diff options
Diffstat (limited to 'sound/soc/codecs')
38 files changed, 3122 insertions, 358 deletions
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 5da30eb6ad0..83f5c67d3c4 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -22,9 +22,11 @@ config SND_SOC_ALL_CODECS select SND_SOC_AK4642 if I2C select SND_SOC_AK4671 if I2C select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC + select SND_SOC_CS42L51 if I2C select SND_SOC_CS4270 if I2C - select SND_SOC_MAX9877 if I2C select SND_SOC_DA7210 if I2C + select SND_SOC_JZ4740 if SOC_JZ4740 + select SND_SOC_MAX9877 if I2C select SND_SOC_PCM3008 select SND_SOC_SPDIF select SND_SOC_SSM2602 if I2C @@ -48,6 +50,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM8727 select SND_SOC_WM8728 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8731 if SND_SOC_I2C_AND_SPI + select SND_SOC_WM8741 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8750 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8753 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8776 if SND_SOC_I2C_AND_SPI @@ -120,13 +123,13 @@ config SND_SOC_AK4671 config SND_SOC_CQ0093VC tristate +config SND_SOC_CS42L51 + tristate + # Cirrus Logic CS4270 Codec config SND_SOC_CS4270 tristate -config SND_SOC_DA7210 - tristate - # Cirrus Logic CS4270 Codec VD = 3.3V Errata # Select if you are affected by the errata where the part will not function # if MCLK divide-by-1.5 is selected and VD is set to 3.3V. The driver will @@ -138,9 +141,15 @@ config SND_SOC_CS4270_VD33_ERRATA config SND_SOC_CX20442 tristate +config SND_SOC_JZ4740_CODEC + tristate + config SND_SOC_L3 tristate +config SND_SOC_DA7210 + tristate + config SND_SOC_PCM3008 tristate @@ -206,6 +215,9 @@ config SND_SOC_WM8728 config SND_SOC_WM8731 tristate +config SND_SOC_WM8741 + tristate + config SND_SOC_WM8750 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 91429eab070..53524095759 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -9,6 +9,7 @@ snd-soc-ak4535-objs := ak4535.o 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-cs4270-objs := cs4270.o snd-soc-cx20442-objs := cx20442.o snd-soc-da7210-objs := da7210.o @@ -34,6 +35,7 @@ snd-soc-wm8711-objs := wm8711.o snd-soc-wm8727-objs := wm8727.o snd-soc-wm8728-objs := wm8728.o snd-soc-wm8731-objs := wm8731.o +snd-soc-wm8741-objs := wm8741.o snd-soc-wm8750-objs := wm8750.o snd-soc-wm8753-objs := wm8753.o snd-soc-wm8776-objs := wm8776.o @@ -56,6 +58,7 @@ snd-soc-wm9705-objs := wm9705.o snd-soc-wm9712-objs := wm9712.o snd-soc-wm9713-objs := wm9713.o snd-soc-wm-hubs-objs := wm_hubs.o +snd-soc-jz4740-codec-objs := jz4740.o # Amp snd-soc-max9877-objs := max9877.o @@ -74,10 +77,12 @@ obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o +obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o +obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o @@ -99,6 +104,7 @@ obj-$(CONFIG_SND_SOC_WM8711) += snd-soc-wm8711.o obj-$(CONFIG_SND_SOC_WM8727) += snd-soc-wm8727.o obj-$(CONFIG_SND_SOC_WM8728) += snd-soc-wm8728.o obj-$(CONFIG_SND_SOC_WM8731) += snd-soc-wm8731.o +obj-$(CONFIG_SND_SOC_WM8741) += snd-soc-wm8741.o obj-$(CONFIG_SND_SOC_WM8750) += snd-soc-wm8750.o obj-$(CONFIG_SND_SOC_WM8753) += snd-soc-wm8753.o obj-$(CONFIG_SND_SOC_WM8776) += snd-soc-wm8776.o diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c index 21753842322..a01006c8c60 100644 --- a/sound/soc/codecs/ad1836.c +++ b/sound/soc/codecs/ad1836.c @@ -272,6 +272,7 @@ static int ad1836_register(struct ad1836_priv *ad1836) if (ad1836_codec) { dev_err(codec->dev, "Another ad1836 is registered\n"); + kfree(ad1836); return -EINVAL; } diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index c8ca1142b2f..1def75e4862 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -24,6 +24,7 @@ /* codec private data */ struct ad193x_priv { + unsigned int sysclk; struct snd_soc_codec codec; u8 reg_cache[AD193X_NUM_REGS]; }; @@ -251,15 +252,32 @@ static int ad193x_set_dai_fmt(struct snd_soc_dai *codec_dai, return 0; } +static int ad193x_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 ad193x_priv *ad193x = snd_soc_codec_get_drvdata(codec); + switch (freq) { + case 12288000: + case 18432000: + case 24576000: + case 36864000: + ad193x->sysclk = freq; + return 0; + } + return -EINVAL; +} + static int ad193x_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - int word_len = 0, reg = 0; + int word_len = 0, reg = 0, master_rate = 0; struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->card->codec; + struct ad193x_priv *ad193x = snd_soc_codec_get_drvdata(codec); /* bit size */ switch (params_format(params)) { @@ -275,6 +293,25 @@ static int ad193x_hw_params(struct snd_pcm_substream *substream, break; } + switch (ad193x->sysclk) { + case 12288000: + master_rate = AD193X_PLL_INPUT_256; + break; + case 18432000: + master_rate = AD193X_PLL_INPUT_384; + break; + case 24576000: + master_rate = AD193X_PLL_INPUT_512; + break; + case 36864000: + master_rate = AD193X_PLL_INPUT_768; + break; + } + + reg = snd_soc_read(codec, AD193X_PLL_CLK_CTRL0); + reg = (reg & AD193X_PLL_INPUT_MASK) | master_rate; + snd_soc_write(codec, AD193X_PLL_CLK_CTRL0, reg); + reg = snd_soc_read(codec, AD193X_DAC_CTRL2); reg = (reg & (~AD193X_DAC_WORD_LEN_MASK)) | word_len; snd_soc_write(codec, AD193X_DAC_CTRL2, reg); @@ -348,6 +385,7 @@ static int ad193x_bus_probe(struct device *dev, void *ctrl_data, int bus_type) /* pll input: mclki/xi */ snd_soc_write(codec, AD193X_PLL_CLK_CTRL0, 0x99); /* mclk=24.576Mhz: 0x9D; mclk=12.288Mhz: 0x99 */ snd_soc_write(codec, AD193X_PLL_CLK_CTRL1, 0x04); + ad193x->sysclk = 12288000; ret = snd_soc_register_codec(codec); if (ret != 0) { @@ -383,6 +421,7 @@ static struct snd_soc_dai_ops ad193x_dai_ops = { .hw_params = ad193x_hw_params, .digital_mute = ad193x_mute, .set_tdm_slot = ad193x_set_tdm_slot, + .set_sysclk = ad193x_set_dai_sysclk, .set_fmt = ad193x_set_dai_fmt, }; diff --git a/sound/soc/codecs/ad193x.h b/sound/soc/codecs/ad193x.h index a03c880d52f..654ba64ae04 100644 --- a/sound/soc/codecs/ad193x.h +++ b/sound/soc/codecs/ad193x.h @@ -11,6 +11,11 @@ #define AD193X_PLL_CLK_CTRL0 0x800 #define AD193X_PLL_POWERDOWN 0x01 +#define AD193X_PLL_INPUT_MASK (~0x6) +#define AD193X_PLL_INPUT_256 (0 << 1) +#define AD193X_PLL_INPUT_384 (1 << 1) +#define AD193X_PLL_INPUT_512 (2 << 1) +#define AD193X_PLL_INPUT_768 (3 << 1) #define AD193X_PLL_CLK_CTRL1 0x801 #define AD193X_DAC_CTRL0 0x802 #define AD193X_DAC_POWERDOWN 0x01 diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 7528a54102b..3d7dc55305e 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -22,20 +22,13 @@ * AK4643 is tested. */ -#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/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 "ak4642.h" @@ -111,6 +104,23 @@ struct snd_soc_codec_device soc_codec_dev_ak4642; +/* + * Playback Volume (table 39) + * + * max : 0x00 : +12.0 dB + * ( 0.5 dB step ) + * min : 0xFE : -115.0 dB + * mute: 0xFF + */ +static const DECLARE_TLV_DB_SCALE(out_tlv, -11500, 50, 1); + +static const struct snd_kcontrol_new ak4642_snd_controls[] = { + + SOC_DOUBLE_R_TLV("Digital Playback Volume", L_DVC, R_DVC, + 0, 0xFF, 1, out_tlv), +}; + + /* codec private data */ struct ak4642_priv { struct snd_soc_codec codec; @@ -204,7 +214,6 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream, * * PLL, Master Mode * Audio I/F Format :MSB justified (ADC & DAC) - * Digital Volume: -8dB * Bass Boost Level : Middle * * This operation came from example code of @@ -214,8 +223,6 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream, ak4642_write(codec, 0x0e, 0x19); ak4642_write(codec, 0x09, 0x91); ak4642_write(codec, 0x0c, 0x91); - ak4642_write(codec, 0x0a, 0x28); - ak4642_write(codec, 0x0d, 0x28); ak4642_write(codec, 0x00, 0x64); snd_soc_update_bits(codec, PW_MGMT2, PMHP_MASK, PMHP); snd_soc_update_bits(codec, PW_MGMT2, HPMTN, HPMTN); @@ -491,8 +498,10 @@ static int ak4642_i2c_probe(struct i2c_client *i2c, codec->control_data = i2c; ret = ak4642_init(ak4642); - if (ret < 0) + if (ret < 0) { printk(KERN_ERR "failed to initialise AK4642\n"); + kfree(ak4642); + } return ret; } @@ -548,6 +557,9 @@ static int ak4642_probe(struct platform_device *pdev) goto pcm_err; } + snd_soc_add_controls(ak4642_codec, ak4642_snd_controls, + ARRAY_SIZE(ak4642_snd_controls)); + dev_info(&pdev->dev, "AK4642 Audio Codec %s", AK4642_VERSION); return ret; diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c new file mode 100644 index 00000000000..dd9b8550c40 --- /dev/null +++ b/sound/soc/codecs/cs42l51.c @@ -0,0 +1,763 @@ +/* + * cs42l51.c + * + * ASoC Driver for Cirrus Logic CS42L51 codecs + * + * Copyright (c) 2010 Arnaud Patard <apatard@mandriva.com> + * + * Based on cs4270.c - Copyright (c) Freescale Semiconductor + * + * 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. + * + * For now: + * - Only I2C is support. Not SPI + * - master mode *NOT* supported + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> +#include <sound/initval.h> +#include <sound/pcm_params.h> +#include <sound/pcm.h> +#include <linux/i2c.h> + +#include "cs42l51.h" + +enum master_slave_mode { + MODE_SLAVE, + MODE_SLAVE_AUTO, + MODE_MASTER, +}; + +struct cs42l51_private { + unsigned int mclk; + unsigned int audio_mode; /* The mode (I2S or left-justified) */ + enum master_slave_mode func; + struct snd_soc_codec codec; + u8 reg_cache[CS42L51_NUMREGS]; +}; + +static struct snd_soc_codec *cs42l51_codec; + +#define CS42L51_FORMATS ( \ + SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | \ + SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S18_3BE | \ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE) + +static int cs42l51_fill_cache(struct snd_soc_codec *codec) +{ + u8 *cache = codec->reg_cache + 1; + struct i2c_client *i2c_client = codec->control_data; + s32 length; + + length = i2c_smbus_read_i2c_block_data(i2c_client, + CS42L51_FIRSTREG | 0x80, CS42L51_NUMREGS, cache); + if (length != CS42L51_NUMREGS) { + dev_err(&i2c_client->dev, + "I2C read failure, addr=0x%x (ret=%d vs %d)\n", + i2c_client->addr, length, CS42L51_NUMREGS); + return -EIO; + } + + return 0; +} + +static int cs42l51_i2c_probe(struct i2c_client *i2c_client, + const struct i2c_device_id *id) +{ + struct snd_soc_codec *codec; + struct cs42l51_private *cs42l51; + int ret = 0; + int reg; + + if (cs42l51_codec) + return -EBUSY; + + /* Verify that we have a CS42L51 */ + ret = i2c_smbus_read_byte_data(i2c_client, CS42L51_CHIP_REV_ID); + if (ret < 0) { + dev_err(&i2c_client->dev, "failed to read I2C\n"); + goto error; + } + + if ((ret != CS42L51_MK_CHIP_REV(CS42L51_CHIP_ID, CS42L51_CHIP_REV_A)) && + (ret != CS42L51_MK_CHIP_REV(CS42L51_CHIP_ID, CS42L51_CHIP_REV_B))) { + dev_err(&i2c_client->dev, "Invalid chip id\n"); + ret = -ENODEV; + goto error; + } + + dev_info(&i2c_client->dev, "found device cs42l51 rev %d\n", + ret & 7); + + cs42l51 = kzalloc(sizeof(struct cs42l51_private), GFP_KERNEL); + if (!cs42l51) { + dev_err(&i2c_client->dev, "could not allocate codec\n"); + return -ENOMEM; + } + codec = &cs42l51->codec; + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->dev = &i2c_client->dev; + codec->name = "CS42L51"; + codec->owner = THIS_MODULE; + codec->dai = &cs42l51_dai; + codec->num_dai = 1; + snd_soc_codec_set_drvdata(codec, cs42l51); + + codec->control_data = i2c_client; + codec->reg_cache = cs42l51->reg_cache; + codec->reg_cache_size = CS42L51_NUMREGS; + i2c_set_clientdata(i2c_client, codec); + + ret = cs42l51_fill_cache(codec); + if (ret < 0) { + dev_err(&i2c_client->dev, "failed to fill register cache\n"); + goto error_alloc; + } + + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); + if (ret < 0) { + dev_err(&i2c_client->dev, "Failed to set cache I/O: %d\n", ret); + goto error_alloc; + } + + /* + * DAC configuration + * - Use signal processor + * - auto mute + * - vol changes immediate + * - no de-emphasize + */ + reg = CS42L51_DAC_CTL_DATA_SEL(1) + | CS42L51_DAC_CTL_AMUTE | CS42L51_DAC_CTL_DACSZ(0); + ret = snd_soc_write(codec, CS42L51_DAC_CTL, reg); + if (ret < 0) + goto error_alloc; + + cs42l51_dai.dev = codec->dev; + cs42l51_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + goto error_alloc; + } + + ret = snd_soc_register_dai(&cs42l51_dai); + if (ret < 0) { + dev_err(&i2c_client->dev, "failed to register DAIe\n"); + goto error_reg; + } + + return 0; + +error_reg: + snd_soc_unregister_codec(codec); +error_alloc: + kfree(cs42l51); +error: + return ret; +} + +static int cs42l51_i2c_remove(struct i2c_client *client) +{ + struct cs42l51_private *cs42l51 = i2c_get_clientdata(client); + snd_soc_unregister_dai(&cs42l51_dai); + snd_soc_unregister_codec(cs42l51_codec); + cs42l51_codec = NULL; + kfree(cs42l51); + return 0; +} + + +static const struct i2c_device_id cs42l51_id[] = { + {"cs42l51", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, cs42l51_id); + +static struct i2c_driver cs42l51_i2c_driver = { + .driver = { + .name = "CS42L51 I2C", + .owner = THIS_MODULE, + }, + .id_table = cs42l51_id, + .probe = cs42l51_i2c_probe, + .remove = cs42l51_i2c_remove, +}; + +static int cs42l51_get_chan_mix(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned long value = snd_soc_read(codec, CS42L51_PCM_MIXER)&3; + + switch (value) { + default: + case 0: + ucontrol->value.integer.value[0] = 0; + break; + /* same value : (L+R)/2 and (R+L)/2 */ + case 1: + case 2: + ucontrol->value.integer.value[0] = 1; + break; + case 3: + ucontrol->value.integer.value[0] = 2; + break; + } + + return 0; +} + +#define CHAN_MIX_NORMAL 0x00 +#define CHAN_MIX_BOTH 0x55 +#define CHAN_MIX_SWAP 0xFF + +static int cs42l51_set_chan_mix(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned char val; + + switch (ucontrol->value.integer.value[0]) { + default: + case 0: + val = CHAN_MIX_NORMAL; + break; + case 1: + val = CHAN_MIX_BOTH; + break; + case 2: + val = CHAN_MIX_SWAP; + break; + } + + snd_soc_write(codec, CS42L51_PCM_MIXER, val); + + return 1; +} + +static const DECLARE_TLV_DB_SCALE(adc_pcm_tlv, -5150, 50, 0); +static const DECLARE_TLV_DB_SCALE(tone_tlv, -1050, 150, 0); +/* This is a lie. after -102 db, it stays at -102 */ +/* maybe a range would be better */ +static const DECLARE_TLV_DB_SCALE(aout_tlv, -11550, 50, 0); + +static const DECLARE_TLV_DB_SCALE(boost_tlv, 1600, 1600, 0); +static const char *chan_mix[] = { + "L R", + "L+R", + "R L", +}; + +static const struct soc_enum cs42l51_chan_mix = + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(chan_mix), 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), + 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), + SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume", + CS42L51_ADCA_VOL, CS42L51_ADCB_VOL, + 7, 0xffffff99, 0x18, 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), + SOC_SINGLE("Auto-Mute Switch", CS42L51_DAC_CTL, 2, 1, 0), + SOC_SINGLE("Soft Ramp Switch", CS42L51_DAC_CTL, 1, 1, 0), + SOC_SINGLE("Zero Cross Switch", CS42L51_DAC_CTL, 0, 0, 0), + SOC_DOUBLE_TLV("Mic Boost Volume", + CS42L51_MIC_CTL, 0, 1, 1, 0, boost_tlv), + SOC_SINGLE_TLV("Bass Volume", CS42L51_TONE_CTL, 0, 0xf, 1, tone_tlv), + SOC_SINGLE_TLV("Treble Volume", CS42L51_TONE_CTL, 4, 0xf, 1, tone_tlv), + SOC_ENUM_EXT("PCM channel mixer", + cs42l51_chan_mix, + cs42l51_get_chan_mix, cs42l51_set_chan_mix), +}; + +/* + * to power down, one must: + * 1.) Enable the PDN bit + * 2.) enable power-down for the select channels + * 3.) disable the PDN bit. + */ +static int cs42l51_pdn_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + unsigned long value; + + value = snd_soc_read(w->codec, CS42L51_POWER_CTL1); + value &= ~CS42L51_POWER_CTL1_PDN; + + switch (event) { + case SND_SOC_DAPM_PRE_PMD: + value |= CS42L51_POWER_CTL1_PDN; + break; + default: + case SND_SOC_DAPM_POST_PMD: + break; + } + snd_soc_update_bits(w->codec, CS42L51_POWER_CTL1, + CS42L51_POWER_CTL1_PDN, value); + + return 0; +} + +static const char *cs42l51_dac_names[] = {"Direct PCM", + "DSP PCM", "ADC"}; +static const struct soc_enum cs42l51_dac_mux_enum = + SOC_ENUM_SINGLE(CS42L51_DAC_CTL, 6, 3, cs42l51_dac_names); +static const struct snd_kcontrol_new cs42l51_dac_mux_controls = + SOC_DAPM_ENUM("Route", cs42l51_dac_mux_enum); + +static const char *cs42l51_adcl_names[] = {"AIN1 Left", "AIN2 Left", + "MIC Left", "MIC+preamp Left"}; +static const struct soc_enum cs42l51_adcl_mux_enum = + SOC_ENUM_SINGLE(CS42L51_ADC_INPUT, 4, 4, cs42l51_adcl_names); +static const struct snd_kcontrol_new cs42l51_adcl_mux_controls = + SOC_DAPM_ENUM("Route", cs42l51_adcl_mux_enum); + +static const char *cs42l51_adcr_names[] = {"AIN1 Right", "AIN2 Right", + "MIC Right", "MIC+preamp