_REG_EARCTL, 0, 0, NULL, 0, twl6040_ep_drv_event, @@ -1339,8 +927,7 @@ static int twl6040_set_bias_level(struct snd_soc_codec *codec, static int twl6040_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); snd_pcm_hw_constraint_list(substream->runtime, 0, @@ -1354,8 +941,7 @@ static int twl6040_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); int rate; @@ -1391,8 +977,7 @@ static int twl6040_hw_params(struct snd_pcm_substream *substream, static int twl6040_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct twl6040 *twl6040 = codec->control_data; struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); int ret; @@ -1570,14 +1155,9 @@ static int twl6040_probe(struct snd_soc_codec *codec) } INIT_DELAYED_WORK(&priv->hs_jack.work, twl6040_accessory_work); - INIT_DELAYED_WORK(&priv->headset.work, twl6040_pga_hs_work); - INIT_DELAYED_WORK(&priv->handsfree.work, twl6040_pga_hf_work); mutex_init(&priv->mutex); - init_completion(&priv->headset.ramp_done); - init_completion(&priv->handsfree.ramp_done); - ret = request_threaded_irq(priv->plug_irq, NULL, twl6040_audio_handler, 0, "twl6040_irq_plug", codec); if (ret) { diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index 797b0dde2c68..6c3d43b8ee85 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -159,8 +159,7 @@ static int uda134x_mute(struct snd_soc_dai *dai, int mute) static int uda134x_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec =rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct uda134x_priv *uda134x = snd_soc_codec_get_drvdata(codec); struct snd_pcm_runtime *master_runtime; @@ -191,8 +190,7 @@ static int uda134x_startup(struct snd_pcm_substream *substream, static void uda134x_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct uda134x_priv *uda134x = snd_soc_codec_get_drvdata(codec); if (uda134x->master_substream == substream) diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index 4f1b23d7e404..2502214b84ab 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -502,8 +502,7 @@ static int uda1380_set_dai_fmt_capture(struct snd_soc_dai *codec_dai, static int uda1380_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct uda1380_priv *uda1380 = snd_soc_codec_get_drvdata(codec); int mixer = uda1380_read_reg_cache(codec, UDA1380_MIXER); @@ -528,8 +527,7 @@ static int uda1380_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK); /* set WSPLL power and divider if running from this clock */ diff --git a/sound/soc/codecs/wl1273.c b/sound/soc/codecs/wl1273.c index 3d868dc40092..7b24d6d192e1 100644 --- a/sound/soc/codecs/wl1273.c +++ b/sound/soc/codecs/wl1273.c @@ -293,8 +293,7 @@ static const struct snd_kcontrol_new wl1273_controls[] = { static int wl1273_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(codec); switch (wl1273->mode) { @@ -329,8 +328,7 @@ static int wl1273_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(rtd->codec); + struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(dai->codec); struct wl1273_core *core = wl1273->core; unsigned int rate, width, r; diff --git a/sound/soc/codecs/wm1250-ev1.c b/sound/soc/codecs/wm1250-ev1.c index aefb4f89be0e..e0b51e9f8b12 100644 --- a/sound/soc/codecs/wm1250-ev1.c +++ b/sound/soc/codecs/wm1250-ev1.c @@ -79,22 +79,65 @@ static const struct snd_soc_dapm_route wm1250_ev1_dapm_routes[] = { { "WM1250 Output", NULL, "DAC" }, }; +static int wm1250_ev1_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct wm1250_priv *wm1250 = snd_soc_codec_get_drvdata(dai->codec); + + switch (params_rate(params)) { + case 8000: + gpio_set_value(wm1250->gpios[WM1250_EV1_GPIO_CLK_SEL0].gpio, + 1); + gpio_set_value(wm1250->gpios[WM1250_EV1_GPIO_CLK_SEL1].gpio, + 1); + break; + case 16000: + gpio_set_value(wm1250->gpios[WM1250_EV1_GPIO_CLK_SEL0].gpio, + 0); + gpio_set_value(wm1250->gpios[WM1250_EV1_GPIO_CLK_SEL1].gpio, + 1); + break; + case 32000: + gpio_set_value(wm1250->gpios[WM1250_EV1_GPIO_CLK_SEL0].gpio, + 1); + gpio_set_value(wm1250->gpios[WM1250_EV1_GPIO_CLK_SEL1].gpio, + 0); + break; + case 64000: + gpio_set_value(wm1250->gpios[WM1250_EV1_GPIO_CLK_SEL0].gpio, + 0); + gpio_set_value(wm1250->gpios[WM1250_EV1_GPIO_CLK_SEL1].gpio, + 0); + break; + default: + return -EINVAL; + } + + return 0; +} + +static const struct snd_soc_dai_ops wm1250_ev1_ops = { + .hw_params = wm1250_ev1_hw_params, +}; + static struct snd_soc_dai_driver wm1250_ev1_dai = { .name = "wm1250-ev1", .playback = { .stream_name = "Playback", .channels_min = 1, - .channels_max = 1, + .channels_max = 2, .rates = SNDRV_PCM_RATE_8000, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .capture = { .stream_name = "Capture", .channels_min = 1, - .channels_max = 1, + .channels_max = 2, .rates = SNDRV_PCM_RATE_8000, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, + .ops = &wm1250_ev1_ops, }; static struct snd_soc_codec_driver soc_codec_dev_wm1250_ev1 = { @@ -215,23 +258,7 @@ static struct i2c_driver wm1250_ev1_i2c_driver = { .id_table = wm1250_ev1_i2c_id, }; -static int __init wm1250_ev1_modinit(void) -{ - int ret = 0; - - ret = i2c_add_driver(&wm1250_ev1_i2c_driver); - if (ret != 0) - pr_err("Failed to register WM1250-EV1 I2C driver: %d\n", ret); - - return ret; -} -module_init(wm1250_ev1_modinit); - -static void __exit wm1250_ev1_exit(void) -{ - i2c_del_driver(&wm1250_ev1_i2c_driver); -} -module_exit(wm1250_ev1_exit); +module_i2c_driver(wm1250_ev1_i2c_driver); MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>"); MODULE_DESCRIPTION("WM1250-EV1 audio I/O module driver"); diff --git a/sound/soc/codecs/wm5100-tables.c b/sound/soc/codecs/wm5100-tables.c index 9a18fae68204..e167207a19cc 100644 --- a/sound/soc/codecs/wm5100-tables.c +++ b/sound/soc/codecs/wm5100-tables.c @@ -32,7 +32,18 @@ bool wm5100_volatile_register(struct device *dev, unsigned int reg) case WM5100_MIC_DETECT_3: return 1; default: - return 0; + if ((reg >= WM5100_DSP1_PM_0 && reg <= WM5100_DSP1_PM_1535) || + (reg >= WM5100_DSP1_ZM_0 && reg <= WM5100_DSP1_ZM_2047) || + (reg >= WM5100_DSP1_DM_0 && reg <= WM5100_DSP1_DM_511) || + (reg >= WM5100_DSP2_PM_0 && reg <= WM5100_DSP2_PM_1535) || + (reg >= WM5100_DSP2_ZM_0 && reg <= WM5100_DSP2_ZM_2047) || + (reg >= WM5100_DSP2_DM_0 && reg <= WM5100_DSP2_DM_511) || + (reg >= WM5100_DSP3_PM_0 && reg <= WM5100_DSP3_PM_1535) || + (reg >= WM5100_DSP3_ZM_0 && reg <= WM5100_DSP3_ZM_2047) || + (reg >= WM5100_DSP3_DM_0 && reg <= WM5100_DSP3_DM_511)) + return 1; + else + return 0; } } @@ -697,9 +708,110 @@ bool wm5100_readable_register(struct device *dev, unsigned int reg) case WM5100_HPLPF3_2: case WM5100_HPLPF4_1: case WM5100_HPLPF4_2: + case WM5100_DSP1_CONTROL_1: + case WM5100_DSP1_CONTROL_2: + case WM5100_DSP1_CONTROL_3: + case WM5100_DSP1_CONTROL_4: + case WM5100_DSP1_CONTROL_5: + case WM5100_DSP1_CONTROL_6: + case WM5100_DSP1_CONTROL_7: + case WM5100_DSP1_CONTROL_8: + case WM5100_DSP1_CONTROL_9: + case WM5100_DSP1_CONTROL_10: + case WM5100_DSP1_CONTROL_11: + case WM5100_DSP1_CONTROL_12: + case WM5100_DSP1_CONTROL_13: + case WM5100_DSP1_CONTROL_14: + case WM5100_DSP1_CONTROL_15: + case WM5100_DSP1_CONTROL_16: + case WM5100_DSP1_CONTROL_17: + case WM5100_DSP1_CONTROL_18: + case WM5100_DSP1_CONTROL_19: + case WM5100_DSP1_CONTROL_20: + case WM5100_DSP1_CONTROL_21: + case WM5100_DSP1_CONTROL_22: + case WM5100_DSP1_CONTROL_23: + case WM5100_DSP1_CONTROL_24: + case WM5100_DSP1_CONTROL_25: + case WM5100_DSP1_CONTROL_26: + case WM5100_DSP1_CONTROL_27: + case WM5100_DSP1_CONTROL_28: + case WM5100_DSP1_CONTROL_29: + case WM5100_DSP1_CONTROL_30: + case WM5100_DSP2_CONTROL_1: + case WM5100_DSP2_CONTROL_2: + case WM5100_DSP2_CONTROL_3: + case WM5100_DSP2_CONTROL_4: + case WM5100_DSP2_CONTROL_5: + case WM5100_DSP2_CONTROL_6: + case WM5100_DSP2_CONTROL_7: + case WM5100_DSP2_CONTROL_8: + case WM5100_DSP2_CONTROL_9: + case WM5100_DSP2_CONTROL_10: + case WM5100_DSP2_CONTROL_11: + case WM5100_DSP2_CONTROL_12: + case WM5100_DSP2_CONTROL_13: + case WM5100_DSP2_CONTROL_14: + case WM5100_DSP2_CONTROL_15: + case WM5100_DSP2_CONTROL_16: + case WM5100_DSP2_CONTROL_17: + case WM5100_DSP2_CONTROL_18: + case WM5100_DSP2_CONTROL_19: + case WM5100_DSP2_CONTROL_20: + case WM5100_DSP2_CONTROL_21: + case WM5100_DSP2_CONTROL_22: + case WM5100_DSP2_CONTROL_23: + case WM5100_DSP2_CONTROL_24: + case WM5100_DSP2_CONTROL_25: + case WM5100_DSP2_CONTROL_26: + case WM5100_DSP2_CONTROL_27: + case WM5100_DSP2_CONTROL_28: + case WM5100_DSP2_CONTROL_29: + case WM5100_DSP2_CONTROL_30: + case WM5100_DSP3_CONTROL_1: + case WM5100_DSP3_CONTROL_2: + case WM5100_DSP3_CONTROL_3: + case WM5100_DSP3_CONTROL_4: + case WM5100_DSP3_CONTROL_5: + case WM5100_DSP3_CONTROL_6: + case WM5100_DSP3_CONTROL_7: + case WM5100_DSP3_CONTROL_8: + case WM5100_DSP3_CONTROL_9: + case WM5100_DSP3_CONTROL_10: + case WM5100_DSP3_CONTROL_11: + case WM5100_DSP3_CONTROL_12: + case WM5100_DSP3_CONTROL_13: + case WM5100_DSP3_CONTROL_14: + case WM5100_DSP3_CONTROL_15: + case WM5100_DSP3_CONTROL_16: + case WM5100_DSP3_CONTROL_17: + case WM5100_DSP3_CONTROL_18: + case WM5100_DSP3_CONTROL_19: + case WM5100_DSP3_CONTROL_20: + case WM5100_DSP3_CONTROL_21: + case WM5100_DSP3_CONTROL_22: + case WM5100_DSP3_CONTROL_23: + case WM5100_DSP3_CONTROL_24: + case WM5100_DSP3_CONTROL_25: + case WM5100_DSP3_CONTROL_26: + case WM5100_DSP3_CONTROL_27: + case WM5100_DSP3_CONTROL_28: + case WM5100_DSP3_CONTROL_29: + case WM5100_DSP3_CONTROL_30: return 1; default: - return 0; + if ((reg >= WM5100_DSP1_PM_0 && reg <= WM5100_DSP1_PM_1535) || + (reg >= WM5100_DSP1_ZM_0 && reg <= WM5100_DSP1_ZM_2047) || + (reg >= WM5100_DSP1_DM_0 && reg <= WM5100_DSP1_DM_511) || + (reg >= WM5100_DSP2_PM_0 && reg <= WM5100_DSP2_PM_1535) || + (reg >= WM5100_DSP2_ZM_0 && reg <= WM5100_DSP2_ZM_2047) || + (reg >= WM5100_DSP2_DM_0 && reg <= WM5100_DSP2_DM_511) || + (reg >= WM5100_DSP3_PM_0 && reg <= WM5100_DSP3_PM_1535) || + (reg >= WM5100_DSP3_ZM_0 && reg <= WM5100_DSP3_ZM_2047) || + (reg >= WM5100_DSP3_DM_0 && reg <= WM5100_DSP3_DM_511)) + return 1; + else + return 0; } } @@ -1361,4 +1473,13 @@ struct reg_default wm5100_reg_defaults[WM5100_REGISTER_COUNT] = { { 0x0EC9, 0x0000 }, /* R3785 - HPLPF3_2 */ { 0x0ECC, 0x0000 }, /* R3788 - HPLPF4_1 */ { 0x0ECD, 0x0000 }, /* R3789 - HPLPF4_2 */ + { 0x0F02, 0x0000 }, /* R3842 - DSP1 Control 2 */ + { 0x0F03, 0x0000 }, /* R3843 - DSP1 Control 3 */ + { 0x0F04, 0x0000 }, /* R3844 - DSP1 Control 4 */ + { 0x1002, 0x0000 }, /* R4098 - DSP2 Control 2 */ + { 0x1003, 0x0000 }, /* R4099 - DSP2 Control 3 */ + { 0x1004, 0x0000 }, /* R4100 - DSP2 Control 4 */ + { 0x1102, 0x0000 }, /* R4354 - DSP3 Control 2 */ + { 0x1103, 0x0000 }, /* R4355 - DSP3 Control 3 */ + { 0x1104, 0x0000 }, /* R4356 - DSP3 Control 4 */ }; diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index b9c185ce64e4..cb6d5372103a 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -1265,29 +1265,12 @@ static const __devinitdata struct reg_default wm5100_reva_patches[] = { { WM5100_AUDIO_IF_3_19, 1 }, }; -static int wm5100_dai_to_base(struct snd_soc_dai *dai) -{ - switch (dai->id) { - case 0: - return WM5100_AUDIO_IF_1_1 - 1; - case 1: - return WM5100_AUDIO_IF_2_1 - 1; - case 2: - return WM5100_AUDIO_IF_3_1 - 1; - default: - BUG(); - return -EINVAL; - } -} - static int wm5100_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct snd_soc_codec *codec = dai->codec; int lrclk, bclk, mask, base; - base = wm5100_dai_to_base(dai); - if (base < 0) - return base; + base = dai->driver->base; lrclk = 0; bclk = 0; @@ -1414,9 +1397,7 @@ static int wm5100_hw_params(struct snd_pcm_substream *substream, int i, base, bclk, aif_rate, lrclk, wl, fl, sr; int *bclk_rates; - base = wm5100_dai_to_base(dai); - if (base < 0) - return base; + base = dai->driver->base; /* Data sizes if not using TDM */ wl = snd_pcm_format_width(params_format(params)); @@ -1897,6 +1878,7 @@ static int wm5100_set_fll(struct snd_soc_codec *codec, int fll_id, int source, static struct snd_soc_dai_driver wm5100_dai[] = { { .name = "wm5100-aif1", + .base = WM5100_AUDIO_IF_1_1 - 1, .playback = { .stream_name = "AIF1 Playback", .channels_min = 2, @@ -1916,6 +1898,7 @@ static struct snd_soc_dai_driver wm5100_dai[] = { { .name = "wm5100-aif2", .id = 1, + .base = WM5100_AUDIO_IF_2_1 - 1, .playback = { .stream_name = "AIF2 Playback", .channels_min = 2, @@ -1935,6 +1918,7 @@ static struct snd_soc_dai_driver wm5100_dai[] = { { .name = "wm5100-aif3", .id = 2, + .base = WM5100_AUDIO_IF_3_1 - 1, .playback = { .stream_name = "AIF3 Playback", .channels_min = 2, @@ -2454,7 +2438,7 @@ static __devinit int wm5100_i2c_probe(struct i2c_client *i2c, wm5100->dev = &i2c->dev; - wm5100->regmap = regmap_init_i2c(i2c, &wm5100_regmap); + wm5100->regmap = devm_regmap_init_i2c(i2c, &wm5100_regmap); if (IS_ERR(wm5100->regmap)) { ret = PTR_ERR(wm5100->regmap); dev_err(&i2c->dev, "Failed to allocate register map: %d\n", @@ -2479,7 +2463,7 @@ static __devinit int wm5100_i2c_probe(struct i2c_client *i2c, if (ret != 0) { dev_err(&i2c->dev, "Failed to request core supplies: %d\n", ret); - goto err_regmap; + goto err; } ret = regulator_bulk_enable(ARRAY_SIZE(wm5100->core_supplies), @@ -2487,7 +2471,7 @@ static __devinit int wm5100_i2c_probe(struct i2c_client *i2c, if (ret != 0) { dev_err(&i2c->dev, "Failed to enable core supplies: %d\n", ret); - goto err_regmap; + goto err; } if (wm5100->pdata.ldo_ena) { @@ -2660,8 +2644,6 @@ err_ldo: err_enable: regulator_bulk_disable(ARRAY_SIZE(wm5100->core_supplies), wm5100->core_supplies); -err_regmap: - regmap_exit(wm5100->regmap); err: return ret; } @@ -2682,7 +2664,6 @@ static __devexit int wm5100_i2c_remove(struct i2c_client *i2c) gpio_set_value_cansleep(wm5100->pdata.ldo_ena, 0); gpio_free(wm5100->pdata.ldo_ena); } - regmap_exit(wm5100->regmap); return 0; } @@ -2749,17 +2730,7 @@ static struct i2c_driver wm5100_i2c_driver = { .id_table = wm5100_i2c_id, }; -static int __init wm5100_modinit(void) -{ - return i2c_add_driver(&wm5100_i2c_driver); -} -module_init(wm5100_modinit); - -static void __exit wm5100_exit(void) -{ - i2c_del_driver(&wm5100_i2c_driver); -} -module_exit(wm5100_exit); +module_i2c_driver(wm5100_i2c_driver); MODULE_DESCRIPTION("ASoC WM5100 driver"); MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>"); diff --git a/sound/soc/codecs/wm5100.h b/sound/soc/codecs/wm5100.h index 25cb6016f9d7..935a9b7fb274 100644 --- a/sound/soc/codecs/wm5100.h +++ b/sound/soc/codecs/wm5100.h @@ -709,6 +709,96 @@ int wm5100_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack); #define WM5100_HPLPF3_2 0xEC9 #define WM5100_HPLPF4_1 0xECC #define WM5100_HPLPF4_2 0xECD +#define WM5100_DSP1_CONTROL_1 0xF00 +#define WM5100_DSP1_CONTROL_2 0xF02 +#define WM5100_DSP1_CONTROL_3 0xF03 +#define WM5100_DSP1_CONTROL_4 0xF04 +#define WM5100_DSP1_CONTROL_5 0xF06 +#define WM5100_DSP1_CONTROL_6 0xF07 +#define WM5100_DSP1_CONTROL_7 0xF08 +#define WM5100_DSP1_CONTROL_8 0xF09 +#define WM5100_DSP1_CONTROL_9 0xF0A +#define WM5100_DSP1_CONTROL_10 0xF0B +#define WM5100_DSP1_CONTROL_11 0xF0C +#define WM5100_DSP1_CONTROL_12 0xF0D +#define WM5100_DSP1_CONTROL_13 0xF0F +#define WM5100_DSP1_CONTROL_14 0xF10 +#define WM5100_DSP1_CONTROL_15 0xF11 +#define WM5100_DSP1_CONTROL_16 0xF12 +#define WM5100_DSP1_CONTROL_17 0xF13 +#define WM5100_DSP1_CONTROL_18 0xF14 +#define WM5100_DSP1_CONTROL_19 0xF16 +#define WM5100_DSP1_CONTROL_20 0xF17 +#define WM5100_DSP1_CONTROL_21 0xF18 +#define WM5100_DSP1_CONTROL_22 0xF1A +#define WM5100_DSP1_CONTROL_23 0xF1B +#define WM5100_DSP1_CONTROL_24 0xF1C +#define WM5100_DSP1_CONTROL_25 0xF1E +#define WM5100_DSP1_CONTROL_26 0xF20 +#define WM5100_DSP1_CONTROL_27 0xF21 +#define WM5100_DSP1_CONTROL_28 0xF22 +#define WM5100_DSP1_CONTROL_29 0xF23 +#define WM5100_DSP1_CONTROL_30 0xF24 +#define WM5100_DSP2_CONTROL_1 0x1000 +#define WM5100_DSP2_CONTROL_2 0x1002 +#define WM5100_DSP2_CONTROL_3 0x1003 +#define WM5100_DSP2_CONTROL_4 0x1004 +#define WM5100_DSP2_CONTROL_5 0x1006 +#define WM5100_DSP2_CONTROL_6 0x1007 +#define WM5100_DSP2_CONTROL_7 0x1008 +#define WM5100_DSP2_CONTROL_8 0x1009 +#define WM5100_DSP2_CONTROL_9 0x100A +#define WM5100_DSP2_CONTROL_10 0x100B +#define WM5100_DSP2_CONTROL_11 0x100C +#define WM5100_DSP2_CONTROL_12 0x100D +#define WM5100_DSP2_CONTROL_13 0x100F +#define WM5100_DSP2_CONTROL_14 0x1010 +#define WM5100_DSP2_CONTROL_15 0x1011 +#define WM5100_DSP2_CONTROL_16 0x1012 +#define WM5100_DSP2_CONTROL_17 0x1013 +#define WM5100_DSP2_CONTROL_18 0x1014 +#define WM5100_DSP2_CONTROL_19 0x1016 +#define WM5100_DSP2_CONTROL_20 0x1017 +#define WM5100_DSP2_CONTROL_21 0x1018 +#define WM5100_DSP2_CONTROL_22 0x101A +#define WM5100_DSP2_CONTROL_23 0x101B +#define WM5100_DSP2_CONTROL_24 0x101C +#define WM5100_DSP2_CONTROL_25 0x101E +#define WM5100_DSP2_CONTROL_26 0x1020 +#define WM5100_DSP2_CONTROL_27 0x1021 +#define WM5100_DSP2_CONTROL_28 0x1022 +#define WM5100_DSP2_CONTROL_29 0x1023 +#define WM5100_DSP2_CONTROL_30 0x1024 +#define WM5100_DSP3_CONTROL_1 0x1100 +#define WM5100_DSP3_CONTROL_2 0x1102 +#define WM5100_DSP3_CONTROL_3 0x1103 +#define WM5100_DSP3_CONTROL_4 0x1104 +#define WM5100_DSP3_CONTROL_5 0x1106 +#define WM5100_DSP3_CONTROL_6 0x1107 +#define WM5100_DSP3_CONTROL_7 0x1108 +#define WM5100_DSP3_CONTROL_8 0x1109 +#define WM5100_DSP3_CONTROL_9 0x110A +#define WM5100_DSP3_CONTROL_10 0x110B +#define WM5100_DSP3_CONTROL_11 0x110C +#define WM5100_DSP3_CONTROL_12 0x110D +#define WM5100_DSP3_CONTROL_13 0x110F +#define WM5100_DSP3_CONTROL_14 0x1110 +#define WM5100_DSP3_CONTROL_15 0x1111 +#define WM5100_DSP3_CONTROL_16 0x1112 +#define WM5100_DSP3_CONTROL_17 0x1113 +#define WM5100_DSP3_CONTROL_18 0x1114 +#define WM5100_DSP3_CONTROL_19 0x1116 +#define WM5100_DSP3_CONTROL_20 0x1117 +#define WM5100_DSP3_CONTROL_21 0x1118 +#define WM5100_DSP3_CONTROL_22 0x111A +#define WM5100_DSP3_CONTROL_23 0x111B +#define WM5100_DSP3_CONTROL_24 0x111C +#define WM5100_DSP3_CONTROL_25 0x111E +#define WM5100_DSP3_CONTROL_26 0x1120 +#define WM5100_DSP3_CONTROL_27 0x1121 +#define WM5100_DSP3_CONTROL_28 0x1122 +#define WM5100_DSP3_CONTROL_29 0x1123 +#define WM5100_DSP3_CONTROL_30 0x1124 #define WM5100_DSP1_DM_0 0x4000 #define WM5100_DSP1_DM_1 0x4001 #define WM5100_DSP1_DM_2 0x4002 @@ -4561,6 +4651,75 @@ int wm5100_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack); #define WM5100_LHPF4_COEFF_WIDTH 16 /* LHPF4_COEFF - [15:0] */ /* + * R4132 (0x1024) - DSP2 Control 30 + */ +#define WM5100_DSP2_RATE_MASK 0xC000 /* DSP2_RATE - [15:14] */ +#define WM5100_DSP2_RATE_SHIFT 14 /* DSP2_RATE - [15:14] */ +#define WM5100_DSP2_RATE_WIDTH 2 /* DSP2_RATE - [15:14] */ +#define WM5100_DSP2_DBG_CLK_ENA 0x0008 /* DSP2_DBG_CLK_ENA */ +#define WM5100_DSP2_DBG_CLK_ENA_MASK 0x0008 /* DSP2_DBG_CLK_ENA */ +#define WM5100_DSP2_DBG_CLK_ENA_SHIFT 3 /* DSP2_DBG_CLK_ENA */ +#define WM5100_DSP2_DBG_CLK_ENA_WIDTH 1 /* DSP2_DBG_CLK_ENA */ +#define WM5100_DSP2_SYS_ENA 0x0004 /* DSP2_SYS_ENA */ +#define WM5100_DSP2_SYS_ENA_MASK 0x0004 /* DSP2_SYS_ENA */ +#define WM5100_DSP2_SYS_ENA_SHIFT 2 /* DSP2_SYS_ENA */ +#define WM5100_DSP2_SYS_ENA_WIDTH 1 /* DSP2_SYS_ENA */ +#define WM5100_DSP2_CORE_ENA 0x0002 /* DSP2_CORE_ENA */ +#define WM5100_DSP2_CORE_ENA_MASK 0x0002 /* DSP2_CORE_ENA */ +#define WM5100_DSP2_CORE_ENA_SHIFT 1 /* DSP2_CORE_ENA */ +#define WM5100_DSP2_CORE_ENA_WIDTH 1 /* DSP2_CORE_ENA */ +#define WM5100_DSP2_START 0x0001 /* DSP2_START */ +#define WM5100_DSP2_START_MASK 0x0001 /* DSP2_START */ +#define WM5100_DSP2_START_SHIFT 0 /* DSP2_START */ +#define WM5100_DSP2_START_WIDTH 1 /* DSP2_START */ + +/* + * R3876 (0xF24) - DSP1 Control 30 + */ +#define WM5100_DSP1_RATE_MASK 0xC000 /* DSP1_RATE - [15:14] */ +#define WM5100_DSP1_RATE_SHIFT 14 /* DSP1_RATE - [15:14] */ +#define WM5100_DSP1_RATE_WIDTH 2 /* DSP1_RATE - [15:14] */ +#define WM5100_DSP1_DBG_CLK_ENA 0x0008 /* DSP1_DBG_CLK_ENA */ +#define WM5100_DSP1_DBG_CLK_ENA_MASK 0x0008 /* DSP1_DBG_CLK_ENA */ +#define WM5100_DSP1_DBG_CLK_ENA_SHIFT 3 /* DSP1_DBG_CLK_ENA */ +#define WM5100_DSP1_DBG_CLK_ENA_WIDTH 1 /* DSP1_DBG_CLK_ENA */ +#define WM5100_DSP1_SYS_ENA 0x0004 /* DSP1_SYS_ENA */ +#define WM5100_DSP1_SYS_ENA_MASK 0x0004 /* DSP1_SYS_ENA */ +#define WM5100_DSP1_SYS_ENA_SHIFT 2 /* DSP1_SYS_ENA */ +#define WM5100_DSP1_SYS_ENA_WIDTH 1 /* DSP1_SYS_ENA */ +#define WM5100_DSP1_CORE_ENA 0x0002 /* DSP1_CORE_ENA */ +#define WM5100_DSP1_CORE_ENA_MASK 0x0002 /* DSP1_CORE_ENA */ +#define WM5100_DSP1_CORE_ENA_SHIFT 1 /* DSP1_CORE_ENA */ +#define WM5100_DSP1_CORE_ENA_WIDTH 1 /* DSP1_CORE_ENA */ +#define WM5100_DSP1_START 0x0001 /* DSP1_START */ +#define WM5100_DSP1_START_MASK 0x0001 /* DSP1_START */ +#define WM5100_DSP1_START_SHIFT 0 /* DSP1_START */ +#define WM5100_DSP1_START_WIDTH 1 /* DSP1_START */ + +/* + * R4388 (0x1124) - DSP3 Control 30 + */ +#define WM5100_DSP3_RATE_MASK 0xC000 /* DSP3_RATE - [15:14] */ +#define WM5100_DSP3_RATE_SHIFT 14 /* DSP3_RATE - [15:14] */ +#define WM5100_DSP3_RATE_WIDTH 2 /* DSP3_RATE - [15:14] */ +#define WM5100_DSP3_DBG_CLK_ENA 0x0008 /* DSP3_DBG_CLK_ENA */ +#define WM5100_DSP3_DBG_CLK_ENA_MASK 0x0008 /* DSP3_DBG_CLK_ENA */ +#define WM5100_DSP3_DBG_CLK_ENA_SHIFT 3 /* DSP3_DBG_CLK_ENA */ +#define WM5100_DSP3_DBG_CLK_ENA_WIDTH 1 /* DSP3_DBG_CLK_ENA */ +#define WM5100_DSP3_SYS_ENA 0x0004 /* DSP3_SYS_ENA */ +#define WM5100_DSP3_SYS_ENA_MASK 0x0004 /* DSP3_SYS_ENA */ +#define WM5100_DSP3_SYS_ENA_SHIFT 2 /* DSP3_SYS_ENA */ +#define WM5100_DSP3_SYS_ENA_WIDTH 1 /* DSP3_SYS_ENA */ +#define WM5100_DSP3_CORE_ENA 0x0002 /* DSP3_CORE_ENA */ +#define WM5100_DSP3_CORE_ENA_MASK 0x0002 /* DSP3_CORE_ENA */ +#define WM5100_DSP3_CORE_ENA_SHIFT 1 /* DSP3_CORE_ENA */ +#define WM5100_DSP3_CORE_ENA_WIDTH 1 /* DSP3_CORE_ENA */ +#define WM5100_DSP3_START 0x0001 /* DSP3_START */ +#define WM5100_DSP3_START_MASK 0x0001 /* DSP3_START */ +#define WM5100_DSP3_START_SHIFT 0 /* DSP3_START */ +#define WM5100_DSP3_START_WIDTH 1 /* DSP3_START */ + +/* * R16384 (0x4000) - DSP1 DM 0 */ #define WM5100_DSP1_DM_START_1_MASK 0x00FF /* DSP1_DM_START - [7:0] */ diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index aa12c6b6beeb..555ee146ae0d 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -71,13 +71,6 @@ struct wm8350_data { int fll_freq_in; }; -static unsigned int wm8350_codec_cache_read(struct snd_soc_codec *codec, - unsigned int reg) -{ - struct wm8350 *wm8350 = codec->control_data; - return wm8350->reg_cache[reg]; -} - static unsigned int wm8350_codec_read(struct snd_soc_codec *codec, unsigned int reg) { @@ -99,7 +92,7 @@ static inline int wm8350_out1_ramp_step(struct snd_soc_codec *codec) { struct wm8350_data *wm8350_data = snd_soc_codec_get_drvdata(codec); struct wm8350_output *out1 = &wm8350_data->out1; - struct wm8350 *wm8350 = codec->control_data; + struct wm8350 *wm8350 = wm8350_data->wm8350; int left_complete = 0, right_complete = 0; u16 reg, val; @@ -165,7 +158,7 @@ static inline int wm8350_out2_ramp_step(struct snd_soc_codec *codec) { struct wm8350_data *wm8350_data = snd_soc_codec_get_drvdata(codec); struct wm8350_output *out2 = &wm8350_data->out2; - struct wm8350 *wm8350 = codec->control_data; + struct wm8350 *wm8350 = wm8350_data->wm8350; int left_complete = 0, right_complete = 0; u16 reg, val; @@ -360,8 +353,8 @@ static int wm8350_put_volsw_2r_vu(struct snd_kcontrol *kcontrol, return ret; /* now hit the volume update bits (always bit 8) */ - val = wm8350_codec_read(codec, reg); - wm8350_codec_write(codec, reg, val | WM8350_OUT1_VU); + val = snd_soc_read(codec, reg); + snd_soc_write(codec, reg, val | WM8350_OUT1_VU); return 1; } @@ -781,7 +774,8 @@ static int wm8350_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { struct snd_soc_codec *codec = codec_dai->codec; - struct wm8350 *wm8350 = codec->control_data; + struct wm8350_data *wm8350_data = snd_soc_codec_get_drvdata(codec); + struct wm8350 *wm8350 = wm8350_data->wm8350; u16 fll_4; switch (clk_id) { @@ -795,9 +789,9 @@ static int wm8350_set_dai_sysclk(struct snd_soc_dai *codec_dai, case WM8350_MCLK_SEL_PLL_32K: wm8350_set_bits(wm8350, WM8350_CLOCK_CONTROL_1, WM8350_MCLK_SEL); - fll_4 = wm8350_codec_read(codec, WM8350_FLL_CONTROL_4) & + fll_4 = snd_soc_read(codec, WM8350_FLL_CONTROL_4) & ~WM8350_FLL_CLK_SRC_MASK; - wm8350_codec_write(codec, WM8350_FLL_CONTROL_4, fll_4 | clk_id); + snd_soc_write(codec, WM8350_FLL_CONTROL_4, fll_4 | clk_id); break; } @@ -819,39 +813,39 @@ static int wm8350_set_clkdiv(struct snd_soc_dai *codec_dai, int div_id, int div) switch (div_id) { case WM8350_ADC_CLKDIV: - val = wm8350_codec_read(codec, WM8350_ADC_DIVIDER) & + val = snd_soc_read(codec, WM8350_ADC_DIVIDER) & ~WM8350_ADC_CLKDIV_MASK; - wm8350_codec_write(codec, WM8350_ADC_DIVIDER, val | div); + snd_soc_write(codec, WM8350_ADC_DIVIDER, val | div); break; case WM8350_DAC_CLKDIV: - val = wm8350_codec_read(codec, WM8350_DAC_CLOCK_CONTROL) & + val = snd_soc_read(codec, WM8350_DAC_CLOCK_CONTROL) & ~WM8350_DAC_CLKDIV_MASK; - wm8350_codec_write(codec, WM8350_DAC_CLOCK_CONTROL, val | div); + snd_soc_write(codec, WM8350_DAC_CLOCK_CONTROL, val | div); break; case WM8350_BCLK_CLKDIV: - val = wm8350_codec_read(codec, WM8350_CLOCK_CONTROL_1) & + val = snd_soc_read(codec, WM8350_CLOCK_CONTROL_1) & ~WM8350_BCLK_DIV_MASK; - wm8350_codec_write(codec, WM8350_CLOCK_CONTROL_1, val | div); + snd_soc_write(codec, WM8350_CLOCK_CONTROL_1, val | div); break; case WM8350_OPCLK_CLKDIV: - val = wm8350_codec_read(codec, WM8350_CLOCK_CONTROL_1) & + val = snd_soc_read(codec, WM8350_CLOCK_CONTROL_1) & ~WM8350_OPCLK_DIV_MASK; - wm8350_codec_write(codec, WM8350_CLOCK_CONTROL_1, val | div); + snd_soc_write(codec, WM8350_CLOCK_CONTROL_1, val | div); break; case WM8350_SYS_CLKDIV: - val = wm8350_codec_read(codec, WM8350_CLOCK_CONTROL_1) & + val = snd_soc_read(codec, WM8350_CLOCK_CONTROL_1) & ~WM8350_MCLK_DIV_MASK; - wm8350_codec_write(codec, WM8350_CLOCK_CONTROL_1, val | div); + snd_soc_write(codec, WM8350_CLOCK_CONTROL_1, val | div); break; case WM8350_DACLR_CLKDIV: - val = wm8350_codec_read(codec, WM8350_DAC_LR_RATE) & + val = snd_soc_read(codec, WM8350_DAC_LR_RATE) & ~WM8350_DACLRC_RATE_MASK; - wm8350_codec_write(codec, WM8350_DAC_LR_RATE, val | div); + snd_soc_write(codec, WM8350_DAC_LR_RATE, val | div); break; case WM8350_ADCLR_CLKDIV: - val = wm8350_codec_read(codec, WM8350_ADC_LR_RATE) & + val = snd_soc_read(codec, WM8350_ADC_LR_RATE) & ~WM8350_ADCLRC_RATE_MASK; - wm8350_codec_write(codec, WM8350_ADC_LR_RATE, val | div); + snd_soc_write(codec, WM8350_ADC_LR_RATE, val | div); break; default: return -EINVAL; @@ -863,13 +857,13 @@ static int wm8350_set_clkdiv(struct snd_soc_dai *codec_dai, int div_id, int div) static int wm8350_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; - u16 iface = wm8350_codec_read(codec, WM8350_AI_FORMATING) & + u16 iface = snd_soc_read(codec, WM8350_AI_FORMATING) & ~(WM8350_AIF_BCLK_INV | WM8350_AIF_LRCLK_INV | WM8350_AIF_FMT_MASK); - u16 master = wm8350_codec_read(codec, WM8350_AI_DAC_CONTROL) & + u16 master = snd_soc_read(codec, WM8350_AI_DAC_CONTROL) & ~WM8350_BCLK_MSTR; - u16 dac_lrc = wm8350_codec_read(codec, WM8350_DAC_LR_RATE) & + u16 dac_lrc = snd_soc_read(codec, WM8350_DAC_LR_RATE) & ~WM8350_DACLRC_ENA; - u16 adc_lrc = wm8350_codec_read(codec, WM8350_ADC_LR_RATE) & + u16 adc_lrc = snd_soc_read(codec, WM8350_ADC_LR_RATE) & ~WM8350_ADCLRC_ENA; /* set master/slave audio interface */ @@ -922,42 +916,10 @@ static int wm8350_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) return -EINVAL; } - wm8350_codec_write(codec, WM8350_AI_FORMATING, iface); - wm8350_codec_write(codec, WM8350_AI_DAC_CONTROL, master); - wm8350_codec_write(codec, WM8350_DAC_LR_RATE, dac_lrc); - wm8350_codec_write(codec, WM8350_ADC_LR_RATE, adc_lrc); - return 0; -} - -static int wm8350_pcm_trigger(struct snd_pcm_substream *substream, - int cmd, struct snd_soc_dai *codec_dai) -{ - struct snd_soc_codec *codec = codec_dai->codec; - int master = wm8350_codec_cache_read(codec, WM8350_AI_DAC_CONTROL) & - WM8350_BCLK_MSTR; - int enabled = 0; - - /* Check that the DACs or ADCs are enabled since they are - * required for LRC in master mode. The DACs or ADCs need a - * valid audio path i.e. pin -> ADC or DAC -> pin before - * the LRC will be enabled in master mode. */ - if (!master || cmd != SNDRV_PCM_TRIGGER_START) - return 0; - - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { - enabled = wm8350_codec_cache_read(codec, WM8350_POWER_MGMT_4) & - (WM8350_ADCR_ENA | WM8350_ADCL_ENA); - } else { - enabled = wm8350_codec_cache_read(codec, WM8350_POWER_MGMT_4) & - (WM8350_DACR_ENA | WM8350_DACL_ENA); - } - - if (!enabled) { - dev_err(codec->dev, - "%s: invalid audio path - no clocks available\n", - __func__); - return -EINVAL; - } + snd_soc_write(codec, WM8350_AI_FORMATING, iface); + snd_soc_write(codec, WM8350_AI_DAC_CONTROL, master); + snd_soc_write(codec, WM8350_DAC_LR_RATE, dac_lrc); + snd_soc_write(codec, WM8350_ADC_LR_RATE, adc_lrc); return 0; } @@ -966,8 +928,9 @@ static int wm8350_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *codec_dai) { struct snd_soc_codec *codec = codec_dai->codec; - struct wm8350 *wm8350 = codec->control_data; - u16 iface = wm8350_codec_read(codec, WM8350_AI_FORMATING) & + struct wm8350_data *wm8350_data = snd_soc_codec_get_drvdata(codec); + struct wm8350 *wm8350 = wm8350_data->wm8350; + u16 iface = snd_soc_read(codec, WM8350_AI_FORMATING) & ~WM8350_AIF_WL_MASK; /* bit size */ @@ -985,7 +948,7 @@ static int wm8350_pcm_hw_params(struct snd_pcm_substream *substream, break; } - wm8350_codec_write(codec, WM8350_AI_FORMATING, iface); + snd_soc_write(codec, WM8350_AI_FORMATING, iface); /* The sloping stopband filter is recommended for use with * lower sample rates to improve performance. @@ -1005,12 +968,15 @@ static int wm8350_pcm_hw_params(struct snd_pcm_substream *substream, static int wm8350_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; - struct wm8350 *wm8350 = codec->control_data; + unsigned int val; if (mute) - wm8350_set_bits(wm8350, WM8350_DAC_MUTE, WM8350_DAC_MUTE_ENA); + val = WM8350_DAC_MUTE_ENA; else - wm8350_clear_bits(wm8350, WM8350_DAC_MUTE, WM8350_DAC_MUTE_ENA); + val = 0; + + snd_soc_update_bits(codec, WM8350_DAC_MUTE, WM8350_DAC_MUTE_ENA, val); + return 0; } @@ -1079,8 +1045,8 @@ static int wm8350_set_fll(struct snd_soc_dai *codec_dai, unsigned int freq_out) { struct snd_soc_codec *codec = codec_dai->codec; - struct wm8350 *wm8350 = codec->control_data; struct wm8350_data *priv = snd_soc_codec_get_drvdata(codec); + struct wm8350 *wm8350 = priv->wm8350; struct _fll_div fll_div; int ret = 0; u16 fll_1, fll_4; @@ -1104,17 +1070,17 @@ static int wm8350_set_fll(struct snd_soc_dai *codec_dai, fll_div.ratio); /* set up N.K & dividers */ - fll_1 = wm8350_codec_read(codec, WM8350_FLL_CONTROL_1) & + fll_1 = snd_soc_read(codec, WM8350_FLL_CONTROL_1) & ~(WM8350_FLL_OUTDIV_MASK | WM8350_FLL_RSP_RATE_MASK | 0xc000); - wm8350_codec_write(codec, WM8350_FLL_CONTROL_1, + snd_soc_write(codec, WM8350_FLL_CONTROL_1, fll_1 | (fll_div.div << 8) | 0x50); - wm8350_codec_write(codec, WM8350_FLL_CONTROL_2, + snd_soc_write(codec, WM8350_FLL_CONTROL_2, (fll_div.ratio << 11) | (fll_div. n & WM8350_FLL_N_MASK)); - wm8350_codec_write(codec, WM8350_FLL_CONTROL_3, fll_div.k); - fll_4 = wm8350_codec_read(codec, WM8350_FLL_CONTROL_4) & + snd_soc_write(codec, WM8350_FLL_CONTROL_3, fll_div.k); + fll_4 = snd_soc_read(codec, WM8350_FLL_CONTROL_4) & ~(WM8350_FLL_FRAC | WM8350_FLL_SLOW_LOCK_REF); - wm8350_codec_write(codec, WM8350_FLL_CONTROL_4, + snd_soc_write(codec, WM8350_FLL_CONTROL_4, fll_4 | (fll_div.k ? WM8350_FLL_FRAC : 0) | (fll_div.ratio == 8 ? WM8350_FLL_SLOW_LOCK_REF : 0)); @@ -1131,8 +1097,8 @@ static int wm8350_set_fll(struct snd_soc_dai *codec_dai, static int wm8350_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - struct wm8350 *wm8350 = codec->control_data; struct wm8350_data *priv = snd_soc_codec_get_drvdata(codec); + struct wm8350 *wm8350 = priv->wm8350; struct wm8350_audio_platform_data *platform = wm8350->codec.platform_data; u16 pm1; @@ -1339,35 +1305,36 @@ static void wm8350_hpr_work(struct work_struct *work) wm8350_hp_work(priv, &priv->hpr, WM8350_JACK_R_LVL); } -static irqreturn_t wm8350_hp_jack_handler(int irq, void *data) +static irqreturn_t wm8350_hpl_jack_handler(int irq, void *data) { struct wm8350_data *priv = data; struct wm8350 *wm8350 = priv->wm8350; - struct wm8350_jack_data *jack = NULL; - switch (irq - wm8350->irq_base) { - case WM8350_IRQ_CODEC_JCK_DET_L: #ifndef CONFIG_SND_SOC_WM8350_MODULE - trace_snd_soc_jack_irq("WM8350 HPL"); + trace_snd_soc_jack_irq("WM8350 HPL"); #endif - jack = &priv->hpl; - break; - case WM8350_IRQ_CODEC_JCK_DET_R: + if (device_may_wakeup(wm8350->dev)) + pm_wakeup_event(wm8350->dev, 250); + + schedule_delayed_work(&priv->hpl.work, 200); + + return IRQ_HANDLED; +} + +static irqreturn_t wm8350_hpr_jack_handler(int irq, void *data) +{ + struct wm8350_data *priv = data; + struct wm8350 *wm8350 = priv->wm8350; + #ifndef CONFIG_SND_SOC_WM8350_MODULE - trace_snd_soc_jack_irq("WM8350 HPR"); + trace_snd_soc_jack_irq("WM8350 HPR"); #endif - jack = &priv->hpr; - break; - - default: - BUG(); - } if (device_may_wakeup(wm8350->dev)) pm_wakeup_event(wm8350->dev, 250); - schedule_delayed_work(&jack->work, 200); + schedule_delayed_work(&priv->hpr.work, 200); return IRQ_HANDLED; } @@ -1387,7 +1354,7 @@ int wm8350_hp_jack_detect(struct snd_soc_codec *codec, enum wm8350_jack which, struct snd_soc_jack *jack, int report) { struct wm8350_data *priv = snd_soc_codec_get_drvdata(codec); - struct wm8350 *wm8350 = codec->control_data; + struct wm8350 *wm8350 = priv->wm8350; int irq; int ena; @@ -1418,7 +1385,14 @@ int wm8350_hp_jack_detect(struct snd_soc_codec *codec, enum wm8350_jack which, } /* Sync status */ - wm8350_hp_jack_handler(irq + wm8350->irq_base, priv); + switch (which) { + case WM8350_JDL: + wm8350_hpl_jack_handler(0, priv); + break; + case WM8350_JDR: + wm8350_hpr_jack_handler(0, priv); + break; + } return 0; } @@ -1463,7 +1437,7 @@ int wm8350_mic_jack_detect(struct snd_soc_codec *codec, int detect_report, int short_report) { struct wm8350_data *priv = snd_soc_codec_get_drvdata(codec); - struct wm8350 *wm8350 = codec->control_data; + struct wm8350 *wm8350 = priv->wm8350; priv->mic.jack = jack; priv->mic.report = detect_report; @@ -1491,7 +1465,6 @@ EXPORT_SYMBOL_GPL(wm8350_mic_jack_detect); static const struct snd_soc_dai_ops wm8350_dai_ops = { .hw_params = wm8350_pcm_hw_params, .digital_mute = wm8350_mute, - .trigger = wm8350_pcm_trigger, .set_fmt = wm8350_set_dai_fmt, .set_sysclk = wm8350_set_dai_sysclk, .set_pll = wm8350_set_fll, @@ -1559,9 +1532,9 @@ static int wm8350_codec_probe(struct snd_soc_codec *codec) wm8350_set_bits(wm8350, WM8350_POWER_MGMT_5, WM8350_CODEC_ENA); /* Enable robust clocking mode in ADC */ - wm8350_codec_write(codec, WM8350_SECURITY, 0xa7); - wm8350_codec_write(codec, 0xde, 0x13); - wm8350_codec_write(codec, WM8350_SECURITY, 0); + snd_soc_write(codec, WM8350_SECURITY, 0xa7); + snd_soc_write(codec, 0xde, 0x13); + snd_soc_write(codec, WM8350_SECURITY, 0); /* read OUT1 & OUT2 volumes */ out1 = &priv->out1; @@ -1601,10 +1574,10 @@ static int wm8350_codec_probe(struct snd_soc_codec *codec) WM8350_JDL_ENA | WM8350_JDR_ENA); wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L, - wm8350_hp_jack_handler, 0, "Left jack detect", + wm8350_hpl_jack_handler, 0, "Left jack detect", priv); wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R, - wm8350_hp_jack_handler, 0, "Right jack detect", + wm8350_hpr_jack_handler, 0, "Right jack detect", priv); wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_MICSCD, wm8350_mic_handler, 0, "Microphone