diff options
Diffstat (limited to 'sound/soc/codecs')
-rw-r--r-- | sound/soc/codecs/Kconfig | 13 | ||||
-rw-r--r-- | sound/soc/codecs/Makefile | 4 | ||||
-rw-r--r-- | sound/soc/codecs/ac97.c | 3 | ||||
-rw-r--r-- | sound/soc/codecs/ad1980.c | 1 | ||||
-rw-r--r-- | sound/soc/codecs/ad73311.c | 107 | ||||
-rw-r--r-- | sound/soc/codecs/ad73311.h | 90 | ||||
-rw-r--r-- | sound/soc/codecs/ak4535.c | 1 | ||||
-rw-r--r-- | sound/soc/codecs/ssm2602.c | 1 | ||||
-rw-r--r-- | sound/soc/codecs/tlv320aic23.c | 714 | ||||
-rw-r--r-- | sound/soc/codecs/tlv320aic23.h | 122 | ||||
-rw-r--r-- | sound/soc/codecs/tlv320aic3x.c | 5 | ||||
-rw-r--r-- | sound/soc/codecs/uda1380.c | 1 | ||||
-rw-r--r-- | sound/soc/codecs/wm8510.c | 111 | ||||
-rw-r--r-- | sound/soc/codecs/wm8510.h | 1 | ||||
-rw-r--r-- | sound/soc/codecs/wm8580.c | 2 | ||||
-rw-r--r-- | sound/soc/codecs/wm8731.c | 1 | ||||
-rw-r--r-- | sound/soc/codecs/wm8750.c | 1 | ||||
-rw-r--r-- | sound/soc/codecs/wm8753.c | 75 | ||||
-rw-r--r-- | sound/soc/codecs/wm8753.h | 4 | ||||
-rw-r--r-- | sound/soc/codecs/wm8900.c | 1 | ||||
-rw-r--r-- | sound/soc/codecs/wm8903.c | 4 | ||||
-rw-r--r-- | sound/soc/codecs/wm8971.c | 1 | ||||
-rw-r--r-- | sound/soc/codecs/wm8990.c | 1 | ||||
-rw-r--r-- | sound/soc/codecs/wm9712.c | 3 | ||||
-rw-r--r-- | sound/soc/codecs/wm9713.c | 3 |
25 files changed, 1224 insertions, 46 deletions
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index e0b9869df0f..38a0e3b620a 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -3,9 +3,11 @@ config SND_SOC_ALL_CODECS depends on I2C select SPI select SPI_MASTER + select SND_SOC_AD73311 select SND_SOC_AK4535 select SND_SOC_CS4270 select SND_SOC_SSM2602 + select SND_SOC_TLV320AIC23 select SND_SOC_TLV320AIC26 select SND_SOC_TLV320AIC3X select SND_SOC_UDA1380 @@ -34,6 +36,9 @@ config SND_SOC_AC97_CODEC config SND_SOC_AD1980 tristate +config SND_SOC_AD73311 + tristate + config SND_SOC_AK4535 tristate @@ -58,9 +63,13 @@ config SND_SOC_CS4270_VD33_ERRATA config SND_SOC_SSM2602 tristate +config SND_SOC_TLV320AIC23 + tristate + depends on I2C + config SND_SOC_TLV320AIC26 - tristate "TI TLV320AIC26 Codec support" - depends on SND_SOC && SPI + tristate "TI TLV320AIC26 Codec support" if SND_SOC_OF_SIMPLE + depends on SPI config SND_SOC_TLV320AIC3X tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index f977978a340..90f0a585fc7 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -1,8 +1,10 @@ snd-soc-ac97-objs := ac97.o snd-soc-ad1980-objs := ad1980.o +snd-soc-ad73311-objs := ad73311.o snd-soc-ak4535-objs := ak4535.o snd-soc-cs4270-objs := cs4270.o snd-soc-ssm2602-objs := ssm2602.o +snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o snd-soc-uda1380-objs := uda1380.o @@ -20,9 +22,11 @@ snd-soc-wm9713-objs := wm9713.o obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o +obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o +obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index 61fd96ca7bc..bd1ebdc6c86 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -2,8 +2,7 @@ * ac97.c -- ALSA Soc AC97 codec support * * Copyright 2005 Wolfson Microelectronics PLC. - * Author: Liam Girdwood - * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com + * Author: Liam Girdwood <lrg@slimlogic.co.uk> * * 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 diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index 4e09c1f2c06..1397b8e06c0 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -13,7 +13,6 @@ #include <linux/init.h> #include <linux/module.h> -#include <linux/version.h> #include <linux/kernel.h> #include <linux/device.h> #include <sound/core.h> diff --git a/sound/soc/codecs/ad73311.c b/sound/soc/codecs/ad73311.c new file mode 100644 index 00000000000..37af8607b00 --- /dev/null +++ b/sound/soc/codecs/ad73311.c @@ -0,0 +1,107 @@ +/* + * ad73311.c -- ALSA Soc AD73311 codec support + * + * Copyright: Analog Device Inc. + * Author: Cliff Cai <cliff.cai@analog.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + * Revision history + * 25th Sep 2008 Initial version. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/version.h> +#include <linux/kernel.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/ac97_codec.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include "ad73311.h" + +struct snd_soc_dai ad73311_dai = { + .name = "AD73311", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, }, +}; +EXPORT_SYMBOL_GPL(ad73311_dai); + +static int ad73311_soc_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) + return -ENOMEM; + mutex_init(&codec->mutex); + codec->name = "AD73311"; + codec->owner = THIS_MODULE; + codec->dai = &ad73311_dai; + codec->num_dai = 1; + socdev->codec = codec; + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "ad73311: failed to create pcms\n"); + goto pcm_err; + } + + ret = snd_soc_register_card(socdev); + if (ret < 0) { + printk(KERN_ERR "ad73311: failed to register card\n"); + goto register_err; + } + + return ret; + +register_err: + snd_soc_free_pcms(socdev); +pcm_err: + kfree(socdev->codec); + socdev->codec = NULL; + return ret; +} + +static int ad73311_soc_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + if (codec == NULL) + return 0; + snd_soc_free_pcms(socdev); + kfree(codec); + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_ad73311 = { + .probe = ad73311_soc_probe, + .remove = ad73311_soc_remove, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_ad73311); + +MODULE_DESCRIPTION("ASoC ad73311 driver"); +MODULE_AUTHOR("Cliff Cai "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ad73311.h b/sound/soc/codecs/ad73311.h new file mode 100644 index 00000000000..507ce0c30ed --- /dev/null +++ b/sound/soc/codecs/ad73311.h @@ -0,0 +1,90 @@ +/* + * File: sound/soc/codec/ad73311.h + * Based on: + * Author: Cliff Cai <cliff.cai@analog.com> + * + * Created: Thur Sep 25, 2008 + * Description: definitions for AD73311 registers + * + * + * Modified: + * Copyright 2006 Analog Devices Inc. + * + * Bugs: Enter bugs at http://blackfin.uclinux.org/ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, see the file COPYING, or write + * to the Free Software Foundation, Inc., + * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#ifndef __AD73311_H__ +#define __AD73311_H__ + +#define AD_CONTROL 0x8000 +#define AD_DATA 0x0000 +#define AD_READ 0x4000 +#define AD_WRITE 0x0000 + +/* Control register A */ +#define CTRL_REG_A (0 << 8) + +#define REGA_MODE_PRO 0x00 +#define REGA_MODE_DATA 0x01 +#define REGA_MODE_MIXED 0x03 +#define REGA_DLB 0x04 +#define REGA_SLB 0x08 +#define REGA_DEVC(x) ((x & 0x7) << 4) +#define REGA_RESET 0x80 + +/* Control register B */ +#define CTRL_REG_B (1 << 8) + +#define REGB_DIRATE(x) (x & 0x3) +#define REGB_SCDIV(x) ((x & 0x3) << 2) +#define REGB_MCDIV(x) ((x & 0x7) << 4) +#define REGB_CEE (1 << 7) + +/* Control register C */ +#define CTRL_REG_C (2 << 8) + +#define REGC_PUDEV (1 << 0) +#define REGC_PUADC (1 << 3) +#define REGC_PUDAC (1 << 4) +#define REGC_PUREF (1 << 5) +#define REGC_REFUSE (1 << 6) + +/* Control register D */ +#define CTRL_REG_D (3 << 8) + +#define REGD_IGS(x) (x & 0x7) +#define REGD_RMOD (1 << 3) +#define REGD_OGS(x) ((x & 0x7) << 4) +#define REGD_MUTE (x << 7) + +/* Control register E */ +#define CTRL_REG_E (4 << 8) + +#define REGE_DA(x) (x & 0x1f) +#define REGE_IBYP (1 << 5) + +/* Control register F */ +#define CTRL_REG_F (5 << 8) + +#define REGF_SEEN (1 << 5) +#define REGF_INV (1 << 6) +#define REGF_ALB (1 << 7) + +extern struct snd_soc_dai ad73311_dai; +extern struct snd_soc_codec_device soc_codec_dev_ad73311; +#endif diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index 088cf992772..2a89b5888e1 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -28,7 +28,6 @@ #include "ak4535.h" -#define AUDIO_NAME "ak4535" #define AK4535_VERSION "0.3" struct snd_soc_codec_device soc_codec_dev_ak4535; diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 940ce1c3522..44ef0dacd56 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -42,7 +42,6 @@ #include "ssm2602.h" -#define AUDIO_NAME "ssm2602" #define SSM2602_VERSION "0.1" struct snd_soc_codec_device soc_codec_dev_ssm2602; diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c new file mode 100644 index 00000000000..44308dac9e1 --- /dev/null +++ b/sound/soc/codecs/tlv320aic23.c @@ -0,0 +1,714 @@ +/* + * ALSA SoC TLV320AIC23 codec driver + * + * Author: Arun KS, <arunks@mistralsolutions.com> + * Copyright: (C) 2008 Mistral Solutions Pvt Ltd., + * + * Based on sound/soc/codecs/wm8731.c by Richard Purdie + * + * 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. + * + * Notes: + * The AIC23 is a driver for a low power stereo audio + * codec tlv320aic23 + * + * The machine layer should disable unsupported inputs/outputs by + * snd_soc_dapm_disable_pin(codec, "LHPOUT"), etc. + */ + +#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 <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 "tlv320aic23.h" + +#define AIC23_VERSION "0.1" + +struct tlv320aic23_srate_reg_info { + u32 sample_rate; + u8 control; /* SR3, SR2, SR1, SR0 and BOSR */ + u8 divider; /* if 0 CLKIN = MCLK, if 1 CLKIN = MCLK/2 */ +}; + +/* + * AIC23 register cache + */ +static const u16 tlv320aic23_reg[] = { + 0x0097, 0x0097, 0x00F9, 0x00F9, /* 0 */ + 0x001A, 0x0004, 0x0007, 0x0001, /* 4 */ + 0x0020, 0x0000, 0x0000, 0x0000, /* 8 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 12 */ +}; + +/* + * read tlv320aic23 register cache + */ +static inline unsigned int tlv320aic23_read_reg_cache(struct snd_soc_codec + *codec, unsigned int reg) +{ + u16 *cache = codec->reg_cache; + if (reg >= ARRAY_SIZE(tlv320aic23_reg)) + return -1; + return cache[reg]; +} + +/* + * write tlv320aic23 register cache + */ +static inline void tlv320aic23_write_reg_cache(struct snd_soc_codec *codec, + u8 reg, u16 value) +{ + u16 *cache = codec->reg_cache; + if (reg >= ARRAY_SIZE(tlv320aic23_reg)) + return; + cache[reg] = value; +} + +/* + * write to the tlv320aic23 register space + */ +static int tlv320aic23_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + + u8 data[2]; + + /* TLV320AIC23 has 7 bit address and 9 bits of data + * so we need to switch one data bit into reg and rest + * of data into val + */ + + if ((reg < 0 || reg > 9) && (reg != 15)) { + printk(KERN_WARNING "%s Invalid register R%d\n", __func__, reg); + return -1; + } + + data[0] = (reg << 1) | (value >> 8 & 0x01); + data[1] = value & 0xff; + + tlv320aic23_write_reg_cache(codec, reg, value); + + if (codec->hw_write(codec->control_data, data, 