From 4dc69be22163bab880384858f30cb8cc76ad47f9 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Tue, 4 Jan 2011 20:16:07 +0530 Subject: ASoC: sst v2: Add sn95031 codec driver This patch adds the sn95031 asoc codec driver. This driver currently supports only playback. Capture and jack detection to be added later Signed-off-by: Vinod Koul Signed-off-by: Harsha Priya Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/sn95031.c | 495 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/sn95031.h | 99 +++++++++ 4 files changed, 600 insertions(+) create mode 100644 sound/soc/codecs/sn95031.c create mode 100644 sound/soc/codecs/sn95031.h (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 054191ed08e..fa42be529a7 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -32,6 +32,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX98088 if I2C select SND_SOC_MAX9877 if I2C select SND_SOC_PCM3008 + select SND_SOC_SN95031 if INTEL_SCU_IPC select SND_SOC_SPDIF select SND_SOC_SSM2602 if I2C select SND_SOC_STAC9766 if SND_SOC_AC97_BUS @@ -173,6 +174,9 @@ config SND_SOC_MAX98088 config SND_SOC_PCM3008 tristate +config SND_SOC_SN95031 + tristate + config SND_SOC_SPDIF tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 6a1e17bbbf8..76304d47891 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -18,6 +18,7 @@ snd-soc-l3-objs := l3.o snd-soc-max98088-objs := max98088.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-alc5623-objs := alc5623.o +snd-soc-sn95031-objs := sn95031.o snd-soc-spdif-objs := spdif_transciever.o snd-soc-ssm2602-objs := ssm2602.o snd-soc-stac9766-objs := stac9766.o @@ -97,6 +98,7 @@ obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_ALC5623) += snd-soc-alc5623.o +obj-$(CONFIG_SND_SOC_SN95031) +=snd-soc-sn95031.o obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c new file mode 100644 index 00000000000..146b74467ae --- /dev/null +++ b/sound/soc/codecs/sn95031.c @@ -0,0 +1,495 @@ +/* + * sn95031.c - TI sn95031 Codec driver + * + * Copyright (C) 2010 Intel Corp + * Author: Vinod Koul + * Author: Harsha Priya + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * + */ +#define pr_fmt(fmt) KBUILD_MODNAME ": " fmt + +#include +#include +#include +#include +#include +#include +#include +#include +#include "sn95031.h" + +#define SN95031_RATES (SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_44100) +#define SN95031_FORMATS (SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE) + +/* + * todo: + * capture paths + * jack detection + * PM functions + */ + +static inline unsigned int sn95031_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 value = 0; + int ret; + + ret = intel_scu_ipc_ioread8(reg, &value); + if (ret) + pr_err("read of %x failed, err %d\n", reg, ret); + return value; + +} + +static inline int sn95031_write(struct snd_soc_codec *codec, + unsigned int reg, unsigned int value) +{ + int ret; + + ret = intel_scu_ipc_iowrite8(reg, value); + if (ret) + pr_err("write of %x failed, err %d\n", reg, ret); + return ret; +} + +static int sn95031_set_vaud_bias(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY) { + pr_debug("vaud_bias powering up pll\n"); + /* power up the pll */ + snd_soc_write(codec, SN95031_AUDPLLCTRL, BIT(5)); + /* enable pcm 2 */ + snd_soc_update_bits(codec, SN95031_PCM2C2, + BIT(0), BIT(0)); + } + break; + + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + pr_debug("vaud_bias power up rail\n"); + /* power up the rail */ + snd_soc_write(codec, SN95031_VAUD, + BIT(2)|BIT(1)|BIT(0)); + msleep(1); + } else if (codec->dapm.bias_level == SND_SOC_BIAS_PREPARE) { + /* turn off pcm */ + pr_debug("vaud_bias power dn pcm\n"); + snd_soc_update_bits(codec, SN95031_PCM2C2, BIT(0), 0); + snd_soc_write(codec, SN95031_AUDPLLCTRL, 0); + } + break; + + + case SND_SOC_BIAS_OFF: + pr_debug("vaud_bias _OFF doing rail shutdown\n"); + snd_soc_write(codec, SN95031_VAUD, BIT(3)); + break; + } + + codec->dapm.bias_level = level; + return 0; +} + +static int sn95031_vhs_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + if (SND_SOC_DAPM_EVENT_ON(event)) { + pr_debug("VHS SND_SOC_DAPM_EVENT_ON doing rail startup now\n"); + /* power up the rail */ + snd_soc_write(w->codec, SN95031_VHSP, 0x3D); + snd_soc_write(w->codec, SN95031_VHSN, 0x3F); + msleep(1); + } else if (SND_SOC_DAPM_EVENT_OFF(event)) { + pr_debug("VHS SND_SOC_DAPM_EVENT_OFF doing rail shutdown\n"); + snd_soc_write(w->codec, SN95031_VHSP, 0xC4); + snd_soc_write(w->codec, SN95031_VHSN, 0x04); + } + return 0; +} + +static int sn95031_vihf_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + if (SND_SOC_DAPM_EVENT_ON(event)) { + pr_debug("VIHF SND_SOC_DAPM_EVENT_ON doing rail startup now\n"); + /* power up the rail */ + snd_soc_write(w->codec, SN95031_VIHF, 0x27); + msleep(1); + } else if (SND_SOC_DAPM_EVENT_OFF(event)) { + pr_debug("VIHF SND_SOC_DAPM_EVENT_OFF doing rail shutdown\n"); + snd_soc_write(w->codec, SN95031_VIHF, 0x24); + } + return 0; +} + +/* DAPM widgets */ +static const struct snd_soc_dapm_widget sn95031_dapm_widgets[] = { + + /* all end points mic, hs etc */ + SND_SOC_DAPM_OUTPUT("HPOUTL"), + SND_SOC_DAPM_OUTPUT("HPOUTR"), + SND_SOC_DAPM_OUTPUT("EPOUT"), + SND_SOC_DAPM_OUTPUT("IHFOUTL"), + SND_SOC_DAPM_OUTPUT("IHFOUTR"), + SND_SOC_DAPM_OUTPUT("LINEOUTL"), + SND_SOC_DAPM_OUTPUT("LINEOUTR"), + SND_SOC_DAPM_OUTPUT("VIB1OUT"), + SND_SOC_DAPM_OUTPUT("VIB2OUT"), + + SND_SOC_DAPM_SUPPLY("Headset Rail", SND_SOC_NOPM, 0, 0, + sn95031_vhs_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY("Speaker Rail", SND_SOC_NOPM, 0, 0, + sn95031_vihf_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + + /* playback path driver enables */ + SND_SOC_DAPM_PGA("Headset Left Playback", + SN95031_DRIVEREN, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Headset Right Playback", + SN95031_DRIVEREN, 1, 0, NULL, 0), + SND_SOC_DAPM_PGA("Speaker Left Playback", + SN95031_DRIVEREN, 2, 0, NULL, 0), + SND_SOC_DAPM_PGA("Speaker Right Playback", + SN95031_DRIVEREN, 3, 0, NULL, 0), + SND_SOC_DAPM_PGA("Vibra1 Playback", + SN95031_DRIVEREN, 4, 0, NULL, 0), + SND_SOC_DAPM_PGA("Vibra2 Playback", + SN95031_DRIVEREN, 5, 0, NULL, 0), + SND_SOC_DAPM_PGA("Earpiece Playback", + SN95031_DRIVEREN, 6, 0, NULL, 0), + SND_SOC_DAPM_PGA("Lineout Left Playback", + SN95031_LOCTL, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Lineout Right Playback", + SN95031_LOCTL, 4, 0, NULL, 0), + + /* playback path filter enable */ + SND_SOC_DAPM_PGA("Headset Left Filter", + SN95031_HSEPRXCTRL, 4, 0, NULL, 0), + SND_SOC_DAPM_PGA("Headset Right Filter", + SN95031_HSEPRXCTRL, 5, 0, NULL, 0), + SND_SOC_DAPM_PGA("Speaker Left Filter", + SN95031_IHFRXCTRL, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Speaker Right Filter", + SN95031_IHFRXCTRL, 1, 0, NULL, 0), + + /* DACs */ + SND_SOC_DAPM_DAC("HSDAC Left", "Headset", + SN95031_DACCONFIG, 0, 0), + SND_SOC_DAPM_DAC("HSDAC Right", "Headset", + SN95031_DACCONFIG, 1, 0), + SND_SOC_DAPM_DAC("IHFDAC Left", "Speaker", + SN95031_DACCONFIG, 2, 0), + SND_SOC_DAPM_DAC("IHFDAC Right", "Speaker", + SN95031_DACCONFIG, 3, 0), + SND_SOC_DAPM_DAC("Vibra1 DAC", "Vibra1", + SN95031_VIB1C5, 1, 0), + SND_SOC_DAPM_DAC("Vibra2 DAC", "Vibra2", + SN95031_VIB2C5, 1, 0), +}; + +static const struct snd_soc_dapm_route sn95031_audio_map[] = { + /* headset and earpiece map */ + { "HPOUTL", NULL, "Headset Left Playback" }, + { "HPOUTR", NULL, "Headset Right Playback" }, + { "EPOUT", NULL, "Earpiece Playback" }, + { "Headset Left Playback", NULL, "Headset Left Filter"}, + { "Headset Right Playback", NULL, "Headset Right Filter"}, + { "Earpiece Playback", NULL, "Headset Left Filter"}, + { "Headset Left Filter", NULL, "HSDAC Left"}, + { "Headset Right Filter", NULL, "HSDAC Right"}, + { "HSDAC Left", NULL, "Headset Rail"}, + { "HSDAC Right", NULL, "Headset Rail"}, + + /* speaker map */ + { "IHFOUTL", "NULL", "Speaker Left Playback"}, + { "IHFOUTR", "NULL", "Speaker Right Playback"}, + { "Speaker Left Playback", NULL, "Speaker Left Filter"}, + { "Speaker Right Playback", NULL, "Speaker Right Filter"}, + { "Speaker Left Filter", NULL, "IHFDAC Left"}, + { "Speaker Right Filter", NULL, "IHFDAC Right"}, + { "IHFDAC Left", NULL, "Speaker Rail"}, + { "IHFDAC Right", NULL, "Speaker Rail"}, + + /* vibra map */ + {"VIB1OUT", NULL, "Vibra1 Playback"}, + {"Vibra1 Playback", NULL, "Vibra1 DAC"}, + + {"VIB2OUT", NULL, "Vibra2 Playback"}, + {"Vibra2 Playback", NULL, "Vibra2 DAC"}, + + /* lineout */ + {"LINEOUTL", NULL, "Lineout Left Playback"}, + {"LINEOUTR", NULL, "Lineout Right Playback"}, + {"Lineout Left Playback", NULL, "Headset Left Filter"}, + {"Lineout Left Playback", NULL, "Speaker Left Filter"}, + {"Lineout Left Playback", NULL, "Vibra1 DAC"}, + {"Lineout Right Playback", NULL, "Headset Right Filter"}, + {"Lineout Right Playback", NULL, "Speaker Right Filter"}, + {"Lineout Right Playback", NULL, "Vibra2 DAC"}, +}; + +/* speaker and headset mutes, for audio pops and clicks */ +static int sn95031_pcm_hs_mute(struct snd_soc_dai *dai, int mute) +{ + snd_soc_update_bits(dai->codec, + SN95031_HSLVOLCTRL, BIT(7), (!mute << 7)); + snd_soc_update_bits(dai->codec, + SN95031_HSRVOLCTRL, BIT(7), (!mute << 7)); + return 0; +} + +static int sn95031_pcm_spkr_mute(struct snd_soc_dai *dai, int mute) +{ + snd_soc_update_bits(dai->codec, + SN95031_IHFLVOLCTRL, BIT(7), (!mute << 7)); + snd_soc_update_bits(dai->codec, + SN95031_IHFRVOLCTRL, BIT(7), (!mute << 7)); + return 0; +} + +int sn95031_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + unsigned int format, rate; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + format = BIT(4)|BIT(5); + break; + + case SNDRV_PCM_FORMAT_S24_LE: + format = 0; + break; + default: + return -EINVAL; + } + snd_soc_update_bits(dai->codec, SN95031_PCM2C2, + BIT(4)|BIT(5), format); + + switch (params_rate(params)) { + case 48000: + pr_debug("RATE_48000\n"); + rate = 0; + break; + + case 44100: + pr_debug("RATE_44100\n"); + rate = BIT(7); + break; + + default: + pr_err("ERR rate %d\n", params_rate(params)); + return -EINVAL; + } + snd_soc_update_bits(dai->codec, SN95031_PCM1C1, BIT(7), rate); + + return 0; +} + +/* Codec DAI section */ +static struct snd_soc_dai_ops sn95031_headset_dai_ops = { + .digital_mute = sn95031_pcm_hs_mute, + .hw_params = sn95031_pcm_hw_params, +}; + +static struct snd_soc_dai_ops sn95031_speaker_dai_ops = { + .digital_mute = sn95031_pcm_spkr_mute, + .hw_params = sn95031_pcm_hw_params, +}; + +static struct snd_soc_dai_ops sn95031_vib1_dai_ops = { + .hw_params = sn95031_pcm_hw_params, +}; + +static struct snd_soc_dai_ops sn95031_vib2_dai_ops = { + .hw_params = sn95031_pcm_hw_params, +}; + +struct snd_soc_dai_driver sn95031_dais[] = { +{ + .name = "SN95031 Headset", + .playback = { + .stream_name = "Headset", + .channels_min = 2, + .channels_max = 2, + .rates = SN95031_RATES, + .formats = SN95031_FORMATS, + }, + .ops = &sn95031_headset_dai_ops, +}, +{ .name = "SN95031 Speaker", + .playback = { + .stream_name = "Speaker", + .channels_min = 2, + .channels_max = 2, + .rates = SN95031_RATES, + .formats = SN95031_FORMATS, + }, + .ops = &sn95031_speaker_dai_ops, +}, +{ .name = "SN95031 Vibra1", + .playback = { + .stream_name = "Vibra1", + .channels_min = 1, + .channels_max = 1, + .rates = SN95031_RATES, + .formats = SN95031_FORMATS, + }, + .ops = &sn95031_vib1_dai_ops, +}, +{ .name = "SN95031 Vibra2", + .playback = { + .stream_name = "Vibra2", + .channels_min = 1, + .channels_max = 1, + .rates = SN95031_RATES, + .formats = SN95031_FORMATS, + }, + .ops = &sn95031_vib2_dai_ops, +}, +}; + +/* codec registration */ +static int sn95031_codec_probe(struct snd_soc_codec *codec) +{ + int ret; + + pr_debug("codec_probe called\n"); + + codec->dapm.bias_level = SND_SOC_BIAS_OFF; + codec->dapm.idle_bias_off = 1; + + /* PCM interface config + * This sets the pcm rx slot conguration to max 6 slots + * for max 4 dais (2 stereo and 2 mono) + */ + snd_soc_write(codec, SN95031_PCM2RXSLOT01, 0x10); + snd_soc_write(codec, SN95031_PCM2RXSLOT23, 0x32); + snd_soc_write(codec, SN95031_PCM2RXSLOT45, 0x54); + /* pcm port setting + * This sets the pcm port to slave and clock at 19.2Mhz which + * can support 6slots, sampling rate set per stream in hw-params + */ + snd_soc_write(codec, SN95031_PCM1C1, 0x00); + snd_soc_write(codec, SN95031_PCM2C1, 0x01); + snd_soc_write(codec, SN95031_PCM2C2, 0x0A); + snd_soc_write(codec, SN95031_HSMIXER, BIT(0)|BIT(4)); + /* vendor vibra workround, the vibras are muted by + * custom register so unmute them + */ + snd_soc_write(codec, SN95031_SSR5, 0x80); + snd_soc_write(codec, SN95031_SSR6, 0x80); + snd_soc_write(codec, SN95031_VIB1C5, 