Right"}; +static const struct soc_enum cs42l51_adcr_mux_enum = + SOC_ENUM_SINGLE(CS42L51_ADC_INPUT, 6, 4, cs42l51_adcr_names); +static const struct snd_kcontrol_new cs42l51_adcr_mux_controls = + SOC_DAPM_ENUM("Route", cs42l51_adcr_mux_enum); + +static const struct snd_soc_dapm_widget cs42l51_dapm_widgets[] = { + SND_SOC_DAPM_MICBIAS("Mic Bias", CS42L51_MIC_POWER_CTL, 1, 1), + SND_SOC_DAPM_PGA_E("Left PGA", CS42L51_POWER_CTL1, 3, 1, NULL, 0, + cs42l51_pdn_event, SND_SOC_DAPM_PRE_POST_PMD), + SND_SOC_DAPM_PGA_E("Right PGA", CS42L51_POWER_CTL1, 4, 1, NULL, 0, + cs42l51_pdn_event, SND_SOC_DAPM_PRE_POST_PMD), + SND_SOC_DAPM_ADC_E("Left ADC", "Left HiFi Capture", + CS42L51_POWER_CTL1, 1, 1, + cs42l51_pdn_event, SND_SOC_DAPM_PRE_POST_PMD), + SND_SOC_DAPM_ADC_E("Right ADC", "Right HiFi Capture", + CS42L51_POWER_CTL1, 2, 1, + cs42l51_pdn_event, SND_SOC_DAPM_PRE_POST_PMD), + SND_SOC_DAPM_DAC_E("Left DAC", "Left HiFi Playback", + CS42L51_POWER_CTL1, 5, 1, + cs42l51_pdn_event, SND_SOC_DAPM_PRE_POST_PMD), + SND_SOC_DAPM_DAC_E("Right DAC", "Right HiFi Playback", + CS42L51_POWER_CTL1, 6, 1, + cs42l51_pdn_event, SND_SOC_DAPM_PRE_POST_PMD), + + /* analog/mic */ + SND_SOC_DAPM_INPUT("AIN1L"), + SND_SOC_DAPM_INPUT("AIN1R"), + SND_SOC_DAPM_INPUT("AIN2L"), + SND_SOC_DAPM_INPUT("AIN2R"), + SND_SOC_DAPM_INPUT("MICL"), + SND_SOC_DAPM_INPUT("MICR"), + + SND_SOC_DAPM_MIXER("Mic Preamp Left", + CS42L51_MIC_POWER_CTL, 2, 1, NULL, 0), + SND_SOC_DAPM_MIXER("Mic Preamp Right", + CS42L51_MIC_POWER_CTL, 3, 1, NULL, 0), + + /* HP */ + SND_SOC_DAPM_OUTPUT("HPL"), + SND_SOC_DAPM_OUTPUT("HPR"), + + /* mux */ + SND_SOC_DAPM_MUX("DAC Mux", SND_SOC_NOPM, 0, 0, + &cs42l51_dac_mux_controls), + SND_SOC_DAPM_MUX("PGA-ADC Mux Left", SND_SOC_NOPM, 0, 0, + &cs42l51_adcl_mux_controls), + SND_SOC_DAPM_MUX("PGA-ADC Mux Right", SND_SOC_NOPM, 0, 0, + &cs42l51_adcr_mux_controls), +}; + +static const struct snd_soc_dapm_route cs42l51_routes[] = { + {"HPL", NULL, "Left DAC"}, + {"HPR", NULL, "Right DAC"}, + + {"Left ADC", NULL, "Left PGA"}, + {"Right ADC", NULL, "Right PGA"}, + + {"Mic Preamp Left", NULL, "MICL"}, + {"Mic Preamp Right", NULL, "MICR"}, + + {"PGA-ADC Mux Left", "AIN1 Left", "AIN1L" }, + {"PGA-ADC Mux Left", "AIN2 Left", "AIN2L" }, + {"PGA-ADC Mux Left", "MIC Left", "MICL" }, + {"PGA-ADC Mux Left", "MIC+preamp Left", "Mic Preamp Left" }, + {"PGA-ADC Mux Right", "AIN1 Right", "AIN1R" }, + {"PGA-ADC Mux Right", "AIN2 Right", "AIN2R" }, + {"PGA-ADC Mux Right", "MIC Right", "MICR" }, + {"PGA-ADC Mux Right", "MIC+preamp Right", "Mic Preamp Right" }, + + {"Left PGA", NULL, "PGA-ADC Mux Left"}, + {"Right PGA", NULL, "PGA-ADC Mux Right"}, +}; + +static int cs42l51_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int format) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs42l51_private *cs42l51 = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + switch (format & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_LEFT_J: + case SND_SOC_DAIFMT_RIGHT_J: + cs42l51->audio_mode = format & SND_SOC_DAIFMT_FORMAT_MASK; + break; + default: + dev_err(codec->dev, "invalid DAI format\n"); + ret = -EINVAL; + } + + switch (format & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + cs42l51->func = MODE_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFS: + cs42l51->func = MODE_SLAVE_AUTO; + break; + default: + ret = -EINVAL; + break; + } + + return ret; +} + +struct cs42l51_ratios { + unsigned int ratio; + unsigned char speed_mode; + unsigned char mclk; +}; + +static struct cs42l51_ratios slave_ratios[] = { + { 512, CS42L51_QSM_MODE, 0 }, { 768, CS42L51_QSM_MODE, 0 }, + { 1024, CS42L51_QSM_MODE, 0 }, { 1536, CS42L51_QSM_MODE, 0 }, + { 2048, CS42L51_QSM_MODE, 0 }, { 3072, CS42L51_QSM_MODE, 0 }, + { 256, CS42L51_HSM_MODE, 0 }, { 384, CS42L51_HSM_MODE, 0 }, + { 512, CS42L51_HSM_MODE, 0 }, { 768, CS42L51_HSM_MODE, 0 }, + { 1024, CS42L51_HSM_MODE, 0 }, { 1536, CS42L51_HSM_MODE, 0 }, + { 128, CS42L51_SSM_MODE, 0 }, { 192, CS42L51_SSM_MODE, 0 }, + { 256, CS42L51_SSM_MODE, 0 }, { 384, CS42L51_SSM_MODE, 0 }, + { 512, CS42L51_SSM_MODE, 0 }, { 768, CS42L51_SSM_MODE, 0 }, + { 128, CS42L51_DSM_MODE, 0 }, { 192, CS42L51_DSM_MODE, 0 }, + { 256, CS42L51_DSM_MODE, 0 }, { 384, CS42L51_DSM_MODE, 0 }, +}; + +static struct cs42l51_ratios slave_auto_ratios[] = { + { 1024, CS42L51_QSM_MODE, 0 }, { 1536, CS42L51_QSM_MODE, 0 }, + { 2048, CS42L51_QSM_MODE, 1 }, { 3072, CS42L51_QSM_MODE, 1 }, + { 512, CS42L51_HSM_MODE, 0 }, { 768, CS42L51_HSM_MODE, 0 }, + { 1024, CS42L51_HSM_MODE, 1 }, { 1536, CS42L51_HSM_MODE, 1 }, + { 256, CS42L51_SSM_MODE, 0 }, { 384, CS42L51_SSM_MODE, 0 }, + { 512, CS42L51_SSM_MODE, 1 }, { 768, CS42L51_SSM_MODE, 1 }, + { 128, CS42L51_DSM_MODE, 0 }, { 192, CS42L51_DSM_MODE, 0 }, + { 256, CS42L51_DSM_MODE, 1 }, { 384, CS42L51_DSM_MODE, 1 }, +}; + +static int cs42l51_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 cs42l51_private *cs42l51 = snd_soc_codec_get_drvdata(codec); + struct cs42l51_ratios *ratios = NULL; + int nr_ratios = 0; + unsigned int rates = 0; + unsigned int rate_min = -1; + unsigned int rate_max = 0; + int i; + + cs42l51->mclk = freq; + + switch (cs42l51->func) { + case MODE_MASTER: + return -EINVAL; + case MODE_SLAVE: + ratios = slave_ratios; + nr_ratios = ARRAY_SIZE(slave_ratios); + break; + case MODE_SLAVE_AUTO: + ratios = slave_auto_ratios; + nr_ratios = ARRAY_SIZE(slave_auto_ratios); + break; + } + + for (i = 0; i < nr_ratios; i++) { + unsigned int rate = freq / ratios[i].ratio; + rates |= snd_pcm_rate_to_rate_bit(rate); + if (rate < rate_min) + rate_min = rate; + if (rate > rate_max) + rate_max = rate; + } + rates &= ~SNDRV_PCM_RATE_KNOT; + + if (!rates) { + dev_err(codec->dev, "could not find a valid sample rate\n"); + return -EINVAL; + } + + codec_dai->playback.rates = rates; + codec_dai->playback.rate_min = rate_min; + codec_dai->playback.rate_max = rate_max; + + codec_dai->capture.rates = rates; + codec_dai->capture.rate_min = rate_min; + codec_dai->capture.rate_max = rate_max; + + return 0; +} + +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_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + struct cs42l51_private *cs42l51 = snd_soc_codec_get_drvdata(codec); + int ret; + unsigned int i; + unsigned int rate; + unsigned int ratio; + struct cs42l51_ratios *ratios = NULL; + int nr_ratios = 0; + int intf_ctl, power_ctl, fmt; + + switch (cs42l51->func) { + case MODE_MASTER: + return -EINVAL; + case MODE_SLAVE: + ratios = slave_ratios; + nr_ratios = ARRAY_SIZE(slave_ratios); + break; + case MODE_SLAVE_AUTO: + ratios = slave_auto_ratios; + nr_ratios = ARRAY_SIZE(slave_auto_ratios); + break; + } + + /* Figure out which MCLK/LRCK ratio to use */ + rate = params_rate(params); /* Sampling rate, in Hz */ + ratio = cs42l51->mclk / rate; /* MCLK/LRCK ratio */ + for (i = 0; i < nr_ratios; i++) { + if (ratios[i].ratio == ratio) + break; + } + + if (i == nr_ratios) { + /* We did not find a matching ratio */ + dev_err(codec->dev, "could not find matching ratio\n"); + return -EINVAL; + } + + intf_ctl = snd_soc_read(codec, CS42L51_INTF_CTL); + power_ctl = snd_soc_read(codec, CS42L51_MIC_POWER_CTL); + + intf_ctl &= ~(CS42L51_INTF_CTL_MASTER | CS42L51_INTF_CTL_ADC_I2S + | CS42L51_INTF_CTL_DAC_FORMAT(7)); + power_ctl &= ~(CS42L51_MIC_POWER_CTL_SPEED(3) + | CS42L51_MIC_POWER_CTL_MCLK_DIV2); + + switch (cs42l51->func) { + case MODE_MASTER: + intf_ctl |= CS42L51_INTF_CTL_MASTER; + power_ctl |= CS42L51_MIC_POWER_CTL_SPEED(ratios[i].speed_mode); + break; + case MODE_SLAVE: + power_ctl |= CS42L51_MIC_POWER_CTL_SPEED(ratios[i].speed_mode); + break; + case MODE_SLAVE_AUTO: + power_ctl |= CS42L51_MIC_POWER_CTL_AUTO; + break; + } + + switch (cs42l51->audio_mode) { + case SND_SOC_DAIFMT_I2S: + intf_ctl |= CS42L51_INTF_CTL_ADC_I2S; + intf_ctl |= CS42L51_INTF_CTL_DAC_FORMAT(CS42L51_DAC_DIF_I2S); + break; + case SND_SOC_DAIFMT_LEFT_J: + intf_ctl |= CS42L51_INTF_CTL_DAC_FORMAT(CS42L51_DAC_DIF_LJ24); + break; + case SND_SOC_DAIFMT_RIGHT_J: + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + case SNDRV_PCM_FORMAT_S16_BE: + fmt = CS42L51_DAC_DIF_RJ16; + break; + case SNDRV_PCM_FORMAT_S18_3LE: + case SNDRV_PCM_FORMAT_S18_3BE: + fmt = CS42L51_DAC_DIF_RJ18; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + case SNDRV_PCM_FORMAT_S20_3BE: + fmt = CS42L51_DAC_DIF_RJ20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + case SNDRV_PCM_FORMAT_S24_BE: + fmt = CS42L51_DAC_DIF_RJ24; + break; + default: + dev_err(codec->dev, "unknown format\n"); + return -EINVAL; + } + intf_ctl |= CS42L51_INTF_CTL_DAC_FORMAT(fmt); + break; + default: + dev_err(codec->dev, "unknown format\n"); + return -EINVAL; + } + + if (ratios[i].mclk) + power_ctl |= CS42L51_MIC_POWER_CTL_MCLK_DIV2; + + ret = snd_soc_write(codec, CS42L51_INTF_CTL, intf_ctl); + if (ret < 0) + return ret; + + ret = snd_soc_write(codec, CS42L51_MIC_POWER_CTL, power_ctl); + if (ret < 0) + return ret; + + return 0; +} + +static int cs42l51_dai_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + int reg; + int mask = CS42L51_DAC_OUT_CTL_DACA_MUTE|CS42L51_DAC_OUT_CTL_DACB_MUTE; + + reg = snd_soc_read(codec, CS42L51_DAC_OUT_CTL); + + if (mute) + reg |= mask; + else + reg &= ~mask; + + return snd_soc_write(codec, CS42L51_DAC_OUT_CTL, reg); +} + +static struct snd_soc_dai_ops cs42l51_dai_ops = { + .hw_params = cs42l51_hw_params, + .set_sysclk = cs42l51_set_dai_sysclk, + .set_fmt = cs42l51_set_dai_fmt, + .digital_mute = cs42l51_dai_mute, +}; + +struct snd_soc_dai cs42l51_dai = { + .name = "CS42L51 HiFi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = CS42L51_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = CS42L51_FORMATS, + }, + .ops = &cs42l51_dai_ops, +}; +EXPORT_SYMBOL_GPL(cs42l51_dai); + + +static int cs42l51_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + if (!cs42l51_codec) { + dev_err(&pdev->dev, "CS42L51 codec not yet registered\n"); + return -EINVAL; + } + + socdev->card->codec = cs42l51_codec; + codec = socdev->card->codec; + + /* Register PCMs */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(&pdev->dev, "failed to create PCMs\n"); + return ret; + } + + snd_soc_add_controls(codec, cs42l51_snd_controls, + ARRAY_SIZE(cs42l51_snd_controls)); + snd_soc_dapm_new_controls(codec, cs42l51_dapm_widgets, + ARRAY_SIZE(cs42l51_dapm_widgets)); + snd_soc_dapm_add_routes(codec, cs42l51_routes, + ARRAY_SIZE(cs42l51_routes)); + + return 0; +} + + +static int cs42l51_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_device_cs42l51 = { + .probe = cs42l51_probe, + .remove = cs42l51_remove +}; +EXPORT_SYMBOL_GPL(soc_codec_device_cs42l51); + +static int __init cs42l51_init(void) +{ + int ret; + + ret = i2c_add_driver(&cs42l51_i2c_driver); + if (ret != 0) { + printk(KERN_ERR "%s: can't add i2c driver\n", __func__); + return ret; + } + return 0; +} +module_init(cs42l51_init); + +static void __exit cs42l51_exit(void) +{ + i2c_del_driver(&cs42l51_i2c_driver); +} +module_exit(cs42l51_exit); + +MODULE_AUTHOR("Arnaud Patard <apatard@mandriva.com>"); +MODULE_DESCRIPTION("Cirrus Logic CS42L51 ALSA SoC Codec Driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs42l51.h b/sound/soc/codecs/cs42l51.h new file mode 100644 index 00000000000..8f0bd9786ad --- /dev/null +++ b/sound/soc/codecs/cs42l51.h @@ -0,0 +1,163 @@ +/* + * cs42l51.h + * + * ASoC Driver for Cirrus Logic CS42L51 codecs + * + * Copyright (c) 2010 Arnaud Patard <apatard@mandriva.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * 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. + */ +#ifndef _CS42L51_H +#define _CS42L51_H + +#define CS42L51_CHIP_ID 0x1B +#define CS42L51_CHIP_REV_A 0x00 +#define CS42L51_CHIP_REV_B 0x01 + +#define CS42L51_CHIP_REV_ID 0x01 +#define CS42L51_MK_CHIP_REV(a, b) ((a)<<3|(b)) + +#define CS42L51_POWER_CTL1 0x02 +#define CS42L51_POWER_CTL1_PDN_DACB (1<<6) +#define CS42L51_POWER_CTL1_PDN_DACA (1<<5) +#define CS42L51_POWER_CTL1_PDN_PGAB (1<<4) +#define CS42L51_POWER_CTL1_PDN_PGAA (1<<3) +#define CS42L51_POWER_CTL1_PDN_ADCB (1<<2) +#define CS42L51_POWER_CTL1_PDN_ADCA (1<<1) +#define CS42L51_POWER_CTL1_PDN (1<<0) + +#define CS42L51_MIC_POWER_CTL 0x03 +#define CS42L51_MIC_POWER_CTL_AUTO (1<<7) +#define CS42L51_MIC_POWER_CTL_SPEED(x) (((x)&3)<<5) +#define CS42L51_QSM_MODE 3 +#define CS42L51_HSM_MODE 2 +#define CS42L51_SSM_MODE 1 +#define CS42L51_DSM_MODE 0 +#define CS42L51_MIC_POWER_CTL_3ST_SP (1<<4) +#define CS42L51_MIC_POWER_CTL_PDN_MICB (1<<3) +#define CS42L51_MIC_POWER_CTL_PDN_MICA (1<<2) +#define CS42L51_MIC_POWER_CTL_PDN_BIAS (1<<1) +#define CS42L51_MIC_POWER_CTL_MCLK_DIV2 (1<<0) + +#define CS42L51_INTF_CTL 0x04 +#define CS42L51_INTF_CTL_LOOPBACK (1<<7) +#define CS42L51_INTF_CTL_MASTER (1<<6) +#define CS42L51_INTF_CTL_DAC_FORMAT(x) (((x)&7)<<3) +#define CS42L51_DAC_DIF_LJ24 0x00 +#define CS42L51_DAC_DIF_I2S 0x01 +#define CS42L51_DAC_DIF_RJ24 0x02 +#define CS42L51_DAC_DIF_RJ20 0x03 +#define CS42L51_DAC_DIF_RJ18 0x04 +#define CS42L51_DAC_DIF_RJ16 0x05 +#define CS42L51_INTF_CTL_ADC_I2S (1<<2) +#define CS42L51_INTF_CTL_DIGMIX (1<<1) +#define CS42L51_INTF_CTL_MICMIX (1<<0) + +#define CS42L51_MIC_CTL 0x05 +#define CS42L51_MIC_CTL_ADC_SNGVOL (1<<7) +#define CS42L51_MIC_CTL_ADCD_DBOOST (1<<6) +#define CS42L51_MIC_CTL_ADCA_DBOOST (1<<5) +#define CS42L51_MIC_CTL_MICBIAS_SEL (1<<4) +#define CS42L51_MIC_CTL_MICBIAS_LVL(x) (((x)&3)<<2) +#define CS42L51_MIC_CTL_MICB_BOOST (1<<1) +#define CS42L51_MIC_CTL_MICA_BOOST (1<<0) + +#define CS42L51_ADC_CTL 0x06 +#define CS42L51_ADC_CTL_ADCB_HPFEN (1<<7) +#define CS42L51_ADC_CTL_ADCB_HPFRZ (1<<6) +#define CS42L51_ADC_CTL_ADCA_HPFEN (1<<5) +#define CS42L51_ADC_CTL_ADCA_HPFRZ (1<<4) +#define CS42L51_ADC_CTL_SOFTB (1<<3) +#define CS42L51_ADC_CTL_ZCROSSB (1<<2) +#define CS42L51_ADC_CTL_SOFTA (1<<1) +#define CS42L51_ADC_CTL_ZCROSSA (1<<0) + +#define CS42L51_ADC_INPUT 0x07 +#define CS42L51_ADC_INPUT_AINB_MUX(x) (((x)&3)<<6) +#define CS42L51_ADC_INPUT_AINA_MUX(x) (((x)&3)<<4) +#define CS42L51_ADC_INPUT_INV_ADCB (1<<3) +#define CS42L51_ADC_INPUT_INV_ADCA (1<<2) +#define CS42L51_ADC_INPUT_ADCB_MUTE (1<<1) +#define CS42L51_ADC_INPUT_ADCA_MUTE (1<<0) + +#define CS42L51_DAC_OUT_CTL 0x08 +#define CS42L51_DAC_OUT_CTL_HP_GAIN(x) (((x)&7)<<5) +#define CS42L51_DAC_OUT_CTL_DAC_SNGVOL (1<<4) +#define CS42L51_DAC_OUT_CTL_INV_PCMB (1<<3) +#define CS42L51_DAC_OUT_CTL_INV_PCMA (1<<2) +#define CS42L51_DAC_OUT_CTL_DACB_MUTE (1<<1) +#define CS42L51_DAC_OUT_CTL_DACA_MUTE (1<<0) + +#define CS42L51_DAC_CTL 0x09 +#define CS42L51_DAC_CTL_DATA_SEL(x) (((x)&3)<<6) +#define CS42L51_DAC_CTL_FREEZE (1<<5) +#define CS42L51_DAC_CTL_DEEMPH (1<<3) +#define CS42L51_DAC_CTL_AMUTE (1<<2) +#define CS42L51_DAC_CTL_DACSZ(x) (((x)&3)<<0) + +#define CS42L51_ALC_PGA_CTL 0x0A +#define CS42L51_ALC_PGB_CTL 0x0B +#define CS42L51_ALC_PGX_ALCX_SRDIS (1<<7) +#define CS42L51_ALC_PGX_ALCX_ZCDIS (1<<6) +#define CS42L51_ALC_PGX_PGX_VOL(x) (((x)&0x1f)<<0) + +#define CS42L51_ADCA_ATT 0x0C +#define CS42L51_ADCB_ATT 0x0D + +#define CS42L51_ADCA_VOL 0x0E +#define CS42L51_ADCB_VOL 0x0F +#define CS42L51_PCMA_VOL 0x10 +#define CS42L51_PCMB_VOL 0x11 +#define CS42L51_MIX_MUTE_ADCMIX (1<<7) +#define CS42L51_MIX_VOLUME(x) (((x)&0x7f)<<0) + +#define CS42L51_BEEP_FREQ 0x12 +#define CS42L51_BEEP_VOL 0x13 +#define CS42L51_BEEP_CONF 0x14 + +#define CS42L51_TONE_CTL 0x15 +#define CS42L51_TONE_CTL_TREB(x) (((x)&0xf)<<4) +#define CS42L51_TONE_CTL_BASS(x) (((x)&0xf)<<0) + +#define CS42L51_AOUTA_VOL 0x16 +#define CS42L51_AOUTB_VOL 0x17 +#define CS42L51_PCM_MIXER 0x18 +#define CS42L51_LIMIT_THRES_DIS 0x19 +#define CS42L51_LIMIT_REL 0x1A +#define CS42L51_LIMIT_ATT 0x1B +#define CS42L51_ALC_EN 0x1C +#define CS42L51_ALC_REL 0x1D +#define CS42L51_ALC_THRES 0x1E +#define CS42L51_NOISE_CONF 0x1F + +#define CS42L51_STATUS 0x20 +#define CS42L51_STATUS_SP_CLKERR (1<<6) +#define CS42L51_STATUS_SPEA_OVFL (1<<5) +#define CS42L51_STATUS_SPEB_OVFL (1<<4) +#define CS42L51_STATUS_PCMA_OVFL (1<<3) +#define CS42L51_STATUS_PCMB_OVFL (1<<2) +#define CS42L51_STATUS_ADCA_OVFL (1<<1) +#define CS42L51_STATUS_ADCB_OVFL (1<<0) + +#define CS42L51_CHARGE_FREQ 0x21 + +#define CS42L51_FIRSTREG 0x01 +/* + * Hack: with register 0x21, it makes 33 registers. Looks like someone in the + * i2c layer doesn't like i2c smbus block read of 33 regs. Workaround by using + * 32 regs + */ +#define CS42L51_LASTREG 0x20 +#define CS42L51_NUMREGS (CS42L51_LASTREG - CS42L51_FIRSTREG + 1) + +extern struct snd_soc_dai cs42l51_dai; +extern struct snd_soc_codec_device soc_codec_device_cs42l51; +#endif diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 75af2d6e0e7..3c51d6a5752 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -15,23 +15,15 @@ * option) any later version. */ -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/kernel.h> -#include <linux/init.h> #include <linux/delay.h> -#include <linux/pm.h> #include <linux/i2c.