2) == 2) + return 0; + + printk(KERN_ERR "%s cannot write %03x to register R%d\n", __func__, + value, reg); + + return -EIO; +} + +static const char *rec_src_text[] = { "Line", "Mic" }; +static const char *deemph_text[] = {"None", "32Khz", "44.1Khz", "48Khz"}; + +static const struct soc_enum rec_src_enum = + SOC_ENUM_SINGLE(TLV320AIC23_ANLG, 2, 2, rec_src_text); + +static const struct snd_kcontrol_new tlv320aic23_rec_src_mux_controls = +SOC_DAPM_ENUM("Input Select", rec_src_enum); + +static const struct soc_enum tlv320aic23_rec_src = + SOC_ENUM_SINGLE(TLV320AIC23_ANLG, 2, 2, rec_src_text); +static const struct soc_enum tlv320aic23_deemph = + SOC_ENUM_SINGLE(TLV320AIC23_DIGT, 1, 4, deemph_text); + +static const DECLARE_TLV_DB_SCALE(out_gain_tlv, -12100, 100, 0); +static const DECLARE_TLV_DB_SCALE(input_gain_tlv, -1725, 75, 0); +static const DECLARE_TLV_DB_SCALE(sidetone_vol_tlv, -1800, 300, 0); + +static int snd_soc_tlv320aic23_put_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + u16 val, reg; + + val = (ucontrol->value.integer.value[0] & 0x07); + + /* linear conversion to userspace + * 000 = -6db + * 001 = -9db + * 010 = -12db + * 011 = -18db (Min) + * 100 = 0db (Max) + */ + val = (val >= 4) ? 4 : (3 - val); + + reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_ANLG) & (~0x1C0); + tlv320aic23_write(codec, TLV320AIC23_ANLG, reg | (val << 6)); + + return 0; +} + +static int snd_soc_tlv320aic23_get_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + u16 val; + + val = tlv320aic23_read_reg_cache(codec, TLV320AIC23_ANLG) & (0x1C0); + val = val >> 6; + val = (val >= 4) ? 4 : (3 - val); + ucontrol->value.integer.value[0] = val; + return 0; + +} + +#define SOC_TLV320AIC23_SINGLE_TLV(xname, reg, shift, max, invert, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ + SNDRV_CTL_ELEM_ACCESS_READWRITE,\ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw, .get = snd_soc_tlv320aic23_get_volsw,\ + .put = snd_soc_tlv320aic23_put_volsw, \ + .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } + +static const struct snd_kcontrol_new tlv320aic23_snd_controls[] = { + SOC_DOUBLE_R_TLV("Digital Playback Volume", TLV320AIC23_LCHNVOL, + TLV320AIC23_RCHNVOL, 0, 127, 0, out_gain_tlv), + SOC_SINGLE("Digital Playback Switch", TLV320AIC23_DIGT, 3, 1, 1), + SOC_DOUBLE_R("Line Input Switch", TLV320AIC23_LINVOL, + TLV320AIC23_RINVOL, 7, 1, 0), + SOC_DOUBLE_R_TLV("Line Input Volume", TLV320AIC23_LINVOL, + TLV320AIC23_RINVOL, 0, 31, 0, input_gain_tlv), + SOC_SINGLE("Mic Input Switch", TLV320AIC23_ANLG, 1, 1, 1), + SOC_SINGLE("Mic Booster Switch", TLV320AIC23_ANLG, 0, 1, 0), + SOC_TLV320AIC23_SINGLE_TLV("Sidetone Volume", TLV320AIC23_ANLG, + 6, 4, 0, sidetone_vol_tlv), + SOC_ENUM("Playback De-emphasis", tlv320aic23_deemph), +}; + +/* add non dapm controls */ +static int tlv320aic23_add_controls(struct snd_soc_codec *codec) +{ + + int err, i; + + for (i = 0; i < ARRAY_SIZE(tlv320aic23_snd_controls); i++) { + err = snd_ctl_add(codec->card, + snd_soc_cnew(&tlv320aic23_snd_controls[i], + codec, NULL)); + if (err < 0) + return err; + } + + return 0; + +} + +/* PGA Mixer controls for Line and Mic switch */ +static const struct snd_kcontrol_new tlv320aic23_output_mixer_controls[] = { + SOC_DAPM_SINGLE("Line Bypass Switch", TLV320AIC23_ANLG, 3, 1, 0), + SOC_DAPM_SINGLE("Mic Sidetone Switch", TLV320AIC23_ANLG, 5, 1, 0), + SOC_DAPM_SINGLE("Playback Switch", TLV320AIC23_ANLG, 4, 1, 0), +}; + +static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC", "Playback", TLV320AIC23_PWR, 3, 1), + SND_SOC_DAPM_ADC("ADC", "Capture", TLV320AIC23_PWR, 2, 1), + SND_SOC_DAPM_MUX("Capture Source", SND_SOC_NOPM, 0, 0, + &tlv320aic23_rec_src_mux_controls), + SND_SOC_DAPM_MIXER("Output Mixer", TLV320AIC23_PWR, 4, 1, + &tlv320aic23_output_mixer_controls[0], + ARRAY_SIZE(tlv320aic23_output_mixer_controls)), + SND_SOC_DAPM_PGA("Line Input", TLV320AIC23_PWR, 0, 1, NULL, 0), + SND_SOC_DAPM_PGA("Mic Input", TLV320AIC23_PWR, 1, 1, NULL, 0), + + SND_SOC_DAPM_OUTPUT("LHPOUT"), + SND_SOC_DAPM_OUTPUT("RHPOUT"), + SND_SOC_DAPM_OUTPUT("LOUT"), + SND_SOC_DAPM_OUTPUT("ROUT"), + + SND_SOC_DAPM_INPUT("LLINEIN"), + SND_SOC_DAPM_INPUT("RLINEIN"), + + SND_SOC_DAPM_INPUT("MICIN"), +}; + +static const struct snd_soc_dapm_route intercon[] = { + /* Output Mixer */ + {"Output Mixer", "Line Bypass Switch", "Line Input"}, + {"Output Mixer", "Playback Switch", "DAC"}, + {"Output Mixer", "Mic Sidetone Switch", "Mic Input"}, + + /* Outputs */ + {"RHPOUT", NULL, "Output Mixer"}, + {"LHPOUT", NULL, "Output Mixer"}, + {"LOUT", NULL, "Output Mixer"}, + {"ROUT", NULL, "Output Mixer"}, + + /* Inputs */ + {"Line Input", "NULL", "LLINEIN"}, + {"Line Input", "NULL", "RLINEIN"}, + + {"Mic Input", "NULL", "MICIN"}, + + /* input mux */ + {"Capture Source", "Line", "Line Input"}, + {"Capture Source", "Mic", "Mic Input"}, + {"ADC", NULL, "Capture Source"}, + +}; + +/* tlv320aic23 related */ +static const struct tlv320aic23_srate_reg_info srate_reg_info[] = { + {4000, 0x06, 1}, /* 4000 */ + {8000, 0x06, 0}, /* 8000 */ + {16000, 0x0C, 1}, /* 16000 */ + {22050, 0x11, 1}, /* 22050 */ + {24000, 0x00, 1}, /* 24000 */ + {32000, 0x0C, 0}, /* 32000 */ + {44100, 0x11, 0}, /* 44100 */ + {48000, 0x00, 0}, /* 48000 */ + {88200, 0x1F, 0}, /* 