0x00); + snd_soc_write(codec, SN95031_VIB2C5, 0x00); + /* configure vibras for pcm port */ + snd_soc_write(codec, SN95031_VIB1C3, 0x00); + snd_soc_write(codec, SN95031_VIB2C3, 0x00); + + /* soft mute ramp time */ + snd_soc_write(codec, SN95031_SOFTMUTE, 0x3); + /* fix the initial volume at 1dB, + * default in +9dB, + * 1dB give optimal swing on DAC, amps + */ + snd_soc_write(codec, SN95031_HSLVOLCTRL, 0x08); + snd_soc_write(codec, SN95031_HSRVOLCTRL, 0x08); + snd_soc_write(codec, SN95031_IHFLVOLCTRL, 0x08); + snd_soc_write(codec, SN95031_IHFRVOLCTRL, 0x08); + /* dac mode and lineout workaround */ + snd_soc_write(codec, SN95031_SSR2, 0x10); + snd_soc_write(codec, SN95031_SSR3, 0x40); + + ret = snd_soc_dapm_new_controls(&codec->dapm, sn95031_dapm_widgets, + ARRAY_SIZE(sn95031_dapm_widgets)); + if (ret) + pr_err("soc_dapm_new_control failed %d", ret); + ret = snd_soc_dapm_add_routes(&codec->dapm, sn95031_audio_map, + ARRAY_SIZE(sn95031_audio_map)); + if (ret) + pr_err("soc_dapm_add_routes failed %d", ret); + + return ret; +} + +static int sn95031_codec_remove(struct snd_soc_codec *codec) +{ + pr_debug("codec_remove called\n"); + sn95031_set_vaud_bias(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +struct snd_soc_codec_driver sn95031_codec = { + .probe = sn95031_codec_probe, + .remove = sn95031_codec_remove, + .read = sn95031_read, + .write = sn95031_write, + .set_bias_level = sn95031_set_vaud_bias, +}; + +static int __devinit sn95031_device_probe(struct platform_device *pdev) +{ + pr_debug("codec device probe called for %s\n", dev_name(&pdev->dev)); + return snd_soc_register_codec(&pdev->dev, &sn95031_codec, + sn95031_dais, ARRAY_SIZE(sn95031_dais)); +} + +static int __devexit sn95031_device_remove(struct platform_device *pdev) +{ + pr_debug("codec device remove called\n"); + snd_soc_unregister_codec(&pdev->dev); + return 0; +} + +static struct platform_driver sn95031_codec_driver = { + .driver = { + .name = "sn95031", + .owner = THIS_MODULE, + }, + .probe = sn95031_device_probe, + .remove = sn95031_device_remove, +}; + +static int __init sn95031_init(void) +{ + pr_debug("driver init called\n"); + return platform_driver_register(&sn95031_codec_driver); +} +module_init(sn95031_init); + +static void __exit sn95031_exit(void) +{ + pr_debug("driver exit called\n"); + platform_driver_unregister(&sn95031_codec_driver); +} +module_exit(sn95031_exit); + +MODULE_DESCRIPTION("ASoC Intel(R) SN95031 codec driver"); +MODULE_AUTHOR("Vinod Koul "); +MODULE_AUTHOR("Harsha Priya "); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:sn95031"); diff --git a/sound/soc/codecs/sn95031.h b/sound/soc/codecs/sn95031.h new file mode 100644 index 00000000000..b17a39b2aef --- /dev/null +++ b/sound/soc/codecs/sn95031.h @@ -0,0 +1,99 @@ +/* + * sn95031.h - TI sn95031 Codec driver + * + * Copyright (C) 2010 Intel Corp + * Author: Vinod Koul + * Author: Harsha Priya + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * + */ +#ifndef _SN95031_H +#define _SN95031_H + +/*register map*/ +#define SN95031_VAUD 0xDB +#define SN95031_VHSP 0xDC +#define SN95031_VHSN 0xDD +#define SN95031_VIHF 0xC9 + +#define SN95031_AUDPLLCTRL 0x240 +#define SN95031_DMICBUF0123 0x241 +#define SN95031_DMICBUF45 0x242 +#define SN95031_DMICGPO 0x244 +#define SN95031_DMICMUX 0x245 +#define SN95031_DMICLK 0x246 +#define SN95031_MICBIAS 0x247 +#define SN95031_ADCCONFIG 0x248 +#define SN95031_MICAMP1 0x249 +#define SN95031_MICAMP2 0x24A +#define SN95031_NOISEMUX 0x24B +#define SN95031_AUDIOMUX12 0x24C +#define SN95031_AUDIOMUX34 0x24D +#define SN95031_AUDIOSINC 0x24E +#define SN95031_AUDIOTXEN 0x24F +#define SN95031_HSEPRXCTRL 0x250 +#define SN95031_IHFRXCTRL 0x251 +#define SN95031_HSMIXER 0x256 +#define SN95031_DACCONFIG 0x257 +#define SN95031_SOFTMUTE 0x258 +#define SN95031_HSLVOLCTRL 0x259 +#define SN95031_HSRVOLCTRL 0x25A +#define SN95031_IHFLVOLCTRL 0x25B +#define SN95031_IHFRVOLCTRL 0x25C +#define SN95031_DRIVEREN 0x25D +#define SN95031_LOCTL 0x25E +#define SN95031_VIB1C1 0x25F +#define SN95031_VIB1C2 0x260 +#define SN95031_VIB1C3 0x261 +#define SN95031_VIB1SPIPCM1 0x262 +#define SN95031_VIB1SPIPCM2 0x263 +#define SN95031_VIB1C5 0x264 +#define SN95031_VIB2C1 0x265 +#define SN95031_VIB2C2 0x266 +#define SN95031_VIB2C3 0x267 +#define SN95031_VIB2SPIPCM1 0x268 +#define SN95031_VIB2SPIPCM2 0x269 +#define SN95031_VIB2C5 0x26A +#define SN95031_BTNCTRL1 0x26B +#define SN95031_BTNCTRL2 0x26C +#define SN95031_PCM1TXSLOT01 0x26D +#define SN95031_PCM1TXSLOT23 0x26E +#define SN95031_PCM1TXSLOT45 0x26F +#define SN95031_PCM1RXSLOT0_3 0x270 +#define SN95031_PCM1RXSLOT45 0x271 +#define SN95031_PCM2TXSLOT01 0x272 +#define SN95031_PCM2TXSLOT23 0x273 +#define SN95031_PCM2TXSLOT45 0x274 +#define SN95031_PCM2RXSLOT01 0x275 +#define SN95031_PCM2RXSLOT23 0x276 +#define SN95031_PCM2RXSLOT45 0x277 +#define SN95031_PCM1C1 0x278 +#define SN95031_PCM1C2 0x279 +#define SN95031_PCM1C3 0x27A +#define SN95031_PCM2C1 0x27B +#define SN95031_PCM2C2 0x27C +/*end codec register defn*/ + +/*vendor defn these are not part of avp*/ +#define SN95031_SSR2 0x381 +#define SN95031_SSR3 0x382 +#define SN95031_SSR5 0x384 +#define SN95031_SSR6 0x385 + +#endif -- cgit v1.2.3-70-g09d2 From b22dab8883250d5fd91c8c4a7bd00c5985ac82ad Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Fri, 7 Jan 2011 16:20:28 +0530 Subject: ASoC: sn95031 fix the code style and format inconsistencies this patch fixes inconsistencies commented by Mark. This also fixes few other style things in audio_map & header file Signed-off-by: Vinod Koul Signed-off-by: Harsha Priya Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/sn95031.c | 30 +++++++++++++++--------------- sound/soc/codecs/sn95031.h | 12 ++++++------ 2 files changed, 21 insertions(+), 21 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index 146b74467ae..593632cf791 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -235,21 +235,21 @@ static const struct snd_soc_dapm_route sn95031_audio_map[] = { { "IHFDAC Right", NULL, "Speaker Rail"}, /* vibra map */ - {"VIB1OUT", NULL, "Vibra1 Playback"}, - {"Vibra1 Playback", NULL, "Vibra1 DAC"}, + { "VIB1OUT", NULL, "Vibra1 Playback"}, + { "Vibra1 Playback", NULL, "Vibra1 DAC"}, - {"VIB2OUT", NULL, "Vibra2 Playback"}, - {"Vibra2 Playback", NULL, "Vibra2 DAC"}, + { "VIB2OUT", NULL, "Vibra2 Playback"}, + { "Vibra2 Playback", NULL, "Vibra2 DAC"}, /* lineout */ - {"LINEOUTL", NULL, "Lineout Left Playback"}, - {"LINEOUTR", NULL, "Lineout Right Playback"}, - {"Lineout Left Playback", NULL, "Headset Left Filter"}, - {"Lineout Left Playback", NULL, "Speaker Left Filter"}, - {"Lineout Left Playback", NULL, "Vibra1 DAC"}, - {"Lineout Right Playback", NULL, "Headset Right Filter"}, - {"Lineout Right Playback", NULL, "Speaker Right Filter"}, - {"Lineout Right Playback", NULL, "Vibra2 DAC"}, + { "LINEOUTL", NULL, "Lineout Left Playback"}, + { "LINEOUTR", NULL, "Lineout Right Playback"}, + { "Lineout Left Playback", NULL, "Headset Left Filter"}, + { "Lineout Left Playback", NULL, "Speaker Left Filter"}, + { "Lineout Left Playback", NULL, "Vibra1 DAC"}, + { "Lineout Right Playback", NULL, "Headset Right Filter"}, + { "Lineout Right Playback", NULL, "Speaker Right Filter"}, + { "Lineout Right Playback", NULL, "Vibra2 DAC"}, }; /* speaker and headset mutes, for audio pops and clicks */ @@ -444,8 +444,8 @@ static int sn95031_codec_remove(struct snd_soc_codec *codec) } struct snd_soc_codec_driver sn95031_codec = { - .probe = sn95031_codec_probe, - .remove = sn95031_codec_remove, + .probe = sn95031_codec_probe, + .remove = sn95031_codec_remove, .read = sn95031_read, .write = sn95031_write, .set_bias_level = sn95031_set_vaud_bias, @@ -488,7 +488,7 @@ static void __exit sn95031_exit(void) } module_exit(sn95031_exit); -MODULE_DESCRIPTION("ASoC Intel(R) SN95031 codec driver"); +MODULE_DESCRIPTION("ASoC TI SN95031 codec driver"); MODULE_AUTHOR("Vinod Koul "); MODULE_AUTHOR("Harsha Priya "); MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/sn95031.h b/sound/soc/codecs/sn95031.h index b17a39b2aef..e2b17d908ae 100644 --- a/sound/soc/codecs/sn95031.h +++ b/sound/soc/codecs/sn95031.h @@ -35,13 +35,13 @@ #define SN95031_AUDPLLCTRL 0x240 #define SN95031_DMICBUF0123 0x241 #define SN95031_DMICBUF45 0x242 -#define SN95031_DMICGPO 0x244 -#define SN95031_DMICMUX 0x245 +#define SN95031_DMICGPO 0x244 +#define SN95031_DMICMUX 0x245 #define SN95031_DMICLK 0x246 -#define SN95031_MICBIAS 0x247 +#define SN95031_MICBIAS 0x247 #define SN95031_ADCCONFIG 0x248 -#define SN95031_MICAMP1 0x249 -#define SN95031_MICAMP2 0x24A +#define SN95031_MICAMP1 0x249 +#define SN95031_MICAMP2 0x24A #define SN95031_NOISEMUX 0x24B #define SN95031_AUDIOMUX12 0x24C #define SN95031_AUDIOMUX34 0x24D @@ -49,7 +49,7 @@ #define SN95031_AUDIOTXEN 0x24F #define SN95031_HSEPRXCTRL 0x250 #define SN95031_IHFRXCTRL 0x251 -#define SN95031_HSMIXER 0x256 +#define SN95031_HSMIXER 0x256 #define SN95031_DACCONFIG 0x257 #define SN95031_SOFTMUTE 0x258 #define SN95031_HSLVOLCTRL 0x259 -- cgit v1.2.3-70-g09d2 From 4fde768ecf39360eaed44152c2556dd10c3e0ce7 Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Tue, 11 Jan 2011 09:28:32 +0000 Subject: ASoC: WM8995: Remember to flush the cache on resume Signed-off-by: Dimitris Papastamos Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/wm8995.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index 3d2110c1d81..96e0dc09f20 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -1498,6 +1498,7 @@ static int wm8995_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_OFF: snd_soc_update_bits(codec, WM8995_POWER_MANAGEMENT_1, WM8995_BG_ENA_MASK, 0); + codec->cache_sync = 1; break; } -- cgit v1.2.3-70-g09d2 From d4754ec91c7b094298f0b2ba02327e6887671edc Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Thu, 13 Jan 2011 12:20:37 +0000 Subject: ASoC: Update users of readable_register()/volatile_register() Signed-off-by: Dimitris Papastamos Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- include/sound/soc.h | 4 ++-- sound/soc/codecs/cs4270.c | 4 ++-- sound/soc/codecs/max98088.c | 2 +- sound/soc/codecs/wm8523.c | 2 +- sound/soc/codecs/wm8804.c | 2 +- sound/soc/codecs/wm8900.c | 2 +- sound/soc/codecs/wm8903.c | 2 +- sound/soc/codecs/wm8904.c | 2 +- sound/soc/codecs/wm8961.c | 2 +- sound/soc/codecs/wm8962.c | 4 ++-- sound/soc/codecs/wm8993.c | 2 +- sound/soc/codecs/wm8994.c | 10 +++++----- sound/soc/codecs/wm8995.c | 2 +- sound/soc/codecs/wm9081.c | 2 +- sound/soc/codecs/wm9090.c | 4 ++-- sound/soc/soc-core.c | 4 ++-- 16 files changed, 25 insertions(+), 25 deletions(-) (limited to 'sound/soc/codecs') diff --git a/include/sound/soc.h b/include/sound/soc.h index b8acf99ac89..97d1832bb9d 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -529,8 +529,8 @@ struct snd_soc_codec_driver { int (*write)(struct snd_soc_codec *, unsigned int, unsigned int); int (*display_register)(struct snd_soc_codec *, char *, size_t, unsigned int); - int (*volatile_register)(unsigned int); - int (*readable_register)(unsigned int); + int (*volatile_register)(struct snd_soc_codec *, unsigned int); + int (*readable_register)(struct snd_soc_codec *, unsigned int); short reg_cache_size; short reg_cache_step; short reg_word_size; diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 8b51245f231..c0fccadaea9 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -193,12 +193,12 @@ static struct cs4270_mode_ratios cs4270_mode_ratios[] = { /* The number of MCLK/LRCK ratios supported by the CS4270 */ #define NUM_MCLK_RATIOS ARRAY_SIZE(cs4270_mode_ratios) -static int cs4270_reg_is_readable(unsigned int reg) +static int cs4270_reg_is_readable(struct snd_soc_codec *codec, unsigned int reg) { return (reg >= CS4270_FIRSTREG) && (reg <= CS4270_LASTREG); } -static int cs4270_reg_is_volatile(unsigned int reg) +static int cs4270_reg_is_volatile(struct snd_soc_codec *codec, unsigned int reg) { /* Unreadable registers are considered volatile */ if ((reg < CS4270_FIRSTREG) || (reg > CS4270_LASTREG)) diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 37133c40e76..b6ecc7e8967 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -608,7 +608,7 @@ static struct { { 0xFF, 0x00, 1 }, /* FF */ }; -static int max98088_volatile_register(unsigned int reg) +static int max98088_volatile_register(struct snd_soc_codec *codec, unsigned int reg) { return max98088_access[reg].vol; } diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c index 5eb2f501ce3..83e86f077ee 100644 --- a/sound/soc/codecs/wm8523.c +++ b/sound/soc/codecs/wm8523.c @@ -58,7 +58,7 @@ static const u16 wm8523_reg[WM8523_REGISTER_COUNT] = { 0x0000, /* R8 - ZERO_DETECT */ }; -static int wm8523_volatile_register(unsigned int