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/tlv.h> #include <sound/initval.h> -#include <asm/div64.h> +#include <sound/tlv.h> #include "da7210.h" @@ -145,6 +137,29 @@ #define DA7210_VERSION "0.0.1" +/* + * Playback Volume + * + * max : 0x3F (+15.0 dB) + * (1.5 dB step) + * min : 0x11 (-54.0 dB) + * mute : 0x10 + * reserved : 0x00 - 0x0F + * + * ** FIXME ** + * + * Reserved area are considered as "mute". + * -> min = -79.5 dB + */ +static const DECLARE_TLV_DB_SCALE(hp_out_tlv, -7950, 150, 1); + +static const struct snd_kcontrol_new da7210_snd_controls[] = { + + SOC_DOUBLE_R_TLV("HeadPhone Playback Volume", + DA7210_HP_L_VOL, DA7210_HP_R_VOL, + 0, 0x3F, 0, hp_out_tlv), +}; + /* Codec private data */ struct da7210_priv { struct snd_soc_codec codec; @@ -227,10 +242,6 @@ static int da7210_startup(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = dai->codec; if (is_play) { - /* PlayBack Volume 40 */ - snd_soc_update_bits(codec, DA7210_HP_L_VOL, 0x3F, 40); - snd_soc_update_bits(codec, DA7210_HP_R_VOL, 0x3F, 40); - /* Enable Out */ snd_soc_update_bits(codec, DA7210_OUTMIX_L, 0x1F, 0x10); snd_soc_update_bits(codec, DA7210_OUTMIX_R, 0x1F, 0x10); @@ -488,7 +499,7 @@ static int da7210_init(struct da7210_priv *da7210) ret = snd_soc_register_dai(&da7210_dai); if (ret) { dev_err(codec->dev, "Failed to register DAI: %d\n", ret); - goto init_err; + goto codec_err; } /* FIXME @@ -574,6 +585,8 @@ static int da7210_init(struct da7210_priv *da7210) return ret; +codec_err: + snd_soc_unregister_codec(codec); init_err: kfree(codec->reg_cache); codec->reg_cache = NULL; @@ -601,8 +614,10 @@ static int __devinit da7210_i2c_probe(struct i2c_client *i2c, codec->control_data = i2c; ret = da7210_init(da7210); - if (ret < 0) + if (ret < 0) { pr_err("Failed to initialise da7210 audio codec\n"); + kfree(da7210); + } return ret; } @@ -656,6 +671,9 @@ static int da7210_probe(struct platform_device *pdev) if (ret < 0) goto pcm_err; + snd_soc_add_controls(da7210_codec, da7210_snd_controls, + ARRAY_SIZE(da7210_snd_controls)); + dev_info(&pdev->dev, "DA7210 Audio Codec %s\n", DA7210_VERSION); pcm_err: diff --git a/sound/soc/codecs/jz4740.c b/sound/soc/codecs/jz4740.c new file mode 100644 index 00000000000..66557de1e4f --- /dev/null +++ b/sound/soc/codecs/jz4740.c @@ -0,0 +1,511 @@ +/* + * Copyright (C) 2009-2010, Lars-Peter Clausen <lars@metafoo.de> + * + * 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. + * + * 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., + * 675 Mass Ave, Cambridge, MA 02139, USA. + * + */ + +#include <linux/kernel.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/slab.h> + +#include <linux/delay.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc-dapm.h> +#include <sound/soc.h> + +#define JZ4740_REG_CODEC_1 0x0 +#define JZ4740_REG_CODEC_2 0x1 + +#define JZ4740_CODEC_1_LINE_ENABLE BIT(29) +#define JZ4740_CODEC_1_MIC_ENABLE BIT(28) +#define JZ4740_CODEC_1_SW1_ENABLE BIT(27) +#define JZ4740_CODEC_1_ADC_ENABLE BIT(26) +#define JZ4740_CODEC_1_SW2_ENABLE BIT(25) +#define JZ4740_CODEC_1_DAC_ENABLE BIT(24) +#define JZ4740_CODEC_1_VREF_DISABLE BIT(20) +#define JZ4740_CODEC_1_VREF_AMP_DISABLE BIT(19) +#define JZ4740_CODEC_1_VREF_PULLDOWN BIT(18) +#define JZ4740_CODEC_1_VREF_LOW_CURRENT BIT(17) +#define JZ4740_CODEC_1_VREF_HIGH_CURRENT BIT(16) +#define JZ4740_CODEC_1_HEADPHONE_DISABLE BIT(14) +#define JZ4740_CODEC_1_HEADPHONE_AMP_CHANGE_ANY BIT(13) +#define JZ4740_CODEC_1_HEADPHONE_CHARGE BIT(12) +#define JZ4740_CODEC_1_HEADPHONE_PULLDOWN (BIT(11) | BIT(10)) +#define JZ4740_CODEC_1_HEADPHONE_POWERDOWN_M BIT(9) +#define JZ4740_CODEC_1_HEADPHONE_POWERDOWN BIT(8) +#define JZ4740_CODEC_1_SUSPEND BIT(1) +#define JZ4740_CODEC_1_RESET BIT(0) + +#define JZ4740_CODEC_1_LINE_ENABLE_OFFSET 29 +#define JZ4740_CODEC_1_MIC_ENABLE_OFFSET 28 +#define JZ4740_CODEC_1_SW1_ENABLE_OFFSET 27 +#define JZ4740_CODEC_1_ADC_ENABLE_OFFSET 26 +#define JZ4740_CODEC_1_SW2_ENABLE_OFFSET 25 +#define JZ4740_CODEC_1_DAC_ENABLE_OFFSET 24 +#define JZ4740_CODEC_1_HEADPHONE_DISABLE_OFFSET 14 +#define JZ4740_CODEC_1_HEADPHONE_POWERDOWN_OFFSET 8 + +#define JZ4740_CODEC_2_INPUT_VOLUME_MASK 0x1f0000 +#define JZ4740_CODEC_2_SAMPLE_RATE_MASK 0x000f00 +#define JZ4740_CODEC_2_MIC_BOOST_GAIN_MASK 0x000030 +#define JZ4740_CODEC_2_HEADPHONE_VOLUME_MASK 0x000003 + +#define JZ4740_CODEC_2_INPUT_VOLUME_OFFSET 16 +#define JZ4740_CODEC_2_SAMPLE_RATE_OFFSET 8 +#define JZ4740_CODEC_2_MIC_BOOST_GAIN_OFFSET 4 +#define JZ4740_CODEC_2_HEADPHONE_VOLUME_OFFSET 0 + +static const uint32_t jz4740_codec_regs[] = { + 0x021b2302, 0x00170803, +}; + +struct jz4740_codec { + void __iomem *base; + struct resource *mem; + + uint32_t reg_cache[2]; + struct snd_soc_codec codec; +}; + +static inline struct jz4740_codec *codec_to_jz4740(struct snd_soc_codec *codec) +{ + return container_of(codec, struct jz4740_codec, codec); +} + +static unsigned int jz4740_codec_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + struct jz4740_codec *jz4740_codec = codec_to_jz4740(codec); + return readl(jz4740_codec->base + (reg << 2)); +} + +static int jz4740_codec_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int val) +{ + struct jz4740_codec *jz4740_codec = codec_to_jz4740(codec); + + jz4740_codec->reg_cache[reg] = val; + writel(val, jz4740_codec->base + (reg << 2)); + + return 0; +} + +static const struct snd_kcontrol_new jz4740_codec_controls[] = { + SOC_SINGLE("Master Playback Volume", JZ4740_REG_CODEC_2, + JZ4740_CODEC_2_HEADPHONE_VOLUME_OFFSET, 3, 0), + SOC_SINGLE("Master Capture Volume", JZ4740_REG_CODEC_2, + JZ4740_CODEC_2_INPUT_VOLUME_OFFSET, 31, 0), + SOC_SINGLE("Master Playback Switch", JZ4740_REG_CODEC_1, + JZ4740_CODEC_1_HEADPHONE_DISABLE_OFFSET, 1, 1), + SOC_SINGLE("Mic Capture Volume", JZ4740_REG_CODEC_2, + JZ4740_CODEC_2_MIC_BOOST_GAIN_OFFSET, 3, 0), +}; + +static const struct snd_kcontrol_new jz4740_codec_output_controls[] = { + SOC_DAPM_SINGLE("Bypass Switch", JZ4740_REG_CODEC_1, + JZ4740_CODEC_1_SW1_ENABLE_OFFSET, 1, 0), + SOC_DAPM_SINGLE("DAC Switch", JZ4740_REG_CODEC_1, + JZ4740_CODEC_1_SW2_ENABLE_OFFSET, 1, 0), +}; + +static const struct snd_kcontrol_new jz4740_codec_input_controls[] = { + SOC_DAPM_SINGLE("Line Capture Switch", JZ4740_REG_CODEC_1, + JZ4740_CODEC_1_LINE_ENABLE_OFFSET, 1, 0), + SOC_DAPM_SINGLE("Mic Capture Switch", JZ4740_REG_CODEC_1, + JZ4740_CODEC_1_MIC_ENABLE_OFFSET, 1, 0), +}; + +static const struct snd_soc_dapm_widget jz4740_codec_dapm_widgets[] = { + SND_SOC_DAPM_ADC("ADC", "Capture", JZ4740_REG_CODEC_1, + JZ4740_CODEC_1_ADC_ENABLE_OFFSET, 0), + SND_SOC_DAPM_DAC("DAC", "Playback", JZ4740_REG_CODEC_1, + JZ4740_CODEC_1_DAC_ENABLE_OFFSET, 0), + + SND_SOC_DAPM_MIXER("Output Mixer", JZ4740_REG_CODEC_1, + JZ4740_CODEC_1_HEADPHONE_POWERDOWN_OFFSET, 1, + jz4740_codec_output_controls, + ARRAY_SIZE(jz4740_codec_output_controls)), + + SND_SOC_DAPM_MIXER_NAMED_CTL("Input Mixer", SND_SOC_NOPM, 0, 0, + jz4740_codec_input_controls, + ARRAY_SIZE(jz4740_codec_input_controls)), + SND_SOC_DAPM_MIXER("Line Input", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_OUTPUT("LOUT"), + SND_SOC_DAPM_OUTPUT("ROUT"), + + SND_SOC_DAPM_INPUT("MIC"), + SND_SOC_DAPM_INPUT("LIN"), + SND_SOC_DAPM_INPUT("RIN"), +}; + +static const struct snd_soc_dapm_route jz4740_codec_dapm_routes[] = { + {"Line Input", NULL, "LIN"}, + {"Line Input", NULL, "RIN"}, + + {"Input Mixer", "Line Capture Switch", "Line Input"}, + {"Input Mixer", "Mic Capture Switch", "MIC"}, + + {"ADC", NULL, "Input Mixer"}, + + {"Output Mixer", "Bypass Switch", "Input Mixer"}, + {"Output Mixer", "DAC Switch", "DAC"}, + + {"LOUT", NULL, "Output Mixer"}, + {"ROUT", NULL, "Output Mixer"}, +}; + +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_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + + switch (params_rate(params)) { + case 8000: + val = 0; + break; + case 11025: + val = 1; + break; + case 12000: + val = 2; + break; + case 16000: + val = 3; + break; + case 22050: + val = 4; + break; + case 24000: + val = 5; + break; + case 32000: + val = 6; + break; + case 44100: + val = 7; + break; + case 48000: + val = 8; + break; + default: + return -EINVAL; + } + + val <<= JZ4740_CODEC_2_SAMPLE_RATE_OFFSET; + + snd_soc_update_bits(codec, JZ4740_REG_CODEC_2, + JZ4740_CODEC_2_SAMPLE_RATE_MASK, val); + + return 0; +} + +static struct snd_soc_dai_ops jz4740_codec_dai_ops = { + .hw_params = jz4740_codec_hw_params, +}; + +struct snd_soc_dai jz4740_codec_dai = { + .name = "jz4740", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8, + }, + .ops = &jz4740_codec_dai_ops, + .symmetric_rates = 1, +}; +EXPORT_SYMBOL_GPL(jz4740_codec_dai); + +static void jz4740_codec_wakeup(struct snd_soc_codec *codec) +{ + int i; + uint32_t *cache = codec->reg_cache; + + snd_soc_update_bits(codec, JZ4740_REG_CODEC_1, + JZ4740_CODEC_1_RESET, JZ4740_CODEC_1_RESET); + udelay(2); + + snd_soc_update_bits(codec, JZ4740_REG_CODEC_1, + JZ4740_CODEC_1_SUSPEND | JZ4740_CODEC_1_RESET, 0); + + for (i = 0; i < ARRAY_SIZE(jz4740_codec_regs); ++i) + jz4740_codec_write(codec, i, cache[i]); +} + +static int jz4740_codec_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + unsigned int mask; + unsigned int value; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + mask = JZ4740_CODEC_1_VREF_DISABLE | + JZ4740_CODEC_1_VREF_AMP_DISABLE | + JZ4740_CODEC_1_HEADPHONE_POWERDOWN_M; + value = 0; + + snd_soc_update_bits(codec, JZ4740_REG_CODEC_1, mask, value); + break; + case SND_SOC_BIAS_STANDBY: + /* The only way to clear the suspend flag is to reset the codec */ + if (codec->bias_level == SND_SOC_BIAS_OFF) + jz4740_codec_wakeup(codec); + + mask = JZ4740_CODEC_1_VREF_DISABLE | + JZ4740_CODEC_1_VREF_AMP_DISABLE | + JZ4740_CODEC_1_HEADPHONE_POWERDOWN_M; + value = JZ4740_CODEC_1_VREF_DISABLE | + JZ4740_CODEC_1_VREF_AMP_DISABLE | + JZ4740_CODEC_1_HEADPHONE_POWERDOWN_M; + + snd_soc_update_bits(codec, JZ4740_REG_CODEC_1, mask, value); + break; + case SND_SOC_BIAS_OFF: + mask = JZ4740_CODEC_1_SUSPEND; + value = JZ4740_CODEC_1_SUSPEND; + + snd_soc_update_bits(codec, JZ4740_REG_CODEC_1, mask, value); + break; + default: + break; + } + + codec->bias_level = level; + + return 0; +} + +static struct snd_soc_codec *jz4740_codec_codec; + +static int jz4740_codec_dev_probe(struct platform_device *pdev) +{ + int ret; + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = jz4740_codec_codec; + + BUG_ON(!codec); + + socdev->card->codec = codec; + + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret) { + dev_err(&pdev->dev, "Failed to create pcms: %d\n", ret); + return ret; + } + + snd_soc_add_controls(codec, jz4740_codec_controls, + ARRAY_SIZE(jz4740_codec_controls)); + + snd_soc_dapm_new_controls(codec, jz4740_codec_dapm_widgets, + ARRAY_SIZE(jz4740_codec_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, jz4740_codec_dapm_routes, + ARRAY_SIZE(jz4740_codec_dapm_routes)); + + snd_soc_dapm_new_widgets(codec); + + return 0; +} + +static int jz4740_codec_dev_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +#ifdef CONFIG_PM_SLEEP + +static int jz4740_codec_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + return jz4740_codec_set_bias_level(codec, SND_SOC_BIAS_OFF); +} + +static int jz4740_codec_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + return jz4740_codec_set_bias_level(codec, SND_SOC_BIAS_STANDBY); +} + +#else +#define jz4740_codec_suspend NULL +#define jz4740_codec_resume NULL +#endif + +struct snd_soc_codec_device soc_codec_dev_jz4740_codec = { + .probe = jz4740_codec_dev_probe, + .remove = jz4740_codec_dev_remove, + .suspend = jz4740_codec_suspend, + .resume = jz4740_codec_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_jz4740_codec); + +static int __devinit jz4740_codec_probe(struct platform_device *pdev) +{ + int ret; + struct jz4740_codec *jz4740_codec; + struct snd_soc_codec *codec; + struct resource *mem; + + jz4740_codec = kzalloc(sizeof(*jz4740_codec), GFP_KERNEL); + if (!jz4740_codec) + return -ENOMEM; + + mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!mem) { + dev_err(&pdev->dev, "Failed to get mmio memory resource\n"); + ret = -ENOENT; + goto err_free_codec; + } + + mem = request_mem_region(mem->start, resource_size(mem), pdev->name); + if (!mem) { + dev_err(&pdev->dev, "Failed to request mmio memory region\n"); + ret = -EBUSY; + goto err_free_codec; + } + + jz4740_codec->base = ioremap(mem->start, resource_size(mem)); + if (!jz4740_codec->base) { + dev_err(&pdev->dev, "Failed to ioremap mmio memory\n"); + ret = -EBUSY; + goto err_release_mem_region; + } + jz4740_codec->mem = mem; + + jz4740_codec_dai.dev = &pdev->dev; + + codec = &jz4740_codec->codec; + + codec->dev = &pdev->dev; + codec->name = "jz4740"; + codec->owner = THIS_MODULE; + + codec->read = jz4740_codec_read; + codec->write = jz4740_codec_write; + codec->set_bias_level = jz4740_codec_set_bias_level; + codec->bias_level = SND_SOC_BIAS_OFF; + + codec->dai = &jz4740_codec_dai; + codec->num_dai = 1; + + codec->reg_cache = jz4740_codec->reg_cache; + codec->reg_cache_size = 2; + memcpy(codec->reg_cache, jz4740_codec_regs, sizeof(jz4740_codec_regs)); + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + jz4740_codec_codec = codec; + + snd_soc_update_bits(codec, JZ4740_REG_CODEC_1, + JZ4740_CODEC_1_SW2_ENABLE, JZ4740_CODEC_1_SW2_ENABLE); + + platform_set_drvdata(pdev, jz4740_codec); + + ret = snd_soc_register_codec(codec); + if (ret) { + dev_err(&pdev->dev, "Failed to register codec\n"); + goto err_iounmap; + } + + ret = snd_soc_register_dai(&jz4740_codec_dai); + if (ret) { + dev_err(&pdev->dev, "Failed to register codec dai\n"); + goto err_unregister_codec; + } + + jz4740_codec_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; + +err_unregister_codec: + snd_soc_unregister_codec(codec); +err_iounmap: + iounmap(jz4740_codec->base); +err_release_mem_region: + release_mem_region(mem->start, resource_size(mem)); +err_free_codec: + kfree(jz4740_codec); + + return ret; +} + +static int __devexit jz4740_codec_remove(struct platform_device *pdev) +{ + struct jz4740_codec *jz4740_codec = platform_get_drvdata(pdev); + struct resource *mem = jz4740_codec->mem; + + snd_soc_unregister_dai(&jz4740_codec_dai); + snd_soc_unregister_codec(&jz4740_codec->codec); + + iounmap(jz4740_codec->base); + release_mem_region(mem->start, resource_size(mem)); + + platform_set_drvdata(pdev, NULL); + kfree(jz4740_codec); + + return 0; +} + +static struct platform_driver jz4740_codec_driver = { + .probe = jz4740_codec_probe, + .remove = __devexit_p(jz4740_codec_remove), + .driver = { + .name = "jz4740-codec", + .owner = THIS_MODULE, + }, +}; + +static int __init jz4740_codec_init(void) +{ + return platform_driver_register(&jz4740_codec_driver); +} +module_init(jz4740_codec_init); + +static void __exit jz4740_codec_exit(void) +{ + platform_driver_unregister(&jz4740_codec_driver); +} +module_exit(jz4740_codec_exit); + +MODULE_DESCRIPTION("JZ4740 SoC internal codec driver"); +MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:jz4740-codec"); diff --git a/sound/soc/codecs/jz4740.h b/sound/soc/codecs/jz4740.h new file mode 100644 index 00000000000..b5a0691be76 --- /dev/null +++ b/sound/soc/codecs/jz4740.h @@ -0,0 +1,20 @@ +/* + * Copyright (C) 2009, Lars-Peter Clausen <lars@metafoo.de> + * + * 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. + * + * 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., + * 675 Mass Ave, Cambridge, MA 02139, USA. + * + */ + +#ifndef __SND_SOC_CODECS_JZ4740_CODEC_H__ +#define __SND_SOC_CODECS_JZ4740_CODEC_H__ + +extern struct snd_soc_dai jz4740_codec_dai; +extern struct snd_soc_codec_device soc_codec_dev_jz4740_codec; + +#endif diff --git a/sound/soc/codecs/spdif_transciever.c b/sound/soc/codecs/spdif_transciever.c index a6319114105..9119836051a 100644 --- a/sound/soc/codecs/spdif_transciever.c +++ b/sound/soc/codecs/spdif_transciever.c @@ -16,8 +16,10 @@ #include <linux/module.h> #include <linux/moduleparam.h> +#include <linux/slab.h> #include <sound/soc.h> #include <sound/pcm.h> +#include <sound/initval.h> #include "spdif_transciever.h" @@ -26,6 +28,48 @@ MODULE_LICENSE("GPL"); #define STUB_RATES SNDRV_PCM_RATE_8000_96000 #define STUB_FORMATS SNDRV_PCM_FMTBIT_S16_LE +static struct snd_soc_codec *spdif_dit_codec; + +static int spdif_dit_codec_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret; + + if (spdif_dit_codec == NULL) { + dev_err(&pdev->dev, "Codec