88200 */ + {96000, 0x0E, 0}, /* 96000 */ +}; + +static int tlv320aic23_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets, + ARRAY_SIZE(tlv320aic23_dapm_widgets)); + + /* set up audio path interconnects */ + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +static int tlv320aic23_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + u16 iface_reg, data; + u8 count = 0; + + iface_reg = + tlv320aic23_read_reg_cache(codec, + TLV320AIC23_DIGT_FMT) & ~(0x03 << 2); + + /* Search for the right sample rate */ + /* Verify what happens if the rate is not supported + * now it goes to 96Khz */ + while ((srate_reg_info[count].sample_rate != params_rate(params)) && + (count < ARRAY_SIZE(srate_reg_info))) { + count++; + } + + data = (srate_reg_info[count].divider << TLV320AIC23_CLKIN_SHIFT) | + (srate_reg_info[count]. control << TLV320AIC23_BOSR_SHIFT) | + TLV320AIC23_USB_CLK_ON; + + tlv320aic23_write(codec, TLV320AIC23_SRATE, data); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + iface_reg |= (0x01 << 2); + break; + case SNDRV_PCM_FORMAT_S24_LE: + iface_reg |= (0x02 << 2); + break; + case SNDRV_PCM_FORMAT_S32_LE: + iface_reg |= (0x03 << 2); + break; + } + tlv320aic23_write(codec, TLV320AIC23_DIGT_FMT, iface_reg); + + return 0; +} + +static int tlv320aic23_pcm_prepare(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->codec; + + /* set active */ + tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0001); + + return 0; +} + +static void tlv320aic23_shutdown(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->codec; + + /* deactivate */ + if (!codec->active) { + udelay(50); + tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0); + } +} + +static int tlv320aic23_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u16 reg; + + reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_DIGT); + if (mute) + reg |= TLV320AIC23_DACM_MUTE; + + else + reg &= ~TLV320AIC23_DACM_MUTE; + + tlv320aic23_write(codec, TLV320AIC23_DIGT, reg); + + return 0; +} + +static int tlv320aic23_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 iface_reg; + + iface_reg = + tlv320aic23_read_reg_cache(codec, TLV320AIC23_DIGT_FMT) & (~0x03); + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + iface_reg |= TLV320AIC23_MS_MASTER; + break; + 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_reg |= TLV320AIC23_FOR_I2S; + break; + case SND_SOC_DAIFMT_DSP_A: + iface_reg |= TLV320AIC23_FOR_DSP; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + iface_reg |= TLV320AIC23_FOR_LJUST; + break; + default: + return -EINVAL; + + } + + tlv320aic23_write(codec, TLV320AIC23_DIGT_FMT, iface_reg); + + return 0; +} + +static int tlv320aic23_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; + + switch (freq) { + case 12000000: + return 0; + } + return -EINVAL; +} + +static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + u16 reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_PWR) & 0xff7f; + + switch (level) { + case SND_SOC_BIAS_ON: + /* vref/mid, osc on, dac unmute */ + 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); + break; + case SND_SOC_BIAS_OFF: + /* everything off, dac mute, inactive */ + tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0); + tlv320aic23_write(codec, TLV320AIC23_PWR, 0xffff); + break; + } + codec->bias_level = level; + return 0; +} + +#define AIC23_RATES SNDRV_PCM_RATE_8000_96000 +#define AIC23_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) + +struct snd_soc_dai tlv320aic23_dai = { + .name = "tlv320aic23", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = AIC23_RATES, + .formats = AIC23_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = AIC23_RATES, + .formats = AIC23_FORMATS,}, + .ops = { + .prepare = tlv320aic23_pcm_prepare, + .hw_params = tlv320aic23_hw_params, + .shutdown = tlv320aic23_shutdown, + }, + .dai_ops = { + .digital_mute = tlv320aic23_mute, + .set_fmt = tlv320aic23_set_dai_fmt, + .set_sysclk = tlv320aic23_set_dai_sysclk, + } +}; +EXPORT_SYMBOL_GPL(tlv320aic23_dai); + +static int tlv320aic23_suspend(struct platform_device *pdev, + pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0); + tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int tlv320aic23_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + int i; + u16 reg; + + /* Sync reg_cache with the hardware */ + for (reg = 0; reg < ARRAY_SIZE(tlv320aic23_reg); i++) { + u16 val = tlv320aic23_read_reg_cache(codec, reg); + tlv320aic23_write(codec, reg, val); + } + + tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + tlv320aic23_set_bias_level(codec, codec->suspend_bias_level); + + return 0; +} + +/* + * initialise the AIC23 driver + * register the mixer and dsp interfaces with the kernel + */ +static int tlv320aic23_init(struct snd_soc_device *socdev) +{ + struct snd_soc_codec *codec = socdev->codec; + int ret = 0; + u16 reg; + + codec->name = "tlv320aic23"; + codec->owner = THIS_MODULE; + codec->read = tlv320aic23_read_reg_cache; + codec->write = tlv320aic23_write; + codec->set_bias_level = tlv320aic23_set_bias_level; + codec->dai = &tlv320aic23_dai; + codec->num_dai = 1; + codec->reg_cache_size = ARRAY_SIZE(tlv320aic23_reg); + codec->reg_cache = + kmemdup(tlv320aic23_reg, sizeof(tlv320aic23_reg), GFP_KERNEL); + if (codec->reg_cache == NULL) + return -ENOMEM; + + /* Reset codec */ + tlv320aic23_write(codec, TLV320AIC23_RESET, 0); + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "tlv320aic23: failed to create pcms\n"); + goto pcm_err; + } + + /* power on device */ + tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + tlv320aic23_write(codec, TLV320AIC23_DIGT, TLV320AIC23_DEEMP_44K); + + /* Unmute input */ + reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_LINVOL); + tlv320aic23_write(codec, TLV320AIC23_LINVOL, + (reg & (~TLV320AIC23_LIM_MUTED)) | + (TLV320AIC23_LRS_ENABLED)); + + reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_RINVOL); + tlv320aic23_write(codec, TLV320AIC23_RINVOL, + (reg & (~TLV320AIC23_LIM_MUTED)) | + TLV320AIC23_LRS_ENABLED); + + reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_ANLG); + tlv320aic23_write(codec, TLV320AIC23_ANLG, + (reg) & (~TLV320AIC23_BYPASS_ON) & + (~TLV320AIC23_MICM_MUTED)); + + /* Default output volume */ + tlv320aic23_write(codec, TLV320AIC23_LCHNVOL, + TLV320AIC23_DEFAULT_OUT_VOL & + TLV320AIC23_OUT_VOL_MASK); + tlv320aic23_write(codec, TLV320AIC23_RCHNVOL, + TLV320AIC23_DEFAULT_OUT_VOL & + TLV320AIC23_OUT_VOL_MASK); + + tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x1); + + tlv320aic23_add_controls(codec); + tlv320aic23_add_widgets(codec); + ret = snd_soc_register_card(socdev); + if (ret < 0) { + printk(KERN_ERR "tlv320aic23: failed to register card\n"); + goto card_err; + } + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + kfree(codec->reg_cache); + return ret; +} +static struct snd_soc_device *tlv320aic23_socdev; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +/* + * If the i2c layer weren't so broken, we could pass this kind of data + * around + */ +static int tlv320aic23_codec_probe(struct i2c_client *i2c, + const struct i2c_device_id *i2c_id) +{ + struct snd_soc_device *socdev = tlv320aic23_socdev; + struct snd_soc_codec *codec = socdev->codec; + int ret; + + if (!i2c_check_functionality(i2c->adapter, I2C_FUNC_SMBUS_BYTE_DATA)) + return -EINVAL; + + i2c_set_clientdata(i2c, codec); + codec->control_data = i2c; + + ret = tlv320aic23_init(socdev); + if (ret < 0) { + printk(KERN_ERR "tlv320aic23: failed to initialise AIC23\n"); + goto err; + } + return ret; + +err: + kfree(codec); + kfree(i2c); + return ret; +} +static int __exit tlv320aic23_i2c_remove(struct i2c_client *i2c) +{ + put_device(&i2c->dev); + return 0; +} + +static const struct i2c_device_id tlv320aic23_id[] = { + {"tlv320aic23", 0}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, tlv320aic23_id); + +static struct i2c_driver tlv320aic23_i2c_driver = { + .driver = { + .name = "tlv320aic23", + }, + .probe = tlv320aic23_codec_probe, + .remove = __exit_p(tlv320aic23_i2c_remove), + .id_table = tlv320aic23_id, +}; + +#endif + +static int tlv320aic23_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + printk(KERN_INFO "AIC23 Audio Codec %s\n", AIC23_VERSION); + + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) + return -ENOMEM; + + socdev->codec = codec; + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + tlv320aic23_socdev = socdev; +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + codec->hw_write = (hw_write_t) i2c_master_send; + codec->hw_read = NULL; + ret = i2c_add_driver(&tlv320aic23_i2c_driver); + if (ret != 0) + printk(KERN_ERR "can't add i2c driver"); +#endif + return ret; +} + +static int tlv320aic23_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + if (codec->control_data) + tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_OFF); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&tlv320aic23_i2c_driver); +#endif + kfree(codec->reg_cache); + kfree(codec); + + return 0; +} +struct snd_soc_codec_device soc_codec_dev_tlv320aic23 = { + .probe = tlv320aic23_probe, + .remove = tlv320aic23_remove, + .suspend = tlv320aic23_suspend, + .resume = tlv320aic23_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_tlv320aic23); + +MODULE_DESCRIPTION("ASoC TLV320AIC23 codec driver"); +MODULE_AUTHOR("Arun KS <arunks@mistralsolutions.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tlv320aic23.h b/sound/soc/codecs/tlv320aic23.h new file mode 100644 index 00000000000..79d1faf8e57 --- /dev/null +++ b/sound/soc/codecs/tlv320aic23.h @@ -0,0 +1,122 @@ +/* + * ALSA SoC TLV320AIC23 codec driver + * + * Author: Arun KS, <arunks@mistralsolutions.com> + * Copyright: (C) 2008 Mistral Solutions Pvt Ltd + * + * 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 _TLV320AIC23_H +#define _TLV320AIC23_H + +/* Codec TLV320AIC23 */ +#define TLV320AIC23_LINVOL 0x00 +#define TLV320AIC23_RINVOL 0x01 +#define TLV320AIC23_LCHNVOL 0x02 +#define TLV320AIC23_RCHNVOL 0x03 +#define TLV320AIC23_ANLG 0x04 +#define TLV320AIC23_DIGT 0x05 +#define TLV320AIC23_PWR 0x06 +#define TLV320AIC23_DIGT_FMT 0x07 +#define TLV320AIC23_SRATE 0x08 +#define TLV320AIC23_ACTIVE 0x09 +#define TLV320AIC23_RESET 0x0F + +/* Left (right) line input volume control register */ +#define TLV320AIC23_LRS_ENABLED 0x0100 +#define TLV320AIC23_LIM_MUTED 0x0080 +#define TLV320AIC23_LIV_DEFAULT 0x0017 +#define TLV320AIC23_LIV_MAX 0x001f +#define TLV320AIC23_LIV_MIN 0x0000 + +/* Left (right) channel headphone volume control register */ +#define TLV320AIC23_LZC_ON 