reg) +static int wm8523_volatile_register(struct snd_soc_codec *codec, unsigned int reg) { switch (reg) { case WM8523_DEVICE_ID: diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index 6dae1b40c9f..6785688f880 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -175,7 +175,7 @@ static int txsrc_put(struct snd_kcontrol *kcontrol, return 0; } -static int wm8804_volatile(unsigned int reg) +static int wm8804_volatile(struct snd_soc_codec *codec, unsigned int reg) { switch (reg) { case WM8804_RST_DEVID1: diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index cd0959926d1..449ea09a193 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -180,7 +180,7 @@ static const u16 wm8900_reg_defaults[WM8900_MAXREG] = { /* Remaining registers all zero */ }; -static int wm8900_volatile_register(unsigned int reg) +static int wm8900_volatile_register(struct snd_soc_codec *codec, unsigned int reg) { switch (reg) { case WM8900_REG_ID: diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 987476a5895..a2a446cb180 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -232,7 +232,7 @@ struct wm8903_priv { int mic_delay; }; -static int wm8903_volatile_register(unsigned int reg) +static int wm8903_volatile_register(struct snd_soc_codec *codec, unsigned int reg) { switch (reg) { case WM8903_SW_RESET_AND_ID: diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 9de44a4c05c..17a8fe9b39b 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -596,7 +596,7 @@ static struct { { 0x003F, 0x003F, 0 }, /* R248 - FLL NCO Test 1 */ }; -static int wm8904_volatile_register(unsigned int reg) +static int wm8904_volatile_register(struct snd_soc_codec *codec, unsigned int reg) { return wm8904_access[reg].vol; } diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index 55252e7d02c..cdee8103d09 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -291,7 +291,7 @@ struct wm8961_priv { int sysclk; }; -static int wm8961_volatile_register(unsigned int reg) +static int wm8961_volatile_register(struct snd_soc_codec *codec, unsigned int reg) { switch (reg) { case WM8961_SOFTWARE_RESET: diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index b9cb1fcf8c9..7c02924bedd 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -1938,7 +1938,7 @@ static const struct wm8962_reg_access { [21139] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21139 - VSS_XTS32_0 */ }; -static int wm8962_volatile_register(unsigned int reg) +static int wm8962_volatile_register(struct snd_soc_codec *codec, unsigned int reg) { if (wm8962_reg_access[reg].vol) return 1; @@ -1946,7 +1946,7 @@ static int wm8962_volatile_register(unsigned int reg) return 0; } -static int wm8962_readable_register(unsigned int reg) +static int wm8962_readable_register(struct snd_soc_codec *codec, unsigned int reg) { if (wm8962_reg_access[reg].read) return 1; diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 18c0d9ce7c3..379fa22c5b6 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -242,7 +242,7 @@ struct wm8993_priv { int fll_src; }; -static int wm8993_volatile(unsigned int reg) +static int wm8993_volatile(struct snd_soc_codec *codec, unsigned int reg) { switch (reg) { case WM8993_SOFTWARE_RESET: diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 247a6a99feb..0bb0bb40b84 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -109,7 +109,7 @@ struct wm8994_priv { struct wm8994_pdata *pdata; }; -static int wm8994_readable(unsigned int reg) +static int wm8994_readable(struct snd_soc_codec *codec, unsigned int reg) { switch (reg) { case WM8994_GPIO_1: @@ -136,7 +136,7 @@ static int wm8994_readable(unsigned int reg) return wm8994_access_masks[reg].readable != 0; } -static int wm8994_volatile(unsigned int reg) +static int wm8994_volatile(struct snd_soc_codec *codec, unsigned int reg) { if (reg >= WM8994_CACHE_SIZE) return 1; @@ -164,7 +164,7 @@ static int wm8994_write(struct snd_soc_codec *codec, unsigned int reg, BUG_ON(reg > WM8994_MAX_REGISTER); - if (!wm8994_volatile(reg)) { + if (!wm8994_volatile(codec, reg)) { ret = snd_soc_cache_write(codec, reg, value); if (ret != 0) dev_err(codec->dev, "Cache write to %x failed: %d\n", @@ -182,7 +182,7 @@ static unsigned int wm8994_read(struct snd_soc_codec *codec, BUG_ON(reg > WM8994_MAX_REGISTER); - if (!wm8994_volatile(reg) && wm8994_readable(reg) && + if (!wm8994_volatile(codec, reg) && wm8994_readable(codec, reg) && reg < codec->driver->reg_cache_size) { ret = snd_soc_cache_read(codec, reg, &val); if (ret >= 0) @@ -2943,7 +2943,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) /* Read our current status back from the chip - we don't want to * reset as this may interfere with the GPIO or LDO operation. */ for (i = 0; i < WM8994_CACHE_SIZE; i++) { - if (!wm8994_readable(i) || wm8994_volatile(i)) + if (!wm8994_readable(codec, i) || wm8994_volatile(codec, i)) continue; ret = wm8994_reg_read(codec->control_data, i); diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index ac210ccebd4..f0f678de489 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -909,7 +909,7 @@ static const struct snd_soc_dapm_route wm8995_intercon[] = { { "SPK2R", NULL, "SPK2R Driver" } }; -static int wm8995_volatile(unsigned int reg) +static int wm8995_volatile(struct snd_soc_codec *codec, unsigned int reg) { /* out of bounds registers are generally considered * volatile to support register banks that are partially diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 43825b2102a..5c224dd917d 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -169,7 +169,7 @@ struct wm9081_priv { struct wm9081_retune_mobile_config *retune; }; -static int wm9081_volatile_register(unsigned int reg) +static int wm9081_volatile_register(struct snd_soc_codec *codec, unsigned int reg) { switch (reg) { case WM9081_SOFTWARE_RESET: diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c index a788c429704..d40bfc9f880 100644 --- a/sound/soc/codecs/wm9090.c +++ b/sound/soc/codecs/wm9090.c @@ -144,7 +144,7 @@ struct wm9090_priv { void *control_data; }; -static int wm9090_volatile(unsigned int reg) +static int wm9090_volatile(struct snd_soc_codec *codec, unsigned int reg) { switch (reg) { case WM9090_SOFTWARE_RESET: @@ -518,7 +518,7 @@ static int wm9090_set_bias_level(struct snd_soc_codec *codec, for (i = 1; i < codec->driver->reg_cache_size; i++) { if (reg_cache[i] == wm9090_reg_defaults[i]) continue; - if (wm9090_volatile(i)) + if (wm9090_volatile(codec, i)) continue; ret = snd_soc_write(codec, i, reg_cache[i]); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index cbac50b69c3..b5e5758456b 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -84,7 +84,7 @@ static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf) count += sprintf(buf, "%s registers\n", codec->name); for (i = 0; i < codec->driver->reg_cache_size; i += step) { - if (codec->driver->readable_register && !codec->driver->readable_register(i)) + if (codec->driver->readable_register && !codec->driver->readable_register(codec, i)) continue; count += sprintf(buf + count, "%2x: ", i); @@ -2030,7 +2030,7 @@ static int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) int snd_soc_codec_volatile_register(struct snd_soc_codec *codec, int reg) { if (codec->driver->volatile_register) - return codec->driver->volatile_register(reg); + return codec->driver->volatile_register(codec, reg); else return 0; } -- cgit v1.2.3-70-g09d2 From 203db220718c735dcb959fddc64e94fff3b52f73 Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Fri, 14 Jan 2011 15:49:40 +0000 Subject: ASoC: WM8991: Add initial WM8991 driver The WM8991 is a highly integrated ultra-low power hi-fi CODEC designed for handsets rich in multimedia features such as GPS, mobile TV, digital audio playback and gaming. This driver was originally written by Graeme Gregory and has been maintained out of tree by Mark Brown and Dimitris Papastamos. Signed-off-by: Graeme Gregory Signed-off-by: Dimitris Papastamos Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wm8991.c | 1427 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8991.h | 833 ++++++++++++++++++++++++++ 4 files changed, 2266 insertions(+) create mode 100644 sound/soc/codecs/wm8991.c create mode 100644 sound/soc/codecs/wm8991.h (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 61e36efbf27..a18cff4afbc 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -77,6 +77,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM8985 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8988 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8990 if I2C + select SND_SOC_WM8991 if I2C select SND_SOC_WM8993 if I2C select SND_SOC_WM8994 if MFD_WM8994 select SND_SOC_WM8995 if SND_SOC_I2C_AND_SPI @@ -308,6 +309,9 @@ config SND_SOC_WM8988 config SND_SOC_WM8990 tristate +config SND_SOC_WM8991 + tristate + config SND_SOC_WM8993 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 333910a9f8f..68e76af894b 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -62,6 +62,7 @@ snd-soc-wm8978-objs := wm8978.o snd-soc-wm8985-objs := wm8985.o snd-soc-wm8988-objs := wm8988.o snd-soc-wm8990-objs := wm8990.o +snd-soc-wm8991-objs := wm8991.o snd-soc-wm8993-objs := wm8993.o snd-soc-wm8994-objs := wm8994.o wm8994-tables.o snd-soc-wm8995-objs := wm8995.o @@ -143,6 +144,7 @@ obj-$(CONFIG_SND_SOC_WM8978) += snd-soc-wm8978.o obj-$(CONFIG_SND_SOC_WM8985) += snd-soc-wm8985.o obj-$(CONFIG_SND_SOC_WM8988) += snd-soc-wm8988.o obj-$(CONFIG_SND_SOC_WM8990) += snd-soc-wm8990.o +obj-$(CONFIG_SND_SOC_WM8991) += snd-soc-wm8991.o obj-$(CONFIG_SND_SOC_WM8993) += snd-soc-wm8993.o obj-$(CONFIG_SND_SOC_WM8994) += snd-soc-wm8994.o obj-$(CONFIG_SND_SOC_WM8995) += snd-soc-wm8995.o diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c new file mode 100644 index 00000000000..28fdfd66661 --- /dev/null +++ b/sound/soc/codecs/wm8991.c @@ -0,0 +1,1427 @@ +/* + * wm8991.c -- WM8991 ALSA Soc Audio driver + * + * Copyright 2007-2010 Wolfson Microelectronics PLC. + * Author: Graeme Gregory + * linux@wolfsonmicro.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. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "wm8991.h" + +struct wm8991_priv { + enum snd_soc_control_type control_type; + unsigned int pcmclk; +}; + +static const u16 wm8991_reg_defs[] = { + 0x8991, /* R0 - Reset */ + 0x0000, /* R1 - Power Management (1) */ + 0x6000, /* R2 - Power Management (2) */ + 0x0000, /* R3 - Power Management (3) */ + 0x4050, /* R4 - Audio Interface (1) */ + 0x4000, /* R5 - Audio Interface (2) */ + 0x01C8, /* R6 - Clocking (1) */ + 0x0000, /* R7 - Clocking (2) */ + 0x0040, /* R8 - Audio Interface (3) */ + 0x0040, /* R9 - Audio Interface (4) */ + 0x0004, /* R10 - DAC CTRL */ + 0x00C0, /* R11 - Left DAC Digital Volume */ + 0x00C0, /* R12 - Right DAC Digital Volume */ + 0x0000, /* R13 - Digital Side Tone */ + 0x0100, /* R14 - ADC CTRL */ + 0x00C0, /* R15 - Left ADC Digital Volume */ + 0x00C0, /* R16 - Right ADC Digital Volume */ + 0x0000, /* R17 */ + 0x0000, /* R18 - GPIO CTRL 1 */ + 0x1000, /* R19 - GPIO1 & GPIO2 */ + 0x1010, /* R20 - GPIO3 & GPIO4 */ + 0x1010, /* R21 - GPIO5 & GPIO6 */ + 0x8000, /* R22 - GPIOCTRL 2 */ + 0x0800, /* R23 - GPIO_POL */ + 0x008B, /* R24 - Left Line Input 1&2 Volume */ + 0x008B, /* R25 - Left Line Input 3&4 Volume */ + 0x008B, /* R26 - Right Line Input 1&2 Volume */ + 0x008B, /* R27 - Right Line Input 3&4 Volume */ + 0x0000, /* R28 - Left Output Volume */ + 0x0000, /* R29 - Right Output Volume */ + 0x0066, /* R30 - Line Outputs Volume */ + 0x0022, /* R31 - Out3/4 Volume */ + 0x0079, /* R32 - Left OPGA Volume */ + 0x0079, /* R33 - Right OPGA Volume */ + 0x0003, /* R34 - Speaker Volume */ + 0x0003, /* R35 - ClassD1 */ + 0x0000, /* R36 */ + 0x0100, /* R37 - ClassD3 */ + 0x0000, /* R38 */ + 0x0000, /* R39 - Input Mixer1 */ + 0x0000, /* R40 - Input Mixer2 */ + 0x0000, /* R41 - Input Mixer3 */ + 0x0000, /* R42 - Input Mixer4 */ + 0x0000, /* R43 - Input Mixer5 */ + 0x0000, /* R44 - Input Mixer6 */ + 0x0000, /* R45 - Output Mixer1 */ + 0x0000, /* R46 - Output Mixer2 */ + 0x0000, /* R47 - Output Mixer3 */ + 0x0000, /* R48 - Output Mixer4 */ + 0x0000, /* R49 - Output Mixer5 */ + 0x0000, /* R50 - Output Mixer6 */ + 0x0180, /* R51 - Out3/4 Mixer */ + 0x0000, /* R52 - Line Mixer1 */ + 0x0000, /* R53 - Line Mixer2 */ + 0x0000, /* R54 - Speaker Mixer */ + 0x0000, /* R55 - Additional Control */ + 0x0000, /* R56 - AntiPOP1 */ + 0x0000, /* R57 - AntiPOP2 */ + 0x0000, /* R58 - MICBIAS */ + 0x0000, /* R59 */ + 0x0008, /* R60 - PLL1 */ + 0x0031, /* R61 - PLL2 */ + 0x0026, /* R62 - PLL3 */ +}; + +#define wm8991_reset(c) snd_soc_write(c, WM8991_RESET, 0) + +static const unsigned int rec_mix_tlv[] = { + TLV_DB_RANGE_HEAD(1), + 0, 