device not registered\n"); + return -ENODEV; + } + + socdev->card->codec = spdif_dit_codec; + codec = spdif_dit_codec; + + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(codec->dev, "failed to create pcms: %d\n", ret); + goto err_create_pcms; + } + + return 0; + +err_create_pcms: + return ret; +} + +static int spdif_dit_codec_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_spdif_dit = { + .probe = spdif_dit_codec_probe, + .remove = spdif_dit_codec_remove, +}; EXPORT_SYMBOL_GPL(soc_codec_dev_spdif_dit); + struct snd_soc_dai dit_stub_dai = { .name = "DIT", .playback = { @@ -40,13 +84,61 @@ EXPORT_SYMBOL_GPL(dit_stub_dai); static int spdif_dit_probe(struct platform_device *pdev) { + struct snd_soc_codec *codec; + int ret; + + if (spdif_dit_codec) { + dev_err(&pdev->dev, "Another Codec is registered\n"); + ret = -EINVAL; + goto err_reg_codec; + } + + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) + return -ENOMEM; + + codec->dev = &pdev->dev; + + mutex_init(&codec->mutex); + + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->name = "spdif-dit"; + codec->owner = THIS_MODULE; + codec->dai = &dit_stub_dai; + codec->num_dai = 1; + + spdif_dit_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret < 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + goto err_reg_codec; + } + dit_stub_dai.dev = &pdev->dev; - return snd_soc_register_dai(&dit_stub_dai); + ret = snd_soc_register_dai(&dit_stub_dai); + if (ret < 0) { + dev_err(codec->dev, "Failed to register dai: %d\n", ret); + goto err_reg_dai; + } + + return 0; + +err_reg_dai: + snd_soc_unregister_codec(codec); +err_reg_codec: + kfree(spdif_dit_codec); + return ret; } static int spdif_dit_remove(struct platform_device *pdev) { snd_soc_unregister_dai(&dit_stub_dai); + snd_soc_unregister_codec(spdif_dit_codec); + kfree(spdif_dit_codec); + spdif_dit_codec = NULL; return 0; } diff --git a/sound/soc/codecs/spdif_transciever.h b/sound/soc/codecs/spdif_transciever.h index 296f2eb6c4e..1e102124f54 100644 --- a/sound/soc/codecs/spdif_transciever.h +++ b/sound/soc/codecs/spdif_transciever.h @@ -12,6 +12,7 @@ #ifndef CODEC_STUBS_H #define CODEC_STUBS_H +extern struct snd_soc_codec_device soc_codec_dev_spdif_dit; extern struct snd_soc_dai dit_stub_dai; #endif /* CODEC_STUBS_H */ diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index b0bae3508b2..0a4b0fef335 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -560,13 +560,16 @@ static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec, switch (level) { case SND_SOC_BIAS_ON: /* vref/mid, osc on, dac unmute */ + reg &= ~(TLV320AIC23_DEVICE_PWR_OFF | TLV320AIC23_OSC_OFF | \ + TLV320AIC23_DAC_OFF); tlv320aic23_write(codec, TLV320AIC23_PWR, reg); break; case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: /* everything off except vref/vmid, */ - tlv320aic23_write(codec, TLV320AIC23_PWR, reg | 0x0040); + tlv320aic23_write(codec, TLV320AIC23_PWR, reg | \ + TLV320AIC23_CLK_OFF); break; case SND_SOC_BIAS_OFF: /* everything off, dac mute, inactive */ @@ -615,7 +618,6 @@ static int tlv320aic23_suspend(struct platform_device *pdev, struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->card->codec; - tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0); tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; @@ -632,7 +634,6 @@ static int tlv320aic23_resume(struct platform_device *pdev) u16 val = tlv320aic23_read_reg_cache(codec, reg); tlv320aic23_write(codec, reg, val); } - tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 65adc77eada..8651b01ed22 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -49,8 +49,6 @@ #define NSAMPLE_MAX 5700 -#define LATENCY_TIME_MS 20 - #define MODE7_LTHR 10 #define MODE7_UTHR (DAC33_BUFFER_SIZE_SAMPLES - 10) @@ -62,6 +60,9 @@ #define US_TO_SAMPLES(rate, us) \ (rate / (1000000 / us)) +#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); @@ -107,6 +108,10 @@ struct tlv320dac33_priv { * this */ enum dac33_fifo_modes fifo_mode;/* FIFO mode selection */ unsigned int nsample; /* burst read amount from host */ + int mode1_latency; /* latency caused by the i2c writes in + * us */ + int auto_fifo_config; /* Configure the FIFO based on the + * period size */ u8 burst_bclkdiv; /* BCLK divider value in burst mode */ unsigned int burst_rate; /* Interface speed in Burst modes */ @@ -120,6 +125,8 @@ struct tlv320dac33_priv { * samples */ unsigned int mode7_us_to_lthr; /* Time to reach lthr from uthr */ + unsigned int uthr; + enum dac33_state state; }; @@ -442,6 +449,39 @@ static int dac33_set_nsample(struct snd_kcontrol *kcontrol, return ret; } +static int dac33_get_uthr(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.integer.value[0] = dac33->uthr; + + return 0; +} + +static int dac33_set_uthr(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + if (dac33->substream) + return -EBUSY; + + if (dac33->uthr == ucontrol->value.integer.value[0]) + return 0; + + if (ucontrol->value.integer.value[0] < (MODE7_LTHR + 10) || + ucontrol->value.integer.value[0] > MODE7_UTHR) + ret = -EINVAL; + else + dac33->uthr = ucontrol->value.integer.value[0]; + + return ret; +} + static int dac33_get_fifo_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -503,13 +543,18 @@ static const struct snd_kcontrol_new dac33_snd_controls[] = { DAC33_LINEL_TO_LLO_VOL, DAC33_LINER_TO_RLO_VOL, 0, 127, 1), }; -static const struct snd_kcontrol_new dac33_nsample_snd_controls[] = { - SOC_SINGLE_EXT("nSample", 0, 0, 5900, 0, - dac33_get_nsample, dac33_set_nsample), +static const struct snd_kcontrol_new dac33_mode_snd_controls[] = { SOC_ENUM_EXT("FIFO Mode", dac33_fifo_mode_enum, dac33_get_fifo_mode, dac33_set_fifo_mode), }; +static const struct snd_kcontrol_new dac33_fifo_snd_controls[] = { + SOC_SINGLE_EXT("nSample", 0, 0, 5900, 0, + dac33_get_nsample, dac33_set_nsample), + SOC_SINGLE_EXT("UTHR", 0, 0, MODE7_UTHR, 0, + dac33_get_uthr, dac33_set_uthr), +}; + /* Analog bypass */ static const struct snd_kcontrol_new dac33_dapm_abypassl_control = SOC_DAPM_SINGLE("Switch", DAC33_LINEL_TO_LLO_VOL, 7, 1, 1); @@ -612,7 +657,7 @@ static inline void dac33_prefill_handler(struct tlv320dac33_priv *dac33) switch (dac33->fifo_mode) { case DAC33_FIFO_MODE1: dac33_write16(codec, DAC33_NSAMPLE_MSB, - DAC33_THRREG(dac33->nsample + dac33->alarm_threshold)); + DAC33_THRREG(dac33->nsample)); /* Take the timestamps */ spin_lock_irq(&dac33->lock); @@ -761,6 +806,10 @@ static void dac33_shutdown(struct snd_pcm_substream *substream, struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); dac33->substream = NULL; + + /* Reset the nSample restrictions */ + dac33->nsample_min = 0; + dac33->nsample_max = NSAMPLE_MAX; } static int dac33_hw_params(struct snd_pcm_substream *substream, @@ -985,7 +1034,7 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream) * Configure the threshold levels, and leave 10 sample space * at the bottom, and also at the top of the FIFO */ - dac33_write16(codec, DAC33_UTHR_MSB, DAC33_THRREG(MODE7_UTHR)); + dac33_write16(codec, DAC33_UTHR_MSB, DAC33_THRREG(dac33->uthr)); dac33_write16(codec, DAC33_LTHR_MSB, DAC33_THRREG(MODE7_LTHR)); break; default: @@ -1003,57 +1052,71 @@ static void dac33_calculate_times(struct snd_pcm_substream *substream) struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->card->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; unsigned int nsample_limit; /* In bypass mode we don't need to calculate */ if (!dac33->fifo_mode) return; - /* Number of samples (16bit, stereo) in one period */ - dac33->nsample_min = snd_pcm_lib_period_bytes(substream) / 4; - - /* Number of samples (16bit, stereo) in ALSA buffer */ - dac33->nsample_max = snd_pcm_lib_buffer_bytes(substream) / 4; - /* Subtract one period from the total */ - dac33->nsample_max -= dac33->nsample_min; - - /* Number of samples for LATENCY_TIME_MS / 2 */ - dac33->alarm_threshold = substream->runtime->rate / - (1000 / (LATENCY_TIME_MS / 2)); - - /* Find and fix up the lowest nsmaple limit */ - nsample_limit = substream->runtime->rate / (1000 / LATENCY_TIME_MS); - - if (dac33->nsample_min < nsample_limit) - dac33->nsample_min = nsample_limit; - - if (dac33->nsample < dac33->nsample_min) - dac33->nsample = dac33->nsample_min; - - /* - * Find and fix up the highest nsmaple limit - * In order to not overflow the DAC33 buffer substract the - * alarm_threshold value from the size of the DAC33 buffer - */ - nsample_limit = DAC33_BUFFER_SIZE_SAMPLES - dac33->alarm_threshold; - - if (dac33->nsample_max > nsample_limit) - dac33->nsample_max = nsample_limit; - - if (dac33->nsample > dac33->nsample_max) - dac33->nsample = dac33->nsample_max; - switch (dac33->fifo_mode) { case DAC33_FIFO_MODE1: + /* Number of samples under i2c latency */ + dac33->alarm_threshold = US_TO_SAMPLES(rate, + dac33->mode1_latency); + if (dac33->auto_fifo_config) { + if (period_size <= dac33->alarm_threshold) + /* + * Configure nSamaple to number of periods, + * which covers the latency requironment. + */ + dac33->nsample = period_size * + ((dac33->alarm_threshold / period_size) + + (dac33->alarm_threshold % period_size ? + 1 : 0)); + else + dac33->nsample = period_size; + } else { + /* nSample time shall not be shorter than i2c latency */ + dac33->nsample_min = dac33->alarm_threshold; + /* + * nSample should not be bigger than alsa buffer minus + * size of one period to avoid overruns + */ + dac33->nsample_max = substream->runtime->buffer_size - + period_size; + nsample_limit = DAC33_BUFFER_SIZE_SAMPLES - + dac33->alarm_threshold; + if (dac33->nsample_max > nsample_limit) + dac33->nsample_max = nsample_limit; + + /* Correct the nSample if it is outside of the ranges */ + if (dac33->nsample < dac33->nsample_min) + dac33->nsample = dac33->nsample_min; + if (dac33->nsample > dac33->nsample_max) + dac33->nsample = dac33->nsample_max; + } + dac33->mode1_us_burst = SAMPLES_TO_US(dac33->burst_rate, dac33->nsample); dac33->t_stamp1 = 0; dac33->t_stamp2 = 0; break; case DAC33_FIFO_MODE7: + if (dac33->auto_fifo_config) { + dac33->uthr = UTHR_FROM_PERIOD_SIZE( + period_size, + rate, + dac33->burst_rate) + 9; + if (dac33->uthr > MODE7_UTHR) + dac33->uthr = MODE7_UTHR; + if (dac33->uthr < (MODE7_LTHR + 10)) + dac33->uthr = (MODE7_LTHR + 10); + } dac33->mode7_us_to_lthr = - SAMPLES_TO_US(substream->runtime->rate, - MODE7_UTHR - MODE7_LTHR + 1); + SAMPLES_TO_US(substream->runtime->rate, + dac33->uthr - MODE7_LTHR + 1); dac33->t_stamp1 = 0; break; default: @@ -1104,7 +1167,7 @@ static snd_pcm_sframes_t dac33_dai_delay( struct snd_soc_codec *codec = socdev->card->codec; struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); unsigned long long t0, t1, t_now; - unsigned int time_delta; + unsigned int time_delta, uthr; int samples_out, samples_in, samples; snd_pcm_sframes_t delay = 0; @@ -1182,6 +1245,7 @@ static snd_pcm_sframes_t dac33_dai_delay( case DAC33_FIFO_MODE7: spin_lock(&dac33->lock); t0 = dac33->t_stamp1; + uthr = dac33->uthr; spin_unlock(&dac33->lock); t_now = ktime_to_us(ktime_get()); @@ -1194,7 +1258,7 @@ static snd_pcm_sframes_t dac33_dai_delay( * Either the timestamps are messed or equal. Report * maximum delay */ - delay = MODE7_UTHR; + delay = uthr; goto out; } @@ -1208,8 +1272,8 @@ static snd_pcm_sframes_t dac33_dai_delay( substream->runtime->rate, time_delta); - if (likely(MODE7_UTHR > samples_out)) - delay = MODE7_UTHR - samples_out; + if (likely(uthr > samples_out)) + delay = uthr - samples_out; else delay = 0; } else { @@ -1227,8 +1291,8 @@ static snd_pcm_sframes_t dac33_dai_delay( time_delta); delay = MODE7_LTHR + samples_in - samples_out; - if (unlikely(delay > MODE7_UTHR)) - delay = MODE7_UTHR; + if (unlikely(delay > uthr)) + delay = uthr; } break; default: @@ -1347,10 +1411,15 @@ static int dac33_soc_probe(struct platform_device *pdev) snd_soc_add_controls(codec, dac33_snd_controls, ARRAY_SIZE(dac33_snd_controls)); - /* Only add the nSample controls, if we have valid IRQ number */ - if (dac33->irq >= 0) - snd_soc_add_controls(codec, dac33_nsample_snd_controls, - ARRAY_SIZE(dac33_nsample_snd_controls)); + /* Only add the FIFO controls, if we have valid IRQ number */ + if (dac33->irq >= 0) { + snd_soc_add_controls(codec, dac33_mode_snd_controls, + ARRAY_SIZE(dac33_mode_snd_controls)); + /* FIFO usage controls only, if autoio config is not selected */ + if (!dac33->auto_fifo_config) + snd_soc_add_controls(codec, dac33_fifo_snd_controls, + ARRAY_SIZE(dac33_fifo_snd_controls)); + } dac33_add_widgets(codec); @@ -1481,9 +1550,14 @@ static int __devinit dac33_i2c_probe(struct i2c_client *client, /* Pre calculate the burst rate */ dac33->burst_rate = BURST_BASEFREQ_HZ / dac33->burst_bclkdiv / 32; dac33->keep_bclk = pdata->keep_bclk; + dac33->auto_fifo_config = pdata->auto_fifo_config; + dac33->mode1_latency = pdata->mode1_latency; + if (!dac33->mode1_latency) + dac33->mode1_latency = 10000; /* 10ms */ dac33->irq = client->irq; dac33->nsample = NSAMPLE_MAX; dac33->nsample_max = NSAMPLE_MAX; + dac33->uthr = MODE7_UTHR; /* Disable FIFO use by default */ dac33->fifo_mode = DAC33_FIFO_BYPASS; diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index b4fcdb01fc4..7b618bbff88 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -43,37 +43,37 @@ */ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = { 0x00, /* this register not used */ - 0x91, /* REG_CODEC_MODE (0x1) */ - 0xc3, /* REG_OPTION (0x2) */ + 0x00, /* REG_CODEC_MODE (0x1) */ + 0x00, /* REG_OPTION (0x2) */ 0x00, /* REG_UNKNOWN (0x3) */ 0x00, /* REG_MICBIAS_CTL (0x4) */ - 0x20, /* REG_ANAMICL (0x5) */ + 0x00, /* REG_ANAMICL (0x5) */ 0x00, /* REG_ANAMICR (0x6) */ 0x00, /* REG_AVADC_CTL (0x7) */ 0x00, /* REG_ADCMICSEL (0x8) */ 0x00, /* REG_DIGMIXING (0x9) */ - 0x0c, /* REG_ATXL1PGA (0xA) */ - 0x0c, /* REG_ATXR1PGA (0xB) */ - 0x00, /* REG_AVTXL2PGA (0xC) */ - 0x00, /* REG_AVTXR2PGA (0xD) */ + 0x0f, /* REG_ATXL1PGA (0xA) */ + 0x0f, /* REG_ATXR1PGA (0xB) */ + 0x0f, /* REG_AVTXL2PGA (0xC) */ + 0x0f, /* REG_AVTXR2PGA (0xD) */ 0x00, /* REG_AUDIO_IF (0xE) */ 0x00, /* REG_VOICE_IF (0xF) */ - 0x00, /* REG_ARXR1PGA (0x10) */ - 0x00, /* REG_ARXL1PGA (0x11) */ - 0x6c, /* REG_ARXR2PGA (0x12) */ - 0x6c, /* REG_ARXL2PGA (0x13) */ - 0x00, /* REG_VRXPGA (0x14) */ + 0x3f, /* REG_ARXR1PGA (0x10) */ + 0x3f, /* REG_ARXL1PGA (0x11) */ + 0x3f, /* REG_ARXR2PGA (0x12) */ + 0x3f, /* REG_ARXL2PGA (0x13) */ + 0x25, /* REG_VRXPGA (0x14) */ 0x00, /* REG_VSTPGA (0x15) */ 0x00, /* REG_VRX2ARXPGA (0x16) */ 0x00, /* REG_AVDAC_CTL (0x17) */ 0x00, /* REG_ARX2VTXPGA (0x18) */ - 0x00, /* REG_ARXL1_APGA_CTL (0x19) */ - 0x00, /* REG_ARXR1_APGA_CTL (0x1A) */ - 0x4a, /* REG_ARXL2_APGA_CTL (0x1B) */ - 0x4a, /* REG_ARXR2_APGA_CTL (0x1C) */ + 0x32, /* REG_ARXL1_APGA_CTL (0x19) */ + 0x32, /* REG_ARXR1_APGA_CTL (0x1A) */ + 0x32, /* REG_ARXL2_APGA_CTL (0x1B) */ + 0x32, /* REG_ARXR2_APGA_CTL (0x1C) */ 0x00, /* REG_ATX2ARXPGA (0x1D) */ 0x00, /* REG_BT_IF (0x1E) */ - 0x00, /* REG_BTPGA (0x1F) */ + 0x55, /* REG_BTPGA (0x1F) */ 0x00, /* REG_BTSTPGA (0x20) */ 0x00, /* REG_EAR_CTL (0x21) */ 0x00, /* REG_HS_SEL (0x22) */ @@ -85,32 +85,32 @@ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = { 0x00, /* REG_PRECKR_CTL (0x28) */ 0x00, /* REG_HFL_CTL (0x29) */ 0x00, /* REG_HFR_CTL (0x2A) */ - 0x00, /* REG_ALC_CTL (0x2B) */ + 0x05, /* REG_ALC_CTL (0x2B) */ 0x00, /* REG_ALC_SET1 (0x2C) */ 0x00, /* REG_ALC_SET2 (0x2D) */ 0x00, /* REG_BOOST_CTL (0x2E) */ 0x00, /* REG_SOFTVOL_CTL (0x2F) */ - 0x00, /* REG_DTMF_FREQSEL (0x30) */ + 0x13, /* REG_DTMF_FREQSEL (0x30) */ 0x00, /* REG_DTMF_TONEXT1H (0x31) */ 0x00, /* REG_DTMF_TONEXT1L (0x32) */ 0x00, /* REG_DTMF_TONEXT2H (0x33) */ 0x00, /* REG_DTMF_TONEXT2L (0x34) */ - 0x00, /* REG_DTMF_TONOFF (0x35) */ - 0x00, /* REG_DTMF_WANONOFF (0x36) */ + 0x79, /* REG_DTMF_TONOFF (0x35) */ + 0x11, /* REG_DTMF_WANONOFF (0x36) */ 0x00, /* REG_I2S_RX_SCRAMBLE_H (0x37) */ 0x00, /* REG_I2S_RX_SCRAMBLE_M (0x38) */ 0x00, /* REG_I2S_RX_SCRAMBLE_L (0x39) */ 0x06, /* REG_APLL_CTL (0x3A) */ 0x00, /* REG_DTMF_CTL (0x3B) */ - 0x00, /* REG_DTMF_PGA_CTL2 (0x3C) */ - 0x00, /* REG_DTMF_PGA_CTL1 (0x3D) */ + 0x44, /* REG_DTMF_PGA_CTL2 (0x3C) */ + 0x69, /* REG_DTMF_PGA_CTL1 (0x3D) */ 0x00, /* REG_MISC_SET_1 (0x3E) */ 0x00, /* REG_PCMBTMUX (0x3F) */ 0x00, /* not used (0x40) */ 0x00, /* not used (0x41) */ 0x00, /* not used (0x42) */ 0x00, /* REG_RX_PATH_SEL (0x43) */ - 0x00, /* REG_VDL_APGA_CTL (0x44) */ + 0x32, /* REG_VDL_APGA_CTL (0x44) */ 0x00, /* REG_VIBRA_CTL (0x45) */ 0x00, /* REG_VIBRA_SET (0x46) */ 0x00, /* REG_VIBRA_PWM_SET (0x47) */ @@ -143,6 +143,9 @@ struct twl4030_priv { u8 earpiece_enabled; u8 predrivel_enabled, predriver_enabled; u8 