0x0080 +#define TLV320AIC23_LHV_DEFAULT 0x0079 +#define TLV320AIC23_LHV_MAX 0x007f +#define TLV320AIC23_LHV_MIN 0x0000 + +/* Analog audio path control register */ +#define TLV320AIC23_STA_REG(x) ((x)<<6) +#define TLV320AIC23_STE_ENABLED 0x0020 +#define TLV320AIC23_DAC_SELECTED 0x0010 +#define TLV320AIC23_BYPASS_ON 0x0008 +#define TLV320AIC23_INSEL_MIC 0x0004 +#define TLV320AIC23_MICM_MUTED 0x0002 +#define TLV320AIC23_MICB_20DB 0x0001 + +/* Digital audio path control register */ +#define TLV320AIC23_DACM_MUTE 0x0008 +#define TLV320AIC23_DEEMP_32K 0x0002 +#define TLV320AIC23_DEEMP_44K 0x0004 +#define TLV320AIC23_DEEMP_48K 0x0006 +#define TLV320AIC23_ADCHP_ON 0x0001 + +/* Power control down register */ +#define TLV320AIC23_DEVICE_PWR_OFF 0x0080 +#define TLV320AIC23_CLK_OFF 0x0040 +#define TLV320AIC23_OSC_OFF 0x0020 +#define TLV320AIC23_OUT_OFF 0x0010 +#define TLV320AIC23_DAC_OFF 0x0008 +#define TLV320AIC23_ADC_OFF 0x0004 +#define TLV320AIC23_MIC_OFF 0x0002 +#define TLV320AIC23_LINE_OFF 0x0001 + +/* Digital audio interface register */ +#define TLV320AIC23_MS_MASTER 0x0040 +#define TLV320AIC23_LRSWAP_ON 0x0020 +#define TLV320AIC23_LRP_ON 0x0010 +#define TLV320AIC23_IWL_16 0x0000 +#define TLV320AIC23_IWL_20 0x0004 +#define TLV320AIC23_IWL_24 0x0008 +#define TLV320AIC23_IWL_32 0x000C +#define TLV320AIC23_FOR_I2S 0x0002 +#define TLV320AIC23_FOR_DSP 0x0003 +#define TLV320AIC23_FOR_LJUST 0x0001 + +/* Sample rate control register */ +#define TLV320AIC23_CLKOUT_HALF 0x0080 +#define TLV320AIC23_CLKIN_HALF 0x0040 +#define TLV320AIC23_BOSR_384fs 0x0002 /* BOSR_272fs in USB mode */ +#define TLV320AIC23_USB_CLK_ON 0x0001 +#define TLV320AIC23_SR_MASK 0xf +#define TLV320AIC23_CLKOUT_SHIFT 7 +#define TLV320AIC23_CLKIN_SHIFT 6 +#define TLV320AIC23_SR_SHIFT 2 +#define TLV320AIC23_BOSR_SHIFT 1 + +/* Digital interface register */ +#define TLV320AIC23_ACT_ON 0x0001 + +/* + * AUDIO related MACROS + */ + +#define TLV320AIC23_DEFAULT_OUT_VOL 0x70 +#define TLV320AIC23_DEFAULT_IN_VOLUME 0x10 + +#define TLV320AIC23_OUT_VOL_MIN TLV320AIC23_LHV_MIN +#define TLV320AIC23_OUT_VOL_MAX TLV320AIC23_LHV_MAX +#define TLV320AIC23_OUT_VO_RANGE (TLV320AIC23_OUT_VOL_MAX - \ + TLV320AIC23_OUT_VOL_MIN) +#define TLV320AIC23_OUT_VOL_MASK TLV320AIC23_OUT_VOL_MAX + +#define TLV320AIC23_IN_VOL_MIN TLV320AIC23_LIV_MIN +#define TLV320AIC23_IN_VOL_MAX TLV320AIC23_LIV_MAX +#define TLV320AIC23_IN_VOL_RANGE (TLV320AIC23_IN_VOL_MAX - \ + TLV320AIC23_IN_VOL_MIN) +#define TLV320AIC23_IN_VOL_MASK TLV320AIC23_IN_VOL_MAX + +#define TLV320AIC23_SIDETONE_MASK 0x1c0 +#define TLV320AIC23_SIDETONE_0 0x100 +#define TLV320AIC23_SIDETONE_6 0x000 +#define TLV320AIC23_SIDETONE_9 0x040 +#define TLV320AIC23_SIDETONE_12 0x080 +#define TLV320AIC23_SIDETONE_18 0x0c0 + +extern struct snd_soc_dai tlv320aic23_dai; +extern struct snd_soc_codec_device soc_codec_dev_tlv320aic23; + +#endif /* _TLV320AIC23_H */ diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 566a427c928..05336ed7e49 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -48,7 +48,6 @@ #include "tlv320aic3x.h" -#define AUDIO_NAME "aic3x" #define AIC3X_VERSION "0.2" /* codec private data */ @@ -991,7 +990,7 @@ EXPORT_SYMBOL_GPL(aic3x_headset_detected); SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) struct snd_soc_dai aic3x_dai = { - .name = "aic3x", + .name = "tlv320aic3x", .playback = { .stream_name = "Playback", .channels_min = 1, @@ -1055,7 +1054,7 @@ static int aic3x_init(struct snd_soc_device *socdev) struct aic3x_setup_data *setup = socdev->codec_data; int reg, ret = 0; - codec->name = "aic3x"; + codec->name = "tlv320aic3x"; codec->owner = THIS_MODULE; codec->read = aic3x_read_reg_cache; codec->write = aic3x_write; diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index d206d7f892b..a69ee72a7af 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -36,7 +36,6 @@ #include "uda1380.h" #define UDA1380_VERSION "0.6" -#define AUDIO_NAME "uda1380" /* * uda1380 register cache diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 9a37c8d95ed..d8ca2da8d63 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -3,7 +3,7 @@ * * Copyright 2006 Wolfson Microelectronics PLC. * - * Author: Liam Girdwood <liam.girdwood@wolfsonmicro.com> + * Author: Liam Girdwood <lrg@slimlogic.co.uk> * * 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 @@ -18,6 +18,7 @@ #include <linux/pm.h> #include <linux/i2c.h> #include <linux/platform_device.h> +#include <linux/spi/spi.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -27,7 +28,6 @@ #include "wm8510.h" -#define AUDIO_NAME "wm8510" #define WM8510_VERSION "0.6" struct snd_soc_codec_device soc_codec_dev_wm8510; @@ -55,6 +55,9 @@ static const u16 wm8510_reg[WM8510_CACHEREGNUM] = { 0x0001, }; +#define WM8510_POWER1_BIASEN 0x08 +#define WM8510_POWER1_BUFIOEN 0x10 + /* * read wm8510 register cache */ @@ -224,9 +227,9 @@ SND_SOC_DAPM_PGA("SpkN Out", WM8510_POWER3, 5, 0, NULL, 0), SND_SOC_DAPM_PGA("SpkP Out", WM8510_POWER3, 6, 0, NULL, 0), SND_SOC_DAPM_PGA("Mono Out", WM8510_POWER3, 7, 0, NULL, 0), -SND_SOC_DAPM_PGA("Mic PGA", WM8510_POWER2, 2, 0, - &wm8510_micpga_controls[0], - ARRAY_SIZE(wm8510_micpga_controls)), +SND_SOC_DAPM_MIXER("Mic PGA", WM8510_POWER2, 2, 0, + &wm8510_micpga_controls[0], + ARRAY_SIZE(wm8510_micpga_controls)), SND_SOC_DAPM_MIXER("Boost