7, TLV_DB_LINEAR_ITEM(-1500, 600), +}; + +static const unsigned int in_pga_tlv[] = { + TLV_DB_RANGE_HEAD(1), + 0, 0x1F, TLV_DB_LINEAR_ITEM(-1650, 3000), +}; + +static const unsigned int out_mix_tlv[] = { + TLV_DB_RANGE_HEAD(1), + 0, 7, TLV_DB_LINEAR_ITEM(0, -2100), +}; + +static const unsigned int out_pga_tlv[] = { + TLV_DB_RANGE_HEAD(1), + 0, 127, TLV_DB_LINEAR_ITEM(-7300, 600), +}; + +static const unsigned int out_omix_tlv[] = { + TLV_DB_RANGE_HEAD(1), + 0, 7, TLV_DB_LINEAR_ITEM(-600, 0), +}; + +static const unsigned int out_dac_tlv[] = { + TLV_DB_RANGE_HEAD(1), + 0, 255, TLV_DB_LINEAR_ITEM(-7163, 0), +}; + +static const unsigned int in_adc_tlv[] = { + TLV_DB_RANGE_HEAD(1), + 0, 255, TLV_DB_LINEAR_ITEM(-7163, 1763), +}; + +static const unsigned int out_sidetone_tlv[] = { + TLV_DB_RANGE_HEAD(1), + 0, 31, TLV_DB_LINEAR_ITEM(-3600, 0), +}; + +static int wm899x_outpga_put_volsw_vu(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int reg = kcontrol->private_value & 0xff; + int ret; + u16 val; + + ret = snd_soc_put_volsw(kcontrol, ucontrol); + if (ret < 0) + return ret; + + /* now hit the volume update bits (always bit 8) */ + val = snd_soc_read(codec, reg); + return snd_soc_write(codec, reg, val | 0x0100); +} + +static const char *wm8991_digital_sidetone[] = +{"None", "Left ADC", "Right ADC", "Reserved"}; + +static const struct soc_enum wm8991_left_digital_sidetone_enum = + SOC_ENUM_SINGLE(WM8991_DIGITAL_SIDE_TONE, + WM8991_ADC_TO_DACL_SHIFT, + WM8991_ADC_TO_DACL_MASK, + wm8991_digital_sidetone); + +static const struct soc_enum wm8991_right_digital_sidetone_enum = + SOC_ENUM_SINGLE(WM8991_DIGITAL_SIDE_TONE, + WM8991_ADC_TO_DACR_SHIFT, + WM8991_ADC_TO_DACR_MASK, + wm8991_digital_sidetone); + +static const char *wm8991_adcmode[] = +{"Hi-fi mode", "Voice mode 1", "Voice mode 2", "Voice mode 3"}; + +static const struct soc_enum wm8991_right_adcmode_enum = + SOC_ENUM_SINGLE(WM8991_ADC_CTRL, + WM8991_ADC_HPF_CUT_SHIFT, + WM8991_ADC_HPF_CUT_MASK, + wm8991_adcmode); + +static const struct snd_kcontrol_new wm8991_snd_controls[] = { + /* INMIXL */ + SOC_SINGLE("LIN12 PGA Boost", WM8991_INPUT_MIXER3, WM8991_L12MNBST_BIT, 1, 0), + SOC_SINGLE("LIN34 PGA Boost", WM8991_INPUT_MIXER3, WM8991_L34MNBST_BIT, 1, 0), + /* INMIXR */ + SOC_SINGLE("RIN12 PGA Boost", WM8991_INPUT_MIXER3, WM8991_R12MNBST_BIT, 1, 0), + SOC_SINGLE("RIN34 PGA Boost", WM8991_INPUT_MIXER3, WM8991_R34MNBST_BIT, 1, 0), + + /* LOMIX */ + SOC_SINGLE_TLV("LOMIX LIN3 Bypass Volume", WM8991_OUTPUT_MIXER3, + WM8991_LLI3LOVOL_SHIFT, WM8991_LLI3LOVOL_MASK, 1, out_mix_tlv), + SOC_SINGLE_TLV("LOMIX RIN12 PGA Bypass Volume", WM8991_OUTPUT_MIXER3, + WM8991_LR12LOVOL_SHIFT, WM8991_LR12LOVOL_MASK, 1, out_mix_tlv), + SOC_SINGLE_TLV("LOMIX LIN12 PGA Bypass Volume", WM8991_OUTPUT_MIXER3, + WM8991_LL12LOVOL_SHIFT, WM8991_LL12LOVOL_MASK, 1, out_mix_tlv), + SOC_SINGLE_TLV("LOMIX RIN3 Bypass Volume", WM8991_OUTPUT_MIXER5, + WM8991_LRI3LOVOL_SHIFT, WM8991_LRI3LOVOL_MASK, 1, out_mix_tlv), + SOC_SINGLE_TLV("LOMIX AINRMUX Bypass Volume", WM8991_OUTPUT_MIXER5, + WM8991_LRBLOVOL_SHIFT, WM8991_LRBLOVOL_MASK, 1, out_mix_tlv), + SOC_SINGLE_TLV("LOMIX AINLMUX Bypass Volume", WM8991_OUTPUT_MIXER5, + WM8991_LRBLOVOL_SHIFT, WM8991_LRBLOVOL_MASK, 1, out_mix_tlv), + + /* ROMIX */ + SOC_SINGLE_TLV("ROMIX RIN3 Bypass Volume", WM8991_OUTPUT_MIXER4, + WM8991_RRI3ROVOL_SHIFT, WM8991_RRI3ROVOL_MASK, 1, out_mix_tlv), + SOC_SINGLE_TLV("ROMIX LIN12 PGA Bypass Volume", WM8991_OUTPUT_MIXER4, + WM8991_RL12ROVOL_SHIFT, WM8991_RL12ROVOL_MASK, 1, out_mix_tlv), + SOC_SINGLE_TLV("ROMIX RIN12 PGA Bypass Volume", WM8991_OUTPUT_MIXER4, + WM8991_RR12ROVOL_SHIFT, WM8991_RR12ROVOL_MASK, 1, out_mix_tlv), + SOC_SINGLE_TLV("ROMIX LIN3 Bypass Volume", WM8991_OUTPUT_MIXER6, + WM8991_RLI3ROVOL_SHIFT, WM8991_RLI3ROVOL_MASK, 1, out_mix_tlv), + SOC_SINGLE_TLV("ROMIX AINLMUX Bypass Volume", WM8991_OUTPUT_MIXER6, + WM8991_RLBROVOL_SHIFT, WM8991_RLBROVOL_MASK, 1, out_mix_tlv), + SOC_SINGLE_TLV("ROMIX AINRMUX Bypass Volume", WM8991_OUTPUT_MIXER6, + WM8991_RRBROVOL_SHIFT, WM8991_RRBROVOL_MASK, 1, out_mix_tlv), + + /* LOUT */ + SOC_WM899X_OUTPGA_SINGLE_R_TLV("LOUT Volume", WM8991_LEFT_OUTPUT_VOLUME, + WM8991_LOUTVOL_SHIFT, WM8991_LOUTVOL_MASK, 0, out_pga_tlv), + SOC_SINGLE("LOUT ZC", WM8991_LEFT_OUTPUT_VOLUME, WM8991_LOZC_BIT, 1, 0), + + /* ROUT */ + SOC_WM899X_OUTPGA_SINGLE_R_TLV("ROUT Volume", WM8991_RIGHT_OUTPUT_VOLUME, + WM8991_ROUTVOL_SHIFT, WM8991_ROUTVOL_MASK, 0, out_pga_tlv), + SOC_SINGLE("ROUT ZC", WM8991_RIGHT_OUTPUT_VOLUME, WM8991_ROZC_BIT, 1, 0), + + /* LOPGA */ + SOC_WM899X_OUTPGA_SINGLE_R_TLV("LOPGA Volume", WM8991_LEFT_OPGA_VOLUME, + WM8991_LOPGAVOL_SHIFT, WM8991_LOPGAVOL_MASK, 0, out_pga_tlv), + SOC_SINGLE("LOPGA ZC Switch", WM8991_LEFT_OPGA_VOLUME, + WM8991_LOPGAZC_BIT, 1, 0), + + /* ROPGA */ + SOC_WM899X_OUTPGA_SINGLE_R_TLV("ROPGA Volume", WM8991_RIGHT_OPGA_VOLUME, + WM8991_ROPGAVOL_SHIFT, WM8991_ROPGAVOL_MASK, 0, out_pga_tlv), + SOC_SINGLE("ROPGA ZC Switch", WM8991_RIGHT_OPGA_VOLUME, + WM8991_ROPGAZC_BIT, 1, 0), + + SOC_SINGLE("LON Mute Switch", WM8991_LINE_OUTPUTS_VOLUME, + WM8991_LONMUTE_BIT, 1, 0), + SOC_SINGLE("LOP Mute Switch", WM8991_LINE_OUTPUTS_VOLUME, + WM8991_LOPMUTE_BIT, 1, 0), + SOC_SINGLE("LOP Attenuation Switch", WM8991_LINE_OUTPUTS_VOLUME, + WM8991_LOATTN_BIT, 1, 0), + SOC_SINGLE("RON Mute Switch", WM8991_LINE_OUTPUTS_VOLUME, + WM8991_RONMUTE_BIT, 1, 0), + SOC_SINGLE("ROP Mute Switch", WM8991_LINE_OUTPUTS_VOLUME, + WM8991_ROPMUTE_BIT, 1, 0), + SOC_SINGLE("ROP Attenuation Switch", WM8991_LINE_OUTPUTS_VOLUME, + WM8991_ROATTN_BIT, 1, 0), + + SOC_SINGLE("OUT3 Mute Switch", WM8991_OUT3_4_VOLUME, + WM8991_OUT3MUTE_BIT, 1, 0), + SOC_SINGLE("OUT3 Attenuation Switch", WM8991_OUT3_4_VOLUME, + WM8991_OUT3ATTN_BIT, 1, 0), + + SOC_SINGLE("OUT4 Mute Switch", WM8991_OUT3_4_VOLUME, + WM8991_OUT4MUTE_BIT, 1, 0), + SOC_SINGLE("OUT4 Attenuation Switch", WM8991_OUT3_4_VOLUME, + WM8991_OUT4ATTN_BIT, 1, 0), + + SOC_SINGLE("Speaker Mode Switch", WM8991_CLASSD1, + WM8991_CDMODE_BIT, 1, 0), + + SOC_SINGLE("Speaker Output Attenuation Volume", WM8991_SPEAKER_VOLUME, + WM8991_SPKVOL_SHIFT, WM8991_SPKVOL_MASK, 0), + SOC_SINGLE("Speaker DC Boost Volume", WM8991_CLASSD3, + WM8991_DCGAIN_SHIFT, WM8991_DCGAIN_MASK, 0), + SOC_SINGLE("Speaker AC Boost Volume", WM8991_CLASSD3, + WM8991_ACGAIN_SHIFT, WM8991_ACGAIN_MASK, 0), + + SOC_WM899X_OUTPGA_SINGLE_R_TLV("Left DAC Digital Volume", + WM8991_LEFT_DAC_DIGITAL_VOLUME, + WM8991_DACL_VOL_SHIFT, + WM8991_DACL_VOL_MASK, + 0, + out_dac_tlv), + + SOC_WM899X_OUTPGA_SINGLE_R_TLV("Right DAC Digital Volume", + WM8991_RIGHT_DAC_DIGITAL_VOLUME, + WM8991_DACR_VOL_SHIFT, + WM8991_DACR_VOL_MASK, + 0, + out_dac_tlv), + + SOC_ENUM("Left Digital Sidetone", wm8991_left_digital_sidetone_enum), + SOC_ENUM("Right Digital Sidetone", wm8991_right_digital_sidetone_enum), + + SOC_SINGLE_TLV("Left Digital Sidetone Volume", WM8991_DIGITAL_SIDE_TONE, + WM8991_ADCL_DAC_SVOL_SHIFT, WM8991_ADCL_DAC_SVOL_MASK, 0, + out_sidetone_tlv), + SOC_SINGLE_TLV("Right Digital Sidetone Volume", WM8991_DIGITAL_SIDE_TONE, + WM8991_ADCR_DAC_SVOL_SHIFT, WM8991_ADCR_DAC_SVOL_MASK, 0, + out_sidetone_tlv), + + SOC_SINGLE("ADC Digital High Pass Filter Switch", WM8991_ADC_CTRL, + WM8991_ADC_HPF_ENA_BIT, 1, 0), + + SOC_ENUM("ADC HPF Mode", wm8991_right_adcmode_enum), + + SOC_WM899X_OUTPGA_SINGLE_R_TLV("Left ADC Digital Volume", + WM8991_LEFT_ADC_DIGITAL_VOLUME, + WM8991_ADCL_VOL_SHIFT, + WM8991_ADCL_VOL_MASK, + 0, + in_adc_tlv), + + SOC_WM899X_OUTPGA_SINGLE_R_TLV("Right ADC Digital Volume", + WM8991_RIGHT_ADC_DIGITAL_VOLUME, + WM8991_ADCR_VOL_SHIFT, + WM8991_ADCR_VOL_MASK, + 0, + in_adc_tlv), + + SOC_WM899X_OUTPGA_SINGLE_R_TLV("LIN12 Volume", + WM8991_LEFT_LINE_INPUT_1_2_VOLUME, + WM8991_LIN12VOL_SHIFT, + WM8991_LIN12VOL_MASK, + 0, + in_pga_tlv), + + SOC_SINGLE("LIN12 ZC Switch", WM8991_LEFT_LINE_INPUT_1_2_VOLUME, + WM8991_LI12ZC_BIT, 1, 0), + + SOC_SINGLE("LIN12 Mute Switch", WM8991_LEFT_LINE_INPUT_1_2_VOLUME, + WM8991_LI12MUTE_BIT, 1, 0), + + SOC_WM899X_OUTPGA_SINGLE_R_TLV("LIN34 Volume", + WM8991_LEFT_LINE_INPUT_3_4_VOLUME, + WM8991_LIN34VOL_SHIFT, + WM8991_LIN34VOL_MASK, + 0, + in_pga_tlv), + + SOC_SINGLE("LIN34 ZC Switch", WM8991_LEFT_LINE_INPUT_3_4_VOLUME, + WM8991_LI34ZC_BIT, 1, 0), + + SOC_SINGLE("LIN34 Mute Switch", WM8991_LEFT_LINE_INPUT_3_4_VOLUME, + WM8991_LI34MUTE_BIT, 1, 0), + + SOC_WM899X_OUTPGA_SINGLE_R_TLV("RIN12 Volume", + WM8991_RIGHT_LINE_INPUT_1_2_VOLUME, + WM8991_RIN12VOL_SHIFT, + WM8991_RIN12VOL_MASK, + 0, + in_pga_tlv), + + SOC_SINGLE("RIN12 ZC Switch", WM8991_RIGHT_LINE_INPUT_1_2_VOLUME, + WM8991_RI12ZC_BIT, 1, 0), + + SOC_SINGLE("RIN12 Mute Switch", WM8991_RIGHT_LINE_INPUT_1_2_VOLUME, + WM8991_RI12MUTE_BIT, 1, 0), + + SOC_WM899X_OUTPGA_SINGLE_R_TLV("RIN34 Volume", + WM8991_RIGHT_LINE_INPUT_3_4_VOLUME, + WM8991_RIN34VOL_SHIFT, + WM8991_RIN34VOL_MASK, + 0, + in_pga_tlv), + + SOC_SINGLE("RIN34 ZC Switch", WM8991_RIGHT_LINE_INPUT_3_4_VOLUME, + WM8991_RI34ZC_BIT, 1, 0), + + SOC_SINGLE("RIN34 Mute Switch", WM8991_RIGHT_LINE_INPUT_3_4_VOLUME, + WM8991_RI34MUTE_BIT, 1, 0), +}; + +/* + * _DAPM_ Controls + */ +static int inmixer_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + u16 reg, fakepower; + + reg = snd_soc_read(w->codec, WM8991_POWER_MANAGEMENT_2); + fakepower = snd_soc_read(w->codec, WM8991_INTDRIVBITS); + + if (fakepower & ((1 << WM8991_INMIXL_PWR_BIT) | + (1 << WM8991_AINLMUX_PWR_BIT))) + reg |= WM8991_AINL_ENA; + else + reg &= ~WM8991_AINL_ENA; + + if (fakepower & ((1 << WM8991_INMIXR_PWR_BIT) | + (1 << WM8991_AINRMUX_PWR_BIT))) + reg |= WM8991_AINR_ENA; + else + reg &= ~WM8991_AINL_ENA; + + snd_soc_write(w->codec, WM8991_POWER_MANAGEMENT_2, reg); + return 0; +} + +static int outmixer_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + u32 reg_shift = kcontrol->private_value & 0xfff; + int ret = 0; + u16 reg; + + switch (reg_shift) { + case WM8991_SPEAKER_MIXER | (WM8991_LDSPK_BIT << 8): + reg = snd_soc_read(w->codec, WM8991_OUTPUT_MIXER1); + if (reg & WM8991_LDLO) { + printk(KERN_WARNING + "Cannot set as Output Mixer 1 LDLO Set\n"); + ret = -1; + } + break; + + case WM8991_SPEAKER_MIXER | (WM8991_RDSPK_BIT << 8): + reg = snd_soc_read(w->codec, WM8991_OUTPUT_MIXER2); + if (reg & WM8991_RDRO) { + printk(KERN_WARNING + "Cannot set as Output Mixer 2 RDRO Set\n"); + ret = -1; + } + break; + + case WM8991_OUTPUT_MIXER1 | (WM8991_LDLO_BIT << 8): + reg = snd_soc_read(w->codec, WM8991_SPEAKER_MIXER); + if (reg & WM8991_LDSPK) { + printk(KERN_WARNING + "Cannot set as Speaker Mixer LDSPK Set\n"); + ret = -1; + } + break; + + case WM8991_OUTPUT_MIXER2 | (WM8991_RDRO_BIT << 8): + reg = snd_soc_read(w->codec, WM8991_SPEAKER_MIXER); + if (reg & WM8991_RDSPK) { + printk(KERN_WARNING + "Cannot set as Speaker Mixer RDSPK Set\n"); + ret = -1; + } + break; + } + + return ret; +} + +/* INMIX dB values */ +static const unsigned int in_mix_tlv[] = { + TLV_DB_RANGE_HEAD(1), + 0, 7, TLV_DB_LINEAR_ITEM(-1200, 600), +}; + +/* Left In PGA Connections */ +static const struct snd_kcontrol_new wm8991_dapm_lin12_pga_controls[] = { + SOC_DAPM_SINGLE("LIN1 Switch", WM8991_INPUT_MIXER2, WM8991_LMN1_BIT, 1, 0), + SOC_DAPM_SINGLE("LIN2 Switch", WM8991_INPUT_MIXER2, WM8991_LMP2_BIT, 1, 0), +}; + +static const struct snd_kcontrol_new wm8991_dapm_lin34_pga_controls[] = { + SOC_DAPM_SINGLE("LIN3 Switch", WM8991_INPUT_MIXER2, WM8991_LMN3_BIT, 1, 0), + SOC_DAPM_SINGLE("LIN4 Switch", WM8991_INPUT_MIXER2, WM8991_LMP4_BIT, 1, 0), +}; + +/* Right In PGA Connections */ +static const struct snd_kcontrol_new wm8991_dapm_rin12_pga_controls[] = { + SOC_DAPM_SINGLE("RIN1 Switch", WM8991_INPUT_MIXER2, WM8991_RMN1_BIT, 1, 0), + SOC_DAPM_SINGLE("RIN2 Switch", WM8991_INPUT_MIXER2, WM8991_RMP2_BIT, 1, 0), +}; + +static const struct snd_kcontrol_new wm8991_dapm_rin34_pga_controls[] = { + SOC_DAPM_SINGLE("RIN3 Switch", WM8991_INPUT_MIXER2, WM8991_RMN3_BIT, 1, 0), + SOC_DAPM_SINGLE("RIN4 Switch", WM8991_INPUT_MIXER2, WM8991_RMP4_BIT, 1, 0), +}; + +/* INMIXL */ +static const struct snd_kcontrol_new wm8991_dapm_inmixl_controls[] = { + SOC_DAPM_SINGLE_TLV("Record Left Volume", WM8991_INPUT_MIXER3, + WM8991_LDBVOL_SHIFT, WM8991_LDBVOL_MASK, 0, in_mix_tlv), + SOC_DAPM_SINGLE_TLV("LIN2 Volume", WM8991_INPUT_MIXER5, WM8991_LI2BVOL_SHIFT, + 7, 0, in_mix_tlv), + SOC_DAPM_SINGLE("LINPGA12 Switch", WM8991_INPUT_MIXER3, WM8991_L12MNB_BIT, + 1, 0), + SOC_DAPM_SINGLE("LINPGA34 Switch", WM8991_INPUT_MIXER3, WM8991_L34MNB_BIT, + 1, 0), +}; + +/* INMIXR */ +static const struct snd_kcontrol_new wm8991_dapm_inmixr_controls[] = { + SOC_DAPM_SINGLE_TLV("Record Right Volume", WM8991_INPUT_MIXER4, + WM8991_RDBVOL_SHIFT, WM8991_RDBVOL_MASK, 