carkitl_enabled, carkitr_enabled; + + /* Delay needed after enabling the digimic interface */ + unsigned int digimic_delay; }; /* @@ -244,58 +247,95 @@ static void twl4030_codec_enable(struct snd_soc_codec *codec, int enable) udelay(10); } -static void twl4030_init_chip(struct snd_soc_codec *codec) +static inline void twl4030_check_defaults(struct snd_soc_codec *codec) { - u8 *cache = codec->reg_cache; - int i; + int i, difference = 0; + u8 val; + + dev_dbg(codec->dev, "Checking TWL audio default configuration\n"); + for (i = 1; i <= TWL4030_REG_MISC_SET_2; i++) { + twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &val, i); + if (val != twl4030_reg[i]) { + difference++; + dev_dbg(codec->dev, + "Reg 0x%02x: chip: 0x%02x driver: 0x%02x\n", + i, val, twl4030_reg[i]); + } + } + dev_dbg(codec->dev, "Found %d non maching registers. %s\n", + difference, difference ? "Not OK" : "OK"); +} - /* clear CODECPDZ prior to setting register defaults */ - twl4030_codec_enable(codec, 0); +static inline void twl4030_reset_registers(struct snd_soc_codec *codec) +{ + int i; /* set all audio section registers to reasonable defaults */ for (i = TWL4030_REG_OPTION; i <= TWL4030_REG_MISC_SET_2; i++) if (i != TWL4030_REG_APLL_CTL) - twl4030_write(codec, i, cache[i]); + twl4030_write(codec, i, twl4030_reg[i]); } -static void twl4030_apll_enable(struct snd_soc_codec *codec, int enable) +static void twl4030_init_chip(struct platform_device *pdev) { + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct twl4030_setup_data *setup = socdev->codec_data; + struct snd_soc_codec *codec = socdev->card->codec; struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); - int status = -1; + u8 reg, byte; + int i = 0; - if (enable) { - twl4030->apll_enabled++; - if (twl4030->apll_enabled == 1) - status = twl4030_codec_enable_resource( - TWL4030_CODEC_RES_APLL); - } else { - twl4030->apll_enabled--; - if (!twl4030->apll_enabled) - status = twl4030_codec_disable_resource( - TWL4030_CODEC_RES_APLL); - } + /* Check defaults, if instructed before anything else */ + if (setup && setup->check_defaults) + twl4030_check_defaults(codec); - if (status >= 0) - twl4030_write_reg_cache(codec, TWL4030_REG_APLL_CTL, status); -} + /* Reset registers, if no setup data or if instructed to do so */ + if (!setup || (setup && setup->reset_registers)) + twl4030_reset_registers(codec); -static void twl4030_power_up(struct snd_soc_codec *codec) -{ - struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); - u8 anamicl, regmisc1, byte; - int i = 0; + /* Refresh APLL_CTL register from HW */ + twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &byte, + TWL4030_REG_APLL_CTL); + twl4030_write_reg_cache(codec, TWL4030_REG_APLL_CTL, byte); + + /* anti-pop when changing analog gain */ + reg = twl4030_read_reg_cache(codec, TWL4030_REG_MISC_SET_1); + twl4030_write(codec, TWL4030_REG_MISC_SET_1, + reg | TWL4030_SMOOTH_ANAVOL_EN); - if (twl4030->codec_powered) + twl4030_write(codec, TWL4030_REG_OPTION, + TWL4030_ATXL1_EN | TWL4030_ATXR1_EN | + TWL4030_ARXL2_EN | TWL4030_ARXR2_EN); + + /* REG_ARXR2_APGA_CTL reset according to the TRM: 0dB, DA_EN */ + twl4030_write(codec, TWL4030_REG_ARXR2_APGA_CTL, 0x32); + + /* Machine dependent setup */ + if (!setup) return; - /* set CODECPDZ to turn on codec */ - twl4030_codec_enable(codec, 1); + twl4030->digimic_delay = setup->digimic_delay; + + /* Configuration for headset ramp delay from setup data */ + if (setup->sysclk != twl4030->sysclk) + dev_warn(codec->dev, + "Mismatch in APLL mclk: %u (configured: %u)\n", + setup->sysclk, twl4030->sysclk); + + reg = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET); + reg &= ~TWL4030_RAMP_DELAY; + reg |= (setup->ramp_delay_value << 2); + twl4030_write_reg_cache(codec, TWL4030_REG_HS_POPN_SET, reg); /* initiate offset cancellation */ - anamicl = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICL); + twl4030_codec_enable(codec, 1); + + reg = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICL); + reg &= ~TWL4030_OFFSET_CNCL_SEL; + reg |= setup->offset_cncl_path; twl4030_write(codec, TWL4030_REG_ANAMICL, - anamicl | TWL4030_CNCL_OFFSET_START); + reg | TWL4030_CNCL_OFFSET_START); /* wait for offset cancellation to complete */ do { @@ -310,23 +350,28 @@ static void twl4030_power_up(struct snd_soc_codec *codec) /* Make sure that the reg_cache has the same value as the HW */ twl4030_write_reg_cache(codec, TWL4030_REG_ANAMICL, byte); - /* anti-pop when changing analog gain */ - regmisc1 = twl4030_read_reg_cache(codec, TWL4030_REG_MISC_SET_1); - twl4030_write(codec, TWL4030_REG_MISC_SET_1, - regmisc1 | TWL4030_SMOOTH_ANAVOL_EN); - - /* toggle CODECPDZ as per TRM */ twl4030_codec_enable(codec, 0); - twl4030_codec_enable(codec, 1); } -/* - * Unconditional power down - */ -static void twl4030_power_down(struct snd_soc_codec *codec) +static void twl4030_apll_enable(struct snd_soc_codec *codec, int enable) { - /* power down */ - twl4030_codec_enable(codec, 0); + struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); + int status = -1; + + if (enable) { + twl4030->apll_enabled++; + if (twl4030->apll_enabled == 1) + status = twl4030_codec_enable_resource( + TWL4030_CODEC_RES_APLL); + } else { + twl4030->apll_enabled--; + if (!twl4030->apll_enabled) + status = twl4030_codec_disable_resource( + TWL4030_CODEC_RES_APLL); + } + + if (status >= 0) + twl4030_write_reg_cache(codec, TWL4030_REG_APLL_CTL, status); } /* Earpiece */ @@ -500,10 +545,11 @@ static const struct snd_kcontrol_new twl4030_dapm_abypassl2_control = static const struct snd_kcontrol_new twl4030_dapm_abypassv_control = SOC_DAPM_SINGLE("Switch", TWL4030_REG_VDL_APGA_CTL, 2, 1, 0); -/* Digital bypass gain, 0 mutes the bypass */ +/* Digital bypass gain, mute instead of -30dB */ static const unsigned int twl4030_dapm_dbypass_tlv[] = { - TLV_DB_RANGE_HEAD(2), - 0, 3, TLV_DB_SCALE_ITEM(-2400, 0, 1), + TLV_DB_RANGE_HEAD(3), + 0, 1, TLV_DB_SCALE_ITEM(-3000, 600, 1), + 2, 3, TLV_DB_SCALE_ITEM(-2400, 0, 0), 4, 7, TLV_DB_SCALE_ITEM(-1800, 600, 0), }; @@ -531,36 +577,6 @@ static const struct snd_kcontrol_new twl4030_dapm_dbypassv_control = TWL4030_REG_VSTPGA, 0, 0x29, 0, twl4030_dapm_dbypassv_tlv); -static int micpath_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) -{ - struct soc_enum *e = (struct soc_enum *)w->kcontrols->private_value; - unsigned char adcmicsel, micbias_ctl; - - adcmicsel = twl4030_read_reg_cache(w->codec, TWL4030_REG_ADCMICSEL); - micbias_ctl = twl4030_read_reg_cache(w->codec, TWL4030_REG_MICBIAS_CTL); - /* Prepare the bits for the given TX path: - * shift_l == 0: TX1 microphone path - * shift_l == 2: TX2 microphone path */ - if (e->shift_l) { - /* TX2 microphone path */ - if (adcmicsel & TWL4030_TX2IN_SEL) - micbias_ctl |= TWL4030_MICBIAS2_CTL; /* digimic */ - else - micbias_ctl &= ~TWL4030_MICBIAS2_CTL; - } else { - /* TX1 microphone path */ - if (adcmicsel & TWL4030_TX1IN_SEL) - micbias_ctl |= TWL4030_MICBIAS1_CTL; /* digimic */ - else - micbias_ctl &= ~TWL4030_MICBIAS1_CTL; - } - - twl4030_write(w->codec, TWL4030_REG_MICBIAS_CTL, micbias_ctl); - - return 0; -} - /* * Output PGA builder: * Handle the muting and unmuting of the given output (turning off the @@ -814,6 +830,16 @@ static int headsetrpga_event(struct snd_soc_dapm_widget *w, return 0; } +static int digimic_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(w->codec); + + if (twl4030->digimic_delay) + mdelay(twl4030->digimic_delay); + return 0; +} + /* * Some of the gain controls in TWL (mostly those which are associated with * the outputs) are implemented in an interesting way: @@ -1374,14 +1400,10 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { /* Analog/Digital mic path selection. TX1 Left/Right: either analog Left/Right or Digimic0 TX2 Left/Right: either analog Left/Right or Digimic1 */ - SND_SOC_DAPM_MUX_E("TX1 Capture Route", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_micpathtx1_control, micpath_event, - SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD| - SND_SOC_DAPM_POST_REG), - SND_SOC_DAPM_MUX_E("TX2 Capture Route", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_micpathtx2_control, micpath_event, - SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD| - SND_SOC_DAPM_POST_REG), + SND_SOC_DAPM_MUX("TX1 Capture Route", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_micpathtx1_control), + SND_SOC_DAPM_MUX("TX2 Capture Route", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_micpathtx2_control), /* Analog input mixers for the capture amplifiers */ SND_SOC_DAPM_MIXER("Analog Left", @@ -1398,10 +1420,17 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_PGA("ADC Physical Right", TWL4030_REG_AVADC_CTL, 1, 0, NULL, 0), - SND_SOC_DAPM_PGA("Digimic0 Enable", - TWL4030_REG_ADCMICSEL, 1, 0, NULL, 0), - SND_SOC_DAPM_PGA("Digimic1 Enable", - TWL4030_REG_ADCMICSEL, 3, 0, NULL, 0), + SND_SOC_DAPM_PGA_E("Digimic0 Enable", + TWL4030_REG_ADCMICSEL, 1, 0, NULL, 0, + digimic_event, SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_E("Digimic1 Enable", + TWL4030_REG_ADCMICSEL, 3, 0, NULL, 0, + digimic_event, SND_SOC_DAPM_POST_PMU), + + SND_SOC_DAPM_SUPPLY("micbias1 select", TWL4030_REG_MICBIAS_CTL, 5, 0, + NULL, 0), + SND_SOC_DAPM_SUPPLY("micbias2 select", TWL4030_REG_MICBIAS_CTL, 6, 0, + NULL, 0), SND_SOC_DAPM_MICBIAS("Mic Bias 1", TWL4030_REG_MICBIAS_CTL, 0, 0), SND_SOC_DAPM_MICBIAS("Mic Bias 2", TWL4030_REG_MICBIAS_CTL, 1, 0), @@ -1419,8 +1448,11 @@ static const struct snd_soc_dapm_route intercon[] = { /* Supply for the digital part (APLL) */ {"Digital Voice Playback Mixer", NULL, "APLL Enable"}, - {"Digital R1 Playback Mixer", NULL, "AIF Enable"}, - {"Digital L1 Playback Mixer", NULL, "AIF Enable"}, + {"DAC Left1", NULL, "AIF Enable"}, + {"DAC Right1", NULL, "AIF Enable"}, + {"DAC Left2", NULL, "AIF Enable"}, + {"DAC Right1", NULL, "AIF Enable"}, + {"Digital R2 Playback Mixer", NULL, "AIF Enable"}, {"Digital L2 Playback Mixer", NULL, "AIF Enable"}, @@ -1491,10 +1523,10 @@ static const struct snd_soc_dapm_route intercon[] = { /* outputs */ /* Must be always connected (for AIF and APLL) */ - {"Virtual HiFi OUT", NULL, "Digital L1 Playback Mixer"}, - {"Virtual HiFi OUT", NULL, "Digital R1 Playback Mixer"}, - {"Virtual HiFi OUT", NULL, "Digital L2 Playback Mixer"}, - {"Virtual HiFi OUT", NULL, "Digital R2 Playback Mixer"}, + {"Virtual HiFi OUT", NULL, "DAC Left1"}, + {"Virtual HiFi OUT", NULL, "DAC Right1"}, + {"Virtual HiFi OUT", NULL, "DAC Left2"}, + {"Virtual HiFi OUT", NULL, "DAC Right2"}, /* Must be always connected (for APLL) */ {"Virtual Voice OUT", NULL, "Digital Voice Playback Mixer"}, /* Physical outputs */ @@ -1531,6 +1563,9 @@ static const struct snd_soc_dapm_route intercon[] = { {"Digimic0 Enable", NULL, "DIGIMIC0"}, {"Digimic1 Enable", NULL, "DIGIMIC1"}, + {"DIGIMIC0", NULL, "micbias1 select"}, + {"DIGIMIC1", NULL, "micbias2 select"}, + /* TX1 Left capture path */ {"TX1 Capture Route", "Analog", "ADC Physical Left"}, {"TX1 Capture Route", "Digimic0", "Digimic0 Enable"}, @@ -1605,10 +1640,10 @@ static int twl4030_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: if (codec->bias_level == SND_SOC_BIAS_OFF) - twl4030_power_up(codec); + twl4030_codec_enable(codec, 1); break; case SND_SOC_BIAS_OFF: - twl4030_power_down(codec); + twl4030_codec_enable(codec, 0); break; } codec->bias_level = level; @@ -1794,13 +1829,6 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - if (mode != old_mode) { - /* change rate and set CODECPDZ */ - twl4030_codec_enable(codec, 0); - twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode); - twl4030_codec_enable(codec, 1); - } - /* sample size */ old_format = twl4030_read_reg_cache(codec, TWL4030_REG_AUDIO_IF); format = old_format; @@ -1818,16 +1846,20 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - if (format != old_format) { - - /* clear CODECPDZ before changing format (codec requirement) */ - twl4030_codec_enable(codec, 0); - - /* change format */ - twl4030_write(codec, TWL4030_REG_AUDIO_IF, format); - - /* set CODECPDZ afterwards */ - twl4030_codec_enable(codec, 1); + if (format != old_format || mode != old_mode) { + if (twl4030->codec_powered) { + /* + * If the codec is powered, than we need to toggle the + * codec power. + */ + twl4030_codec_enable(codec, 0); + twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode); + twl4030_write(codec, TWL4030_REG_AUDIO_IF, format); + twl4030_codec_enable(codec, 1); + } else { + twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode); + twl4030_write(codec, TWL4030_REG_AUDIO_IF, format); + } } /* Store the important parameters for the DAI configuration and set @@ -1877,6 +1909,7 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; + struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); u8 old_format, format; /* get format */ @@ -1911,15 +1944,17 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai, } if (format != old_format) { - - /* clear CODECPDZ before changing format (codec requirement) */ - twl4030_codec_enable(codec, 0); - - /* change format */ - twl4030_write(codec, TWL4030_REG_AUDIO_IF, format); - - /* set CODECPDZ afterwards */ - twl4030_codec_enable(codec, 1); + if (twl4030->codec_powered) { + /* + * If the codec is powered, than we need to toggle the + * codec power. + */ + twl4030_codec_enable(codec, 0); + twl4030_write(codec, TWL4030_REG_AUDIO_IF, format); + twl4030_codec_enable(codec, 1); + } else { + twl4030_write(codec, TWL4030_REG_AUDIO_IF, format); + } } return 0; @@ -2011,6 +2046,7 @@ static int twl4030_voice_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->card->codec; + struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); u8 old_mode, mode; /* Enable voice digital filters */ @@ -2035,10 +2071,17 @@ static int twl4030_voice_hw_params(struct snd_pcm_substream *substream, } if (mode != old_mode) { - /* change rate and set CODECPDZ */ - twl4030_codec_enable(codec, 0); - twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode); - twl4030_codec_enable(codec, 1); + if (twl4030->codec_powered) { + /* + * If the codec is powered, than we need to toggle the + * codec power. + */ + twl4030_codec_enable(codec, 0); + twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode); + twl4030_codec_enable(codec, 1); + } else { + twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode); + } } return 0; @@ -2068,6 +2111,7 @@ static int twl4030_voice_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; + struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); u8 old_format, format; /* get format */ @@ -2099,10 +2143,17 @@ static int twl4030_voice_set_dai_fmt(struct snd_soc_dai *codec_dai, } if (format != old_format) { - /* change format and set CODECPDZ */ - twl4030_codec_enable(codec, 0); - twl4030_write(codec, TWL4030_REG_VOICE_IF, format); - twl4030_codec_enable(codec, 1); + if (twl4030->codec_powered) { + /* + * If the codec is powered, than we need to toggle the + * codec power. + */ + twl4030_codec_enable(codec, 0); + twl4030_write(codec, TWL4030_REG_VOICE_IF, format); + twl4030_codec_enable(codec, 1); + } else { + twl4030_write(codec, TWL4030_REG_VOICE_IF, format); + } } return 0; @@ -2202,31 +2253,15 @@ static struct snd_soc_codec *twl4030_codec; static int twl4030_soc_probe(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct twl4030_setup_data *setup = socdev->codec_data; struct snd_soc_codec *codec; - struct twl4030_priv *twl4030; int ret; BUG_ON(!twl4030_codec); codec = twl4030_codec; - twl4030 = snd_soc_codec_get_drvdata(codec); socdev->card->codec = codec; - /* Configuration for headset ramp delay from setup data */ - if (setup) { - unsigned char hs_pop; - - if (setup->sysclk != twl4030->sysclk) - dev_warn(&pdev->dev, - "Mismatch in APLL mclk: %u (configured: %u)\n", - setup->sysclk, twl4030->sysclk); - - hs_pop = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET); - hs_pop &= ~TWL4030_RAMP_DELAY; - hs_pop |= (setup->ramp_delay_value << 2); - twl4030_write_reg_cache(codec, TWL4030_REG_HS_POPN_SET, hs_pop); - } + twl4030_init_chip(pdev); /* register pcms */ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); @@ -2247,6 +2282,8 @@ static int twl4030_soc_remove(struct platform_device *pdev) struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->card->codec; + /* Reset registers to their chip default before leaving */ + twl4030_reset_registers(codec); twl4030_set_bias_level(codec, SND_SOC_BIAS_OFF); snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); @@ -2287,6 +2324,7 @@ static int __devinit twl4030_codec_probe(struct platform_device *pdev) codec->read = twl4030_read_reg_cache; codec->write = twl4030_write; codec->set_bias_level = twl4030_set_bias_level; + codec->idle_bias_off = 1; codec->dai = twl4030_dai; codec->num_dai = ARRAY_SIZE(twl4030_dai); codec->reg_cache_size = sizeof(twl4030_reg); @@ -2302,9 +2340,7 @@ static int __devinit twl4030_codec_probe(struct platform_device *pdev) /* Set the defaults, and power up the codec */ twl4030->sysclk = twl4030_codec_get_mclk() / 1000; - twl4030_init_chip(codec); codec->bias_level = SND_SOC_BIAS_OFF; - twl4030_set_bias_level(codec, SND_SOC_BIAS_STANDBY); ret = snd_soc_register_codec(codec); if (ret != 0) { @@ -2322,7 +2358,7 @@ static int __devinit twl4030_codec_probe(struct platform_device *pdev) return 0; error_codec: - twl4030_power_down(codec); + twl4030_codec_enable(codec, 0); kfree(codec->reg_cache); error_cache: kfree(twl4030); diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h index f206d242ca3..6c57430f6e2 100644 --- a/sound/soc/codecs/twl4030.h +++ b/sound/soc/codecs/twl4030.h @@ -41,7 +41,11 @@ extern struct snd_soc_codec_device soc_codec_dev_twl4030; struct twl4030_setup_data { unsigned int ramp_delay_value; + unsigned int digimic_delay; /* in ms */ unsigned int sysclk; + unsigned int offset_cncl_path; + unsigned int check_defaults:1; + unsigned int reset_registers:1; unsigned int hs_extmute:1; void (*set_hs_extmute)(int mute); }; diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index af36346ff33..64a807f1a8a 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -360,6 +360,13 @@ static int headset_power_mode(struct snd_soc_codec *codec, int high_perf) return 0; } +static int twl6040_hs_dac_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + msleep(1); + return 0; +} + static int twl6040_power_mode_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -371,6 +378,8 @@ static int twl6040_power_mode_event(struct snd_soc_dapm_widget *w, else priv->non_lp--; + msleep(1); + return 0; } @@ -471,20 +480,6 @@ static const struct snd_kcontrol_new hfdacl_switch_controls = static const struct snd_kcontrol_new hfdacr_switch_controls = SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFRCTL, 2, 1, 0); -/* Headset driver switches */ -static const struct snd_kcontrol_new hsl_driver_switch_controls = - SOC_DAPM_SINGLE("Switch", TWL6040_REG_HSLCTL, 2, 1, 0); - -static const struct snd_kcontrol_new hsr_driver_switch_controls = - SOC_DAPM_SINGLE("Switch", TWL6040_REG_HSRCTL, 2, 1, 0); - -/* Handsfree driver switches */ -static const struct snd_kcontrol_new hfl_driver_switch_controls = - SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFLCTL, 4, 1, 0); - -static const struct snd_kcontrol_new hfr_driver_switch_controls = - SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFRCTL, 4, 1, 0); - static const struct snd_kcontrol_new ep_driver_switch_controls = SOC_DAPM_SINGLE("Switch", TWL6040_REG_EARCTL, 0, 1, 0); @@ -548,10 +543,14 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { TWL6040_REG_DMICBCTL, 4, 0), /* DACs */ - SND_SOC_DAPM_DAC("HSDAC Left", "Headset Playback", - TWL6040_REG_HSLCTL, 0, 0), - SND_SOC_DAPM_DAC("HSDAC Right", "Headset Playback", - TWL6040_REG_HSRCTL, 0, 0), + SND_SOC_DAPM_DAC_E("HSDAC Left", "Headset Playback", + TWL6040_REG_HSLCTL, 0, 0, + twl6040_hs_dac_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_DAC_E("HSDAC Right", "Headset Playback", + TWL6040_REG_HSRCTL, 0, 0, + twl6040_hs_dac_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_DAC_E("HFDAC Left", "Handsfree Playback", TWL6040_REG_HFLCTL, 0, 0, twl6040_power_mode_event, @@ -571,18 +570,19 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { SND_SOC_DAPM_SWITCH("HFDAC Right Playback", SND_SOC_NOPM, 0, 0, &hfdacr_switch_controls), - SND_SOC_DAPM_SWITCH("Headset Left Driver", - SND_SOC_NOPM, 0, 0, &hsl_driver_switch_controls), - SND_SOC_DAPM_SWITCH("Headset Right Driver", - SND_SOC_NOPM, 0, 0, &hsr_driver_switch_controls), - SND_SOC_DAPM_SWITCH_E("Handsfree Left Driver", - SND_SOC_NOPM, 0, 0, &hfl_driver_switch_controls, + /* Analog playback drivers */ + SND_SOC_DAPM_PGA_E("Handsfree Left Driver", + TWL6040_REG_HFLCTL, 4, 0, NULL, 0, twl6040_power_mode_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), - SND_SOC_DAPM_SWITCH_E("Handsfree Right Driver", - SND_SOC_NOPM, 0, 0, &hfr_driver_switch_controls, + SND_SOC_DAPM_PGA_E("Handsfree Right Driver", + TWL6040_REG_HFRCTL, 4, 0, NULL, 0, twl6040_power_mode_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_PGA("Headset Left Driver", + TWL6040_REG_HSLCTL, 2, 0, NULL, 0), + SND_SOC_DAPM_PGA("Headset Right Driver", + TWL6040_REG_HSRCTL, 2, 0, NULL, 0), SND_SOC_DAPM_SWITCH_E("Earphone Driver", SND_SOC_NOPM, 0, 0, &ep_driver_switch_controls, twl6040_power_mode_event, @@ -616,8 +616,8 @@ static const struct snd_soc_dapm_route intercon[] = { {"HSDAC Left Playback", "Switch", "HSDAC Left"}, {"HSDAC Right Playback", "Switch", "HSDAC Right"}, - {"Headset Left Driver", "Switch", "HSDAC Left Playback"}, - {"Headset Right Driver", "Switch", "HSDAC Right Playback"}, + {"Headset Left Driver", NULL, "HSDAC Left Playback"}, + {"Headset Right Driver", NULL, "HSDAC Right Playback"}, {"HSOL", NULL, "Headset Left Driver"}, {"HSOR", NULL, "Headset Right Driver"}, @@ -928,7 +928,7 @@ static int twl6040_set_dai_sysclk(struct snd_soc_dai *codec_dai, case 19200000: /* mclk input, pll disabled */ hppllctl |= TWL6040_MCLK_19200KHZ | - TWL6040_HPLLSQRBP | + TWL6040_HPLLSQRENA | TWL6040_HPLLBP; break; case 26000000: diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index 28aac53c97b..f3b4c1d6a82 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -28,19 +28,6 @@ #include "uda134x.h" -#define POWER_OFF_ON_STANDBY 1 -/* - ALSA SOC usually puts the device in standby mode when it's not used - for sometime. If you define POWER_OFF_ON_STANDBY the driver will - turn off the ADC/DAC when this callback is invoked and turn it back - on when needed. Unfortunately this will result in a very light bump - (it can be audible only with good earphones). If this bothers you - just comment this line, you will have slightly higher power - consumption . Please note that sending the L3 command for ADC is - enough to make the bump, so it doesn't make difference if you - completely take off power from the codec. - */ - #define UDA134X_RATES SNDRV_PCM_RATE_8000_48000 #define UDA134X_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S20_3LE) @@ -58,7 +45,7 @@ static const char uda134x_reg[UDA134X_REGS_NUM] = { /* Extended address registers */ 0x04, 0x04, 0x04, 0x00, 0x00, 0x00, 0x00, 0x00, /* Status, data regs */ - 0x00, 0x83, 0x00, 0x40, 0x80, 0x00, + 0x00, 0x83, 0x00, 0x40, 0x80, 0xC0, 0x00, }; /* @@ -117,6 +104,7 @@ static int uda134x_write(struct snd_soc_codec *codec, unsigned int reg, case UDA134X_DATA000: case UDA134X_DATA001: case UDA134X_DATA010: + case UDA134X_DATA011: addr = UDA134X_DATA0_ADDR; break; case UDA134X_DATA1: @@ -353,8 +341,22 @@ static int uda134x_set_bias_level(struct snd_soc_codec *codec, switch (level) { case SND_SOC_BIAS_ON: /* ADC, DAC on */ - reg = uda134x_read_reg_cache(codec, UDA134X_STATUS1); - uda134x_write(codec, UDA134X_STATUS1, reg | 0x03); + switch (pd->model) { + case UDA134X_UDA1340: + case UDA134X_UDA1344: + case UDA134X_UDA1345: + reg = uda134x_read_reg_cache(codec, UDA134X_DATA011); + uda134x_write(codec, UDA134X_DATA011, reg | 0x03); + break; + case UDA134X_UDA1341: + reg = uda134x_read_reg_cache(codec, UDA134X_STATUS1); + uda134x_write(codec, UDA134X_STATUS1, reg | 0x03); + break; + default: + printk(KERN_ERR "UDA134X SoC codec: " + "unsupported model %d\n", pd->model); + return -EINVAL; + } break; case SND_SOC_BIAS_PREPARE: /* power on */ @@ -367,8 +369,22 @@ static int uda134x_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: /* ADC, DAC power off */ - reg = uda134x_read_reg_cache(codec, UDA134X_STATUS1); - uda134x_write(codec, UDA134X_STATUS1, reg & ~(0x03)); + switch (pd->model) { + case UDA134X_UDA1340: + case UDA134X_UDA1344: + case UDA134X_UDA1345: + reg = uda134x_read_reg_cache(codec, UDA134X_DATA011); + uda134x_write(codec, UDA134X_DATA011, reg & ~(0x03)); + break; + case UDA134X_UDA1341: + reg = uda134x_read_reg_cache(codec, UDA134X_STATUS1); + uda134x_write(codec, UDA134X_STATUS1, reg & ~(0x03)); + break; + default: + printk(KERN_ERR "UDA134X SoC codec: " + "unsupported model %d\n", pd->model); + return -EINVAL; + } break; case SND_SOC_BIAS_OFF: /* power off */ @@ -531,9 +547,7 @@ static int uda134x_soc_probe(struct platform_device *pdev) codec->num_dai = 1; codec->read = uda134x_read_reg_cache; codec->write = uda134x_write; -#ifdef POWER_OFF_ON_STANDBY - codec->set_bias_level = uda134x_set_bias_level; -#endif + INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); @@ -544,6 +558,14 @@ static int uda134x_soc_probe(struct platform_device *pdev) uda134x_reset(codec); + if (pd->is_powered_on_standby) { + codec->set_bias_level = NULL; + uda134x_set_bias_level(codec, SND_SOC_BIAS_ON); + } else { + codec->set_bias_level = uda134x_set_bias_level; + uda134x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + } + /* register pcms */ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); if (ret < 0) { diff --git a/sound/soc/codecs/uda134x.h b/sound/soc/codecs/uda134x.h index 94f440490b3..205f03b3eaf 100644 --- a/sound/soc/codecs/uda134x.h +++ b/sound/soc/codecs/uda134x.h @@ -23,9 +23,10 @@ #define UDA134X_DATA000 10 #define UDA134X_DATA001 11 #define UDA134X_DATA010 12 -#define UDA134X_DATA1 13 +#define UDA134X_DATA011 13 +#define UDA134X_DATA1 14 -#define UDA134X_REGS_NUM 14 +#define UDA134X_REGS_NUM 15 #define STATUS0_DAIFMT_MASK (~(7<<1)) #define STATUS0_SYSCLK_MASK (~(3<<4)) diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index 002e289d125..4bcd168794e 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -795,6 +795,8 @@ static int __devinit wm2000_i2c_probe(struct i2c_client *i2c, dev_set_drvdata(&i2c->dev, wm2000); wm2000->anc_eng_ena = 1; + wm2000->anc_active = 1; + wm2000->spk_ena = 1; wm2000->i2c = i2c; wm2000_reset(wm2000); diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c index 37242a7d307..0ad039b4adf 100644 --- a/sound/soc/codecs/wm8523.c +++ b/sound/soc/codecs/wm8523.c @@ -482,7 +482,8 @@ static int wm8523_register(struct wm8523_priv *wm8523, if (wm8523_codec) { dev_err(codec->dev, "Another WM8523 is registered\n"); - return -EINVAL; + ret = -EINVAL; + goto err; } mutex_init(&codec->mutex); @@ -570,18 +571,19 @@ static int wm8523_register(struct wm8523_priv *wm8523, ret = snd_soc_register_codec(codec); if (ret != 0) { dev_err(codec->dev, "Failed to register codec: %d\n", ret); - return ret; + goto err_enable; } ret = snd_soc_register_dai(&wm8523_dai); if (ret != 0) { dev_err(codec->dev, "Failed to register DAI: %d\n", ret); - snd_soc_unregister_codec(codec); - return ret; + goto err_codec; } return 0; +err_codec: + snd_soc_unregister_codec(codec); err_enable: regulator_bulk_disable(ARRAY_SIZE(wm8523->supplies), wm8523->supplies); err_get: diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c index effb14eee7d..e2dba07f026 100644 --- a/sound/soc/codecs/wm8711.c +++ b/sound/soc/codecs/wm8711.c @@ -439,7 +439,8 @@ static int wm8711_register(struct wm8711_priv *wm8711, if (wm8711_codec) { dev_err(codec->dev, "Another WM8711 is registered\n"); - return -EINVAL; + ret = -EINVAL; + goto err; } mutex_init(&codec->mutex); diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c new file mode 100644 index 00000000000..b9ea8904ad4 --- /dev/null +++ b/sound/soc/codecs/wm8741.c @@ -0,0 +1,579 @@ +/* + * wm8741.c -- WM8741 ALSA SoC Audio driver + * + * Copyright 2010 Wolfson Microelectronics plc + * + * Author: Ian Lartey <ian@opensource.wolfsonmicro.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/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/regulator/consumer.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 "wm8741.h" + +static struct snd_soc_codec *wm8741_codec; +struct snd_soc_codec_device soc_codec_dev_wm8741; + +#define WM8741_NUM_SUPPLIES 2 +static const char *wm8741_supply_names[WM8741_NUM_SUPPLIES] = { + "AVDD", + "DVDD", +}; + +#define WM8741_NUM_RATES 4 + +/* codec private data */ +struct wm8741_priv { + struct snd_soc_codec codec; + u16 reg_cache[WM8741_REGISTER_COUNT]; + struct regulator_bulk_data supplies[WM8741_NUM_SUPPLIES]; + unsigned int sysclk; + unsigned int rate_constraint_list[WM8741_NUM_RATES]; + struct snd_pcm_hw_constraint_list rate_constraint; +}; + +static const u16 wm8741_reg_defaults[WM8741_REGISTER_COUNT] = { + 0x0000, /* R0 - DACLLSB Attenuation */ + 0x0000, /* R1 - DACLMSB Attenuation */ + 0x0000, /* R2 - DACRLSB Attenuation */ + 0x0000, /* R3 - DACRMSB Attenuation */ + 0x0000, /* R4 - Volume Control */ + 0x000A, /* R5 - Format Control */ + 0x0000, /* R6 - Filter Control */ + 0x0000, /* R7 - Mode Control 1 */ + 0x0002, /* R8 - Mode Control 2 */ + 0x0000, /* R9 - Reset */ + 0x0002, /* R32 - ADDITONAL_CONTROL_1 */ +}; + + +static int wm8741_reset(struct snd_soc_codec *codec) +{ + return snd_soc_write(codec, WM8741_RESET, 0); +} + +static const DECLARE_TLV_DB_SCALE(dac_tlv_fine, -12700, 13, 0); +static const DECLARE_TLV_DB_SCALE(dac_tlv, -12700, 400, 0); + +static const struct snd_kcontrol_new wm8741_snd_controls[] = { +SOC_DOUBLE_R_TLV("Fine Playback Volume", WM8741_DACLLSB_ATTENUATION, + WM8741_DACRLSB_ATTENUATION, 1, 255, 1, dac_tlv_fine), +SOC_DOUBLE_R_TLV("Playback Volume", WM8741_DACLMSB_ATTENUATION, + WM8741_DACRMSB_ATTENUATION, 0, 511, 1, dac_tlv), +}; + +static const struct snd_soc_dapm_widget wm8741_dapm_widgets[] = { +SND_SOC_DAPM_DAC("DACL", "Playback", SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_DAC("DACR", "Playback", SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_OUTPUT("VOUTLP"), +SND_SOC_DAPM_OUTPUT("VOUTLN"), +SND_SOC_DAPM_OUTPUT("VOUTRP"), +SND_SOC_DAPM_OUTPUT("VOUTRN"), +}; + +static const struct snd_soc_dapm_route intercon[] = { + { "VOUTLP", NULL, "DACL" }, + { "VOUTLN", NULL, "DACL" }, + { "VOUTRP", NULL, "DACR" }, + { "VOUTRN", NULL, "DACR" }, +}; + +static int wm8741_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, wm8741_dapm_widgets, + ARRAY_SIZE(wm8741_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + + return 0; +} + +static struct { + int value; + int ratio; +} lrclk_ratios[WM8741_NUM_RATES] = { + { 1, 256 }, + { 2, 384 }, + { 3, 512 }, + { 4, 768 }, +}; + + +static int wm8741_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm8741_priv *wm8741 = snd_soc_codec_get_drvdata(codec); + + /* The set of sample rates that can be supported depends on the + * MCLK supplied to the CODEC - enforce this. + */ + if (!wm8741->sysclk) { + dev_err(codec->dev, + "No MCLK configured, call set_sysclk() on init\n"); + return -EINVAL; + } + + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &wm8741->rate_constraint); + + return 0; +} + +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_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + struct wm8741_priv *wm8741 = snd_soc_codec_get_drvdata(codec); + u16 iface = snd_soc_read(codec, WM8741_FORMAT_CONTROL) & 0x1FC; + int i; + + /* Find a supported LRCLK ratio */ + for (i = 0; i < ARRAY_SIZE(lrclk_ratios); i++) { + if (wm8741->sysclk / params_rate(params) == + lrclk_ratios[i].ratio) + break; + } + + /* Should never happen, should be handled by constraints */ + if (i == ARRAY_SIZE(lrclk_ratios)) { + dev_err(codec->dev, "MCLK/fs ratio %d unsupported\n", + wm8741->sysclk / params_rate(params)); + return -EINVAL; + } + + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + iface |= 0x0001; + break; + case SNDRV_PCM_FORMAT_S24_LE: + iface |= 0x0002; + break; + case SNDRV_PCM_FORMAT_S32_LE: + iface |= 0x0003; + break; + default: + dev_dbg(codec->dev, "wm8741_hw_params: Unsupported bit size param = %d", + params_format(params)); + return -EINVAL; + } + + dev_dbg(codec->dev, "wm8741_hw_params: bit size param = %d", + params_format(params)); + + snd_soc_write(codec, WM8741_FORMAT_CONTROL, iface); + return 0; +} + +static int wm8741_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 wm8741_priv *wm8741 = snd_soc_codec_get_drvdata(codec); + unsigned int val; + int i; + + dev_dbg(codec->dev, "wm8741_set_dai_sysclk info: freq=%dHz\n", freq); + + wm8741->sysclk = freq; + + wm8741->rate_constraint.count = 0; + + for (i = 0; i < ARRAY_SIZE(lrclk_ratios); i++) { + dev_dbg(codec->dev, "index = %d, ratio = %d, freq = %d", + i, lrclk_ratios[i].ratio, freq); + + val = freq / lrclk_ratios[i].ratio; + /* Check that it's a standard rate since core can't + * cope with others and having the odd rates confuses + * constraint