Mixer", WM8510_POWER2, 4, 0, &wm8510_boost_controls[0], ARRAY_SIZE(wm8510_boost_controls)), @@ -526,23 +529,35 @@ static int wm8510_mute(struct snd_soc_dai *dai, int mute) static int wm8510_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + u16 power1 = wm8510_read_reg_cache(codec, WM8510_POWER1) & ~0x3; switch (level) { case SND_SOC_BIAS_ON: - wm8510_write(codec, WM8510_POWER1, 0x1ff); - wm8510_write(codec, WM8510_POWER2, 0x1ff); - wm8510_write(codec, WM8510_POWER3, 0x1ff); - break; case SND_SOC_BIAS_PREPARE: + power1 |= 0x1; /* VMID 50k */ + wm8510_write(codec, WM8510_POWER1, power1); + break; + case SND_SOC_BIAS_STANDBY: + power1 |= WM8510_POWER1_BIASEN | WM8510_POWER1_BUFIOEN; + + if (codec->bias_level == SND_SOC_BIAS_OFF) { + /* Initial cap charge at VMID 5k */ + wm8510_write(codec, WM8510_POWER1, power1 | 0x3); + mdelay(100); + } + + power1 |= 0x2; /* VMID 500k */ + wm8510_write(codec, WM8510_POWER1, power1); break; + case SND_SOC_BIAS_OFF: - /* everything off, dac mute, inactive */ - wm8510_write(codec, WM8510_POWER1, 0x0); - wm8510_write(codec, WM8510_POWER2, 0x0); - wm8510_write(codec, WM8510_POWER3, 0x0); + wm8510_write(codec, WM8510_POWER1, 0); + wm8510_write(codec, WM8510_POWER2, 0); + wm8510_write(codec, WM8510_POWER3, 0); break; } + codec->bias_level = level; return 0; } @@ -640,6 +655,7 @@ static int wm8510_init(struct snd_soc_device *socdev) } /* power on device */ + codec->bias_level = SND_SOC_BIAS_OFF; wm8510_set_bias_level(codec, SND_SOC_BIAS_STANDBY); wm8510_add_controls(codec); wm8510_add_widgets(codec); @@ -747,6 +763,62 @@ err_driver: } #endif +#if defined(CONFIG_SPI_MASTER) +static int __devinit wm8510_spi_probe(struct spi_device *spi) +{ + struct snd_soc_device *socdev = wm8510_socdev; + struct snd_soc_codec *codec = socdev->codec; + int ret; + + codec->control_data = spi; + + ret = wm8510_init(socdev); + if (ret < 0) + dev_err(&spi->dev, "failed to initialise WM8510\n"); + + return ret; +} + +static int __devexit wm8510_spi_remove(struct spi_device *spi) +{ + return 0; +} + +static struct spi_driver wm8510_spi_driver = { + .driver = { + .name = "wm8510", + .bus = &spi_bus_type, + .owner = THIS_MODULE, + }, + .probe = wm8510_spi_probe, + .remove = __devexit_p(wm8510_spi_remove), +}; + +static int wm8510_spi_write(struct spi_device *spi, const char *data, int len) +{ + struct spi_transfer t; + struct spi_message m; + u8 msg[2]; + + if (len <= 0) + return 0; + + msg[0] = data[0]; + msg[1] = data[1]; + + spi_message_init(&m); + memset(&t, 0, (sizeof t)); + + t.tx_buf = &msg[0]; + t.len = len; + + spi_message_add_tail(&t, &m); + spi_sync(spi, &m); + + return len; +} +#endif /* CONFIG_SPI_MASTER */ + static int wm8510_probe(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); @@ -772,8 +844,14 @@ static int wm8510_probe(struct platform_device *pdev) codec->hw_write = (hw_write_t)i2c_master_send; ret = wm8510_add_i2c_device(pdev, setup); } -#else - /* Add other interfaces here */ +#endif +#if defined(CONFIG_SPI_MASTER) + if (setup->spi) { + codec->hw_write = (hw_write_t)wm8510_spi_write; + ret = spi_register_driver(&wm8510_spi_driver); + if (ret != 0) + printk(KERN_ERR "can't add spi driver"); + } #endif if (ret != 0) @@ -796,6 +874,9 @@ static int wm8510_remove(struct platform_device *pdev) i2c_unregister_device(codec->control_data); i2c_del_driver(&wm8510_i2c_driver); #endif +#if defined(CONFIG_SPI_MASTER) + spi_unregister_driver(&wm8510_spi_driver); +#endif kfree(codec); return 0; diff --git a/sound/soc/codecs/wm8510.h b/sound/soc/codecs/wm8510.h index c5368396045..bdefcf5c69f 100644 --- a/sound/soc/codecs/wm8510.h +++ b/sound/soc/codecs/wm8510.h @@ -94,6 +94,7 @@ #define WM8510_MCLKDIV_12 (7 << 5) struct wm8510_setup_data { + int spi; int i2c_bus; unsigned short i2c_address; }; diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index df1ffbe305b..627ebfb4209 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -18,7 +18,6 @@ #include <linux/module.h> #include <linux/moduleparam.h> -#include <linux/version.h> #include <linux/kernel.h> #include <linux/init.h> #include <linux/delay.h> @@ -36,7 +35,6 @@ #include "wm8580.h" -#define AUDIO_NAME "wm8580" #define WM8580_VERSION "0.1" struct pll_state { diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 7b64d9a7ff7..7f8a7e36b33 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -29,7 +29,6 @@ #include "wm8731.h" -#define AUDIO_NAME "wm8731" #define WM8731_VERSION "0.13" struct snd_soc_codec_device soc_codec_dev_wm8731; diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 4892e398a59..9b7296ee5b0 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -29,7 +29,6 @@ #include "wm8750.h" -#define AUDIO_NAME "WM8750" #define WM8750_VERSION "0.12" /* codec private data */ diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 8c4df44f334..d426eaa2218 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -2,8 +2,7 @@ * wm8753.c -- WM8753 ALSA Soc Audio driver * * Copyright 2003 Wolfson Microelectronics PLC. - * Author: Liam Girdwood - * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com + * Author: Liam Girdwood <lrg@slimlogic.co.uk> * * 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 @@ -40,6 +39,7 @@ #include <linux/pm.h> #include <linux/i2c.h> #include <linux/platform_device.h> +#include <linux/spi/spi.