0, in_mix_tlv), + SOC_DAPM_SINGLE_TLV("RIN2 Volume", WM8991_INPUT_MIXER6, WM8991_RI2BVOL_SHIFT, + 7, 0, in_mix_tlv), + SOC_DAPM_SINGLE("RINPGA12 Switch", WM8991_INPUT_MIXER3, WM8991_L12MNB_BIT, + 1, 0), + SOC_DAPM_SINGLE("RINPGA34 Switch", WM8991_INPUT_MIXER3, WM8991_L34MNB_BIT, + 1, 0), +}; + +/* AINLMUX */ +static const char *wm8991_ainlmux[] = +{"INMIXL Mix", "RXVOICE Mix", "DIFFINL Mix"}; + +static const struct soc_enum wm8991_ainlmux_enum = + SOC_ENUM_SINGLE(WM8991_INPUT_MIXER1, WM8991_AINLMODE_SHIFT, + ARRAY_SIZE(wm8991_ainlmux), wm8991_ainlmux); + +static const struct snd_kcontrol_new wm8991_dapm_ainlmux_controls = + SOC_DAPM_ENUM("Route", wm8991_ainlmux_enum); + +/* DIFFINL */ + +/* AINRMUX */ +static const char *wm8991_ainrmux[] = +{"INMIXR Mix", "RXVOICE Mix", "DIFFINR Mix"}; + +static const struct soc_enum wm8991_ainrmux_enum = + SOC_ENUM_SINGLE(WM8991_INPUT_MIXER1, WM8991_AINRMODE_SHIFT, + ARRAY_SIZE(wm8991_ainrmux), wm8991_ainrmux); + +static const struct snd_kcontrol_new wm8991_dapm_ainrmux_controls = + SOC_DAPM_ENUM("Route", wm8991_ainrmux_enum); + +/* RXVOICE */ +static const struct snd_kcontrol_new wm8991_dapm_rxvoice_controls[] = { + SOC_DAPM_SINGLE_TLV("LIN4RXN", WM8991_INPUT_MIXER5, WM8991_LR4BVOL_SHIFT, + WM8991_LR4BVOL_MASK, 0, in_mix_tlv), + SOC_DAPM_SINGLE_TLV("RIN4RXP", WM8991_INPUT_MIXER6, WM8991_RL4BVOL_SHIFT, + WM8991_RL4BVOL_MASK, 0, in_mix_tlv), +}; + +/* LOMIX */ +static const struct snd_kcontrol_new wm8991_dapm_lomix_controls[] = { + SOC_DAPM_SINGLE("LOMIX Right ADC Bypass Switch", WM8991_OUTPUT_MIXER1, + WM8991_LRBLO_BIT, 1, 0), + SOC_DAPM_SINGLE("LOMIX Left ADC Bypass Switch", WM8991_OUTPUT_MIXER1, + WM8991_LLBLO_BIT, 1, 0), + SOC_DAPM_SINGLE("LOMIX RIN3 Bypass Switch", WM8991_OUTPUT_MIXER1, + WM8991_LRI3LO_BIT, 1, 0), + SOC_DAPM_SINGLE("LOMIX LIN3 Bypass Switch", WM8991_OUTPUT_MIXER1, + WM8991_LLI3LO_BIT, 1, 0), + SOC_DAPM_SINGLE("LOMIX RIN12 PGA Bypass Switch", WM8991_OUTPUT_MIXER1, + WM8991_LR12LO_BIT, 1, 0), + SOC_DAPM_SINGLE("LOMIX LIN12 PGA Bypass Switch", WM8991_OUTPUT_MIXER1, + WM8991_LL12LO_BIT, 1, 0), + SOC_DAPM_SINGLE("LOMIX Left DAC Switch", WM8991_OUTPUT_MIXER1, + WM8991_LDLO_BIT, 1, 0), +}; + +/* ROMIX */ +static const struct snd_kcontrol_new wm8991_dapm_romix_controls[] = { + SOC_DAPM_SINGLE("ROMIX Left ADC Bypass Switch", WM8991_OUTPUT_MIXER2, + WM8991_RLBRO_BIT, 1, 0), + SOC_DAPM_SINGLE("ROMIX Right ADC Bypass Switch", WM8991_OUTPUT_MIXER2, + WM8991_RRBRO_BIT, 1, 0), + SOC_DAPM_SINGLE("ROMIX LIN3 Bypass Switch", WM8991_OUTPUT_MIXER2, + WM8991_RLI3RO_BIT, 1, 0), + SOC_DAPM_SINGLE("ROMIX RIN3 Bypass Switch", WM8991_OUTPUT_MIXER2, + WM8991_RRI3RO_BIT, 1, 0), + SOC_DAPM_SINGLE("ROMIX LIN12 PGA Bypass Switch", WM8991_OUTPUT_MIXER2, + WM8991_RL12RO_BIT, 1, 0), + SOC_DAPM_SINGLE("ROMIX RIN12 PGA Bypass Switch", WM8991_OUTPUT_MIXER2, + WM8991_RR12RO_BIT, 1, 0), + SOC_DAPM_SINGLE("ROMIX Right DAC Switch", WM8991_OUTPUT_MIXER2, + WM8991_RDRO_BIT, 1, 0), +}; + +/* LONMIX */ +static const struct snd_kcontrol_new wm8991_dapm_lonmix_controls[] = { + SOC_DAPM_SINGLE("LONMIX Left Mixer PGA Switch", WM8991_LINE_MIXER1, + WM8991_LLOPGALON_BIT, 1, 0), + SOC_DAPM_SINGLE("LONMIX Right Mixer PGA Switch", WM8991_LINE_MIXER1, + WM8991_LROPGALON_BIT, 1, 0), + SOC_DAPM_SINGLE("LONMIX Inverted LOP Switch", WM8991_LINE_MIXER1, + WM8991_LOPLON_BIT, 1, 0), +}; + +/* LOPMIX */ +static const struct snd_kcontrol_new wm8991_dapm_lopmix_controls[] = { + SOC_DAPM_SINGLE("LOPMIX Right Mic Bypass Switch", WM8991_LINE_MIXER1, + WM8991_LR12LOP_BIT, 1, 0), + SOC_DAPM_SINGLE("LOPMIX Left Mic Bypass Switch", WM8991_LINE_MIXER1, + WM8991_LL12LOP_BIT, 1, 0), + SOC_DAPM_SINGLE("LOPMIX Left Mixer PGA Switch", WM8991_LINE_MIXER1, + WM8991_LLOPGALOP_BIT, 1, 0), +}; + +/* RONMIX */ +static const struct snd_kcontrol_new wm8991_dapm_ronmix_controls[] = { + SOC_DAPM_SINGLE("RONMIX Right Mixer PGA Switch", WM8991_LINE_MIXER2, + WM8991_RROPGARON_BIT, 1, 0), + SOC_DAPM_SINGLE("RONMIX Left Mixer PGA Switch", WM8991_LINE_MIXER2, + WM8991_RLOPGARON_BIT, 1, 0), + SOC_DAPM_SINGLE("RONMIX Inverted ROP Switch", WM8991_LINE_MIXER2, + WM8991_ROPRON_BIT, 1, 0), +}; + +/* ROPMIX */ +static const struct snd_kcontrol_new wm8991_dapm_ropmix_controls[] = { + SOC_DAPM_SINGLE("ROPMIX Left Mic Bypass Switch", WM8991_LINE_MIXER2, + WM8991_RL12ROP_BIT, 1, 0), + SOC_DAPM_SINGLE("ROPMIX Right Mic Bypass Switch", WM8991_LINE_MIXER2, + WM8991_RR12ROP_BIT, 1, 0), + SOC_DAPM_SINGLE("ROPMIX Right Mixer PGA Switch", WM8991_LINE_MIXER2, + WM8991_RROPGAROP_BIT, 1, 0), +}; + +/* OUT3MIX */ +static const struct snd_kcontrol_new wm8991_dapm_out3mix_controls[] = { + SOC_DAPM_SINGLE("OUT3MIX LIN4RXN Bypass Switch", WM8991_OUT3_4_MIXER, + WM8991_LI4O3_BIT, 1, 0), + SOC_DAPM_SINGLE("OUT3MIX Left Out PGA Switch", WM8991_OUT3_4_MIXER, + WM8991_LPGAO3_BIT, 1, 0), +}; + +/* OUT4MIX */ +static const struct snd_kcontrol_new wm8991_dapm_out4mix_controls[] = { + SOC_DAPM_SINGLE("OUT4MIX Right Out PGA Switch", WM8991_OUT3_4_MIXER, + WM8991_RPGAO4_BIT, 1, 0), + SOC_DAPM_SINGLE("OUT4MIX RIN4RXP Bypass Switch", WM8991_OUT3_4_MIXER, + WM8991_RI4O4_BIT, 1, 0), +}; + +/* SPKMIX */ +static const struct snd_kcontrol_new wm8991_dapm_spkmix_controls[] = { + SOC_DAPM_SINGLE("SPKMIX LIN2 Bypass Switch", WM8991_SPEAKER_MIXER, + WM8991_LI2SPK_BIT, 1, 0), + SOC_DAPM_SINGLE("SPKMIX LADC Bypass Switch", WM8991_SPEAKER_MIXER, + WM8991_LB2SPK_BIT, 1, 0), + SOC_DAPM_SINGLE("SPKMIX Left Mixer PGA Switch", WM8991_SPEAKER_MIXER, + WM8991_LOPGASPK_BIT, 1, 0), + SOC_DAPM_SINGLE("SPKMIX Left DAC Switch", WM8991_SPEAKER_MIXER, + WM8991_LDSPK_BIT, 1, 0), + SOC_DAPM_SINGLE("SPKMIX Right DAC Switch", WM8991_SPEAKER_MIXER, + WM8991_RDSPK_BIT, 1, 0), + SOC_DAPM_SINGLE("SPKMIX Right Mixer PGA Switch", WM8991_SPEAKER_MIXER, + WM8991_ROPGASPK_BIT, 1, 0), + SOC_DAPM_SINGLE("SPKMIX RADC Bypass Switch", WM8991_SPEAKER_MIXER, + WM8991_RL12ROP_BIT, 1, 0), + SOC_DAPM_SINGLE("SPKMIX RIN2 Bypass Switch", WM8991_SPEAKER_MIXER, + WM8991_RI2SPK_BIT, 1, 0), +}; + +static const struct snd_soc_dapm_widget wm8991_dapm_widgets[] = { + /* Input Side */ + /* Input Lines */ + SND_SOC_DAPM_INPUT("LIN1"), + SND_SOC_DAPM_INPUT("LIN2"), + SND_SOC_DAPM_INPUT("LIN3"), + SND_SOC_DAPM_INPUT("LIN4RXN"), + SND_SOC_DAPM_INPUT("RIN3"), + SND_SOC_DAPM_INPUT("RIN4RXP"), + SND_SOC_DAPM_INPUT("RIN1"), + SND_SOC_DAPM_INPUT("RIN2"), + SND_SOC_DAPM_INPUT("Internal ADC Source"), + + /* DACs */ + SND_SOC_DAPM_ADC("Left ADC", "Left Capture", WM8991_POWER_MANAGEMENT_2, + WM8991_ADCL_ENA_BIT, 0), + SND_SOC_DAPM_ADC("Right ADC", "Right Capture", WM8991_POWER_MANAGEMENT_2, + WM8991_ADCR_ENA_BIT, 0), + + /* Input PGAs */ + SND_SOC_DAPM_MIXER("LIN12 PGA", WM8991_POWER_MANAGEMENT_2, WM8991_LIN12_ENA_BIT, + 0, &wm8991_dapm_lin12_pga_controls[0], + ARRAY_SIZE(wm8991_dapm_lin12_pga_controls)), + SND_SOC_DAPM_MIXER("LIN34 PGA", WM8991_POWER_MANAGEMENT_2, WM8991_LIN34_ENA_BIT, + 0, &wm8991_dapm_lin34_pga_controls[0], + ARRAY_SIZE(wm8991_dapm_lin34_pga_controls)), + SND_SOC_DAPM_MIXER("RIN12 PGA", WM8991_POWER_MANAGEMENT_2, WM8991_RIN12_ENA_BIT, + 0, &wm8991_dapm_rin12_pga_controls[0], + ARRAY_SIZE(wm8991_dapm_rin12_pga_controls)), + SND_SOC_DAPM_MIXER("RIN34 PGA", WM8991_POWER_MANAGEMENT_2, WM8991_RIN34_ENA_BIT, + 0, &wm8991_dapm_rin34_pga_controls[0], + ARRAY_SIZE(wm8991_dapm_rin34_pga_controls)), + + /* INMIXL */ + SND_SOC_DAPM_MIXER_E("INMIXL", WM8991_INTDRIVBITS, WM8991_INMIXL_PWR_BIT, 0, + &wm8991_dapm_inmixl_controls[0], + ARRAY_SIZE(wm8991_dapm_inmixl_controls), + inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + /* AINLMUX */ + SND_SOC_DAPM_MUX_E("AINLMUX", WM8991_INTDRIVBITS, WM8991_AINLMUX_PWR_BIT, 0, + &wm8991_dapm_ainlmux_controls, inmixer_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + /* INMIXR */ + SND_SOC_DAPM_MIXER_E("INMIXR", WM8991_INTDRIVBITS, WM8991_INMIXR_PWR_BIT, 0, + &wm8991_dapm_inmixr_controls[0], + ARRAY_SIZE(wm8991_dapm_inmixr_controls), + inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + /* AINRMUX */ + SND_SOC_DAPM_MUX_E("AINRMUX", WM8991_INTDRIVBITS, WM8991_AINRMUX_PWR_BIT, 0, + &wm8991_dapm_ainrmux_controls, inmixer_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + /* Output Side */ + /* DACs */ + SND_SOC_DAPM_DAC("Left DAC", "Left Playback", WM8991_POWER_MANAGEMENT_3, + WM8991_DACL_ENA_BIT, 0), + SND_SOC_DAPM_DAC("Right DAC", "Right Playback", WM8991_POWER_MANAGEMENT_3, + WM8991_DACR_ENA_BIT, 0), + + /* LOMIX */ + SND_SOC_DAPM_MIXER_E("LOMIX", WM8991_POWER_MANAGEMENT_3, WM8991_LOMIX_ENA_BIT, + 0, &wm8991_dapm_lomix_controls[0], + ARRAY_SIZE(wm8991_dapm_lomix_controls), + outmixer_event, SND_SOC_DAPM_PRE_REG), + + /* LONMIX */ + SND_SOC_DAPM_MIXER("LONMIX", WM8991_POWER_MANAGEMENT_3, WM8991_LON_ENA_BIT, 0, + &wm8991_dapm_lonmix_controls[0], + ARRAY_SIZE(wm8991_dapm_lonmix_controls)), + + /* LOPMIX */ + SND_SOC_DAPM_MIXER("LOPMIX", WM8991_POWER_MANAGEMENT_3, WM8991_LOP_ENA_BIT, 0, + &wm8991_dapm_lopmix_controls[0], + ARRAY_SIZE(wm8991_dapm_lopmix_controls)), + + /* OUT3MIX */ + SND_SOC_DAPM_MIXER("OUT3MIX", WM8991_POWER_MANAGEMENT_1, WM8991_OUT3_ENA_BIT, 0, + &wm8991_dapm_out3mix_controls[0], + ARRAY_SIZE(wm8991_dapm_out3mix_controls)), + + /* SPKMIX */ + SND_SOC_DAPM_MIXER_E("SPKMIX", WM8991_POWER_MANAGEMENT_1, WM8991_SPK_ENA_BIT, 0, + &wm8991_dapm_spkmix_controls[0], + ARRAY_SIZE(wm8991_dapm_spkmix_controls), outmixer_event, + SND_SOC_DAPM_PRE_REG), + + /* OUT4MIX */ + SND_SOC_DAPM_MIXER("OUT4MIX", WM8991_POWER_MANAGEMENT_1, WM8991_OUT4_ENA_BIT, 0, + &wm8991_dapm_out4mix_controls[0], + ARRAY_SIZE(wm8991_dapm_out4mix_controls)), + + /* ROPMIX */ + SND_SOC_DAPM_MIXER("ROPMIX", WM8991_POWER_MANAGEMENT_3, WM8991_ROP_ENA_BIT, 0, + &wm8991_dapm_ropmix_controls[0], + ARRAY_SIZE(wm8991_dapm_ropmix_controls)), + + /* RONMIX */ + SND_SOC_DAPM_MIXER("RONMIX", WM8991_POWER_MANAGEMENT_3, WM8991_RON_ENA_BIT, 0, + &wm8991_dapm_ronmix_controls[0], + ARRAY_SIZE(wm8991_dapm_ronmix_controls)), + + /* ROMIX */ + SND_SOC_DAPM_MIXER_E("ROMIX", WM8991_POWER_MANAGEMENT_3, WM8991_ROMIX_ENA_BIT, + 0, &wm8991_dapm_romix_controls[0], + ARRAY_SIZE(wm8991_dapm_romix_controls), + outmixer_event, SND_SOC_DAPM_PRE_REG), + + /* LOUT PGA */ + SND_SOC_DAPM_PGA("LOUT PGA", WM8991_POWER_MANAGEMENT_1, WM8991_LOUT_ENA_BIT, 0, + NULL, 0), + + /* ROUT PGA */ + SND_SOC_DAPM_PGA("ROUT PGA", WM8991_POWER_MANAGEMENT_1, WM8991_ROUT_ENA_BIT, 0, + NULL, 0), + + /* LOPGA */ + SND_SOC_DAPM_PGA("LOPGA", WM8991_POWER_MANAGEMENT_3, WM8991_LOPGA_ENA_BIT, 0, + NULL, 0), + + /* ROPGA */ + SND_SOC_DAPM_PGA("ROPGA", WM8991_POWER_MANAGEMENT_3, WM8991_ROPGA_ENA_BIT, 0, + NULL, 0), + + /* MICBIAS */ + SND_SOC_DAPM_MICBIAS("MICBIAS", WM8991_POWER_MANAGEMENT_1, + WM8991_MICBIAS_ENA_BIT, 0), + + SND_SOC_DAPM_OUTPUT("LON"), + SND_SOC_DAPM_OUTPUT("LOP"), + SND_SOC_DAPM_OUTPUT("OUT3"), + SND_SOC_DAPM_OUTPUT("LOUT"), + SND_SOC_DAPM_OUTPUT("SPKN"), + SND_SOC_DAPM_OUTPUT("SPKP"), + SND_SOC_DAPM_OUTPUT("ROUT"), + SND_SOC_DAPM_OUTPUT("OUT4"), + SND_SOC_DAPM_OUTPUT("ROP"), + SND_SOC_DAPM_OUTPUT("RON"), + SND_SOC_DAPM_OUTPUT("OUT"), + + SND_SOC_DAPM_OUTPUT("Internal DAC Sink"), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + /* Make DACs turn on when playing even if not mixed into any outputs */ + {"Internal DAC Sink", NULL, "Left DAC"}, + {"Internal DAC Sink", NULL, "Right DAC"}, + + /* Make ADCs turn on when recording even if not mixed from any inputs */ + {"Left ADC", NULL, "Internal ADC Source"}, + {"Right ADC", NULL, "Internal ADC Source"}, + + /* Input Side */ + /* LIN12 PGA */ + {"LIN12 PGA", "LIN1 Switch", "LIN1"}, + {"LIN12 PGA", "LIN2 Switch", "LIN2"}, + /* LIN34 PGA */ + {"LIN34 PGA", "LIN3 Switch", "LIN3"}, + {"LIN34 PGA", "LIN4 Switch", "LIN4RXN"}, + /* INMIXL */ + {"INMIXL", "Record Left Volume", "LOMIX"}, + {"INMIXL", "LIN2 Volume", "LIN2"}, + {"INMIXL", "LINPGA12 Switch", "LIN12 PGA"}, + {"INMIXL", "LINPGA34 Switch", "LIN34 PGA"}, + /* AINLMUX */ + {"AINLMUX", "INMIXL Mix", "INMIXL"}, + {"AINLMUX", "DIFFINL Mix", "LIN12 PGA"}, + {"AINLMUX", "DIFFINL Mix", "LIN34 PGA"}, + {"AINLMUX", "RXVOICE Mix", "LIN4RXN"}, + {"AINLMUX", "RXVOICE Mix", "RIN4RXP"}, + /* ADC */ + {"Left ADC", NULL, "AINLMUX"}, + + /* RIN12 PGA */ + {"RIN12 PGA", "RIN1 Switch", "RIN1"}, + {"RIN12 PGA", "RIN2 Switch", "RIN2"}, + /* RIN34 PGA */ + {"RIN34 PGA", "RIN3 Switch", "RIN3"}, + {"RIN34 PGA", "RIN4 Switch", "RIN4RXP"}, + /* INMIXL */ + {"INMIXR", "Record Right Volume", "ROMIX"}, + {"INMIXR", "RIN2 Volume", "RIN2"}, + {"INMIXR", "RINPGA12 Switch", "RIN12 PGA"}, + {"INMIXR", "RINPGA34 Switch", "RIN34 PGA"}, + /* AINRMUX */ + {"AINRMUX", "INMIXR Mix", "INMIXR"}, + {"AINRMUX", "DIFFINR Mix", "RIN12 PGA"}, + {"AINRMUX", "DIFFINR