matching. + */ + switch (val) { + case 32000: + case 44100: + case 48000: + case 64000: + case 88200: + case 96000: + dev_dbg(codec->dev, "Supported sample rate: %dHz\n", + val); + wm8741->rate_constraint_list[i] = val; + wm8741->rate_constraint.count++; + break; + default: + dev_dbg(codec->dev, "Skipping sample rate: %dHz\n", + val); + } + } + + /* Need at least one supported rate... */ + if (wm8741->rate_constraint.count == 0) + return -EINVAL; + + return 0; +} + +static int wm8741_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 iface = snd_soc_read(codec, WM8741_FORMAT_CONTROL) & 0x1C3; + + /* check master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface |= 0x0008; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + iface |= 0x0004; + break; + case SND_SOC_DAIFMT_DSP_A: + iface |= 0x0003; + break; + case SND_SOC_DAIFMT_DSP_B: + iface |= 0x0013; + 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 |= 0x0010; + break; + case SND_SOC_DAIFMT_IB_NF: + iface |= 0x0020; + break; + case SND_SOC_DAIFMT_NB_IF: + iface |= 0x0030; + break; + default: + return -EINVAL; + } + + + dev_dbg(codec->dev, "wm8741_set_dai_fmt: Format=%x, Clock Inv=%x\n", + fmt & SND_SOC_DAIFMT_FORMAT_MASK, + ((fmt & SND_SOC_DAIFMT_INV_MASK))); + + snd_soc_write(codec, WM8741_FORMAT_CONTROL, iface); + return 0; +} + +#define WM8741_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) + +#define WM8741_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 wm8741_dai_ops = { + .startup = wm8741_startup, + .hw_params = wm8741_hw_params, + .set_sysclk = wm8741_set_dai_sysclk, + .set_fmt = wm8741_set_dai_fmt, +}; + +struct snd_soc_dai wm8741_dai = { + .name = "WM8741", + .playback = { + .stream_name = "Playback", + .channels_min = 2, /* Mono modes not yet supported */ + .channels_max = 2, + .rates = WM8741_RATES, + .formats = WM8741_FORMATS, + }, + .ops = &wm8741_dai_ops, +}; +EXPORT_SYMBOL_GPL(wm8741_dai); + +#ifdef CONFIG_PM +static int wm8741_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + u16 *cache = codec->reg_cache; + int i; + + /* RESTORE REG Cache */ + for (i = 0; i < WM8741_REGISTER_COUNT; i++) { + if (cache[i] == wm8741_reg_defaults[i] || WM8741_RESET == i) + continue; + snd_soc_write(codec, i, cache[i]); + } + return 0; +} +#else +#define wm8741_suspend NULL +#define wm8741_resume NULL +#endif + +static int wm8741_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + if (wm8741_codec == NULL) { + dev_err(&pdev->dev, "Codec device not registered\n"); + return -ENODEV; + } + + socdev->card->codec = wm8741_codec; + codec = wm8741_codec; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(codec->dev, "failed to create pcms: %d\n", ret); + goto pcm_err; + } + + snd_soc_add_controls(codec, wm8741_snd_controls, + ARRAY_SIZE(wm8741_snd_controls)); + wm8741_add_widgets(codec); + + return ret; + +pcm_err: + return ret; +} + +static int wm8741_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_wm8741 = { + .probe = wm8741_probe, + .remove = wm8741_remove, + .resume = wm8741_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm8741); + +static int wm8741_register(struct wm8741_priv *wm8741, + enum snd_soc_control_type control) +{ + int ret; + struct snd_soc_codec *codec = &wm8741->codec; + int i; + + if (wm8741_codec) { + dev_err(codec->dev, "Another WM8741 is registered\n"); + return -EINVAL; + } + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + snd_soc_codec_set_drvdata(codec, wm8741); + codec->name = "WM8741"; + codec->owner = THIS_MODULE; + codec->bias_level = SND_SOC_BIAS_OFF; + codec->set_bias_level = NULL; + codec->dai = &wm8741_dai; + codec->num_dai = 1; + codec->reg_cache_size = WM8741_REGISTER_COUNT; + codec->reg_cache = &wm8741->reg_cache; + + wm8741->rate_constraint.list = &wm8741->rate_constraint_list[0]; + wm8741->rate_constraint.count = + ARRAY_SIZE(wm8741->rate_constraint_list); + + memcpy(codec->reg_cache, wm8741_reg_defaults, + sizeof(wm8741->reg_cache)); + + ret = snd_soc_codec_set_cache_io(codec, 7, 9, control); + if (ret != 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + goto err; + } + + for (i = 0; i < ARRAY_SIZE(wm8741->supplies); i++) + wm8741->supplies[i].supply = wm8741_supply_names[i]; + + ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(wm8741->supplies), + wm8741->supplies); + if (ret != 0) { + dev_err(codec->dev, "Failed to request supplies: %d\n", ret); + goto err; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(wm8741->supplies), + wm8741->supplies); + if (ret != 0) { + dev_err(codec->dev, "Failed to enable supplies: %d\n", ret); + goto err_get; + } + + ret = wm8741_reset(codec); + if (ret < 0) { + dev_err(codec->dev, "Failed to issue reset\n"); + goto err_enable; + } + + wm8741_dai.dev = codec->dev; + + /* Change some default settings - latch VU */ + wm8741->reg_cache[WM8741_DACLLSB_ATTENUATION] |= WM8741_UPDATELL; + wm8741->reg_cache[WM8741_DACLMSB_ATTENUATION] |= WM8741_UPDATELM; + wm8741->reg_cache[WM8741_DACRLSB_ATTENUATION] |= WM8741_UPDATERL; + wm8741->reg_cache[WM8741_DACRLSB_ATTENUATION] |= WM8741_UPDATERM; + + wm8741_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + return ret; + } + + ret = snd_soc_register_dai(&wm8741_dai); + if (ret != 0) { + dev_err(codec->dev, "Failed to register DAI: %d\n", ret); + snd_soc_unregister_codec(codec); + return ret; + } + + dev_dbg(codec->dev, "Successful registration\n"); + return 0; + +err_enable: + regulator_bulk_disable(ARRAY_SIZE(wm8741->supplies), wm8741->supplies); + +err_get: + regulator_bulk_free(ARRAY_SIZE(wm8741->supplies), wm8741->supplies); + +err: + kfree(wm8741); + return ret; +} + +static void wm8741_unregister(struct wm8741_priv *wm8741) +{ + regulator_bulk_free(ARRAY_SIZE(wm8741->supplies), wm8741->supplies); + + snd_soc_unregister_dai(&wm8741_dai); + snd_soc_unregister_codec(&wm8741->codec); + kfree(wm8741); + wm8741_codec = NULL; +} + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +static __devinit int wm8741_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct wm8741_priv *wm8741; + struct snd_soc_codec *codec; + + wm8741 = kzalloc(sizeof(struct wm8741_priv), GFP_KERNEL); + if (wm8741 == NULL) + return -ENOMEM; + + codec = &wm8741->codec; + codec->hw_write = (hw_write_t)i2c_master_send; + + i2c_set_clientdata(i2c, wm8741); + codec->control_data = i2c; + + codec->dev = &i2c->dev; + + return wm8741_register(wm8741, SND_SOC_I2C); +} + +static __devexit int wm8741_i2c_remove(struct i2c_client *client) +{ + struct wm8741_priv *wm8741 = i2c_get_clientdata(client); + wm8741_unregister(wm8741); + return 0; +} + +static const struct i2c_device_id wm8741_i2c_id[] = { + { "wm8741", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, wm8741_i2c_id); + + +static struct i2c_driver wm8741_i2c_driver = { + .driver = { + .name = "WM8741", + .owner = THIS_MODULE, + }, + .probe = wm8741_i2c_probe, + .remove = __devexit_p(wm8741_i2c_remove), + .id_table = wm8741_i2c_id, +}; +#endif + +static int __init wm8741_modinit(void) +{ + int ret; +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + ret = i2c_add_driver(&wm8741_i2c_driver); + if (ret != 0) { + printk(KERN_ERR "Failed to register WM8741 I2C driver: %d\n", + ret); + } +#endif + return 0; +} +module_init(wm8741_modinit); + +static void __exit wm8741_exit(void) +{ +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&wm8741_i2c_driver); +#endif +} +module_exit(wm8741_exit); + +MODULE_DESCRIPTION("ASoC WM8741 driver"); +MODULE_AUTHOR("Ian Lartey <ian@opensource.wolfsonmicro.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8741.h b/sound/soc/codecs/wm8741.h new file mode 100644 index 00000000000..fdef6ecd1f6 --- /dev/null +++ b/sound/soc/codecs/wm8741.h @@ -0,0 +1,214 @@ +/* + * wm8741.h -- WM8423 ASoC driver + * + * Copyright 2010 Wolfson Microelectronics, plc + * + * Author: Ian Lartey <ian@opensource.wolfsonmicro.com> + * + * Based on wm8753.h + * + * 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 _WM8741_H +#define _WM8741_H + +/* + * Register values. + */ +#define WM8741_DACLLSB_ATTENUATION 0x00 +#define WM8741_DACLMSB_ATTENUATION 0x01 +#define WM8741_DACRLSB_ATTENUATION 0x02 +#define WM8741_DACRMSB_ATTENUATION 0x03 +#define WM8741_VOLUME_CONTROL 0x04 +#define WM8741_FORMAT_CONTROL 0x05 +#define WM8741_FILTER_CONTROL 0x06 +#define WM8741_MODE_CONTROL_1 0x07 +#define WM8741_MODE_CONTROL_2 0x08 +#define WM8741_RESET 0x09 +#define WM8741_ADDITIONAL_CONTROL_1 0x20 + +#define WM8741_REGISTER_COUNT 11 +#define WM8741_MAX_REGISTER 0x20 + +/* + * Field Definitions. + */ + +/* + * R0 (0x00) - DACLLSB_ATTENUATION + */ +#define WM8741_UPDATELL 0x0020 /* UPDATELL */ +#define WM8741_UPDATELL_MASK 0x0020 /* UPDATELL */ +#define WM8741_UPDATELL_SHIFT 5 /* UPDATELL */ +#define WM8741_UPDATELL_WIDTH 1 /* UPDATELL */ +#define WM8741_LAT_4_0_MASK 0x001F /* LAT[4:0] - [4:0] */ +#define WM8741_LAT_4_0_SHIFT 0 /* LAT[4:0] - [4:0] */ +#define WM8741_LAT_4_0_WIDTH 5 /* LAT[4:0] - [4:0] */ + +/* + * R1 (0x01) - DACLMSB_ATTENUATION + */ +#define WM8741_UPDATELM 0x0020 /* UPDATELM */ +#define WM8741_UPDATELM_MASK 0x0020 /* UPDATELM */ +#define WM8741_UPDATELM_SHIFT 5 /* UPDATELM */ +#define WM8741_UPDATELM_WIDTH 1 /* UPDATELM */ +#define WM8741_LAT_9_5_0_MASK 0x001F /* LAT[9:5] - [4:0] */ +#define WM8741_LAT_9_5_0_SHIFT 0 /* LAT[9:5] - [4:0] */ +#define WM8741_LAT_9_5_0_WIDTH 5 /* LAT[9:5] - [4:0] */ + +/* + * R2 (0x02) - DACRLSB_ATTENUATION + */ +#define WM8741_UPDATERL 0x0020 /* UPDATERL */ +#define WM8741_UPDATERL_MASK 0x0020 /* UPDATERL */ +#define WM8741_UPDATERL_SHIFT 5 /* UPDATERL */ +#define WM8741_UPDATERL_WIDTH 1 /* UPDATERL */ +#define WM8741_RAT_4_0_MASK 0x001F /* RAT[4:0] - [4:0] */ +#define WM8741_RAT_4_0_SHIFT 0 /* RAT[4:0] - [4:0] */ +#define WM8741_RAT_4_0_WIDTH 5 /* RAT[4:0] - [4:0] */ + +/* + * R3 (0x03) - DACRMSB_ATTENUATION + */ +#define WM8741_UPDATERM 0x0020 /* UPDATERM */ +#define WM8741_UPDATERM_MASK 0x0020 /* UPDATERM */ +#define WM8741_UPDATERM_SHIFT 5 /* UPDATERM */ +#define WM8741_UPDATERM_WIDTH 1 /* UPDATERM */ +#define WM8741_RAT_9_5_0_MASK 0x001F /* RAT[9:5] - [4:0] */ +#define WM8741_RAT_9_5_0_SHIFT 0 /* RAT[9:5] - [4:0] */ +#define WM8741_RAT_9_5_0_WIDTH 5 /* RAT[9:5] - [4:0] */ + +/* + * R4 (0x04) - VOLUME_CONTROL + */ +#define WM8741_AMUTE 0x0080 /* AMUTE */ +#define WM8741_AMUTE_MASK 0x0080 /* AMUTE */ +#define WM8741_AMUTE_SHIFT 7 /* AMUTE */ +#define WM8741_AMUTE_WIDTH 1 /* AMUTE */ +#define WM8741_ZFLAG_MASK 0x0060 /* ZFLAG - [6:5] */ +#define WM8741_ZFLAG_SHIFT 5 /* ZFLAG - [6:5] */ +#define WM8741_ZFLAG_WIDTH 2 /* ZFLAG - [6:5] */ +#define WM8741_IZD 0x0010 /* IZD */ +#define WM8741_IZD_MASK 0x0010 /* IZD */ +#define WM8741_IZD_SHIFT 4 /* IZD */ +#define WM8741_IZD_WIDTH 1 /* IZD */ +#define WM8741_SOFT 0x0008 /* SOFT MUTE */ +#define WM8741_SOFT_MASK 0x0008 /* SOFT MUTE */ +#define WM8741_SOFT_SHIFT 3 /* SOFT MUTE */ +#define WM8741_SOFT_WIDTH 1 /* SOFT MUTE */ +#define WM8741_ATC 0x0004 /* ATC */ +#define WM8741_ATC_MASK 0x0004 /* ATC */ +#define WM8741_ATC_SHIFT 2 /* ATC */ +#define WM8741_ATC_WIDTH 1 /* ATC */ +#define WM8741_ATT2DB 0x0002 /* ATT2DB */ +#define WM8741_ATT2DB_MASK 0x0002 /* ATT2DB */ +#define WM8741_ATT2DB_SHIFT 1 /* ATT2DB */ +#define WM8741_ATT2DB_WIDTH 1 /* ATT2DB */ +#define WM8741_VOL_RAMP 0x0001 /* VOL_RAMP */ +#define WM8741_VOL_RAMP_MASK 0x0001 /* VOL_RAMP */ +#define WM8741_VOL_RAMP_SHIFT 0 /* VOL_RAMP */ +#define WM8741_VOL_RAMP_WIDTH 1 /* VOL_RAMP */ + +/* + * R5 (0x05) - FORMAT_CONTROL + */ +#define WM8741_PWDN 0x0080 /* PWDN */ +#define WM8741_PWDN_MASK 0x0080 /* PWDN */ +#define WM8741_PWDN_SHIFT 7 /* PWDN */ +#define WM8741_PWDN_WIDTH 1 /* PWDN */ +#define WM8741_REV 0x0040 /* REV */ +#define WM8741_REV_MASK 0x0040 /* REV */ +#define WM8741_REV_SHIFT 6 /* REV */ +#define WM8741_REV_WIDTH 1 /* REV */ +#define WM8741_BCP 0x0020 /* BCP */ +#define WM8741_BCP_MASK 0x0020 /* BCP */ +#define WM8741_BCP_SHIFT 5 /* BCP */ +#define WM8741_BCP_WIDTH 1 /* BCP */ +#define WM8741_LRP 0x0010 /* LRP */ +#define WM8741_LRP_MASK 0x0010 /* LRP */ +#define WM8741_LRP_SHIFT 4 /* LRP */ +#define WM8741_LRP_WIDTH 1 /* LRP */ +#define WM8741_FMT_MASK 0x000C /* FMT - [3:2] */ +#define WM8741_FMT_SHIFT 2 /* FMT - [3:2] */ +#define WM8741_FMT_WIDTH 2 /* FMT - [3:2] */ +#define WM8741_IWL_MASK 0x0003 /* IWL - [1:0] */ +#define WM8741_IWL_SHIFT 0 /* IWL - [1:0] */ +#define WM8741_IWL_WIDTH 2 /* IWL - [1:0] */ + +/* + * R6 (0x06) - FILTER_CONTROL + */ +#define WM8741_ZFLAG_HI 0x0080 /* ZFLAG_HI */ +#define WM8741_ZFLAG_HI_MASK 0x0080 /* ZFLAG_HI */ +#define WM8741_ZFLAG_HI_SHIFT 7 /* ZFLAG_HI */ +#define WM8741_ZFLAG_HI_WIDTH 1 /* ZFLAG_HI */ +#define WM8741_DEEMPH_MASK 0x0060 /* DEEMPH - [6:5] */ +#define WM8741_DEEMPH_SHIFT 5 /* DEEMPH - [6:5] */ +#define WM8741_DEEMPH_WIDTH 2 /* DEEMPH - [6:5] */ +#define WM8741_DSDFILT_MASK 0x0018 /* DSDFILT - [4:3] */ +#define WM8741_DSDFILT_SHIFT 3 /* DSDFILT - [4:3] */ +#define WM8741_DSDFILT_WIDTH 2 /* DSDFILT - [4:3] */ +#define WM8741_FIRSEL_MASK 0x0007 /* FIRSEL - [2:0] */ +#define WM8741_FIRSEL_SHIFT 0 /* FIRSEL - [2:0] */ +#define WM8741_FIRSEL_WIDTH 3 /* FIRSEL - [2:0] */ + +/* + * R7 (0x07) - MODE_CONTROL_1 + */ +#define WM8741_MODE8X 0x0080 /* MODE8X */ +#define WM8741_MODE8X_MASK 0x0080 /* MODE8X */ +#define WM8741_MODE8X_SHIFT 7 /* MODE8X */ +#define WM8741_MODE8X_WIDTH 1 /* MODE8X */ +#define WM8741_OSR_MASK 0x0060 /* OSR - [6:5] */ +#define WM8741_OSR_SHIFT 5 /* OSR - [6:5] */ +#define WM8741_OSR_WIDTH 2 /* OSR - [6:5] */ +#define WM8741_SR_MASK 0x001C /* SR - [4:2] */ +#define WM8741_SR_SHIFT 2 /* SR - [4:2] */ +#define WM8741_SR_WIDTH 3 /* SR - [4:2] */ +#define WM8741_MODESEL_MASK 0x0003 /* MODESEL - [1:0] */ +#define WM8741_MODESEL_SHIFT 0 /* MODESEL - [1:0] */ +#define WM8741_MODESEL_WIDTH 2 /* MODESEL - [1:0] */ + +/* + * R8 (0x08) - MODE_CONTROL_2 + */ +#define WM8741_DSD_GAIN 0x0040 /* DSD_GAIN */ +#define WM8741_DSD_GAIN_MASK 0x0040 /* DSD_GAIN */ +#define WM8741_DSD_GAIN_SHIFT 6 /* DSD_GAIN */ +#define WM8741_DSD_GAIN_WIDTH 1 /* DSD_GAIN */ +#define WM8741_SDOUT 0x0020 /* SDOUT */ +#define WM8741_SDOUT_MASK 0x0020 /* SDOUT */ +#define WM8741_SDOUT_SHIFT 5 /* SDOUT */ +#define WM8741_SDOUT_WIDTH 1 /* SDOUT */ +#define WM8741_DOUT 0x0010 /* DOUT */ +#define WM8741_DOUT_MASK 0x0010 /* DOUT */ +#define WM8741_DOUT_SHIFT 4 /* DOUT */ +#define WM8741_DOUT_WIDTH 1 /* DOUT */ +#define WM8741_DIFF_MASK 0x000C /* DIFF - [3:2] */ +#define WM8741_DIFF_SHIFT 2 /* DIFF - [3:2] */ +#define WM8741_DIFF_WIDTH 2 /* DIFF - [3:2] */ +#define WM8741_DITHER_MASK 0x0003 /* DITHER - [1:0] */ +#define WM8741_DITHER_SHIFT 0 /* DITHER - [1:0] */ +#define WM8741_DITHER_WIDTH 2 /* DITHER - [1:0] */ + +/* + * R32 (0x20) - ADDITONAL_CONTROL_1 + */ +#define WM8741_DSD_LEVEL 0x0002 /* DSD_LEVEL */ +#define WM8741_DSD_LEVEL_MASK 0x0002 /* DSD_LEVEL */ +#define WM8741_DSD_LEVEL_SHIFT 1 /* DSD_LEVEL */ +#define WM8741_DSD_LEVEL_WIDTH 1 /* DSD_LEVEL */ +#define WM8741_DSD_NO_NOTCH 0x0001 /* DSD_NO_NOTCH */ +#define WM8741_DSD_NO_NOTCH_MASK 0x0001 /* DSD_NO_NOTCH */ +#define WM8741_DSD_NO_NOTCH_SHIFT 0 /* DSD_NO_NOTCH */ +#define WM8741_DSD_NO_NOTCH_WIDTH 1 /* DSD_NO_NOTCH */ + +#define WM8741_SYSCLK 0 + +extern struct snd_soc_dai wm8741_dai; +extern struct snd_soc_codec_device soc_codec_dev_wm8741; + +#endif diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 9407e193fcc..e2c05e3e323 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -884,6 +884,7 @@ static int wm8750_i2c_remove(struct i2c_client *client) static const struct i2c_device_id wm8750_i2c_id[] = { { "wm8750", 0 }, + { "wm8987", 0 }, /* WM8987 is register compatible with WM8750 */ { } }; MODULE_DEVICE_TABLE(i2c, wm8750_i2c_id); @@ -925,14 +926,22 @@ static int __devexit wm8750_spi_remove(struct spi_device *spi) return 0; } +static const struct spi_device_id wm8750_spi_id[] = { + { "wm8750", 0 }, + { "wm8987", 0 }, + { } +}; +MODULE_DEVICE_TABLE(spi, wm8750_spi_id); + static struct spi_driver wm8750_spi_driver = { .driver = { - .name = "wm8750", + .name = "WM8750 SPI Codec", .bus = &spi_bus_type, .owner = THIS_MODULE, }, .probe = wm8750_spi_probe, .remove = __devexit_p(wm8750_spi_remove), + .id_table = wm8750_spi_id, }; #endif diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 87f14f8675f..f7dcabf6283 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -2433,7 +2433,8 @@ static int wm8904_register(struct wm8904_priv *wm8904, if (wm8904_codec) { dev_err(codec->dev, "Another WM8904 is registered\n"); - return -EINVAL; + ret = -EINVAL; + goto err; } mutex_init(&codec->mutex); @@ -2462,7 +2463,8 @@ static int wm8904_register(struct