short", priv); diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index 898979d23010..5dc31ebcd0e7 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -138,8 +138,8 @@ static int wm8400_outpga_put_volsw_vu(struct snd_kcontrol *kcontrol, return ret; /* now hit the volume update bits (always bit 8) */ - val = wm8400_read(codec, reg); - return wm8400_write(codec, reg, val | 0x0100); + val = snd_soc_read(codec, reg); + return snd_soc_write(codec, reg, val | 0x0100); } #define WM8400_OUTPGA_SINGLE_R_TLV(xname, reg, shift, max, invert, tlv_array) \ @@ -362,8 +362,8 @@ static int inmixer_event (struct snd_soc_dapm_widget *w, { u16 reg, fakepower; - reg = wm8400_read(w->codec, WM8400_POWER_MANAGEMENT_2); - fakepower = wm8400_read(w->codec, WM8400_INTDRIVBITS); + reg = snd_soc_read(w->codec, WM8400_POWER_MANAGEMENT_2); + fakepower = snd_soc_read(w->codec, WM8400_INTDRIVBITS); if (fakepower & ((1 << WM8400_INMIXL_PWR) | (1 << WM8400_AINLMUX_PWR))) { @@ -378,7 +378,7 @@ static int inmixer_event (struct snd_soc_dapm_widget *w, } else { reg &= ~WM8400_AINR_ENA; } - wm8400_write(w->codec, WM8400_POWER_MANAGEMENT_2, reg); + snd_soc_write(w->codec, WM8400_POWER_MANAGEMENT_2, reg); return 0; } @@ -394,7 +394,7 @@ static int outmixer_event (struct snd_soc_dapm_widget *w, switch (reg_shift) { case WM8400_SPEAKER_MIXER | (WM8400_LDSPK << 8) : - reg = wm8400_read(w->codec, WM8400_OUTPUT_MIXER1); + reg = snd_soc_read(w->codec, WM8400_OUTPUT_MIXER1); if (reg & WM8400_LDLO) { printk(KERN_WARNING "Cannot set as Output Mixer 1 LDLO Set\n"); @@ -402,7 +402,7 @@ static int outmixer_event (struct snd_soc_dapm_widget *w, } break; case WM8400_SPEAKER_MIXER | (WM8400_RDSPK << 8): - reg = wm8400_read(w->codec, WM8400_OUTPUT_MIXER2); + reg = snd_soc_read(w->codec, WM8400_OUTPUT_MIXER2); if (reg & WM8400_RDRO) { printk(KERN_WARNING "Cannot set as Output Mixer 2 RDRO Set\n"); @@ -410,7 +410,7 @@ static int outmixer_event (struct snd_soc_dapm_widget *w, } break; case WM8400_OUTPUT_MIXER1 | (WM8400_LDLO << 8): - reg = wm8400_read(w->codec, WM8400_SPEAKER_MIXER); + reg = snd_soc_read(w->codec, WM8400_SPEAKER_MIXER); if (reg & WM8400_LDSPK) { printk(KERN_WARNING "Cannot set as Speaker Mixer LDSPK Set\n"); @@ -418,7 +418,7 @@ static int outmixer_event (struct snd_soc_dapm_widget *w, } break; case WM8400_OUTPUT_MIXER2 | (WM8400_RDRO << 8): - reg = wm8400_read(w->codec, WM8400_SPEAKER_MIXER); + reg = snd_soc_read(w->codec, WM8400_SPEAKER_MIXER); if (reg & WM8400_RDSPK) { printk(KERN_WARNING "Cannot set as Speaker Mixer RDSPK Set\n"); @@ -1021,13 +1021,13 @@ static int wm8400_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, wm8400->fll_in = freq_in; /* We *must* disable the FLL before any changes */ - reg = wm8400_read(codec, WM8400_POWER_MANAGEMENT_2); + reg = snd_soc_read(codec, WM8400_POWER_MANAGEMENT_2); reg &= ~WM8400_FLL_ENA; - wm8400_write(codec, WM8400_POWER_MANAGEMENT_2, reg); + snd_soc_write(codec, WM8400_POWER_MANAGEMENT_2, reg); - reg = wm8400_read(codec, WM8400_FLL_CONTROL_1); + reg = snd_soc_read(codec, WM8400_FLL_CONTROL_1); reg &= ~WM8400_FLL_OSC_ENA; - wm8400_write(codec, WM8400_FLL_CONTROL_1, reg); + snd_soc_write(codec, WM8400_FLL_CONTROL_1, reg); if (!freq_out) return 0; @@ -1035,15 +1035,15 @@ static int wm8400_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, reg &= ~(WM8400_FLL_REF_FREQ | WM8400_FLL_FRATIO_MASK); reg |= WM8400_FLL_FRAC | factors.fratio; reg |= factors.freq_ref << WM8400_FLL_REF_FREQ_SHIFT; - wm8400_write(codec, WM8400_FLL_CONTROL_1, reg); + snd_soc_write(codec, WM8400_FLL_CONTROL_1, reg); - wm8400_write(codec, WM8400_FLL_CONTROL_2, factors.k); - wm8400_write(codec, WM8400_FLL_CONTROL_3, factors.n); + snd_soc_write(codec, WM8400_FLL_CONTROL_2, factors.k); + snd_soc_write(codec, WM8400_FLL_CONTROL_3, factors.n); - reg = wm8400_read(codec, WM8400_FLL_CONTROL_4); + reg = snd_soc_read(codec, WM8400_FLL_CONTROL_4); reg &= ~WM8400_FLL_OUTDIV_MASK; reg |= factors.outdiv; - wm8400_write(codec, WM8400_FLL_CONTROL_4, reg); + snd_soc_write(codec, WM8400_FLL_CONTROL_4, reg); return 0; } @@ -1057,8 +1057,8 @@ static int wm8400_set_dai_fmt(struct snd_soc_dai *codec_dai, struct snd_soc_codec *codec = codec_dai->codec; u16 audio1, audio3; - audio1 = wm8400_read(codec, WM8400_AUDIO_INTERFACE_1); - audio3 = wm8400_read(codec, WM8400_AUDIO_INTERFACE_3); + audio1 = snd_soc_read(codec, WM8400_AUDIO_INTERFACE_1); + audio3 = snd_soc_read(codec, WM8400_AUDIO_INTERFACE_3); /* set master/slave audio interface */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { @@ -1099,8 +1099,8 @@ static int wm8400_set_dai_fmt(struct snd_soc_dai *codec_dai, return -EINVAL; } - wm8400_write(codec, WM8400_AUDIO_INTERFACE_1, audio1); - wm8400_write(codec, WM8400_AUDIO_INTERFACE_3, audio3); + snd_soc_write(codec, WM8400_AUDIO_INTERFACE_1, audio1); + snd_soc_write(codec, WM8400_AUDIO_INTERFACE_3, audio3); return 0; } @@ -1112,24 +1112,24 @@ static int wm8400_set_dai_clkdiv(struct snd_soc_dai *codec_dai, switch (div_id) { case WM8400_MCLK_DIV: - reg = wm8400_read(codec, WM8400_CLOCKING_2) & + reg = snd_soc_read(codec, WM8400_CLOCKING_2) & ~WM8400_MCLK_DIV_MASK; - wm8400_write(codec, WM8400_CLOCKING_2, reg | div); + snd_soc_write(codec, WM8400_CLOCKING_2, reg | div); break; case WM8400_DACCLK_DIV: - reg = wm8400_read(codec, WM8400_CLOCKING_2) & + reg = snd_soc_read(codec, WM8400_CLOCKING_2) & ~WM8400_DAC_CLKDIV_MASK; - wm8400_write(codec, WM8400_CLOCKING_2, reg | div); + snd_soc_write(codec, WM8400_CLOCKING_2, reg | div); break; case WM8400_ADCCLK_DIV: - reg = wm8400_read(codec, WM8400_CLOCKING_2) & + reg = snd_soc_read(codec, WM8400_CLOCKING_2) & ~WM8400_ADC_CLKDIV_MASK; - wm8400_write(codec, WM8400_CLOCKING_2, reg | div); + snd_soc_write(codec, WM8400_CLOCKING_2, reg | div); break; case WM8400_BCLK_DIV: - reg = wm8400_read(codec, WM8400_CLOCKING_1) & + reg = snd_soc_read(codec, WM8400_CLOCKING_1) & ~WM8400_BCLK_DIV_MASK; - wm8400_write(codec, WM8400_CLOCKING_1, reg | div); + snd_soc_write(codec, WM8400_CLOCKING_1, reg | div); break; default: return -EINVAL; @@ -1145,9 +1145,8 @@ static int wm8400_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; - u16 audio1 = wm8400_read(codec, WM8400_AUDIO_INTERFACE_1); + struct snd_soc_codec *codec = dai->codec; + u16 audio1 = snd_soc_read(codec, WM8400_AUDIO_INTERFACE_1); audio1 &= ~WM8400_AIF_WL_MASK; /* bit size */ @@ -1165,19 +1164,19 @@ static int wm8400_hw_params(struct snd_pcm_substream *substream, break; } - wm8400_write(codec, WM8400_AUDIO_INTERFACE_1, audio1); + snd_soc_write(codec, WM8400_AUDIO_INTERFACE_1, audio1); return 0; } static int wm8400_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; - u16 val = wm8400_read(codec, WM8400_DAC_CTRL) & ~WM8400_DAC_MUTE; + u16 val = snd_soc_read(codec, WM8400_DAC_CTRL) & ~WM8400_DAC_MUTE; if (mute) - wm8400_write(codec, WM8400_DAC_CTRL, val | WM8400_DAC_MUTE); + snd_soc_write(codec, WM8400_DAC_CTRL, val | WM8400_DAC_MUTE); else - wm8400_write(codec, WM8400_DAC_CTRL, val); + snd_soc_write(codec, WM8400_DAC_CTRL, val); return 0; } @@ -1196,9 +1195,9 @@ static int wm8400_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_PREPARE: /* VMID=2*50k */ - val = wm8400_read(codec, WM8400_POWER_MANAGEMENT_1) & + val = snd_soc_read(codec, WM8400_POWER_MANAGEMENT_1) & ~WM8400_VMID_MODE_MASK; - wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val | 0x2); + snd_soc_write(codec, WM8400_POWER_MANAGEMENT_1, val | 0x2); break; case SND_SOC_BIAS_STANDBY: @@ -1212,74 +1211,74 @@ static int wm8400_set_bias_level(struct snd_soc_codec *codec, return ret; } - wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, + snd_soc_write(codec, WM8400_POWER_MANAGEMENT_1, WM8400_CODEC_ENA | WM8400_SYSCLK_ENA); /* Enable POBCTRL, SOFT_ST, VMIDTOG and BUFDCOPEN */ - wm8400_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST | + snd_soc_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST | WM8400_BUFDCOPEN | WM8400_POBCTRL); msleep(50); /* Enable VREF & VMID at 2x50k */ - val = wm8400_read(codec, WM8400_POWER_MANAGEMENT_1); + val = snd_soc_read(codec, WM8400_POWER_MANAGEMENT_1); val |= 0x2 | WM8400_VREF_ENA; - wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val); + snd_soc_write(codec, WM8400_POWER_MANAGEMENT_1, val); /* Enable BUFIOEN */ - wm8400_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST | + snd_soc_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST | WM8400_BUFDCOPEN | WM8400_POBCTRL | WM8400_BUFIOEN); /* disable POBCTRL, SOFT_ST and BUFDCOPEN */ - wm8400_write(codec, WM8400_ANTIPOP2, WM8400_BUFIOEN); + snd_soc_write(codec, WM8400_ANTIPOP2, WM8400_BUFIOEN); } /* VMID=2*300k */ - val = wm8400_read(codec, WM8400_POWER_MANAGEMENT_1) & + val = snd_soc_read(codec, WM8400_POWER_MANAGEMENT_1) & ~WM8400_VMID_MODE_MASK; - wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val | 0x4); + snd_soc_write(codec, WM8400_POWER_MANAGEMENT_1, val | 0x4); break; case SND_SOC_BIAS_OFF: /* Enable POBCTRL and SOFT_ST */ - wm8400_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST | + snd_soc_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST | WM8400_POBCTRL | WM8400_BUFIOEN); /* Enable POBCTRL, SOFT_ST and BUFDCOPEN */ - wm8400_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST | + snd_soc_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST | WM8400_BUFDCOPEN | WM8400_POBCTRL | WM8400_BUFIOEN); /* mute DAC */ - val = wm8400_read(codec, WM8400_DAC_CTRL); - wm8400_write(codec, WM8400_DAC_CTRL, val | WM8400_DAC_MUTE); + val = snd_soc_read(codec, WM8400_DAC_CTRL); + snd_soc_write(codec, WM8400_DAC_CTRL, val | WM8400_DAC_MUTE); /* Enable any disabled outputs */ - val = wm8400_read(codec, WM8400_POWER_MANAGEMENT_1); + val = snd_soc_read(codec, WM8400_POWER_MANAGEMENT_1); val |= WM8400_SPK_ENA | WM8400_OUT3_ENA | WM8400_OUT4_ENA | WM8400_LOUT_ENA | WM8400_ROUT_ENA; - wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val); + snd_soc_write(codec, WM8400_POWER_MANAGEMENT_1, val); /* Disable VMID */ val &= ~WM8400_VMID_MODE_MASK; - wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val); + snd_soc_write(codec, WM8400_POWER_MANAGEMENT_1, val); msleep(300); /* Enable all output discharge bits */ - wm8400_write(codec, WM8400_ANTIPOP1, WM8400_DIS_LLINE | + snd_soc_write(codec, WM8400_ANTIPOP1, WM8400_DIS_LLINE | WM8400_DIS_RLINE | WM8400_DIS_OUT3 | WM8400_DIS_OUT4 | WM8400_DIS_LOUT | WM8400_DIS_ROUT); /* Disable VREF */ val &= ~WM8400_VREF_ENA; - wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val); + snd_soc_write(codec, WM8400_POWER_MANAGEMENT_1, val); /* disable POBCTRL, SOFT_ST and BUFDCOPEN */ - wm8400_write(codec, WM8400_ANTIPOP2, 0x0); + snd_soc_write(codec, WM8400_ANTIPOP2, 0x0); ret = regulator_bulk_disable(ARRAY_SIZE(power), &power[0]); @@ -1385,19 +1384,19 @@ static int wm8400_codec_probe(struct snd_soc_codec *codec) wm8400_codec_reset(codec); - reg = wm8400_read(codec, WM8400_POWER_MANAGEMENT_1); - wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, reg | WM8400_CODEC_ENA); + reg = snd_soc_read(codec, WM8400_POWER_MANAGEMENT_1); + snd_soc_write(codec, WM8400_POWER_MANAGEMENT_1, reg | WM8400_CODEC_ENA); /* Latch volume update bits */ - reg = wm8400_read(codec, WM8400_LEFT_LINE_INPUT_1_2_VOLUME); - wm8400_write(codec, WM8400_LEFT_LINE_INPUT_1_2_VOLUME, + reg = snd_soc_read(codec, WM8400_LEFT_LINE_INPUT_1_2_VOLUME); + snd_soc_write(codec, WM8400_LEFT_LINE_INPUT_1_2_VOLUME, reg & WM8400_IPVU); - reg = wm8400_read(codec, WM8400_RIGHT_LINE_INPUT_1_2_VOLUME); - wm8400_write(codec, WM8400_RIGHT_LINE_INPUT_1_2_VOLUME, + reg = snd_soc_read(codec, WM8400_RIGHT_LINE_INPUT_1_2_VOLUME); + snd_soc_write(codec, WM8400_RIGHT_LINE_INPUT_1_2_VOLUME, reg & WM8400_IPVU); - wm8400_write(codec, WM8400_LEFT_OUTPUT_VOLUME, 0x50 | (1<<8)); - wm8400_write(codec, WM8400_RIGHT_OUTPUT_VOLUME, 0x50 | (1<<8)); + snd_soc_write(codec, WM8400_LEFT_OUTPUT_VOLUME, 