Mix", "RIN34 PGA"}, + {"AINRMUX", "RXVOICE Mix", "LIN4RXN"}, + {"AINRMUX", "RXVOICE Mix", "RIN4RXP"}, + /* ADC */ + {"Right ADC", NULL, "AINRMUX"}, + + /* LOMIX */ + {"LOMIX", "LOMIX RIN3 Bypass Switch", "RIN3"}, + {"LOMIX", "LOMIX LIN3 Bypass Switch", "LIN3"}, + {"LOMIX", "LOMIX LIN12 PGA Bypass Switch", "LIN12 PGA"}, + {"LOMIX", "LOMIX RIN12 PGA Bypass Switch", "RIN12 PGA"}, + {"LOMIX", "LOMIX Right ADC Bypass Switch", "AINRMUX"}, + {"LOMIX", "LOMIX Left ADC Bypass Switch", "AINLMUX"}, + {"LOMIX", "LOMIX Left DAC Switch", "Left DAC"}, + + /* ROMIX */ + {"ROMIX", "ROMIX RIN3 Bypass Switch", "RIN3"}, + {"ROMIX", "ROMIX LIN3 Bypass Switch", "LIN3"}, + {"ROMIX", "ROMIX LIN12 PGA Bypass Switch", "LIN12 PGA"}, + {"ROMIX", "ROMIX RIN12 PGA Bypass Switch", "RIN12 PGA"}, + {"ROMIX", "ROMIX Right ADC Bypass Switch", "AINRMUX"}, + {"ROMIX", "ROMIX Left ADC Bypass Switch", "AINLMUX"}, + {"ROMIX", "ROMIX Right DAC Switch", "Right DAC"}, + + /* SPKMIX */ + {"SPKMIX", "SPKMIX LIN2 Bypass Switch", "LIN2"}, + {"SPKMIX", "SPKMIX RIN2 Bypass Switch", "RIN2"}, + {"SPKMIX", "SPKMIX LADC Bypass Switch", "AINLMUX"}, + {"SPKMIX", "SPKMIX RADC Bypass Switch", "AINRMUX"}, + {"SPKMIX", "SPKMIX Left Mixer PGA Switch", "LOPGA"}, + {"SPKMIX", "SPKMIX Right Mixer PGA Switch", "ROPGA"}, + {"SPKMIX", "SPKMIX Right DAC Switch", "Right DAC"}, + {"SPKMIX", "SPKMIX Left DAC Switch", "Right DAC"}, + + /* LONMIX */ + {"LONMIX", "LONMIX Left Mixer PGA Switch", "LOPGA"}, + {"LONMIX", "LONMIX Right Mixer PGA Switch", "ROPGA"}, + {"LONMIX", "LONMIX Inverted LOP Switch", "LOPMIX"}, + + /* LOPMIX */ + {"LOPMIX", "LOPMIX Right Mic Bypass Switch", "RIN12 PGA"}, + {"LOPMIX", "LOPMIX Left Mic Bypass Switch", "LIN12 PGA"}, + {"LOPMIX", "LOPMIX Left Mixer PGA Switch", "LOPGA"}, + + /* OUT3MIX */ + {"OUT3MIX", "OUT3MIX LIN4RXN Bypass Switch", "LIN4RXN"}, + {"OUT3MIX", "OUT3MIX Left Out PGA Switch", "LOPGA"}, + + /* OUT4MIX */ + {"OUT4MIX", "OUT4MIX Right Out PGA Switch", "ROPGA"}, + {"OUT4MIX", "OUT4MIX RIN4RXP Bypass Switch", "RIN4RXP"}, + + /* RONMIX */ + {"RONMIX", "RONMIX Right Mixer PGA Switch", "ROPGA"}, + {"RONMIX", "RONMIX Left Mixer PGA Switch", "LOPGA"}, + {"RONMIX", "RONMIX Inverted ROP Switch", "ROPMIX"}, + + /* ROPMIX */ + {"ROPMIX", "ROPMIX Left Mic Bypass Switch", "LIN12 PGA"}, + {"ROPMIX", "ROPMIX Right Mic Bypass Switch", "RIN12 PGA"}, + {"ROPMIX", "ROPMIX Right Mixer PGA Switch", "ROPGA"}, + + /* Out Mixer PGAs */ + {"LOPGA", NULL, "LOMIX"}, + {"ROPGA", NULL, "ROMIX"}, + + {"LOUT PGA", NULL, "LOMIX"}, + {"ROUT PGA", NULL, "ROMIX"}, + + /* Output Pins */ + {"LON", NULL, "LONMIX"}, + {"LOP", NULL, "LOPMIX"}, + {"OUT", NULL, "OUT3MIX"}, + {"LOUT", NULL, "LOUT PGA"}, + {"SPKN", NULL, "SPKMIX"}, + {"ROUT", NULL, "ROUT PGA"}, + {"OUT4", NULL, "OUT4MIX"}, + {"ROP", NULL, "ROPMIX"}, + {"RON", NULL, "RONMIX"}, +}; + +/* PLL divisors */ +struct _pll_div { + u32 div2; + u32 n; + u32 k; +}; + +/* The size in bits of the pll divide multiplied by 10 + * to allow rounding later */ +#define FIXED_PLL_SIZE ((1 << 16) * 10) + +static void pll_factors(struct _pll_div *pll_div, unsigned int target, + unsigned int source) +{ + u64 Kpart; + unsigned int K, Ndiv, Nmod; + + + Ndiv = target / source; + if (Ndiv < 6) { + source >>= 1; + pll_div->div2 = 1; + Ndiv = target / source; + } else + pll_div->div2 = 0; + + if ((Ndiv < 6) || (Ndiv > 12)) + printk(KERN_WARNING + "WM8991 N value outwith recommended range! N = %d\n", Ndiv); + + pll_div->n = Ndiv; + Nmod = target % source; + Kpart = FIXED_PLL_SIZE * (long long)Nmod; + + do_div(Kpart, source); + + K = Kpart & 0xFFFFFFFF; + + /* Check if we need to round */ + if ((K % 10) >= 5) + K += 5; + + /* Move down to proper range now rounding is done */ + K /= 10; + + pll_div->k = K; +} + +static int wm8991_set_dai_pll(struct snd_soc_dai *codec_dai, + int pll_id, int src, unsigned int freq_in, unsigned int freq_out) +{ + u16 reg; + struct snd_soc_codec *codec = codec_dai->codec; + struct _pll_div pll_div; + + if (freq_in && freq_out) { + pll_factors(&pll_div, freq_out * 4, freq_in); + + /* Turn on PLL */ + reg = snd_soc_read(codec, WM8991_POWER_MANAGEMENT_2); + reg |= WM8991_PLL_ENA; + snd_soc_write(codec, WM8991_POWER_MANAGEMENT_2, reg); + + /* sysclk comes from PLL */ + reg = snd_soc_read(codec, WM8991_CLOCKING_2); + snd_soc_write(codec, WM8991_CLOCKING_2, reg | WM8991_SYSCLK_SRC); + + /* set up N , fractional mode and pre-divisor if neccessary */ + snd_soc_write(codec, WM8991_PLL1, pll_div.n | WM8991_SDM | + (pll_div.div2 ? WM8991_PRESCALE : 0)); + snd_soc_write(codec, WM8991_PLL2, (u8)(pll_div.k>>8)); + snd_soc_write(codec, WM8991_PLL3, (u8)(pll_div.k & 0xFF)); + } else { + /* Turn on PLL */ + reg = snd_soc_read(codec, WM8991_POWER_MANAGEMENT_2); + reg &= ~WM8991_PLL_ENA; + snd_soc_write(codec, WM8991_POWER_MANAGEMENT_2, reg); + } + return 0; +} + +/* + * Set's ADC and Voice DAC format. + */ +static int wm8991_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 audio1, audio3; + + audio1 = snd_soc_read(codec, WM8991_AUDIO_INTERFACE_1); + audio3 = snd_soc_read(codec, WM8991_AUDIO_INTERFACE_3); + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + audio3 &= ~WM8991_AIF_MSTR1; + break; + case SND_SOC_DAIFMT_CBM_CFM: + audio3 |= WM8991_AIF_MSTR1; + break; + default: + return -EINVAL; + } + + audio1 &= ~WM8991_AIF_FMT_MASK; + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + audio1 |= WM8991_AIF_TMF_I2S; + audio1 &= ~WM8991_AIF_LRCLK_INV; + break; + case SND_SOC_DAIFMT_RIGHT_J: + audio1 |= WM8991_AIF_TMF_RIGHTJ; + audio1 &= ~WM8991_AIF_LRCLK_INV; + break; + case SND_SOC_DAIFMT_LEFT_J: + audio1 |= WM8991_AIF_TMF_LEFTJ; + audio1 &= ~WM8991_AIF_LRCLK_INV; + break; + case SND_SOC_DAIFMT_DSP_A: + audio1 |= WM8991_AIF_TMF_DSP; + audio1 &= ~WM8991_AIF_LRCLK_INV; + break; + case SND_SOC_DAIFMT_DSP_B: + audio1 |= WM8991_AIF_TMF_DSP | WM8991_AIF_LRCLK_INV; + break; + default: + return -EINVAL; + } + + snd_soc_write(codec, WM8991_AUDIO_INTERFACE_1, audio1); + snd_soc_write(codec, WM8991_AUDIO_INTERFACE_3, audio3); + return 0; +} + +static int wm8991_set_dai_clkdiv(struct snd_soc_dai *codec_dai, + int div_id, int div) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 reg; + + switch (div_id) { + case WM8991_MCLK_DIV: + reg = snd_soc_read(codec, WM8991_CLOCKING_2) & + ~WM8991_MCLK_DIV_MASK; + snd_soc_write(codec, WM8991_CLOCKING_2, reg | div); + break; + case WM8991_DACCLK_DIV: + reg = snd_soc_read(codec, WM8991_CLOCKING_2) & + ~WM8991_DAC_CLKDIV_MASK; + snd_soc_write(codec, WM8991_CLOCKING_2, reg | div); + break; + case WM8991_ADCCLK_DIV: + reg = snd_soc_read(codec, WM8991_CLOCKING_2) & + ~WM8991_ADC_CLKDIV_MASK; + snd_soc_write(codec, WM8991_CLOCKING_2, reg | div); + break; + case WM8991_BCLK_DIV: + reg = snd_soc_read(codec, WM8991_CLOCKING_1) & + ~WM8991_BCLK_DIV_MASK; + snd_soc_write(codec, WM8991_CLOCKING_1, reg | div); + break; + default: + return -EINVAL; + } + + return 0; +} + +/* + * Set PCM DAI bit size and sample rate. + */ +static int wm8991_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + u16 audio1 = snd_soc_read(codec, WM8991_AUDIO_INTERFACE_1); + + audio1 &= ~WM8991_AIF_WL_MASK; + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + audio1 |= WM8991_AIF_WL_20BITS; + break; + case SNDRV_PCM_FORMAT_S24_LE: + audio1 |= WM8991_AIF_WL_24BITS; + break; + case SNDRV_PCM_FORMAT_S32_LE: + audio1 |= WM8991_AIF_WL_32BITS; + break; + } + + snd_soc_write(codec, WM8991_AUDIO_INTERFACE_1, audio1); + return 0; +} + +static int wm8991_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u16 val; + + val = snd_soc_read(codec, WM8991_DAC_CTRL) & ~WM8991_DAC_MUTE; + if (mute) + snd_soc_write(codec, WM8991_DAC_CTRL, val | WM8991_DAC_MUTE); + else + snd_soc_write(codec, WM8991_DAC_CTRL, val); + return 0; +} + +static int wm8991_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + u16 val; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + /* VMID=2*50k */ + val = snd_soc_read(codec, WM8991_POWER_MANAGEMENT_1) & + ~WM8991_VMID_MODE_MASK; + snd_soc_write(codec, WM8991_POWER_MANAGEMENT_1, val | 0x2); + break; + + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + snd_soc_cache_sync(codec); + /* Enable all output discharge bits */ + snd_soc_write(codec, WM8991_ANTIPOP1, WM8991_DIS_LLINE | + WM8991_DIS_RLINE | WM8991_DIS_OUT3 | + WM8991_DIS_OUT4 | WM8991_DIS_LOUT | + WM8991_DIS_ROUT); + + /* Enable POBCTRL, SOFT_ST, VMIDTOG and BUFDCOPEN */ + snd_soc_write(codec, WM8991_ANTIPOP2, WM8991_SOFTST | + WM8991_BUFDCOPEN | WM8991_POBCTRL | + WM8991_VMIDTOG); + + /* Delay to allow output caps to discharge */ + msleep(300); + + /* Disable VMIDTOG */ + snd_soc_write(codec, WM8991_ANTIPOP2, WM8991_SOFTST | + WM8991_BUFDCOPEN | WM8991_POBCTRL); + + /* disable all output discharge bits */ + snd_soc_write(codec, WM8991_ANTIPOP1, 0); + + /* Enable outputs */ + snd_soc_write(codec, WM8991_POWER_MANAGEMENT_1, 0x1b00); + + msleep(50); + + /* Enable VMID at 2x50k */ + snd_soc_write(codec, WM8991_POWER_MANAGEMENT_1, 0x1f02); + + msleep(100); + + /* Enable VREF */ + snd_soc_write(codec, WM8991_POWER_MANAGEMENT_1, 0x1f03); + + msleep(600); + + /* Enable BUFIOEN */ + snd_soc_write(codec, WM8991_ANTIPOP2, WM8991_SOFTST | + WM8991_BUFDCOPEN | WM8991_POBCTRL | + WM8991_BUFIOEN); + + /* Disable outputs */ + snd_soc_write(codec, WM8991_POWER_MANAGEMENT_1, 0x3); + + /* disable POBCTRL, SOFT_ST and BUFDCOPEN */ + snd_soc_write(codec, WM8991_ANTIPOP2, WM8991_BUFIOEN); + } + + /* VMID=2*250k */ + val = snd_soc_read(codec, WM8991_POWER_MANAGEMENT_1) & + ~WM8991_VMID_MODE_MASK; + snd_soc_write(codec, WM8991_POWER_MANAGEMENT_1, val | 0x4); + break; + + case SND_SOC_BIAS_OFF: + /* Enable POBCTRL and SOFT_ST */ + snd_soc_write(codec, WM8991_ANTIPOP2, WM8991_SOFTST | + WM8991_POBCTRL | WM8991_BUFIOEN); + + /* Enable POBCTRL, SOFT_ST and BUFDCOPEN */ + snd_soc_write(codec, WM8991_ANTIPOP2, WM8991_SOFTST | + WM8991_BUFDCOPEN | WM8991_POBCTRL | + WM8991_BUFIOEN); + + /* mute DAC */ + val = snd_soc_read(codec, WM8991_DAC_CTRL); + snd_soc_write(codec, WM8991_DAC_CTRL, val | WM8991_DAC_MUTE); + + /* Enable any disabled outputs */ + snd_soc_write(codec, WM8991_POWER_MANAGEMENT_1, 0x1f03); + + /* Disable VMID */ + snd_soc_write(codec, WM8991_POWER_MANAGEMENT_1, 0x1f01); + + msleep(300); + + /* Enable all output discharge bits */ + snd_soc_write(codec, WM8991_ANTIPOP1, WM8991_DIS_LLINE | + WM8991_DIS_RLINE | WM8991_DIS_OUT3 | + WM8991_DIS_OUT4 | WM8991_DIS_LOUT | + WM8991_DIS_ROUT); + + /* Disable VREF */ + snd_soc_write(codec, WM8991_POWER_MANAGEMENT_1, 0x0); + + /* disable POBCTRL, SOFT_ST and BUFDCOPEN */ + snd_soc_write(codec, WM8991_ANTIPOP2, 0x0); + codec->cache_sync = 1; + break; + } + + codec->dapm.bias_level = level; + return 0; +} + +static int wm8991_suspend(struct snd_soc_codec *codec, pm_message_t state) +{ + wm8991_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int wm8991_resume(struct snd_soc_codec *codec) +{ + wm8991_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + return 0; +} + +/* power down chip */ +static int wm8991_remove(struct snd_soc_codec *codec) +{ + wm8991_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int wm8991_probe(struct snd_soc_codec *codec) +{ + struct wm8991_priv *wm8991; + int ret; + unsigned int reg; + + wm8991 = snd_soc_codec_get_drvdata(codec); + + ret = snd_soc_codec_set_cache_io(codec, 8, 16, wm8991->control_type); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache i/o: %d\n", ret); + return ret; + } + + ret = wm8991_reset(codec); + if (ret < 0) { + dev_err(codec->dev, "Failed to issue reset\n"); + return ret; + } + + wm8991_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + reg = snd_soc_read(codec, WM8991_AUDIO_INTERFACE_4); + snd_soc_write(codec, WM8991_AUDIO_INTERFACE_4, reg | WM8991_ALRCGPIO1); + + reg = snd_soc_read(codec, WM8991_GPIO1_GPIO2) & + ~WM8991_GPIO1_SEL_MASK; + snd_soc_write(codec, WM8991_GPIO1_GPIO2, reg | 1); + + reg = snd_soc_read(codec, WM8991_POWER_MANAGEMENT_1); + snd_soc_write(codec, WM8991_POWER_MANAGEMENT_1, reg | WM8991_VREF_ENA| + WM8991_VMID_MODE_MASK); + + reg = snd_soc_read(codec, WM8991_POWER_MANAGEMENT_2); + snd_soc_write(codec, WM8991_POWER_MANAGEMENT_2, reg | WM8991_OPCLK_ENA); + + snd_soc_write(codec, WM8991_DAC_CTRL, 0); + snd_soc_write(codec, WM8991_LEFT_OUTPUT_VOLUME, 0x50 | (1<<8)); + snd_soc_write(codec, WM8991_RIGHT_OUTPUT_VOLUME, 0x50 | (1<<8)); + + snd_soc_add_controls(codec, wm8991_snd_controls, + ARRAY_SIZE(wm8991_snd_controls)); + + snd_soc_dapm_new_controls(&codec->dapm, wm8991_dapm_widgets, + ARRAY_SIZE(wm8991_dapm_widgets)); + snd_soc_dapm_add_routes(&codec->dapm, audio_map, + ARRAY_SIZE(audio_map)); + return 