wm8904_priv *wm8904, default: dev_err(codec->dev, "Unknown device type %d\n", wm8904->devtype); - return -EINVAL; + ret = -EINVAL; + goto err; } memcpy(codec->reg_cache, wm8904_reg, sizeof(wm8904_reg)); @@ -2566,18 +2568,19 @@ static int wm8904_register(struct wm8904_priv *wm8904, ret = snd_soc_register_codec(codec); if (ret != 0) { dev_err(codec->dev, "Failed to register codec: %d\n", ret); - return ret; + goto err_enable; } ret = snd_soc_register_dai(&wm8904_dai); if (ret != 0) { dev_err(codec->dev, "Failed to register DAI: %d\n", ret); - snd_soc_unregister_codec(codec); - return ret; + goto err_codec; } return 0; +err_codec: + snd_soc_unregister_codec(codec); err_enable: regulator_bulk_disable(ARRAY_SIZE(wm8904->supplies), wm8904->supplies); err_get: diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index e3c4bbfaae2..f0c11138e61 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -845,6 +845,7 @@ static void wm8940_unregister(struct wm8940_priv *wm8940) static int wm8940_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { + int ret; struct wm8940_priv *wm8940; struct snd_soc_codec *codec; @@ -858,7 +859,11 @@ static int wm8940_i2c_probe(struct i2c_client *i2c, codec->control_data = i2c; codec->dev = &i2c->dev; - return wm8940_register(wm8940, SND_SOC_I2C); + ret = wm8940_register(wm8940, SND_SOC_I2C); + if (ret < 0) + kfree(wm8940); + + return ret; } static int __devexit wm8940_i2c_remove(struct i2c_client *client) diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c index fedb76452f1..5f025593d84 100644 --- a/sound/soc/codecs/wm8955.c +++ b/sound/soc/codecs/wm8955.c @@ -964,7 +964,8 @@ static int wm8955_register(struct wm8955_priv *wm8955, if (wm8955_codec) { dev_err(codec->dev, "Another WM8955 is registered\n"); - return -EINVAL; + ret = -EINVAL; + goto err; } mutex_init(&codec->mutex); @@ -1047,18 +1048,19 @@ static int wm8955_register(struct wm8955_priv *wm8955, ret = snd_soc_register_codec(codec); if (ret != 0) { dev_err(codec->dev, "Failed to register codec: %d\n", ret); - return ret; + goto err_enable; } ret = snd_soc_register_dai(&wm8955_dai); if (ret != 0) { dev_err(codec->dev, "Failed to register DAI: %d\n", ret); - snd_soc_unregister_codec(codec); - return ret; + goto err_codec; } return 0; +err_codec: + snd_soc_unregister_codec(codec); err_enable: regulator_bulk_disable(ARRAY_SIZE(wm8955->supplies), wm8955->supplies); err_get: diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 7233cc68435..3c6ee61f6c9 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -79,12 +79,13 @@ struct wm8960_priv { struct snd_soc_dapm_widget *lout1; struct snd_soc_dapm_widget *rout1; struct snd_soc_dapm_widget *out3; + bool deemph; + int playback_fs; }; #define wm8960_reset(c) snd_soc_write(c, WM8960_RESET, 0) /* enumerated controls */ -static const char *wm8960_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"}; static const char *wm8960_polarity[] = {"No Inversion", "Left Inverted", "Right Inverted", "Stereo Inversion"}; static const char *wm8960_3d_upper_cutoff[] = {"High", "Low"}; @@ -93,7 +94,6 @@ static const char *wm8960_alcfunc[] = {"Off", "Right", "Left", "Stereo"}; static const char *wm8960_alcmode[] = {"ALC", "Limiter"}; static const struct soc_enum wm8960_enum[] = { - SOC_ENUM_SINGLE(WM8960_DACCTL1, 1, 4, wm8960_deemph), SOC_ENUM_SINGLE(WM8960_DACCTL1, 5, 4, wm8960_polarity), SOC_ENUM_SINGLE(WM8960_DACCTL2, 5, 4, wm8960_polarity), SOC_ENUM_SINGLE(WM8960_3D, 6, 2, wm8960_3d_upper_cutoff), @@ -102,6 +102,59 @@ static const struct soc_enum wm8960_enum[] = { SOC_ENUM_SINGLE(WM8960_ALC3, 8, 2, wm8960_alcmode), }; +static const int deemph_settings[] = { 0, 32000, 44100, 48000 }; + +static int wm8960_set_deemph(struct snd_soc_codec *codec) +{ + struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + int val, i, best; + + /* If we're using deemphasis select the nearest available sample + * rate. + */ + if (wm8960->deemph) { + best = 1; + for (i = 2; i < ARRAY_SIZE(deemph_settings); i++) { + if (abs(deemph_settings[i] - wm8960->playback_fs) < + abs(deemph_settings[best] - wm8960->playback_fs)) + best = i; + } + + val = best << 1; + } else { + val = 0; + } + + dev_dbg(codec->dev, "Set deemphasis %d\n", val); + + return snd_soc_update_bits(codec, WM8960_DACCTL1, + 0x6, val); +} + +static int wm8960_get_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + + return wm8960->deemph; +} + +static int wm8960_put_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + int deemph = ucontrol->value.enumerated.item[0]; + + if (deemph > 1) + return -EINVAL; + + wm8960->deemph = deemph; + + return wm8960_set_deemph(codec); +} + static const DECLARE_TLV_DB_SCALE(adc_tlv, -9700, 50, 0); static const DECLARE_TLV_DB_SCALE(dac_tlv, -12700, 50, 1); static const DECLARE_TLV_DB_SCALE(bypass_tlv, -2100, 300, 0); @@ -131,23 +184,24 @@ SOC_SINGLE("Speaker DC Volume", WM8960_CLASSD3, 3, 5, 0), SOC_SINGLE("Speaker AC Volume", WM8960_CLASSD3, 0, 5, 0), SOC_SINGLE("PCM Playback -6dB Switch", WM8960_DACCTL1, 7, 1, 0), -SOC_ENUM("ADC Polarity", wm8960_enum[1]), -SOC_ENUM("Playback De-emphasis", wm8960_enum[0]), +SOC_ENUM("ADC Polarity", wm8960_enum[0]), SOC_SINGLE("ADC High Pass Filter Switch", WM8960_DACCTL1, 0, 1, 0), SOC_ENUM("DAC Polarity", wm8960_enum[2]), +SOC_SINGLE_BOOL_EXT("DAC Deemphasis Switch", 0, + wm8960_get_deemph, wm8960_put_deemph), -SOC_ENUM("3D Filter Upper Cut-Off", wm8960_enum[3]), -SOC_ENUM("3D Filter Lower Cut-Off", wm8960_enum[4]), +SOC_ENUM("3D Filter Upper Cut-Off", wm8960_enum[2]), +SOC_ENUM("3D Filter Lower Cut-Off", wm8960_enum[3]), SOC_SINGLE("3D Volume", WM8960_3D, 1, 15, 0), SOC_SINGLE("3D Switch", WM8960_3D, 0, 1, 0), -SOC_ENUM("ALC Function", wm8960_enum[5]), +SOC_ENUM("ALC Function", wm8960_enum[4]), SOC_SINGLE("ALC Max Gain", WM8960_ALC1, 4, 7, 0), SOC_SINGLE("ALC Target", WM8960_ALC1, 0, 15, 1), SOC_SINGLE("ALC Min Gain", WM8960_ALC2, 4, 7, 0), SOC_SINGLE("ALC Hold Time", WM8960_ALC2, 0, 15, 0), -SOC_ENUM("ALC Mode", wm8960_enum[6]), +SOC_ENUM("ALC Mode", wm8960_enum[5]), SOC_SINGLE("ALC Decay", WM8960_ALC3, 4, 15, 0), SOC_SINGLE("ALC Attack", WM8960_ALC3, 0, 15, 0), @@ -433,6 +487,21 @@ static int wm8960_set_dai_fmt(struct snd_soc_dai *codec_dai, return 0; } +static struct { + int rate; + unsigned int val; +} alc_rates[] = { + { 48000, 0 }, + { 44100, 0 }, + { 32000, 1 }, + { 22050, 2 }, + { 24000, 2 }, + { 16000, 3 }, + { 11250, 4 }, + { 12000, 4 }, + { 8000, 5 }, +}; + static int wm8960_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -440,7 +509,9 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->card->codec; + struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); u16 iface = snd_soc_read(codec, WM8960_IFACE1) & 0xfff3; + int i; /* bit size */ switch (params_format(params)) { @@ -454,6 +525,18 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream, break; } + /* Update filters for the new rate */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + wm8960->playback_fs = params_rate(params); + wm8960_set_deemph(codec); + } else { + for (i = 0; i < ARRAY_SIZE(alc_rates); i++) + if (alc_rates[i].rate == params_rate(params)) + snd_soc_update_bits(codec, + WM8960_ADDCTL3, 0x7, + alc_rates[i].val); + } + /* set iface */ snd_soc_write(codec, WM8960_IFACE1, iface); return 0; diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index 5b9a756242f..2549d3a297a 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -1102,7 +1102,7 @@ static int wm8961_register(struct wm8961_priv *wm8961) ret = wm8961_reset(codec); if (ret < 0) { dev_err(codec->dev, "Failed to issue reset\n"); - return ret; + goto err; } /* Enable class W */ @@ -1147,18 +1147,19 @@ static int wm8961_register(struct wm8961_priv *wm8961) ret = snd_soc_register_codec(codec); if (ret != 0) { dev_err(codec->dev, "Failed to register codec: %d\n", ret); - return ret; + goto err; } ret = snd_soc_register_dai(&wm8961_dai); if (ret != 0) { dev_err(codec->dev, "Failed to register DAI: %d\n", ret); - snd_soc_unregister_codec(codec); - return ret; + goto err_codec; } return 0; +err_codec: + snd_soc_unregister_codec(codec); err: kfree(wm8961); return ret; diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index a2c4b2f37cc..1468fe10cbb 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -670,7 +670,8 @@ static __devinit int wm8974_register(struct wm8974_priv *wm8974) if (wm8974_codec) { dev_err(codec->dev, "Another WM8974 is registered\n"); - return -EINVAL; + ret = -EINVAL; + goto err; } mutex_init(&codec->mutex); diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index 51d5f433215..8a1ad778e7e 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -1076,7 +1076,6 @@ static __devinit int wm8978_register(struct wm8978_priv *wm8978) err_codec: snd_soc_unregister_codec(codec); err: - kfree(wm8978); return ret; } @@ -1085,13 +1084,13 @@ static __devexit void wm8978_unregister(struct wm8978_priv *wm8978) wm8978_set_bias_level(&wm8978->codec, SND_SOC_BIAS_OFF); snd_soc_unregister_dai(&wm8978_dai); snd_soc_unregister_codec(&wm8978->codec); - kfree(wm8978); wm8978_codec = NULL; } static __devinit int wm8978_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { + int ret; struct wm8978_priv *wm8978; struct snd_soc_codec *codec; @@ -1107,13 +1106,18 @@ static __devinit int wm8978_i2c_probe(struct i2c_client *i2c, codec->dev = &i2c->dev; - return wm8978_register(wm8978); + ret = wm8978_register(wm8978); + if (ret < 0) + kfree(wm8978); + + return ret; } static __devexit int wm8978_i2c_remove(struct i2c_client *client) { struct wm8978_priv *wm8978 = i2c_get_clientdata(client); wm8978_unregister(wm8978); + kfree(wm8978); return 0; } diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index c018772cc43..dd8d909788c 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -30,8 +30,6 @@ #include "wm8990.h" -#define WM8990_VERSION "0.2" - /* codec private data */ struct wm8990_priv { unsigned int sysclk; @@ -1511,8 +1509,6 @@ static int wm8990_probe(struct platform_device *pdev) struct wm8990_priv *wm8990; int ret; - pr_info("WM8990 Audio Codec %s\n", WM8990_VERSION); - setup = socdev->codec_data; codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); if (codec == NULL) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index e84a1177f35..a87046a96f2 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1677,6 +1677,26 @@ static struct { static int wm8994_readable(unsigned int reg) { + switch (reg) { + case WM8994_GPIO_1: + case WM8994_GPIO_2: + case WM8994_GPIO_3: + case WM8994_GPIO_4: + case WM8994_GPIO_5: + case WM8994_GPIO_6: + case WM8994_GPIO_7: + case WM8994_GPIO_8: + case WM8994_GPIO_9: + case WM8994_GPIO_10: + case WM8994_GPIO_11: + case WM8994_INTERRUPT_STATUS_1: + case WM8994_INTERRUPT_STATUS_2: + case WM8994_INTERRUPT_RAW_STATUS_2: + return 1; + default: + break; + } + if (reg >= ARRAY_SIZE(access_masks)) return 0; return access_masks[reg].readable != 0; @@ -2341,6 +2361,20 @@ SOC_DAPM_SINGLE("AIF2 Switch", WM8994_AIF1_ADC1_RIGHT_MIXER_ROUTING, 0, 1, 0), }; +static const struct snd_kcontrol_new aif1adc2l_mix[] = { +SOC_DAPM_SINGLE("DMIC Switch", WM8994_AIF1_ADC2_LEFT_MIXER_ROUTING, + 1, 1, 0), +SOC_DAPM_SINGLE("AIF2 Switch", WM8994_AIF1_ADC2_LEFT_MIXER_ROUTING, + 0, 1, 0), +}; + +static const struct snd_kcontrol_new aif1adc2r_mix[] = { +SOC_DAPM_SINGLE("DMIC Switch", WM8994_AIF1_ADC2_RIGHT_MIXER_ROUTING, + 1, 1, 0), +SOC_DAPM_SINGLE("AIF2 Switch", WM8994_AIF1_ADC2_RIGHT_MIXER_ROUTING, + 0, 1, 0), +}; + static const struct snd_kcontrol_new aif2dac2l_mix[] = { SOC_DAPM_SINGLE("Right Sidetone Switch", WM8994_DAC2_LEFT_MIXER_ROUTING, 5, 1, 0), @@ -2472,6 +2506,7 @@ static const struct snd_kcontrol_new aif3adc_mux = static const struct snd_soc_dapm_widget wm8994_dapm_widgets[] = { SND_SOC_DAPM_INPUT("DMIC1DAT"), SND_SOC_DAPM_INPUT("DMIC2DAT"), +SND_SOC_DAPM_INPUT("Clock"), SND_SOC_DAPM_SUPPLY("CLK_SYS", SND_SOC_NOPM, 0, 0, clk_sys_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), @@ -2506,6 +2541,11 @@ SND_SOC_DAPM_MIXER("AIF1ADC1L Mixer", SND_SOC_NOPM, 0, 0, SND_SOC_DAPM_MIXER("AIF1ADC1R Mixer", SND_SOC_NOPM, 0, 0, aif1adc1r_mix, ARRAY_SIZE(aif1adc1r_mix)), +SND_SOC_DAPM_MIXER("AIF1ADC2L Mixer", SND_SOC_NOPM, 0, 0, + aif1adc2l_mix, ARRAY_SIZE(aif1adc2l_mix)), +SND_SOC_DAPM_MIXER("AIF1ADC2R Mixer", SND_SOC_NOPM, 0, 0, + aif1adc2r_mix, ARRAY_SIZE(aif1adc2r_mix)), + SND_SOC_DAPM_MIXER("AIF2DAC2L Mixer", SND_SOC_NOPM, 0, 0, aif2dac2l_mix, ARRAY_SIZE(aif2dac2l_mix)), SND_SOC_DAPM_MIXER("AIF2DAC2R Mixer", SND_SOC_NOPM, 0, 0, @@ -2668,6 +2708,14 @@ static const struct snd_soc_dapm_route intercon[] = { { "AIF1ADC1R Mixer", "ADC/DMIC Switch", "ADCR Mux" }, { "AIF1ADC1R Mixer", "AIF2 Switch", "AIF2DACR" }, + { "AIF1ADC2L", NULL, "AIF1ADC2L Mixer" }, + { "AIF1ADC2L Mixer", "DMIC Switch", "DMIC2L" }, + { "AIF1ADC2L Mixer", "AIF2 Switch", "AIF2DACL" }, + + { "AIF1ADC2R", NULL, "AIF1ADC2R Mixer" }, + { "AIF1ADC2R Mixer", "DMIC Switch", "DMIC2R" }, + { "AIF1ADC2R Mixer", "AIF2 Switch", "AIF2DACR" }, + /* Pin level routing for AIF3 */ { "AIF1DAC1L", NULL, "AIF1DAC Mux" }, { "AIF1DAC1R", NULL, "AIF1DAC Mux" }, @@ -2946,11 +2994,14 @@ static int wm8994_set_fll(struct snd_soc_dai *dai, int id, int src, return 0; } +static int opclk_divs[] = { 10, 20, 30, 40, 55, 60, 80, 120, 160 }; + static int wm8994_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, unsigned int freq, int dir) { struct snd_soc_codec *codec = dai->codec; struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + int i; switch (dai->id) { case 1: @@ -2988,6 +3039,25 @@ static int wm8994_set_dai_sysclk(struct snd_soc_dai *dai, dev_dbg(dai->dev, "AIF%d using FLL2\n", dai->id); break; + case WM8994_SYSCLK_OPCLK: + /* Special case - a division (times 10) is given and + * no effect on main clocking. + */ + if (freq) { + for (i = 0; i < ARRAY_SIZE(opclk_divs); i++) + if (opclk_divs[i] == freq) + break; + if (i == ARRAY_SIZE(opclk_divs)) + return -EINVAL; + snd_soc_update_bits(codec, WM8994_CLOCKING_2, + WM8994_OPCLK_DIV_MASK, i); + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_2, + WM8994_OPCLK_ENA, WM8994_OPCLK_ENA); + } else { + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_2, + WM8994_OPCLK_ENA, 0); + } + default: return -EINVAL; } @@ -4004,6 +4074,11 @@ static int wm8994_codec_probe(struct platform_device *pdev) 1 << WM8994_AIF2DAC_3D_GAIN_SHIFT, 1 << WM8994_AIF2DAC_3D_GAIN_SHIFT); + /* Unconditionally enable AIF1 ADC TDM mode; it only affects + * behaviour on idle TDM clock cycles. */ + snd_soc_update_bits(codec, WM8994_AIF1_CONTROL_1, + WM8994_AIF1ADC_TDM, WM8994_AIF1ADC_TDM); + wm8994_update_class_w(codec); ret = snd_soc_register_codec(codec); diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h index 7072dc53935..2e0ca67a8df 100644 --- a/sound/soc/codecs/wm8994.h +++ b/sound/soc/codecs/wm8994.h @@ -20,6 +20,9 @@ extern struct snd_soc_dai wm8994_dai[]; #define WM8994_SYSCLK_FLL1 3 #define WM8994_SYSCLK_FLL2 4 +/* OPCLK is also configured with set_dai_sysclk, specify division*10 as rate. */ +#define WM8994_SYSCLK_OPCLK 5 + #define WM8994_FLL1 1 #define WM8994_FLL2 2 diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 13186fb4dcb..76b37ff6c26 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -1356,7 +1356,7 @@ static int wm9081_register(struct wm9081_priv *wm9081, ret = snd_soc_codec_set_cache_io(codec, 8, 16, control); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; + goto err; } reg = snd_soc_read(codec, WM9081_SOFTWARE_RESET); @@ -1369,7 +1369,7 @@ static int wm9081_register(struct wm9081_priv *wm9081, ret = wm9081_reset(codec); if (ret < 0) { dev_err(codec->dev, "Failed to issue reset\n"); - return ret; + goto err; } wm9081_set_bias_level(codec, SND_SOC_BIAS_STANDBY); @@ -1388,18 +1388,19 @@ static int wm9081_register(struct wm9081_priv *wm9081, ret = snd_soc_register_codec(codec); if (ret != 0) { dev_err(codec->dev, "Failed to register codec: %d\n", ret); - return ret; + goto err; } ret = snd_soc_register_dai(&wm9081_dai); if (ret != 0) { dev_err(codec->dev, "Failed to register DAI: %d\n", ret); - snd_soc_unregister_codec(codec); - return ret; + goto err_codec; } return 0; +err_codec: + snd_soc_unregister_codec(codec); err: kfree(wm9081); return ret; diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 16f1a57da08..2cb81538cd9 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -410,6 +410,8 @@ static int hp_event(struct snd_soc_dapm_widget *w, WM8993_HPOUT1L_DLY | WM8993_HPOUT1R_DLY, 0); + snd_soc_write(codec, WM8993_DC_SERVO_0, 0); + snd_soc_update_bits(codec, WM8993_POWER_MANAGEMENT_1, WM8993_HPOUT1L_ENA | WM8993_HPOUT1R_ENA, 0); |