0x50 | (1<<8)); + snd_soc_write(codec, WM8400_RIGHT_OUTPUT_VOLUME, 0x50 | (1<<8)); if (!schedule_work(&priv->work)) { ret = -EINVAL; @@ -1414,8 +1413,8 @@ static int wm8400_codec_remove(struct snd_soc_codec *codec) { u16 reg; - reg = wm8400_read(codec, WM8400_POWER_MANAGEMENT_1); - wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, + reg = snd_soc_read(codec, WM8400_POWER_MANAGEMENT_1); + snd_soc_write(codec, WM8400_POWER_MANAGEMENT_1, reg & (~WM8400_CODEC_ENA)); regulator_bulk_free(ARRAY_SIZE(power), power); @@ -1428,7 +1427,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8400 = { .remove = wm8400_codec_remove, .suspend = wm8400_suspend, .resume = wm8400_resume, - .read = wm8400_read, + .read = snd_soc_read, .write = wm8400_write, .set_bias_level = wm8400_set_bias_level, diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 9166126bd312..56a049555e2c 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -392,8 +392,7 @@ static int wm8510_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; u16 iface = snd_soc_read(codec, WM8510_IFACE) & 0x19f; u16 adn = snd_soc_read(codec, WM8510_ADD) & 0x1f1; diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c index 7fea2c3bf7e7..1c3ffb290cdc 100644 --- a/sound/soc/codecs/wm8523.c +++ b/sound/soc/codecs/wm8523.c @@ -145,8 +145,7 @@ static int wm8523_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8523_priv *wm8523 = snd_soc_codec_get_drvdata(codec); int i; u16 aifctrl1 = snd_soc_read(codec, WM8523_AIF_CTRL1); diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c index fc3d59e49084..1467f97dce21 100644 --- a/sound/soc/codecs/wm8728.c +++ b/sound/soc/codecs/wm8728.c @@ -88,8 +88,7 @@ static int wm8728_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; u16 dac = snd_soc_read(codec, WM8728_DACCTL); dac &= ~0x18; diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index a32caa72bd7d..9d1b9b0271f1 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -635,16 +635,17 @@ static int __devinit wm8731_spi_probe(struct spi_device *spi) struct wm8731_priv *wm8731; int ret; - wm8731 = kzalloc(sizeof(struct wm8731_priv), GFP_KERNEL); + wm8731 = devm_kzalloc(&spi->dev, sizeof(struct wm8731_priv), + GFP_KERNEL); if (wm8731 == NULL) return -ENOMEM; - wm8731->regmap = regmap_init_spi(spi, &wm8731_regmap); + wm8731->regmap = devm_regmap_init_spi(spi, &wm8731_regmap); if (IS_ERR(wm8731->regmap)) { ret = PTR_ERR(wm8731->regmap); dev_err(&spi->dev, "Failed to allocate register map: %d\n", ret); - goto err; + return ret; } spi_set_drvdata(spi, wm8731); @@ -653,25 +654,15 @@ static int __devinit wm8731_spi_probe(struct spi_device *spi) &soc_codec_dev_wm8731, &wm8731_dai, 1); if (ret != 0) { dev_err(&spi->dev, "Failed to register CODEC: %d\n", ret); - goto err_regmap; + return ret; } return 0; - -err_regmap: - regmap_exit(wm8731->regmap); -err: - kfree(wm8731); - return ret; } static int __devexit wm8731_spi_remove(struct spi_device *spi) { - struct wm8731_priv *wm8731 = spi_get_drvdata(spi); - snd_soc_unregister_codec(&spi->dev); - regmap_exit(wm8731->regmap); - kfree(wm8731); return 0; } @@ -693,16 +684,17 @@ static __devinit int wm8731_i2c_probe(struct i2c_client *i2c, struct wm8731_priv *wm8731; int ret; - wm8731 = kzalloc(sizeof(struct wm8731_priv), GFP_KERNEL); + wm8731 = devm_kzalloc(&i2c->dev, sizeof(struct wm8731_priv), + GFP_KERNEL); if (wm8731 == NULL) return -ENOMEM; - wm8731->regmap = regmap_init_i2c(i2c, &wm8731_regmap); + wm8731->regmap = devm_regmap_init_i2c(i2c, &wm8731_regmap); if (IS_ERR(wm8731->regmap)) { ret = PTR_ERR(wm8731->regmap); dev_err(&i2c->dev, "Failed to allocate register map: %d\n", ret); - goto err; + return ret; } i2c_set_clientdata(i2c, wm8731); @@ -711,24 +703,15 @@ static __devinit int wm8731_i2c_probe(struct i2c_client *i2c, &soc_codec_dev_wm8731, &wm8731_dai, 1); if (ret != 0) { dev_err(&i2c->dev, "Failed to register CODEC: %d\n", ret); - goto err_regmap; + return ret; } return 0; - -err_regmap: - regmap_exit(wm8731->regmap); -err: - kfree(wm8731); - return ret; } static __devexit int wm8731_i2c_remove(struct i2c_client *client) { - struct wm8731_priv *wm8731 = i2c_get_clientdata(client); snd_soc_unregister_codec(&client->dev); - regmap_exit(wm8731->regmap); - kfree(wm8731); return 0; } diff --git a/sound/soc/codecs/wm8737.c b/sound/soc/codecs/wm8737.c index 4fe9d191e277..d0520124616d 100644 --- a/sound/soc/codecs/wm8737.c +++ b/sound/soc/codecs/wm8737.c @@ -329,8 +329,7 @@ static int wm8737_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8737_priv *wm8737 = snd_soc_codec_get_drvdata(codec); int i; u16 clocking = 0; diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index 3941f50bf187..6e849cb04243 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -203,8 +203,7 @@ static int wm8741_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8741_priv *wm8741 = snd_soc_codec_get_drvdata(codec); u16 iface = snd_soc_read(codec, WM8741_FORMAT_CONTROL) & 0x1FC; int i; diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index e4c50ce7d9c0..89151ca5e776 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -547,8 +547,7 @@ static int wm8750_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8750_priv *wm8750 = snd_soc_codec_get_drvdata(codec); u16 iface = snd_soc_read(codec, WM8750_IFACE) & 0x1f3; u16 srate = snd_soc_read(codec, WM8750_SRATE) & 0x1c0; diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index e27e7b62b365..a26482cd7654 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -931,8 +931,7 @@ static int wm8753_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec); u16 voice = snd_soc_read(codec, WM8753_PCM) & 0x01f3; u16 srate = snd_soc_read(codec, WM8753_SRATE1) & 0x017f; @@ -1161,8 +1160,7 @@ static int wm8753_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec); u16 srate = snd_soc_read(codec, WM8753_SRATE1) & 0x01c0; u16 hifi = snd_soc_read(codec, WM8753_HIFI) & 0x01f3; diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index f18c554efc98..077c9628c70d 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -610,8 +610,7 @@ static int wm8900_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; u16 reg; reg = snd_soc_read(codec, WM8900_REG_AUDIO1) & ~0x60; diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index c91fb2f99c13..86b8a2926591 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1432,8 +1432,7 @@ static int wm8903_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec =rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); int fs = params_rate(params); int bclk; diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index d2883affea3b..481a3d9cfe48 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -371,8 +371,7 @@ static int wm8940_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; u16 iface = snd_soc_read(codec, WM8940_IFACE) & 0xFD9F; u16 addcntrl = snd_soc_read(codec, WM8940_ADDCNTRL) & 0xFFF1; u16 companding = snd_soc_read(codec, diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 840d72086d04..8bc659d8dd2e 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -505,8 +505,7 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); u16 iface = snd_soc_read(codec, WM8960_IFACE1) & 0xfff3; int i; diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 15d467ff91b4..0cfce9999c89 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -1478,7 +1478,8 @@ static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); static int wm8962_dsp2_write_config(struct snd_soc_codec *codec) { - return 0; + return regcache_sync_region(codec->control_data, + WM8962_HDBASS_AI_1, WM8962_MAX_REGISTER); } static int wm8962_dsp2_set_enable(struct snd_soc_codec *codec, u16 val) @@ -1755,10 +1756,22 @@ SOC_DOUBLE_R_TLV("EQ4 Volume", WM8962_EQ3, WM8962_EQ23, SOC_DOUBLE_R_TLV("EQ5 Volume", WM8962_EQ3, WM8962_EQ23, WM8962_EQL_B5_GAIN_SHIFT, 31, 0, eq_tlv), +SOC_SINGLE("3D Switch", WM8962_THREED1, 0, 1, 0), +SND_SOC_BYTES_MASK("3D Coefficients", WM8962_THREED1, 4, WM8962_THREED_ENA), + +SOC_SINGLE("DF1 Switch", WM8962_DF1, 0, 1, 0), +SND_SOC_BYTES_MASK("DF1 Coefficients", WM8962_DF1, 7, WM8962_DF1_ENA), + +SOC_SINGLE("DRC Switch", WM8962_DRC_1, 0, 1, 0), +SND_SOC_BYTES_MASK("DRC Coefficients", WM8962_DRC_1, 5, WM8962_DRC_ENA), + WM8962_DSP2_ENABLE("VSS Switch", WM8962_VSS_ENA_SHIFT), +SND_SOC_BYTES("VSS Coefficients", WM8962_VSS_XHD2_1, 148), WM8962_DSP2_ENABLE("HPF1 Switch", WM8962_HPF1_ENA_SHIFT), WM8962_DSP2_ENABLE("HPF2 Switch", WM8962_HPF2_ENA_SHIFT), +SND_SOC_BYTES("HPF Coefficients", WM8962_LHPF2, 1), WM8962_DSP2_ENABLE("HD Bass Switch", WM8962_HDBASS_ENA_SHIFT), +SND_SOC_BYTES("HD Bass Coefficients", WM8962_HDBASS_AI_1, 30), }; static const struct snd_kcontrol_new wm8962_spk_mono_controls[] = { @@ -2519,8 +2532,7 @@ static int wm8962_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); int i; int aif0 = 0; diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index 28fe59e3ce01..eef783f6b6d6 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -478,8 +478,7 @@ static int wm8971_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8971_priv *wm8971 = snd_soc_codec_get_drvdata(codec); u16 iface = snd_soc_read(codec, WM8971_IFACE) & 0x1f3; u16 srate = snd_soc_read(codec, WM8971_SRATE) & 0x1c0; diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index 72d5fdcd3cc2..a5be3adecf75 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -723,8 +723,7 @@ static int wm8978_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8978_priv *wm8978 = snd_soc_codec_get_drvdata(codec); /* Word length mask = 0x60 */ u16 iface_ctl = snd_soc_read(codec, WM8978_AUDIO_INTERFACE) & ~0x60; diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c index 6cdf6a2bc283..1d4c5cf47b06 100644 --- a/sound/soc/codecs/wm8988.c +++ b/sound/soc/codecs/wm8988.c @@ -668,8 +668,7 @@ static int wm8988_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8988_priv *wm8988 = snd_soc_codec_get_drvdata(codec); u16 iface = snd_soc_read(codec, WM8988_IFACE) & 0x1f3; u16 srate = snd_soc_read(codec, WM8988_SRATE) & 0x180; diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 9d242351e6e8..db63c97ddf51 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -1112,8 +1112,7 @@ static int wm8990_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; u16 audio1 = snd_soc_read(codec, WM8990_AUDIO_INTERFACE_1); audio1 &= ~WM8990_AIF_WL_MASK; diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index d256a9340644..36acfccab999 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -218,7 +218,6 @@ struct wm8993_priv { unsigned int sysclk_rate; unsigned int fs; unsigned int bclk; - int class_w_users; unsigned int fll_fref; unsigned int fll_fout; int fll_src; @@ -824,84 +823,6 @@ static int clk_sys_event(struct snd_soc_dapm_widget *w, return 0; } -/* - * When used with DAC outputs only the WM8993 charge pump supports - * operation in class W mode, providing very low power consumption - * when used with digital sources. Enable and disable this mode - * automatically depending on the mixer configuration. - * - * Currently the only supported paths are the direct DAC->headphone - * paths (which provide minimum power consumption anyway). - */ -static int class_w_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); - struct snd_soc_dapm_widget *widget = wlist->widgets[0]; - struct snd_soc_codec *codec = widget->codec; - struct wm8993_priv *wm8993 = snd_soc_codec_get_drvdata(codec); - int ret; - - /* Turn it off if we're using the main output mixer */ - if (ucontrol->value.integer.value[0] == 0) { - if (wm8993->class_w_users == 0) { - dev_dbg(codec->dev, "Disabling Class W\n"); - snd_soc_update_bits(codec, WM8993_CLASS_W_0, - WM8993_CP_DYN_FREQ | - WM8993_CP_DYN_V, - 0); - } - wm8993->class_w_users++; - wm8993->hubs_data.class_w = true; - } - - /* Implement the change */ - ret = snd_soc_dapm_put_enum_double(kcontrol, ucontrol); - - /* Enable it if we're using the direct DAC path */ - if (ucontrol->value.integer.value[0] == 1) { - if (wm8993->class_w_users == 1) { - dev_dbg(codec->dev, "Enabling Class W\n"); - snd_soc_update_bits(codec, WM8993_CLASS_W_0, - WM8993_CP_DYN_FREQ | - WM8993_CP_DYN_V, - WM8993_CP_DYN_FREQ | - WM8993_CP_DYN_V); - } - wm8993->class_w_users--; - wm8993->hubs_data.class_w = false; - } - - dev_dbg(codec->dev, "Indirect DAC use count now %d\n", - wm8993->class_w_users); - - return ret; -} - -#define SOC_DAPM_ENUM_W(xname, xenum) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ - .info = snd_soc_info_enum_double, \ - .get = snd_soc_dapm_get_enum_double, \ - .put = class_w_put, \ - .private_value = (unsigned long)&xenum } - -static const char *hp_mux_text[] = { - "Mixer", - "DAC", -}; - -static const struct soc_enum hpl_enum = - SOC_ENUM_SINGLE(WM8993_OUTPUT_MIXER1, 8, 2, hp_mux_text); - -static const struct snd_kcontrol_new hpl_mux = - SOC_DAPM_ENUM_W("Left Headphone Mux", hpl_enum); - -static const struct soc_enum hpr_enum = - SOC_ENUM_SINGLE(WM8993_OUTPUT_MIXER2, 8, 2, hp_mux_text); - -static const struct snd_kcontrol_new hpr_mux = - SOC_DAPM_ENUM_W("Right Headphone Mux", hpr_enum); - static const struct snd_kcontrol_new left_speaker_mixer[] = { SOC_DAPM_SINGLE("Input Switch", WM8993_SPEAKER_MIXER, 7, 1, 0), SOC_DAPM_SINGLE("IN1LP Switch", WM8993_SPEAKER_MIXER, 5, 1, 0), @@ -988,8 +909,8 @@ SND_SOC_DAPM_MUX("DACR Sidetone", SND_SOC_NOPM, 0, 0, &sidetoner_mux), SND_SOC_DAPM_DAC("DACL", NULL, WM8993_POWER_MANAGEMENT_3, 1, 0), SND_SOC_DAPM_DAC("DACR", NULL, WM8993_POWER_MANAGEMENT_3, 0, 0), -SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0, &hpl_mux), -SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0, &hpr_mux), +SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0, &wm_hubs_hpl_mux), +SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0, &wm_hubs_hpr_mux), SND_SOC_DAPM_MIXER("SPKL", WM8993_POWER_MANAGEMENT_3, 8, 0, left_speaker_mixer, ARRAY_SIZE(left_speaker_mixer)), @@ -1579,9 +1500,6 @@ static int wm8993_probe(struct snd_soc_codec *codec) return ret; } - /* By default we're using the output mixers */ - wm8993->class_w_users = 2; - /* Latch volume update bits and default ZC on */ snd_soc_update_bits(codec, WM8993_RIGHT_DAC_DIGITAL_VOLUME, WM8993_DAC_VU, WM8993_DAC_VU); diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 2de12ebe43b5..993639d694ce 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -70,8 +70,8 @@ static const struct wm8958_micd_rate micdet_rates[] = { static const struct wm8958_micd_rate jackdet_rates[] = { { 32768, true, 0, 1 }, { 32768, false, 0, 1 }, - { 44100 * 256, true, 7, 10 }, - { 44100 * 256, false, 7, 10 }, + { 44100 * 256, true, 10, 10 }, + { 44100 * 256, false, 7, 8 }, }; static void wm8958_micd_set_rate(struct snd_soc_codec *codec) @@ -82,7 +82,8 @@ static void wm8958_micd_set_rate(struct snd_soc_codec *codec) const struct wm8958_micd_rate *rates; int num_rates; - if (wm8994->jack_cb != wm8958_default_micdet) + if (!(wm8994->pdata && wm8994->pdata->micd_rates) && + wm8994->jack_cb != wm8958_default_micdet) return; idle = !wm8994->jack_mic; @@ -118,6 +119,10 @@ static void wm8958_micd_set_rate(struct snd_soc_codec *codec) val = rates[best].start << WM8958_MICD_BIAS_STARTTIME_SHIFT | rates[best].rate << WM8958_MICD_RATE_SHIFT; + dev_dbg(codec->dev, "MICD rate %d,%d for %dHz %s\n", + rates[best].start, rates[best].rate, sysclk, + idle ? "idle" : "active"); + snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, WM8958_MICD_BIAS_STARTTIME_MASK | WM8958_MICD_RATE_MASK, val); @@ -398,7 +403,7 @@ static void wm8994_set_retune_mobile(struct snd_soc_codec *codec, int block) wm8994->dac_rates[iface]); /* The EQ will be disabled while reconfiguring it, remember the - * current configuration. + * current configuration. */ save = snd_soc_read(codec, base); save &= WM8994_AIF1DAC1_EQ_ENA; @@ -689,6 +694,9 @@ static void wm1811_jackdet_set_mode(struct snd_soc_codec *codec, u16 mode) if (!wm8994->jackdet || !wm8994->jack_cb) return; + if (!wm8994->jackdet || !wm8994->jack_cb) + return; + if (wm8994->active_refcount) mode = WM1811_JACKDET_MODE_AUDIO; @@ -784,7 +792,7 @@ static void vmid_reference(struct snd_soc_codec *codec) switch (wm8994->vmid_mode) { default: - WARN_ON(0 == "Invalid VMID mode"); + WARN_ON(NULL == "Invalid VMID mode"); case WM8994_VMID_NORMAL: /* Startup bias, VMID ramp & buffer */ snd_soc_update_bits(codec, WM8994_ANTIPOP_2, @@ -937,27 +945,12 @@ static int vmid_event(struct snd_soc_dapm_widget *w, return 0; } -static void wm8994_update_class_w(struct snd_soc_codec *codec) +static bool wm8994_check_class_w_digital(struct snd_soc_codec *codec) { - struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); - int enable = 1; int source = 0; /* GCC flow analysis can't track enable */ int reg, reg_r; - /* Only support direct DAC->headphone paths */ - reg = snd_soc_read(codec, WM8994_OUTPUT_MIXER_1); - if (!(reg & WM8994_DAC1L_TO_HPOUT1L)) { - dev_vdbg(codec->dev, "HPL connected to output mixer\n"); - enable = 0; - } - - reg = snd_soc_read(codec, WM8994_OUTPUT_MIXER_2); - if (!(reg & WM8994_DAC1R_TO_HPOUT1R)) { - dev_vdbg(codec->dev, "HPR connected to output mixer\n"); - enable = 0; - } - - /* We also need the same setting for L/R and only one path */ + /* We also need the same AIF source for L/R and only one path */ reg = snd_soc_read(codec, WM8994_DAC1_LEFT_MIXER_ROUTING); switch (reg) { case WM8994_AIF2DACL_TO_DAC1L: @@ -974,30 +967,20 @@ static void wm8994_update_class_w(struct snd_soc_codec *codec) break; default: dev_vdbg(codec->dev, "DAC mixer setting: %x\n", reg); - enable = 0; - break; + return false; } reg_r = snd_soc_read(codec, WM8994_DAC1_RIGHT_MIXER_ROUTING); if (reg_r != reg) { dev_vdbg(codec->dev, "Left and right DAC mixers different\n"); - enable = 0; + return false; } - if (enable) { - dev_dbg(codec->dev, "Class W enabled\n"); - snd_soc_update_bits(codec, WM8994_CLASS_W_1, - WM8994_CP_DYN_PWR | - WM8994_CP_DYN_SRC_SEL_MASK, - source | WM8994_CP_DYN_PWR); - wm8994->hubs.class_w = true; - - } else { - dev_dbg(codec->dev, "Class W disabled\n"); - snd_soc_update_bits(codec, WM8994_CLASS_W_1, - WM8994_CP_DYN_PWR, 0); - wm8994->hubs.class_w = false; - } + /* Set the source up */ + snd_soc_update_bits(codec, WM8994_CLASS_W_1, + WM8994_CP_DYN_SRC_SEL_MASK, source); + + return true; } static int aif1clk_ev(struct snd_soc_dapm_widget *w, @@ -1280,45 +1263,6 @@ static int dac_ev(struct snd_soc_dapm_widget *w, return 0; } -static const char *hp_mux_text[] = { - "Mixer", - "DAC", -}; - -#define WM8994_HP_ENUM(xname, xenum) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ - .info = snd_soc_info_enum_double, \ - .get = snd_soc_dapm_get_enum_double, \ - .put = wm8994_put_hp_enum, \ - .private_value = (unsigned long)&xenum } - -static int wm8994_put_hp_enum(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); - struct snd_soc_dapm_widget *w = wlist->widgets[0]; - struct snd_soc_codec *codec = w->codec; - int ret; - - ret = snd_soc_dapm_put_enum_double(kcontrol, ucontrol); - - wm8994_update_class_w(codec); - - return ret; -} - -static const struct soc_enum hpl_enum = - SOC_ENUM_SINGLE(WM8994_OUTPUT_MIXER_1, 8, 2, hp_mux_text); - -static const struct snd_kcontrol_new hpl_mux = - WM8994_HP_ENUM("Left Headphone Mux", hpl_enum); - -static const struct soc_enum hpr_enum = - SOC_ENUM_SINGLE(WM8994_OUTPUT_MIXER_2, 8, 2, hp_mux_text); - -static const struct snd_kcontrol_new hpr_mux = - WM8994_HP_ENUM("Right Headphone Mux", hpr_enum); - static const char *adc_mux_text[] = { "ADC", "DMIC", @@ -1430,7 +1374,7 @@ static int wm8994_put_class_w(struct snd_kcontrol *kcontrol, ret = snd_soc_dapm_put_volsw(kcontrol, ucontrol); - wm8994_update_class_w(codec); + wm_hubs_update_class_w(codec); return ret; } @@ -1524,7 +1468,7 @@ static const struct snd_kcontrol_new wm8958_aif3adc_mux = SOC_DAPM_ENUM("AIF3ADC Mux", wm8958_aif3adc_enum); static const char *mono_pcm_out_text[] = { - "None", "AIF2ADCL", "AIF2ADCR", + "None", "AIF2ADCL", "AIF2ADCR", }; static const struct soc_enum mono_pcm_out_enum = @@ -1573,9 +1517,9 @@ SND_SOC_DAPM_MIXER_E("SPKL", WM8994_POWER_MANAGEMENT_3, 8, 0, SND_SOC_DAPM_MIXER_E("SPKR", WM8994_POWER_MANAGEMENT_3, 9, 0, right_speaker_mixer, ARRAY_SIZE(right_speaker_mixer), late_enable_ev, SND_SOC_DAPM_PRE_PMU), -SND_SOC_DAPM_MUX_E("Left Headphone Mux", SND_SOC_NOPM, 0, 0, &hpl_mux, +SND_SOC_DAPM_MUX_E("Left Headphone Mux", SND_SOC_NOPM, 0, 0, &wm_hubs_hpl_mux, late_enable_ev, SND_SOC_DAPM_PRE_PMU), -SND_SOC_DAPM_MUX_E("Right Headphone Mux", SND_SOC_NOPM, 0, 0, &hpr_mux, +SND_SOC_DAPM_MUX_E("Right Headphone Mux", SND_SOC_NOPM, 0, 0, &wm_hubs_hpr_mux, late_enable_ev, SND_SOC_DAPM_PRE_PMU), SND_SOC_DAPM_POST("Late Disable PGA", late_disable_ev) @@ -1591,8 +1535,8 @@ SND_SOC_DAPM_MIXER("SPKL", WM8994_POWER_MANAGEMENT_3, 8, 0, left_speaker_mixer, ARRAY_SIZE(left_speaker_mixer)), SND_SOC_DAPM_MIXER("SPKR", WM8994_POWER_MANAGEMENT_3, 9, 0, right_speaker_mixer, ARRAY_SIZE(right_speaker_mixer)), -SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0, &hpl_mux), -SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0, &hpr_mux), +SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0, &wm_hubs_hpl_mux), +SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0, &wm_hubs_hpr_mux), }; static const struct snd_soc_dapm_widget wm8994_dac_revd_widgets[] = { @@ -1732,6 +1676,7 @@ SND_SOC_DAPM_MUX("AIF3ADC Mux", SND_SOC_NOPM, 0, 0, &wm8994_aif3adc_mux), }; static const struct snd_soc_dapm_widget wm8958_dapm_widgets[] = { +SND_SOC_DAPM_SUPPLY("AIF3", WM8994_POWER_MANAGEMENT_6, 5, 1, NULL, 0), SND_SOC_DAPM_MUX("Mono PCM Out Mux", SND_SOC_NOPM, 0, 0, &mono_pcm_out_mux), SND_SOC_DAPM_MUX("AIF2DACL Mux", SND_SOC_NOPM, 0, 0, &aif2dacl_src_mux), SND_SOC_DAPM_MUX("AIF2DACR Mux", SND_SOC_NOPM, 0, 0, &aif2dacr_src_mux), @@ -1972,6 +1917,9 @@ static const struct snd_soc_dapm_route wm8958_intercon[] = { { "AIF2DACR Mux", "AIF2", "AIF2DAC Mux" }, { "AIF2DACR Mux", "AIF3", "AIF3DACDAT" }, + { "AIF3DACDAT", NULL, "AIF3" }, + { "AIF3ADCDAT", NULL, "AIF3" }, + { "Mono PCM Out Mux", "AIF2ADCL", "AIF2ADCL" }, { "Mono PCM Out Mux", "AIF2ADCR", "AIF2ADCR" }, @@ -2068,24 +2016,20 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, struct wm8994 *control = wm8994->wm8994; int reg_offset, ret; struct fll_div fll; - u16 reg, aif1, aif2; + u16 reg, clk1, aif_reg, aif_src; unsigned long timeout; bool was_enabled; - aif1 = snd_soc_read(codec, WM8994_AIF1_CLOCKING_1) - & WM8994_AIF1CLK_ENA; - - aif2 = snd_soc_read(codec, WM8994_AIF2_CLOCKING_1) - & WM8994_AIF2CLK_ENA; - switch (id) { case WM8994_FLL1: reg_offset = 0; id = 0; + aif_src = 0x10; break; case WM8994_FLL2: reg_offset = 0x20; id = 1; + aif_src = 0x18; break; default: return -EINVAL; @@ -2127,16 +2071,33 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, if (ret < 0) return ret; - /* Gate the AIF clocks while we reclock */ - snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1, - WM8994_AIF1CLK_ENA, 0); - snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1, - WM8994_AIF2CLK_ENA, 0); + /* Make sure that we're not providing SYSCLK right now */ + clk1 = snd_soc_read(codec, WM8994_CLOCKING_1); + if (clk1 & WM8994_SYSCLK_SRC) + aif_reg = WM8994_AIF2_CLOCKING_1; + else + aif_reg = WM8994_AIF1_CLOCKING_1; + reg = snd_soc_read(codec, aif_reg); + + if ((reg & WM8994_AIF1CLK_ENA) && + (reg & WM8994_AIF1CLK_SRC_MASK) == aif_src) { + dev_err(codec->dev, "FLL%d is currently providing SYSCLK\n", + id + 1); + return -EBUSY; + } /* We always need to disable the FLL while reconfiguring */ snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_1 + reg_offset, WM8994_FLL1_ENA, 0); + if (wm8994->fll_byp && src == WM8994_FLL_SRC_BCLK && + freq_in == freq_out && freq_out) { + dev_dbg(codec->dev, "Bypassing FLL%d\n", id + 1); + snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_5 + reg_offset, + WM8958_FLL1_BYP, WM8958_FLL1_BYP); + goto out; + } + reg = (fll.outdiv << WM8994_FLL1_OUTDIV_SHIFT) | (fll.fll_fratio << WM8994_FLL1_FRATIO_SHIFT); snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_2 + reg_offset, @@ -2151,6 +2112,7 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, fll.n << WM8994_FLL1_N_SHIFT); snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_5 + reg_offset, + WM8958_FLL1_BYP | WM8994_FLL1_REFCLK_DIV_MASK | WM8994_FLL1_REFCLK_SRC_MASK, (fll.clk_ref_div << WM8994_FLL1_REFCLK_DIV_SHIFT) | @@ -2213,16 +2175,11 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, } } +out: wm8994->fll[id].in = freq_in; wm8994->fll[id].out = freq_out; wm8994->fll[id].src = src; - /* Enable any gated AIF clocks */ - snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1, - WM8994_AIF1CLK_ENA, aif1); - snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1, - WM8994_AIF2CLK_ENA, aif2); - configure_clock(codec); return 0; @@ -2290,7 +2247,7 @@ static int wm8994_set_dai_sysclk(struct snd_soc_dai *dai, case WM8994_SYSCLK_OPCLK: /* Special case - a division (times 10) is given and - * no effect on main clocking. + * no effect on main clocking. */ if (freq) { for (i = 0; i < ARRAY_SIZE(opclk_divs); i++) @@ -2792,33 +2749,6 @@ static int wm8994_aif3_hw_params(struct snd_pcm_substream *substream, return snd_soc_update_bits(codec, aif1_reg, WM8994_AIF1_WL_MASK, aif1); } -static void wm8994_aif_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_codec *codec = dai->codec; - int rate_reg = 0; - - switch (dai->id) { - case 1: - rate_reg = WM8994_AIF1_RATE; - break; - case 2: - rate_reg = WM8994_AIF2_RATE; - break; - default: - break; - } - - /* If the DAI is idle then configure the divider tree for the - * lowest output rate to save a little power if the clock is - * still active (eg, because it is system clock). - */ - if (rate_reg && !dai->playback_active && !dai->capture_active) - snd_soc_update_bits(codec, rate_reg, - WM8994_AIF1_SR_MASK | - WM8994_AIF1CLK_RATE_MASK, 0x9); -} - static int wm8994_aif_mute(struct snd_soc_dai *codec_dai, int mute) { struct snd_soc_codec *codec = codec_dai->codec; @@ -2860,10 +2790,6 @@ static int wm8994_set_tristate(struct snd_soc_dai *codec_dai, int tristate) reg = WM8994_AIF2_MASTER_SLAVE; mask = WM8994_AIF2_TRI; break; - case 3: - reg = WM8994_POWER_MANAGEMENT_6; - mask = WM8994_AIF3_TRI; - break; default: return -EINVAL; } @@ -2900,7 +2826,6 @@ static const struct snd_soc_dai_ops wm8994_aif1_dai_ops = { .set_sysclk = wm8994_set_dai_sysclk, .set_fmt = wm8994_set_dai_fmt, .hw_params = wm8994_hw_params, - .shutdown = wm8994_aif_shutdown, .digital_mute = wm8994_aif_mute, .set_pll = wm8994_set_fll, .set_tristate = wm8994_set_tristate, @@ -2910,7 +2835,6 @@ static const struct snd_soc_dai_ops wm8994_aif2_dai_ops = { .set_sysclk = wm8994_set_dai_sysclk, .set_fmt = wm8994_set_dai_fmt, .hw_params = wm8994_hw_params, - .shutdown = wm8994_aif_shutdown, .digital_mute = wm8994_aif_mute, .set_pll = wm8994_set_fll, .set_tristate = wm8994_set_tristate, @@ -2918,7 +2842,6 @@ static const struct snd_soc_dai_ops wm8994_aif2_dai_ops = { static const struct snd_soc_dai_ops wm8994_aif3_dai_ops = { .hw_params = wm8994_aif3_hw_params, - .set_tristate = wm8994_set_tristate, }; static struct snd_soc_dai_driver wm8994_dai[] = { @@ -3126,14 +3049,14 @@ static void wm8994_handle_retune_mobile_pdata(struct wm8994_priv *wm8994) /* Expand the array... */ t = krealloc(wm8994->retune_mobile_texts, - sizeof(char *) * + sizeof(char *) * (wm8994->num_retune_mobile_texts + 1), GFP_KERNEL); if (t == NULL) continue; /* ...store the new entry... */ - t[wm8994->num_retune_mobile_texts] = + t[wm8994->num_retune_mobile_texts] = pdata->retune_mobile_cfgs[i].name; /* ...and remember the new version. */ @@ -3304,25 +3227,25 @@ int wm8994_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, } EXPORT_SYMBOL_GPL(wm8994_mic_detect); -static irqreturn_t wm8994_mic_irq(int irq, void *data) +static void wm8994_mic_work(struct work_struct *work) { - struct wm8994_priv *priv = data; - struct snd_soc_codec *codec = priv->codec; - int reg; + struct wm8994_priv *priv = container_of(work, + struct wm8994_priv, + mic_work.work); + struct regmap *regmap = priv->wm8994->regmap; + struct device *dev = priv->wm8994->dev; + unsigned int reg; + int ret; int report; -#ifndef CONFIG_SND_SOC_WM8994_MODULE - trace_snd_soc_jack_irq(dev_name(codec->dev)); -#endif - - reg = snd_soc_read(codec, WM8994_INTERRUPT_RAW_STATUS_2); - if (reg < 0) { - dev_err(codec->dev, "Failed to read microphone status: %d\n", - reg); - return IRQ_HANDLED; + ret = regmap_read(regmap, WM8994_INTERRUPT_RAW_STATUS_2, ®); + if (ret < 0) { + dev_err(dev, "Failed to read microphone status: %d\n", + ret); + return; } - dev_dbg(codec->dev, "Microphone status: %x\n", reg); + dev_dbg(dev, "Microphone status: %x\n", reg); report = 0; if (reg & WM8994_MIC1_DET_STS) { @@ -3361,6 +3284,20 @@ static irqreturn_t wm8994_mic_irq(int irq, void *data) snd_soc_jack_report(priv->micdet[1].jack, report, SND_JACK_HEADSET | SND_JACK_BTN_0); +} + +static irqreturn_t wm8994_mic_irq(int irq, void *data) +{ + struct wm8994_priv *priv = data; + struct snd_soc_codec *codec = priv->codec; + +#ifndef CONFIG_SND_SOC_WM8994_MODULE + trace_snd_soc_jack_irq(dev_name(codec->dev)); +#endif + + pm_wakeup_event(codec->dev, 300); + + schedule_delayed_work(&priv->mic_work, msecs_to_jiffies(250)); return IRQ_HANDLED; } @@ -3415,9 +3352,6 @@ static void wm8958_default_micdet(u16 status, void *data) wm8958_micd_set_rate(codec); - snd_soc_jack_report(wm8994->micdet[0].jack, SND_JACK_HEADPHONE, - SND_JACK_HEADSET); - /* If we have jackdet that will detect removal */ if (wm8994->jackdet) { mutex_lock(&wm8994->accdet_lock); @@ -3430,14 +3364,13 @@ static void wm8958_default_micdet(u16 status, void *data) mutex_unlock(&wm8994->accdet_lock); - if (wm8994->pdata->jd_ext_cap) { - mutex_lock(&codec->mutex); + if (wm8994->pdata->jd_ext_cap) snd_soc_dapm_disable_pin(&codec->dapm, "MICBIAS2"); - snd_soc_dapm_sync(&codec->dapm); - mutex_unlock(&codec->mutex); - } } + + snd_soc_jack_report(wm8994->micdet[0].jack, SND_JACK_HEADPHONE, + SND_JACK_HEADSET); } /* Report short circuit as a button */ @@ -3489,6 +3422,8 @@ static irqreturn_t wm1811_jackdet_irq(int irq, void *data) if (present) { dev_dbg(codec->dev, "Jack detected\n"); + wm8958_micd_set_rate(codec); + snd_soc_update_bits(codec, WM8958_MICBIAS2, WM8958_MICB2_DISCH, 0); @@ -3526,16 +3461,11 @@ static irqreturn_t wm1811_jackdet_irq(int irq, void *data) /* If required for an external cap force MICBIAS on */ if (wm8994->pdata->jd_ext_cap) { - mutex_lock(&codec->mutex); - if (present) snd_soc_dapm_force_enable_pin(&codec->dapm, "MICBIAS2"); else snd_soc_dapm_disable_pin(&codec->dapm, "MICBIAS2"); - - snd_soc_dapm_sync(&codec->dapm); - mutex_unlock(&codec->mutex); } if (present) @@ -3740,6 +3670,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) wm8994->codec = codec; mutex_init(&wm8994->accdet_lock); + INIT_DELAYED_WORK(&wm8994->mic_work, wm8994_mic_work); for (i = 0; i < ARRAY_SIZE(wm8994->fll_locked); i++) init_completion(&wm8994->fll_locked[i]); @@ -3783,13 +3714,22 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) case WM8958: wm8994->hubs.dcs_readback_mode = 1; wm8994->hubs.hp_startup_mode = 1; + + switch (wm8994->revision) { + case 0: + break; + default: + wm8994->fll_byp = true; + break; + } break; case WM1811: wm8994->hubs.dcs_readback_mode = 2; wm8994->hubs.no_series_update = 1; wm8994->hubs.hp_startup_mode = 1; - wm8994->hubs.no_cache_class_w = true; + wm8994->hubs.no_cache_dac_hp_direct = true; + wm8994->fll_byp = true; switch (wm8994->revision) { case 0: @@ -4010,7 +3950,8 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) break; } - wm8994_update_class_w(codec); + wm8994->hubs.check_class_w_digital = wm8994_check_class_w_digital; + wm_hubs_update_class_w(codec); wm8994_handle_pdata(wm8994); @@ -4075,7 +4016,6 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) ARRAY_SIZE(wm8994_dac_widgets)); break; } - wm_hubs_add_analogue_routes(codec, 0, 0); snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); @@ -4140,7 +4080,7 @@ err_irq: return ret; } -static int wm8994_codec_remove(struct snd_soc_codec *codec) +static int wm8994_codec_remove(struct snd_soc_codec *codec) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); struct wm8994 *control = wm8994->wm8994; @@ -4181,14 +4121,10 @@ static int wm8994_codec_remove(struct snd_soc_codec *codec) free_irq(wm8994->micdet_irq, wm8994); break; } - if (wm8994->mbc) - release_firmware(wm8994->mbc); - if (wm8994->mbc_vss) - release_firmware(wm8994->mbc_vss); - if (wm8994->enh_eq) - release_firmware(wm8994->enh_eq); + release_firmware(wm8994->mbc); + release_firmware(wm8994->mbc_vss); + release_firmware(wm8994->enh_eq); kfree(wm8994->retune_mobile_texts); - return 0; } diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h index c724112998d8..d77e06f0a675 100644 --- a/sound/soc/codecs/wm8994.h +++ b/sound/soc/codecs/wm8994.h @@ -12,6 +12,7 @@ #include <sound/soc.h> #include <linux/firmware.h> #include <linux/completion.h> +#include <linux/workqueue.h> #include "wm_hubs.h" @@ -79,6 +80,7 @@ struct wm8994_priv { struct wm8994_fll_config fll[2], fll_suspend[2]; struct completion fll_locked[2]; bool fll_locked_irq; + bool fll_byp; int vmid_refcount; int active_refcount; @@ -126,6 +128,7 @@ struct wm8994_priv { struct mutex accdet_lock; struct wm8994_micdet micdet[2]; + struct delayed_work mic_work; bool mic_detecting; bool jack_mic; int btn_mask; diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 1fd635494045..8af422e38fd0 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -1770,7 +1770,13 @@ static int wm8996_set_bias_level(struct snd_soc_codec *codec, switch (level) { case SND_SOC_BIAS_ON: + break; case SND_SOC_BIAS_PREPARE: + /* Put the MICBIASes into regulating mode */ + snd_soc_update_bits(codec, WM8996_MICBIAS_1, + WM8996_MICB1_MODE, 0); + snd_soc_update_bits(codec, WM8996_MICBIAS_2, + WM8996_MICB2_MODE, 0); break; case SND_SOC_BIAS_STANDBY: @@ -1793,6 +1799,12 @@ static int wm8996_set_bias_level(struct snd_soc_codec *codec, regcache_cache_only(codec->control_data, false); regcache_sync(codec->control_data); } + + /* Bypass the MICBIASes for lowest power */ + snd_soc_update_bits(codec, WM8996_MICBIAS_1, + WM8996_MICB1_MODE, WM8996_MICB1_MODE); + snd_soc_update_bits(codec, WM8996_MICBIAS_2, + WM8996_MICB2_MODE, WM8996_MICB2_MODE); break; case SND_SOC_BIAS_OFF: diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 076c126ed9b1..9328270df16c 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -774,7 +774,7 @@ static const struct snd_soc_dapm_widget wm9081_dapm_widgets[] = { SND_SOC_DAPM_INPUT("IN1"), SND_SOC_DAPM_INPUT("IN2"), -SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM9081_POWER_MANAGEMENT, 0, 0), +SND_SOC_DAPM_DAC("DAC", NULL, WM9081_POWER_MANAGEMENT, 0, 0), SND_SOC_DAPM_MIXER_NAMED_CTL("Mixer", SND_SOC_NOPM, 0, 0, mixer, ARRAY_SIZE(mixer)), @@ -799,6 +799,7 @@ SND_SOC_DAPM_SUPPLY("TSENSE", WM9081_POWER_MANAGEMENT, 7, 0, NULL, 0), static const struct snd_soc_dapm_route wm9081_audio_paths[] = { { "DAC", NULL, "CLK_SYS" }, { "DAC", NULL, "CLK_DSP" }, + { "DAC", NULL, "AIF" }, { "Mixer", "IN1 Switch", "IN1" }, { "Mixer", "IN2 Switch", "IN2" }, @@ -1252,7 +1253,7 @@ static const struct snd_soc_dai_ops wm9081_dai_ops = { static struct snd_soc_dai_driver wm9081_dai = { .name = "wm9081-hifi", .playback = { - .stream_name = "HiFi Playback", + .stream_name = "AIF", .channels_min = 1, .channels_max = 2, .rates = WM9081_RATES, diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index cacc6a86b46f..e8e782a0c78d 