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -51,7 +51,6 @@ #include "wm8753.h" -#define AUDIO_NAME "wm8753" #define WM8753_VERSION "0.16" static int caps_charge = 2000; @@ -1719,6 +1718,63 @@ err_driver: } #endif +#if defined(CONFIG_SPI_MASTER) +static int __devinit wm8753_spi_probe(struct spi_device *spi) +{ + struct snd_soc_device *socdev = wm8753_socdev; + struct snd_soc_codec *codec = socdev->codec; + int ret; + + codec->control_data = spi; + + ret = wm8753_init(socdev); + if (ret < 0) + dev_err(&spi->dev, "failed to initialise WM8753\n"); + + return ret; +} + +static int __devexit wm8753_spi_remove(struct spi_device *spi) +{ + return 0; +} + +static struct spi_driver wm8753_spi_driver = { + .driver = { + .name = "wm8753", + .bus = &spi_bus_type, + .owner = THIS_MODULE, + }, + .probe = wm8753_spi_probe, + .remove = __devexit_p(wm8753_spi_remove), +}; + +static int wm8753_spi_write(struct spi_device *spi, const char *data, int len) +{ + struct spi_transfer t; + struct spi_message m; + u8 msg[2]; + + if (len <= 0) + return 0; + + msg[0] = data[0]; + msg[1] = data[1]; + + spi_message_init(&m); + memset(&t, 0, (sizeof t)); + + t.tx_buf = &msg[0]; + t.len = len; + + spi_message_add_tail(&t, &m); + spi_sync(spi, &m); + + return len; +} +#endif + + static int wm8753_probe(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); @@ -1753,8 +1809,14 @@ static int wm8753_probe(struct platform_device *pdev) codec->hw_write = (hw_write_t)i2c_master_send; ret = wm8753_add_i2c_device(pdev, setup); } -#else - /* Add other interfaces here */ +#endif +#if defined(CONFIG_SPI_MASTER) + if (setup->spi) { + codec->hw_write = (hw_write_t)wm8753_spi_write; + ret = spi_register_driver(&wm8753_spi_driver); + if (ret != 0) + printk(KERN_ERR "can't add spi driver"); + } #endif if (ret != 0) { @@ -1798,6 +1860,9 @@ static int wm8753_remove(struct platform_device *pdev) i2c_unregister_device(codec->control_data); i2c_del_driver(&wm8753_i2c_driver); #endif +#if defined(CONFIG_SPI_MASTER) + spi_unregister_driver(&wm8753_spi_driver); +#endif kfree(codec->private_data); kfree(codec); diff --git a/sound/soc/codecs/wm8753.h b/sound/soc/codecs/wm8753.h index 7defde069f1..f55704ce931 100644 --- a/sound/soc/codecs/wm8753.h +++ b/sound/soc/codecs/wm8753.h @@ -2,8 +2,7 @@ * wm8753.h -- audio driver for WM8753 * * Copyright 2003 Wolfson Microelectronics PLC. - * Author: Liam Girdwood - * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com + * Author: Liam Girdwood <lrg@slimlogic.co.uk> * * 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 @@ -79,6 +78,7 @@ #define WM8753_ADCTL2 0x3f struct wm8753_setup_data { + int spi; int i2c_bus; unsigned short i2c_address; }; diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index 0b8c6d38b48..3b326c9b558 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -18,7 +18,6 @@ #include <linux/module.h> #include <linux/moduleparam.h> -#include <linux/version.h> #include <linux/kernel.h> #include <linux/init.h> #include <linux/delay.h> diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index a3f54ec4226..ce40d787760 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -653,14 +653,14 @@ static const struct snd_kcontrol_new wm8903_snd_controls[] = { /* Input PGAs - No TLV since the scale depends on PGA mode */ SOC_SINGLE("Left Input PGA Switch", WM8903_ANALOGUE_LEFT_INPUT_0, - 7, 1, 0), + 7, 1, 1), SOC_SINGLE("Left Input PGA Volume", WM8903_ANALOGUE_LEFT_INPUT_0, 0, 31, 0), SOC_SINGLE("Left Input PGA Common Mode Switch", WM8903_ANALOGUE_LEFT_INPUT_1, 6, 1, 0), SOC_SINGLE("Right Input PGA Switch", WM8903_ANALOGUE_RIGHT_INPUT_0, - 7, 1, 0), + 7, 1, 1), SOC_SINGLE("Right Input PGA Volume", WM8903_ANALOGUE_RIGHT_INPUT_0, 0, 31, 0), SOC_SINGLE("Right Input PGA Common Mode Switch", WM8903_ANALOGUE_RIGHT_INPUT_1, diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index 974a4cd0f3f..f41a578ddd4 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -29,7 +29,6 @@ #include "wm8971.h" -#define AUDIO_NAME "wm8971" #define WM8971_VERSION "0.9" #define WM8971_REG_COUNT 43 diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 63410d7b5ef..572d22b0880 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -30,7 +30,6 @@ #include "wm8990.h" -#define AUDIO_NAME "wm8990" #define WM8990_VERSION "0.2" /* codec private data */ diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 2f1c91b1d55..ffb471e420e 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -2,8 +2,7 @@ * wm9712.c -- ALSA Soc WM9712 codec support * * Copyright 2006 Wolfson Microelectronics PLC. - * Author: Liam Girdwood - * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com + * Author: Liam Girdwood <lrg@slimlogic.co.uk> * * 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 diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 441d0580db1..aba402b3c99 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -2,8 +2,7 @@ * wm9713.c -- ALSA Soc WM9713 codec support * * Copyright 2006 Wolfson Microelectronics PLC. - * Author: Liam Girdwood - * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com + * Author: Liam Girdwood <lrg@slimlogic.co.uk> * * 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 |