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -236,9 +236,7 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, static int ac97_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; int reg; u16 vra; @@ -250,7 +248,7 @@ static int ac97_prepare(struct snd_pcm_substream *substream, else reg = AC97_PCM_LR_ADC_RATE; - return ac97_write(codec, reg, runtime->rate); + return ac97_write(codec, reg, substream->runtime->rate); } #define WM9705_AC97_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | \ diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index b342ae50bcd6..a1541414d904 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -467,11 +467,10 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, static int ac97_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec =rtd->codec; + struct snd_soc_codec *codec = dai->codec; int reg; u16 vra; + struct snd_pcm_runtime *runtime = substream->runtime; vra = ac97_read(codec, AC97_EXTENDED_STATUS); ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1); @@ -487,10 +486,9 @@ static int ac97_prepare(struct snd_pcm_substream *substream, static int ac97_aux_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; u16 vra, xsle; + struct snd_pcm_runtime *runtime = substream->runtime; vra = ac97_read(codec, AC97_EXTENDED_STATUS); ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1); diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 6c028c470601..dfe957a47f29 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -109,12 +109,103 @@ irqreturn_t wm_hubs_dcs_done(int irq, void *data) } EXPORT_SYMBOL_GPL(wm_hubs_dcs_done); +static bool wm_hubs_dac_hp_direct(struct snd_soc_codec *codec) +{ + int reg; + + /* If we're going via the mixer we'll need to do additional checks */ + reg = snd_soc_read(codec, WM8993_OUTPUT_MIXER1); + if (!(reg & WM8993_DACL_TO_HPOUT1L)) { + if (reg & ~WM8993_DACL_TO_MIXOUTL) { + dev_vdbg(codec->dev, "Analogue paths connected: %x\n", + reg & ~WM8993_DACL_TO_HPOUT1L); + return false; + } else { + dev_vdbg(codec->dev, "HPL connected to mixer\n"); + } + } else { + dev_vdbg(codec->dev, "HPL connected to DAC\n"); + } + + reg = snd_soc_read(codec, WM8993_OUTPUT_MIXER2); + if (!(reg & WM8993_DACR_TO_HPOUT1R)) { + if (reg & ~WM8993_DACR_TO_MIXOUTR) { + dev_vdbg(codec->dev, "Analogue paths connected: %x\n", + reg & ~WM8993_DACR_TO_HPOUT1R); + return false; + } else { + dev_vdbg(codec->dev, "HPR connected to mixer\n"); + } + } else { + dev_vdbg(codec->dev, "HPR connected to DAC\n"); + } + + return true; +} + +struct wm_hubs_dcs_cache { + struct list_head list; + unsigned int left; + unsigned int right; + u16 dcs_cfg; +}; + +static bool wm_hubs_dcs_cache_get(struct snd_soc_codec *codec, + struct wm_hubs_dcs_cache **entry) +{ + struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); + struct wm_hubs_dcs_cache *cache; + unsigned int left, right; + + left = snd_soc_read(codec, WM8993_LEFT_OUTPUT_VOLUME); + left &= WM8993_HPOUT1L_VOL_MASK; + + right = snd_soc_read(codec, WM8993_RIGHT_OUTPUT_VOLUME); + right &= WM8993_HPOUT1R_VOL_MASK; + + list_for_each_entry(cache, &hubs->dcs_cache, list) { + if (cache->left != left || cache->right != right) + continue; + + *entry = cache; + return true; + } + + return false; +} + +static void wm_hubs_dcs_cache_set(struct snd_soc_codec *codec, u16 dcs_cfg) +{ + struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); + struct wm_hubs_dcs_cache *cache; + + if (hubs->no_cache_dac_hp_direct) + return; + + cache = devm_kzalloc(codec->dev, sizeof(*cache), GFP_KERNEL); + if (!cache) { + dev_err(codec->dev, "Failed to allocate DCS cache entry\n"); + return; + } + + cache->left = snd_soc_read(codec, WM8993_LEFT_OUTPUT_VOLUME); + cache->left &= WM8993_HPOUT1L_VOL_MASK; + + cache->right = snd_soc_read(codec, WM8993_RIGHT_OUTPUT_VOLUME); + cache->right &= WM8993_HPOUT1R_VOL_MASK; + + cache->dcs_cfg = dcs_cfg; + + list_add_tail(&cache->list, &hubs->dcs_cache); +} + /* * Startup calibration of the DC servo */ static void calibrate_dc_servo(struct snd_soc_codec *codec) { struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); + struct wm_hubs_dcs_cache *cache; s8 offset; u16 reg, reg_l, reg_r, dcs_cfg, dcs_reg; @@ -129,10 +220,11 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec) /* If we're using a digital only path and have a previously * callibrated DC servo offset stored then use that. */ - if (hubs->class_w && hubs->class_w_dcs) { - dev_dbg(codec->dev, "Using cached DC servo offset %x\n", - hubs->class_w_dcs); - snd_soc_write(codec, dcs_reg, hubs->class_w_dcs); + if (wm_hubs_dac_hp_direct(codec) && + wm_hubs_dcs_cache_get(codec, &cache)) { + dev_dbg(codec->dev, "Using cached DCS offset %x for %d,%d\n", + cache->dcs_cfg, cache->left, cache->right); + snd_soc_write(codec, dcs_reg, cache->dcs_cfg); wait_for_dc_servo(codec, WM8993_DCS_TRIG_DAC_WR_0 | WM8993_DCS_TRIG_DAC_WR_1); @@ -207,8 +299,8 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec) /* Save the callibrated offset if we're in class W mode and * therefore don't have any analogue signal mixed in. */ - if (hubs->class_w && !hubs->no_cache_class_w) - hubs->class_w_dcs = dcs_cfg; + if (wm_hubs_dac_hp_direct(codec)) + wm_hubs_dcs_cache_set(codec, dcs_cfg); } /* @@ -223,9 +315,6 @@ static int wm8993_put_dc_servo(struct snd_kcontrol *kcontrol, ret = snd_soc_put_volsw(kcontrol, ucontrol); - /* Updating the analogue gains invalidates the DC servo cache */ - hubs->class_w_dcs = 0; - /* If we're applying an offset correction then updating the * callibration would be likely to introduce further offsets. */ if (hubs->dcs_codes_l || hubs->dcs_codes_r || hubs->no_series_update) @@ -530,6 +619,86 @@ static int lineout_event(struct snd_soc_dapm_widget *w, return 0; } +void wm_hubs_update_class_w(struct snd_soc_codec *codec) +{ + struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); + int enable = WM8993_CP_DYN_V | WM8993_CP_DYN_FREQ; + + if (!wm_hubs_dac_hp_direct(codec)) + enable = false; + + if (hubs->check_class_w_digital && !hubs->check_class_w_digital(codec)) + enable = false; + + dev_vdbg(codec->dev, "Class W %s\n", enable ? "enabled" : "disabled"); + + snd_soc_update_bits(codec, WM8993_CLASS_W_0, + WM8993_CP_DYN_V | WM8993_CP_DYN_FREQ, enable); +} +EXPORT_SYMBOL_GPL(wm_hubs_update_class_w); + +#define WM_HUBS_SINGLE_W(xname, reg, shift, max, invert) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = snd_soc_info_volsw, \ + .get = snd_soc_dapm_get_volsw, .put = class_w_put_volsw, \ + .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } + +static int class_w_put_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget *widget = wlist->widgets[0]; + struct snd_soc_codec *codec = widget->codec; + int ret; + + ret = snd_soc_dapm_put_volsw(kcontrol, ucontrol); + + wm_hubs_update_class_w(codec); + + return ret; +} + +#define WM_HUBS_ENUM_W(xname, xenum) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = snd_soc_info_enum_double, \ + .get = snd_soc_dapm_get_enum_double, \ + .put = class_w_put_double, \ + .private_value = (unsigned long)&xenum } + +static int class_w_put_double(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget *widget = wlist->widgets[0]; + struct snd_soc_codec *codec = widget->codec; + int ret; + + ret = snd_soc_dapm_put_enum_double(kcontrol, ucontrol); + + wm_hubs_update_class_w(codec); + + return ret; +} + +static const char *hp_mux_text[] = { + "Mixer", + "DAC", +}; + +static const struct soc_enum hpl_enum = + SOC_ENUM_SINGLE(WM8993_OUTPUT_MIXER1, 8, 2, hp_mux_text); + +const struct snd_kcontrol_new wm_hubs_hpl_mux = + WM_HUBS_ENUM_W("Left Headphone Mux", hpl_enum); +EXPORT_SYMBOL_GPL(wm_hubs_hpl_mux); + +static const struct soc_enum hpr_enum = + SOC_ENUM_SINGLE(WM8993_OUTPUT_MIXER2, 8, 2, hp_mux_text); + +const struct snd_kcontrol_new wm_hubs_hpr_mux = + WM_HUBS_ENUM_W("Right Headphone Mux", hpr_enum); +EXPORT_SYMBOL_GPL(wm_hubs_hpr_mux); + static const struct snd_kcontrol_new in1l_pga[] = { SOC_DAPM_SINGLE("IN1LP Switch", WM8993_INPUT_MIXER2, 5, 1, 0), SOC_DAPM_SINGLE("IN1LN Switch", WM8993_INPUT_MIXER2, 4, 1, 0), @@ -561,25 +730,25 @@ SOC_DAPM_SINGLE("IN1R Switch", WM8993_INPUT_MIXER4, 5, 1, 0), }; static const struct snd_kcontrol_new left_output_mixer[] = { -SOC_DAPM_SINGLE("Right Input Switch", WM8993_OUTPUT_MIXER1, 7, 1, 0), -SOC_DAPM_SINGLE("Left Input Switch", WM8993_OUTPUT_MIXER1, 6, 1, 0), -SOC_DAPM_SINGLE("IN2RN Switch", WM8993_OUTPUT_MIXER1, 5, 1, 0), -SOC_DAPM_SINGLE("IN2LN Switch", WM8993_OUTPUT_MIXER1, 4, 1, 0), -SOC_DAPM_SINGLE("IN2LP Switch", WM8993_OUTPUT_MIXER1, 1, 1, 0), -SOC_DAPM_SINGLE("IN1R Switch", WM8993_OUTPUT_MIXER1, 3, 1, 0), -SOC_DAPM_SINGLE("IN1L Switch", WM8993_OUTPUT_MIXER1, 2, 1, 0), -SOC_DAPM_SINGLE("DAC Switch", WM8993_OUTPUT_MIXER1, 0, 1, 0), +WM_HUBS_SINGLE_W("Right Input Switch", WM8993_OUTPUT_MIXER1, 7, 1, 0), +WM_HUBS_SINGLE_W("Left Input Switch", WM8993_OUTPUT_MIXER1, 6, 1, 0), +WM_HUBS_SINGLE_W("IN2RN Switch", WM8993_OUTPUT_MIXER1, 5, 1, 0), +WM_HUBS_SINGLE_W("IN2LN Switch", WM8993_OUTPUT_MIXER1, 4, 1, 0), +WM_HUBS_SINGLE_W("IN2LP Switch", WM8993_OUTPUT_MIXER1, 1, 1, 0), +WM_HUBS_SINGLE_W("IN1R Switch", WM8993_OUTPUT_MIXER1, 3, 1, 0), +WM_HUBS_SINGLE_W("IN1L Switch", WM8993_OUTPUT_MIXER1, 2, 1, 0), +WM_HUBS_SINGLE_W("DAC Switch", WM8993_OUTPUT_MIXER1, 0, 1, 0), }; static const struct snd_kcontrol_new right_output_mixer[] = { -SOC_DAPM_SINGLE("Left Input Switch", WM8993_OUTPUT_MIXER2, 7, 1, 0), -SOC_DAPM_SINGLE("Right Input Switch", WM8993_OUTPUT_MIXER2, 6, 1, 0), -SOC_DAPM_SINGLE("IN2LN Switch", WM8993_OUTPUT_MIXER2, 5, 1, 0), -SOC_DAPM_SINGLE("IN2RN Switch", WM8993_OUTPUT_MIXER2, 4, 1, 0), -SOC_DAPM_SINGLE("IN1L Switch", WM8993_OUTPUT_MIXER2, 3, 1, 0), -SOC_DAPM_SINGLE("IN1R Switch", WM8993_OUTPUT_MIXER2, 2, 1, 0), -SOC_DAPM_SINGLE("IN2RP Switch", WM8993_OUTPUT_MIXER2, 1, 1, 0), -SOC_DAPM_SINGLE("DAC Switch", WM8993_OUTPUT_MIXER2, 0, 1, 0), +WM_HUBS_SINGLE_W("Left Input Switch", WM8993_OUTPUT_MIXER2, 7, 1, 0), +WM_HUBS_SINGLE_W("Right Input Switch", WM8993_OUTPUT_MIXER2, 6, 1, 0), +WM_HUBS_SINGLE_W("IN2LN Switch", WM8993_OUTPUT_MIXER2, 5, 1, 0), +WM_HUBS_SINGLE_W("IN2RN Switch", WM8993_OUTPUT_MIXER2, 4, 1, 0), +WM_HUBS_SINGLE_W("IN1L Switch", WM8993_OUTPUT_MIXER2, 3, 1, 0), +WM_HUBS_SINGLE_W("IN1R Switch", WM8993_OUTPUT_MIXER2, 2, 1, 0), +WM_HUBS_SINGLE_W("IN2RP Switch", WM8993_OUTPUT_MIXER2, 1, 1, 0), +WM_HUBS_SINGLE_W("DAC Switch", WM8993_OUTPUT_MIXER2, 0, 1, 0), }; static const struct snd_kcontrol_new earpiece_mixer[] = { @@ -943,6 +1112,7 @@ int wm_hubs_add_analogue_routes(struct snd_soc_codec *codec, struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); struct snd_soc_dapm_context *dapm = &codec->dapm; + INIT_LIST_HEAD(&hubs->dcs_cache); init_completion(&hubs->dcs_done); snd_soc_dapm_add_routes(dapm, analogue_routes, diff --git a/sound/soc/codecs/wm_hubs.h b/sound/soc/codecs/wm_hubs.h index 5705276f4943..da2dc899ce6d 100644 --- a/sound/soc/codecs/wm_hubs.h +++ b/sound/soc/codecs/wm_hubs.h @@ -16,6 +16,8 @@ #include <linux/completion.h> #include <linux/interrupt.h> +#include <linux/list.h> +#include <sound/control.h> struct snd_soc_codec; @@ -30,9 +32,9 @@ struct wm_hubs_data { int series_startup; int no_series_update; - bool no_cache_class_w; - bool class_w; - u16 class_w_dcs; + bool no_cache_dac_hp_direct; + struct list_head dcs_cache; + bool (*check_class_w_digital)(struct snd_soc_codec *); bool lineout1_se; bool lineout1n_ena; @@ -58,5 +60,9 @@ extern irqreturn_t wm_hubs_dcs_done(int irq, void *data); extern void wm_hubs_vmid_ena(struct snd_soc_codec *codec); extern void wm_hubs_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level); +extern void wm_hubs_update_class_w(struct snd_soc_codec *codec); + +extern const struct snd_kcontrol_new wm_hubs_hpl_mux; +extern const struct snd_kcontrol_new wm_hubs_hpr_mux; #endif |