0; +} + +#define WM8991_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_ops wm8991_ops = { + .hw_params = wm8991_hw_params, + .digital_mute = wm8991_mute, + .set_fmt = wm8991_set_dai_fmt, + .set_clkdiv = wm8991_set_dai_clkdiv, + .set_pll = wm8991_set_dai_pll +}; + +/* + * The WM8991 supports 2 different and mutually exclusive DAI + * configurations. + * + * 1. ADC/DAC on Primary Interface + * 2. ADC on Primary Interface/DAC on secondary + */ +static struct snd_soc_dai_driver wm8991_dai = { + /* ADC/DAC on primary */ + .name = "wm8991", + .id = 1, + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = WM8991_FORMATS + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = WM8991_FORMATS + }, + .ops = &wm8991_ops +}; + +static struct snd_soc_codec_driver soc_codec_dev_wm8991 = { + .probe = wm8991_probe, + .remove = wm8991_remove, + .suspend = wm8991_suspend, + .resume = wm8991_resume, + .set_bias_level = wm8991_set_bias_level, + .reg_cache_size = WM8991_MAX_REGISTER + 1, + .reg_word_size = sizeof(u16), + .reg_cache_default = wm8991_reg_defs +}; + +static __devinit int wm8991_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct wm8991_priv *wm8991; + int ret; + + wm8991 = kzalloc(sizeof *wm8991, GFP_KERNEL); + if (!wm8991) + return -ENOMEM; + + wm8991->control_type = SND_SOC_I2C; + i2c_set_clientdata(i2c, wm8991); + + ret = snd_soc_register_codec(&i2c->dev, + &soc_codec_dev_wm8991, &wm8991_dai, 1); + if (ret < 0) + kfree(wm8991); + return ret; +} + +static __devexit int wm8991_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + kfree(i2c_get_clientdata(client)); + return 0; +} + +static const struct i2c_device_id wm8991_i2c_id[] = { + { "wm8991", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, wm8991_i2c_id); + +static struct i2c_driver wm8991_i2c_driver = { + .driver = { + .name = "wm8991", + .owner = THIS_MODULE, + }, + .probe = wm8991_i2c_probe, + .remove = __devexit_p(wm8991_i2c_remove), + .id_table = wm8991_i2c_id, +}; + +static int __init wm8991_modinit(void) +{ + int ret; + ret = i2c_add_driver(&wm8991_i2c_driver); + if (ret != 0) { + printk(KERN_ERR "Failed to register WM8991 I2C driver: %d\n", + ret); + } + return 0; +} +module_init(wm8991_modinit); + +static void __exit wm8991_exit(void) +{ + i2c_del_driver(&wm8991_i2c_driver); +} +module_exit(wm8991_exit); + +MODULE_DESCRIPTION("ASoC WM8991 driver"); +MODULE_AUTHOR("Graeme Gregory"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8991.h b/sound/soc/codecs/wm8991.h new file mode 100644 index 00000000000..8a942efd18a --- /dev/null +++ b/sound/soc/codecs/wm8991.h @@ -0,0 +1,833 @@ +/* + * wm8991.h -- audio driver for WM8991 + * + * Copyright 2007 Wolfson Microelectronics PLC. + * Author: Graeme Gregory + * graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.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. + */ + +#ifndef _WM8991_H +#define _WM8991_H + +/* + * Register values. + */ +#define WM8991_RESET 0x00 +#define WM8991_POWER_MANAGEMENT_1 0x01 +#define WM8991_POWER_MANAGEMENT_2 0x02 +#define WM8991_POWER_MANAGEMENT_3 0x03 +#define WM8991_AUDIO_INTERFACE_1 0x04 +#define WM8991_AUDIO_INTERFACE_2 0x05 +#define WM8991_CLOCKING_1 0x06 +#define WM8991_CLOCKING_2 0x07 +#define WM8991_AUDIO_INTERFACE_3 0x08 +#define WM8991_AUDIO_INTERFACE_4 0x09 +#define WM8991_DAC_CTRL 0x0A +#define WM8991_LEFT_DAC_DIGITAL_VOLUME 0x0B +#define WM8991_RIGHT_DAC_DIGITAL_VOLUME 0x0C +#define WM8991_DIGITAL_SIDE_TONE 0x0D +#define WM8991_ADC_CTRL 0x0E +#define WM8991_LEFT_ADC_DIGITAL_VOLUME 0x0F +#define WM8991_RIGHT_ADC_DIGITAL_VOLUME 0x10 +#define WM8991_GPIO_CTRL_1 0x12 +#define WM8991_GPIO1_GPIO2 0x13 +#define WM8991_GPIO3_GPIO4 0x14 +#define WM8991_GPIO5_GPIO6 0x15 +#define WM8991_GPIOCTRL_2 0x16 +#define WM8991_GPIO_POL 0x17 +#define WM8991_LEFT_LINE_INPUT_1_2_VOLUME 0x18 +#define WM8991_LEFT_LINE_INPUT_3_4_VOLUME 0x19 +#define WM8991_RIGHT_LINE_INPUT_1_2_VOLUME 0x1A +#define WM8991_RIGHT_LINE_INPUT_3_4_VOLUME 0x1B +#define WM8991_LEFT_OUTPUT_VOLUME 0x1C +#define WM8991_RIGHT_OUTPUT_VOLUME 0x1D +#define WM8991_LINE_OUTPUTS_VOLUME 0x1E +#define WM8991_OUT3_4_VOLUME 0x1F +#define WM8991_LEFT_OPGA_VOLUME 0x20 +#define WM8991_RIGHT_OPGA_VOLUME 0x21 +#define WM8991_SPEAKER_VOLUME 0x22 +#define WM8991_CLASSD1 0x23 +#define WM8991_CLASSD3 0x25 +#define WM8991_INPUT_MIXER1 0x27 +#define WM8991_INPUT_MIXER2 0x28 +#define WM8991_INPUT_MIXER3 0x29 +#define WM8991_INPUT_MIXER4 0x2A +#define WM8991_INPUT_MIXER5 0x2B +#define WM8991_INPUT_MIXER6 0x2C +#define WM8991_OUTPUT_MIXER1 0x2D +#define WM8991_OUTPUT_MIXER2 0x2E +#define WM8991_OUTPUT_MIXER3 0x2F +#define WM8991_OUTPUT_MIXER4 0x30 +#define WM8991_OUTPUT_MIXER5 0x31 +#define WM8991_OUTPUT_MIXER6 0x32 +#define WM8991_OUT3_4_MIXER 0x33 +#define WM8991_LINE_MIXER1 0x34 +#define WM8991_LINE_MIXER2 0x35 +#define WM8991_SPEAKER_MIXER 0x36 +#define WM8991_ADDITIONAL_CONTROL 0x37 +#define WM8991_ANTIPOP1 0x38 +#define WM8991_ANTIPOP2 0x39 +#define WM8991_MICBIAS 0x3A +#define WM8991_PLL1 0x3C +#define WM8991_PLL2 0x3D +#define WM8991_PLL3 0x3E +#define WM8991_INTDRIVBITS 0x3F + +#define WM8991_REGISTER_COUNT 60 +#define WM8991_MAX_REGISTER 0x3F + +/* + * Field Definitions. + */ + +/* + * R0 (0x00) - Reset + */ +#define WM8991_SW_RESET_CHIP_ID_MASK 0xFFFF /* SW_RESET_CHIP_ID - [15:0] */ + +/* + * R1 (0x01) - Power Management (1) + */ +#define WM8991_SPK_ENA 0x1000 /* SPK_ENA */ +#define WM8991_SPK_ENA_BIT 12 +#define WM8991_OUT3_ENA 0x0800 /* OUT3_ENA */ +#define WM8991_OUT3_ENA_BIT 11 +#define WM8991_OUT4_ENA 0x0400 /* OUT4_ENA */ +#define WM8991_OUT4_ENA_BIT 10 +#define WM8991_LOUT_ENA 0x0200 /* LOUT_ENA */ +#define WM8991_LOUT_ENA_BIT 9 +#define WM8991_ROUT_ENA 0x0100 /* ROUT_ENA */ +#define WM8991_ROUT_ENA_BIT 8 +#define WM8991_MICBIAS_ENA 0x0010 /* MICBIAS_ENA */ +#define WM8991_MICBIAS_ENA_BIT 4 +#define WM8991_VMID_MODE_MASK 0x0006 /* VMID_MODE - [2:1] */ +#define WM8991_VREF_ENA 0x0001 /* VREF_ENA */ +#define WM8991_VREF_ENA_BIT 0 + +/* + * R2 (0x02) - Power Management (2) + */ +#define WM8991_PLL_ENA 0x8000 /* PLL_ENA */ +#define WM8991_PLL_ENA_BIT 15 +#define WM8991_TSHUT_ENA 0x4000 /* TSHUT_ENA */ +#define WM8991_TSHUT_ENA_BIT 14 +#define WM8991_TSHUT_OPDIS 0x2000 /* TSHUT_OPDIS */ +#define WM8991_TSHUT_OPDIS_BIT 13 +#define WM8991_OPCLK_ENA 0x0800 /* OPCLK_ENA */ +#define WM8991_OPCLK_ENA_BIT 11 +#define WM8991_AINL_ENA 0x0200 /* AINL_ENA */ +#define WM8991_AINL_ENA_BIT 9 +#define WM8991_AINR_ENA 0x0100 /* AINR_ENA */ +#define WM8991_AINR_ENA_BIT 8 +#define WM8991_LIN34_ENA 0x0080 /* LIN34_ENA */ +#define WM8991_LIN34_ENA_BIT 7 +#define WM8991_LIN12_ENA 0x0040 /* LIN12_ENA */ +#define WM8991_LIN12_ENA_BIT 6 +#define WM8991_RIN34_ENA 0x0020 /* RIN34_ENA */ +#define WM8991_RIN34_ENA_BIT 5 +#define WM8991_RIN12_ENA 0x0010 /* RIN12_ENA */ +#define WM8991_RIN12_ENA_BIT 4 +#define WM8991_ADCL_ENA 0x0002 /* ADCL_ENA */ +#define WM8991_ADCL_ENA_BIT 1 +#define WM8991_ADCR_ENA 0x0001 /* ADCR_ENA */ +#define WM8991_ADCR_ENA_BIT 0 + +/* + * R3 (0x03) - Power Management (3) + */ +#define WM8991_LON_ENA 0x2000 /* LON_ENA */ +#define WM8991_LON_ENA_BIT 13 +#define WM8991_LOP_ENA 0x1000 /* LOP_ENA */ +#define WM8991_LOP_ENA_BIT 12 +#define WM8991_RON_ENA 0x0800 /* RON_ENA */ +#define WM8991_RON_ENA_BIT 11 +#define WM8991_ROP_ENA 0x0400 /* ROP_ENA */ +#define WM8991_ROP_ENA_BIT 10 +#define WM8991_LOPGA_ENA 0x0080 /* LOPGA_ENA */ +#define WM8991_LOPGA_ENA_BIT 7 +#define WM8991_ROPGA_ENA 0x0040 /* ROPGA_ENA */ +#define WM8991_ROPGA_ENA_BIT 6 +#define WM8991_LOMIX_ENA 0x0020 /* LOMIX_ENA */ +#define WM8991_LOMIX_ENA_BIT 5 +#define WM8991_ROMIX_ENA 0x0010 /* ROMIX_ENA */ +#define WM8991_ROMIX_ENA_BIT 4 +#define WM8991_DACL_ENA 0x0002 /* DACL_ENA */ +#define WM8991_DACL_ENA_BIT 1 +#define WM8991_DACR_ENA 0x0001 /* DACR_ENA */ +#define WM8991_DACR_ENA_BIT 0 + +/* + * R4 (0x04) - Audio Interface (1) + */ +#define WM8991_AIFADCL_SRC 0x8000 /* AIFADCL_SRC */ +#define WM8991_AIFADCR_SRC 0x4000 /* AIFADCR_SRC */ +#define WM8991_AIFADC_TDM 0x2000 /* AIFADC_TDM */ +#define WM8991_AIFADC_TDM_CHAN 0x1000 /* AIFADC_TDM_CHAN */ +#define WM8991_AIF_BCLK_INV 0x0100 /* AIF_BCLK_INV */ +#define WM8991_AIF_LRCLK_INV 0x0080 /* AIF_LRCLK_INV */ +#define WM8991_AIF_WL_MASK 0x0060 /* AIF_WL - [6:5] */ +#define WM8991_AIF_WL_16BITS (0 << 5) +#define WM8991_AIF_WL_20BITS (1 << 5) +#define WM8991_AIF_WL_24BITS (2 << 5) +#define WM8991_AIF_WL_32BITS (3 << 5) +#define WM8991_AIF_FMT_MASK 0x0018 /* AIF_FMT - [4:3] */ +#define WM8991_AIF_TMF_RIGHTJ (0 << 3) +#define WM8991_AIF_TMF_LEFTJ (1 << 3) +#define WM8991_AIF_TMF_I2S (2 << 3) +#define WM8991_AIF_TMF_DSP (3 << 3) + +/* + * R5 (0x05) - Audio Interface (2) + */ +#define WM8991_DACL_SRC 0x8000 /* DACL_SRC */ +#define WM8991_DACR_SRC 0x4000 /* DACR_SRC */ +#define WM8991_AIFDAC_TDM 0x2000 /* AIFDAC_TDM */ +#define WM8991_AIFDAC_TDM_CHAN 0x1000 /* AIFDAC_TDM_CHAN */ +#define WM8991_DAC_BOOST_MASK 0x0C00 /* DAC_BOOST - [11:10] */ +#define WM8991_DAC_COMP 0x0010 /* DAC_COMP */ +#define WM8991_DAC_COMPMODE 0x0008 /* DAC_COMPMODE */ +#define WM8991_ADC_COMP 0x0004 /* ADC_COMP */ +#define WM8991_ADC_COMPMODE 0x0002 /* ADC_COMPMODE */ +#define WM8991_LOOPBACK 0x0001 /* LOOPBACK */ + +/* + * R6 (0x06) - Clocking (1) + */ +#define WM8991_TOCLK_RATE 0x8000 /* TOCLK_RATE */ +#define WM8991_TOCLK_ENA 0x4000 /* TOCLK_ENA */ +#define WM8991_OPCLKDIV_MASK 0x1E00 /* OPCLKDIV - [12:9] */ +#define WM8991_DCLKDIV_MASK 0x01C0 /* DCLKDIV - [8:6] */ +#define WM8991_BCLK_DIV_MASK 0x001E /* BCLK_DIV - [4:1] */ +#define WM8991_BCLK_DIV_1 (0x0 << 1) +#define WM8991_BCLK_DIV_1_5 (0x1 << 1) +#define WM8991_BCLK_DIV_2 (0x2 << 1) +#define WM8991_BCLK_DIV_3 (0x3 << 1) +#define WM8991_BCLK_DIV_4 (0x4 << 1) +#define WM8991_BCLK_DIV_5_5 (0x5 << 1) +#define WM8991_BCLK_DIV_6 (0x6 << 1) +#define WM8991_BCLK_DIV_8 (0x7 << 1) +#define WM8991_BCLK_DIV_11 (0x8 << 1) +#define WM8991_BCLK_DIV_12 (0x9 << 1) +#define WM8991_BCLK_DIV_16 (0xA << 1) +#define WM8991_BCLK_DIV_22 (0xB << 1) +#define WM8991_BCLK_DIV_24 (0xC << 1) +#define WM8991_BCLK_DIV_32 (0xD << 1) +#define WM8991_BCLK_DIV_44 (0xE << 1) +#define WM8991_BCLK_DIV_48 (0xF << 1) + +/* + * R7 (0x07) - Clocking (2) + */ +#define WM8991_MCLK_SRC 0x8000 /* MCLK_SRC */ +#define WM8991_SYSCLK_SRC 0x4000 /* SYSCLK_SRC */ +#define WM8991_CLK_FORCE 0x2000 /* CLK_FORCE */ +#define WM8991_MCLK_DIV_MASK 0x1800 /* MCLK_DIV - [12:11] */ +#define WM8991_MCLK_DIV_1 (0 << 11) +#define WM8991_MCLK_DIV_2 ( 2 << 11) +#define WM8991_MCLK_INV 0x0400 /* MCLK_INV */ +#define WM8991_ADC_CLKDIV_MASK 0x00E0 /* ADC_CLKDIV - [7:5] */ +#define WM8991_ADC_CLKDIV_1 (0 << 5) +#define WM8991_ADC_CLKDIV_1_5 (1 << 5) +#define WM8991_ADC_CLKDIV_2 (2 << 5) +#define WM8991_ADC_CLKDIV_3 (3 << 5) +#define WM8991_ADC_CLKDIV_4 (4 << 5) +#define WM8991_ADC_CLKDIV_5_5 (5 << 5) +#define WM8991_ADC_CLKDIV_6 (6 << 5) +#define WM8991_DAC_CLKDIV_MASK 0x001C /* DAC_CLKDIV - [4:2] */ +#define WM8991_DAC_CLKDIV_1 (0 << 2) +#define WM8991_DAC_CLKDIV_1_5 (1 << 2) +#define WM8991_DAC_CLKDIV_2 (2 << 2) +#define WM8991_DAC_CLKDIV_3 (3 << 2) +#define WM8991_DAC_CLKDIV_4 (4 << 2) +#define WM8991_DAC_CLKDIV_5_5 (5 << 2) +#define WM8991_DAC_CLKDIV_6 (6 << 2) + +/* + * R8 (0x08) - Audio Interface (3) + */ +#define WM8991_AIF_MSTR1 0x8000 /* AIF_MSTR1 */ +#define WM8991_AIF_MSTR2 0x4000 /* AIF_MSTR2 */ +#define WM8991_AIF_SEL 0x2000 /* AIF_SEL */ +#define WM8991_ADCLRC_DIR 0x0800 /* ADCLRC_DIR */ +#define WM8991_ADCLRC_RATE_MASK 0x07FF /* ADCLRC_RATE - [10:0] */ + +/* + * R9 (0x09) - Audio Interface (4) + */ +#define WM8991_ALRCGPIO1 0x8000 /* ALRCGPIO1 */ +#define WM8991_ALRCBGPIO6 0x4000 /* ALRCBGPIO6 */ +#define WM8991_AIF_TRIS 0x2000 /* AIF_TRIS */ +#define WM8991_DACLRC_DIR 0x0800 /* DACLRC_DIR */ +#define WM8991_DACLRC_RATE_MASK 0x07FF /* DACLRC_RATE - [10:0] */ + +/* + * R10 (0x0A) - DAC CTRL + */ +#define WM8991_AIF_LRCLKRATE 0x0400 /* AIF_LRCLKRATE */ +#define WM8991_DAC_MONO 0x0200 /* DAC_MONO */ +#define WM8991_DAC_SB_FILT 0x0100 /* DAC_SB_FILT */ +#define WM8991_DAC_MUTERATE 0x0080 /* DAC_MUTERATE */ +#define WM8991_DAC_MUTEMODE 0x0040 /* DAC_MUTEMODE */ +#define WM8991_DEEMP_MASK 0x0030 /* DEEMP - [5:4] */ +#define WM8991_DAC_MUTE 0x0004 /* DAC_MUTE */ +#define WM8991_DACL_DATINV 0x0002 /* DACL_DATINV */ +#define WM8991_DACR_DATINV 0x0001 /* DACR_DATINV */ + +/* + * R11 (0x0B) - Left DAC Digital Volume + */ +#define WM8991_DAC_VU 0x0100 /* DAC_VU */ +#define WM8991_DACL_VOL_MASK 0x00FF /* DACL_VOL - [7:0] */ +#define WM8991_DACL_VOL_SHIFT 0 +/* + * R12 (0x0C) - Right DAC Digital Volume + */ +#define WM8991_DAC_VU 0x0100 /* DAC_VU */ +#define WM8991_DACR_VOL_MASK 0x00FF /* DACR_VOL - [7:0] */ +#define WM8991_DACR_VOL_SHIFT 0 +/* + * R13 (0x0D) - Digital Side Tone + */ +#define WM8991_ADCL_DAC_SVOL_MASK 0x0F /* ADCL_DAC_SVOL - [12:9] */ +#define WM8991_ADCL_DAC_SVOL_SHIFT 9 +#define WM8991_ADCR_DAC_SVOL_MASK 0x0F /* ADCR_DAC_SVOL - [8:5] */ +#define WM8991_ADCR_DAC_SVOL_SHIFT 5 +#define WM8991_ADC_TO_DACL_MASK 0x03 /* ADC_TO_DACL - [3:2] */ +#define WM8991_ADC_TO_DACL_SHIFT 2 +#define WM8991_ADC_TO_DACR_MASK 0x03 /* ADC_TO_DACR - [1:0] */ +#define WM8991_ADC_TO_DACR_SHIFT 0 + +/* + * R14 (0x0E) - ADC CTRL + */ +#define WM8991_ADC_HPF_ENA 0x0100 /* ADC_HPF_ENA */ +#define WM8991_ADC_HPF_ENA_BIT 8 +#define WM8991_ADC_HPF_CUT_MASK 0x03 /* ADC_HPF_CUT - [6:5] */ +#define WM8991_ADC_HPF_CUT_SHIFT 5 +#define WM8991_ADCL_DATINV 0x0002 /* ADCL_DATINV */ +#define WM8991_ADCL_DATINV_BIT 1 +#define WM8991_ADCR_DATINV 0x0001 /* ADCR_DATINV */ +#define WM8991_ADCR_DATINV_BIT 0 + +/* + * R15 (0x0F) - Left ADC Digital Volume + */ +#define WM8991_ADC_VU 0x0100 /* ADC_VU */ +#define WM8991_ADCL_VOL_MASK 0x00FF /* ADCL_VOL - [7:0] */ +#define WM8991_ADCL_VOL_SHIFT 0 + +/* + * R16 (0x10) - Right ADC Digital Volume + */ +#define WM8991_ADC_VU 0x0100 /* ADC_VU */ +#define WM8991_ADCR_VOL_MASK 0x00FF /* ADCR_VOL - [7:0] */ +#define WM8991_ADCR_VOL_SHIFT 0 + +/* + * R18 (0x12) - GPIO CTRL 1 + */ +#define WM8991_IRQ 0x1000 /* IRQ */ +#define WM8991_TEMPOK 0x0800 /* TEMPOK */ +#define WM8991_MICSHRT 0x0400 /* MICSHRT */ +#define WM8991_MICDET 0x0200 /* MICDET */ +#define WM8991_PLL_LCK 0x0100 /* PLL_LCK */ +#define WM8991_GPI8_STATUS 0x0080 /* GPI8_STATUS */ +#define WM8991_GPI7_STATUS 0x0040 /* GPI7_STATUS */ +#define WM8991_GPIO6_STATUS 0x0020 /* GPIO6_STATUS */ +#define WM8991_GPIO5_STATUS 0x0010 /* GPIO5_STATUS */ +#define WM8991_GPIO4_STATUS 0x0008 /* GPIO4_STATUS */ +#define WM8991_GPIO3_STATUS 0x0004 /* GPIO3_STATUS */ +#define WM8991_GPIO2_STATUS 0x0002 /* GPIO2_STATUS */ +#define WM8991_GPIO1_STATUS 0x0001 /* GPIO1_STATUS */ + +/* + * R19 (0x13) - GPIO1 & GPIO2 + */ +#define WM8991_GPIO2_DEB_ENA 0x8000 /* GPIO2_DEB_ENA */ +#define WM8991_GPIO2_IRQ_ENA 0x4000 /* GPIO2_IRQ_ENA */ +#define WM8991_GPIO2_PU 0x2000 /* GPIO2_PU */ +#define WM8991_GPIO2_PD 0x1000 /* GPIO2_PD */ +#define WM8991_GPIO2_SEL_MASK 0x0F00 /* GPIO2_SEL - [11:8] */ +#define WM8991_GPIO1_DEB_ENA 0x0080 /* GPIO1_DEB_ENA */ +#define WM8991_GPIO1_IRQ_ENA 0x0040 /* GPIO1_IRQ_ENA */ +#define WM8991_GPIO1_PU 0x0020 /* GPIO1_PU */ +#define WM8991_GPIO1_PD 0x0010 /* GPIO1_PD */ +#define WM8991_GPIO1_SEL_MASK 0x000F /* GPIO1_SEL - [3:0] */ + +/* + * R20 (0x14) - GPIO3 & GPIO4 + */ +#define WM8991_GPIO4_DEB_ENA 0x8000 /* GPIO4_DEB_ENA */ +#define WM8991_GPIO4_IRQ_ENA 0x4000 /* GPIO4_IRQ_ENA */ +#define WM8991_GPIO4_PU 0x2000 /* GPIO4_PU */ +#define WM8991_GPIO4_PD 0x1000 /* GPIO4_PD */ +#define WM8991_GPIO4_SEL_MASK 0x0F00 /* GPIO4_SEL - [11:8] */ +#define WM8991_GPIO3_DEB_ENA 0x0080 /* GPIO3_DEB_ENA */ +#define WM8991_GPIO3_IRQ_ENA 0x0040 /* GPIO3_IRQ_ENA */ +#define WM8991_GPIO3_PU 0x0020 /* GPIO3_PU */ +#define WM8991_GPIO3_PD 0x0010 /* GPIO3_PD */ +#define WM8991_GPIO3_SEL_MASK 0x000F /* GPIO3_SEL - [3:0] */ + +/* + * R21 (0x15) - GPIO5 & GPIO6 + */ +#define WM8991_GPIO6_DEB_ENA 0x8000 /* GPIO6_DEB_ENA */ +#define WM8991_GPIO6_IRQ_ENA 0x4000 /* GPIO6_IRQ_ENA */ +#define WM8991_GPIO6_PU 0x2000 /* GPIO6_PU */ +#define WM8991_GPIO6_PD 0x1000 /* GPIO6_PD */ +#define WM8991_GPIO6_SEL_MASK 0x0F00 /* GPIO6_SEL - [11:8] */ +#define WM8991_GPIO5_DEB_ENA 0x0080 /* GPIO5_DEB_ENA */ +#define WM8991_GPIO5_IRQ_ENA 0x0040 /* GPIO5_IRQ_ENA */ +#define WM8991_GPIO5_PU 0x0020 /* GPIO5_PU */ +#define WM8991_GPIO5_PD 0x0010 /* GPIO5_PD */ +#define WM8991_GPIO5_SEL_MASK 0x000F /* GPIO5_SEL - [3:0] */ + +/* + * R22 (0x16) - GPIOCTRL 2 + */ +#define WM8991_RD_3W_ENA 0x8000 /* RD_3W_ENA */ +#define WM8991_MODE_3W4W 0x4000 /* MODE_3W4W */ +#define WM8991_TEMPOK_IRQ_ENA 0x0800 /* TEMPOK_IRQ_ENA */ +#define WM8991_MICSHRT_IRQ_ENA 0x0400 /* MICSHRT_IRQ_ENA */ +#define WM8991_MICDET_IRQ_ENA 0x0200 /* MICDET_IRQ_ENA */ +#define WM8991_PLL_LCK_IRQ_ENA 0x0100 /* PLL_LCK_IRQ_ENA */ +#define WM8991_GPI8_DEB_ENA 0x0080 /* GPI8_DEB_ENA */ +#define WM8991_GPI8_IRQ_ENA 0x0040 /* GPI8_IRQ_ENA */ +#define WM8991_GPI8_ENA 0x0010 /* GPI8_ENA */ +#define WM8991_GPI7_DEB_ENA 0x0008 /* GPI7_DEB_ENA */ +#define WM8991_GPI7_IRQ_ENA 0x0004 /* GPI7_IRQ_ENA */ +#define WM8991_GPI7_ENA 0x0001 /* GPI7_ENA */ + +/* + * R23 (0x17) - GPIO_POL + */ +#define WM8991_IRQ_INV 0x1000 /* IRQ_INV */ +#define WM8991_TEMPOK_POL 0x0800 /* TEMPOK_POL */ +#define WM8991_MICSHRT_POL 0x0400 /* MICSHRT_POL */ +#define WM8991_MICDET_POL 0x0200 /* MICDET_POL */ +#define WM8991_PLL_LCK_POL 0x0100 /* PLL_LCK_POL */ +#define WM8991_GPI8_POL 0x0080 /* GPI8_POL */ +#define WM8991_GPI7_POL 0x0040 /* GPI7_POL */ +#define WM8991_GPIO6_POL 0x0020 /* GPIO6_POL */ +#define WM8991_GPIO5_POL 0x0010 /* GPIO5_POL */ +#define WM8991_GPIO4_POL 0x0008 /* GPIO4_POL */ +#define WM8991_GPIO3_POL 0x0004 /* GPIO3_POL */ +#define WM8991_GPIO2_POL 0x0002 /* GPIO2_POL */ +#define WM8991_GPIO1_POL 0x0001 /* GPIO1_POL */ + +/* + * R24 (0x18) - Left Line Input 1&2 Volume + */ +#define WM8991_IPVU 0x0100 /* IPVU */ +#define WM8991_LI12MUTE 0x0080 /* LI12MUTE */ +#define WM8991_LI12MUTE_BIT 7 +#define WM8991_LI12ZC 0x0040 /* LI12ZC */ +#define WM8991_LI12ZC_BIT 6 +#define WM8991_LIN12VOL_MASK 0x001F /* LIN12VOL - [4:0] */ +#define WM8991_LIN12VOL_SHIFT 0 +/* + * R25 (0x19) - Left Line Input 3&4 Volume + */ +#define WM8991_IPVU 0x0100 /* IPVU */ +#define WM8991_LI34MUTE 0x0080 /* LI34MUTE */ +#define WM8991_LI34MUTE_BIT 7 +#define WM8991_LI34ZC 0x0040 /* LI34ZC */ +#define WM8991_LI34ZC_BIT 6 +#define WM8991_LIN34VOL_MASK 0x001F /* LIN34VOL - [4:0] */ +#define WM8991_LIN34VOL_SHIFT 0 + +/* + * R26 (0x1A) - Right Line Input 1&2 Volume + */ +#define WM8991_IPVU 0x0100 /* IPVU */ +#define WM8991_RI12MUTE 0x0080 /* RI12MUTE */ +#define WM8991_RI12MUTE_BIT 7 +#define WM8991_RI12ZC 0x0040 /* RI12ZC */ +#define WM8991_RI12ZC_BIT 6 +#define WM8991_RIN12VOL_MASK 0x001F /* RIN12VOL - [4:0] */ +#define WM8991_RIN12VOL_SHIFT 0 + +/* + * R27 (0x1B) - Right Line Input 3&4 Volume + */ +#define WM8991_IPVU 0x0100 /* IPVU */ +#define WM8991_RI34MUTE 0x0080 /* RI34MUTE */ +#define WM8991_RI34MUTE_BIT 7 +#define WM8991_RI34ZC 0x0040 /* RI34ZC */ +#define WM8991_RI34ZC_BIT 6 +#define WM8991_RIN34VOL_MASK 0x001F /* RIN34VOL - [4:0] */ +#define WM8991_RIN34VOL_SHIFT 0 + +/* + * R28 (0x1C) - Left Output Volume + */ +#define WM8991_OPVU 0x0100 /* OPVU */ +#define WM8991_LOZC 0x0080 /* LOZC */ +#define WM8991_LOZC_BIT 7 +#define WM8991_LOUTVOL_MASK 0x007F /* LOUTVOL - [6:0] */ +#define WM8991_LOUTVOL_SHIFT 0 +/* + * R29 (0x1D) - Right Output Volume + */ +#define WM8991_OPVU 0x0100 /* OPVU */ +#define WM8991_ROZC 0x0080 /* ROZC */ +#define WM8991_ROZC_BIT 7 +#define WM8991_ROUTVOL_MASK 0x007F /* ROUTVOL - [6:0] */ +#define WM8991_ROUTVOL_SHIFT 0 +/* + * R30 (0x1E) - Line Outputs Volume + */ +#define WM8991_LONMUTE 0x0040 /* LONMUTE */ +#define WM8991_LONMUTE_BIT 6 +#define WM8991_LOPMUTE 0x0020 /* LOPMUTE */ +#define WM8991_LOPMUTE_BIT 5 +#define WM8991_LOATTN 0x0010 /* LOATTN */ +#define WM8991_LOATTN_BIT 4 +#define WM8991_RONMUTE 0x0004 /* RONMUTE */ +#define WM8991_RONMUTE_BIT 2 +#define WM8991_ROPMUTE 0x0002 /* ROPMUTE */ +#define WM8991_ROPMUTE_BIT 1 +#define WM8991_ROATTN 0x0001 /* ROATTN */ +#define WM8991_ROATTN_BIT 0 + +/* + * R31 (0x1F) - Out3/4 Volume + */ +#define WM8991_OUT3MUTE 0x0020 /* OUT3MUTE */ +#define WM8991_OUT3MUTE_BIT 5 +#define WM8991_OUT3ATTN 0x0010 /* OUT3ATTN */ +#define WM8991_OUT3ATTN_BIT 4 +#define WM8991_OUT4MUTE 0x0002 /* OUT4MUTE */ +#define WM8991_OUT4MUTE_BIT 1 +#define WM8991_OUT4ATTN 0x0001 /* OUT4ATTN */ +#define WM8991_OUT4ATTN_BIT 0 + +/* + * R32 (0x20) - Left OPGA Volume + */ +#define WM8991_OPVU 0x0100 /* OPVU */ +#define WM8991_LOPGAZC 0x0080 /* LOPGAZC */ +#define WM8991_LOPGAZC_BIT 7 +#define WM8991_LOPGAVOL_MASK 0x007F /* LOPGAVOL - [6:0] */ +#define WM8991_LOPGAVOL_SHIFT 0 + +/* + * R33 (0x21) - Right OPGA Volume + */ +#define WM8991_OPVU 0x0100 /* OPVU */ +#define WM8991_ROPGAZC 0x0080 /* ROPGAZC */ +#define WM8991_ROPGAZC_BIT 7 +#define WM8991_ROPGAVOL_MASK 0x007F /* ROPGAVOL - [6:0] */ +#define WM8991_ROPGAVOL_SHIFT 0 +/* + * R34 (0x22) - Speaker Volume + */ +#define WM8991_SPKVOL_MASK 0x0003 /* SPKVOL - [1:0] */ +#define WM8991_SPKVOL_SHIFT 0 + +/* + * R35 (0x23) - ClassD1 + */ +#define WM8991_CDMODE 0x0100 /* CDMODE */ +#define WM8991_CDMODE_BIT 8 + +/* + * R37 (0x25) - ClassD3 + */ +#define WM8991_DCGAIN_MASK 0x0007 /* DCGAIN - [5:3] */ +#define WM8991_DCGAIN_SHIFT 3 +#define WM8991_ACGAIN_MASK 0x0007 /* ACGAIN - [2:0] */ +#define WM8991_ACGAIN_SHIFT 0 +/* + * R39 (0x27) - Input Mixer1 + */ +#define WM8991_AINLMODE_MASK 0x000C /* AINLMODE - [3:2] */ +#define WM8991_AINLMODE_SHIFT 2 +#define WM8991_AINRMODE_MASK 0x0003 /* AINRMODE - [1:0] */ +#define WM8991_AINRMODE_SHIFT 0 + +/* + * R40 (0x28) - Input Mixer2 + */ +#define WM8991_LMP4 0x0080 /* LMP4 */ +#define WM8991_LMP4_BIT 7 /* LMP4 */ +#define WM8991_LMN3 0x0040 /* LMN3 */ +#define WM8991_LMN3_BIT 6 /* LMN3 */ +#define WM8991_LMP2 0x0020 /* LMP2 */ +#define WM8991_LMP2_BIT 5 /* LMP2 */ +#define WM8991_LMN1 0x0010 /* LMN1 */ +#define WM8991_LMN1_BIT 4 /* LMN1 */ +#define WM8991_RMP4 0x0008 /* RMP4 */ +#define WM8991_RMP4_BIT 3 /* RMP4 */ +#define WM8991_RMN3 0x0004 /* RMN3 */ +#define WM8991_RMN3_BIT 2 /* RMN3 */ +#define WM8991_RMP2 0x0002 /* RMP2 */ +#define WM8991_RMP2_BIT 1 /* RMP2 */ +#define WM8991_RMN1 0x0001 /* RMN1 */ +#define WM8991_RMN1_BIT 0 /* RMN1 */ + +/* + * R41 (0x29) - Input Mixer3 + */ +#define WM8991_L34MNB 0x0100 /* L34MNB */ +#define WM8991_L34MNB_BIT 8 +#define WM8991_L34MNBST 0x0080 /* L34MNBST */ +#define WM8991_L34MNBST_BIT 7 +#define WM8991_L12MNB 0x0020 /* L12MNB */ +#define WM8991_L12MNB_BIT 5 +#define WM8991_L12MNBST 0x0010 /* L12MNBST */ +#define WM8991_L12MNBST_BIT 4 +#define WM8991_LDBVOL_MASK 0x0007 /* LDBVOL - [2:0] */ +#define WM8991_LDBVOL_SHIFT 0 + +/* + * R42 (0x2A) - Input Mixer4 + */ +#define WM8991_R34MNB 0x0100 /* R34MNB */ +#define WM8991_R34MNB_BIT 8 +#define WM8991_R34MNBST 0x0080 /* R34MNBST */ +#define WM8991_R34MNBST_BIT 7 +#define WM8991_R12MNB 0x0020 /* R12MNB */ +#define WM8991_R12MNB_BIT 5 +#define WM8991_R12MNBST 0x0010 /* R12MNBST */ +#define WM8991_R12MNBST_BIT 4 +#define WM8991_RDBVOL_MASK 0x0007 /* RDBVOL - [2:0] */ +#define WM8991_RDBVOL_SHIFT 0 + +/* + * R43 (0x2B) - Input Mixer5 + */ +#define WM8991_LI2BVOL_MASK 0x07 /* LI2BVOL - [8:6] */ +#define WM8991_LI2BVOL_SHIFT 6 +#define WM8991_LR4BVOL_MASK 0x07 /* LR4BVOL - [5:3] */ +#define WM8991_LR4BVOL_SHIFT 3 +#define WM8991_LL4BVOL_MASK 0x07 /* LL4BVOL - [2:0] */ +#define WM8991_LL4BVOL_SHIFT 0 + +/* + * R44 (0x2C) - Input Mixer6 + */ +#define WM8991_RI2BVOL_MASK 0x07 /* RI2BVOL - [8:6] */ +#define WM8991_RI2BVOL_SHIFT 6 +#define WM8991_RL4BVOL_MASK 0x07 /* RL4BVOL - [5:3] */ +#define WM8991_RL4BVOL_SHIFT 3 +#define WM8991_RR4BVOL_MASK 0x07 /* RR4BVOL - [2:0] */ +#define WM8991_RR4BVOL_SHIFT 0 + +/* + * R45 (0x2D) - Output Mixer1 + */ +#define WM8991_LRBLO 0x0080 /* LRBLO */ +#define WM8991_LRBLO_BIT 7 +#define WM8991_LLBLO 0x0040 /* LLBLO */ +#define WM8991_LLBLO_BIT 6 +#define WM8991_LRI3LO 0x0020 /* LRI3LO */ +#define WM8991_LRI3LO_BIT 5 +#define WM8991_LLI3LO 0x0010 /* LLI3LO */ +#define WM8991_LLI3LO_BIT 4 +#define WM8991_LR12LO 0x0008 /* LR12LO */ +#define WM8991_LR12LO_BIT 3 +#define WM8991_LL12LO 0x0004 /* LL12LO */ +#define WM8991_LL12LO_BIT 2 +#define WM8991_LDLO 0x0001 /* LDLO */ +#define WM8991_LDLO_BIT 0 + +/* + * R46 (0x2E) - Output Mixer2 + */ +#define WM8991_RLBRO 0x0080 /* RLBRO */ +#define WM8991_RLBRO_BIT 7 +#define WM8991_RRBRO 0x0040 /* RRBRO */ +#define WM8991_RRBRO_BIT 6 +#define WM8991_RLI3RO 0x0020 /* RLI3RO */ +#define WM8991_RLI3RO_BIT 5 +#define WM8991_RRI3RO 0x0010 /* RRI3RO */ +#define WM8991_RRI3RO_BIT 4 +#define WM8991_RL12RO 0x0008 /* RL12RO */ +#define WM8991_RL12RO_BIT 3 +#define WM8991_RR12RO 0x0004 /* RR12RO */ +#define WM8991_RR12RO_BIT 2 +#define WM8991_RDRO 0x0001 /* RDRO */ +#define WM8991_RDRO_BIT 0 + +/* + * R47 (0x2F) - Output Mixer3 + */ +#define WM8991_LLI3LOVOL_MASK 0x07 /* LLI3LOVOL - [8:6] */ +#define WM8991_LLI3LOVOL_SHIFT 6 +#define WM8991_LR12LOVOL_MASK 0x07 /* LR12LOVOL - [5:3] */ +#define WM8991_LR12LOVOL_SHIFT 3 +#define WM8991_LL12LOVOL_MASK 0x07 /* LL12LOVOL - [2:0] */ +#define WM8991_LL12LOVOL_SHIFT 0 + +/* + * R48 (0x30) - Output Mixer4 + */ +#define WM8991_RRI3ROVOL_MASK 0x07 /* RRI3ROVOL - [8:6] */ +#define WM8991_RRI3ROVOL_SHIFT 6 +#define WM8991_RL12ROVOL_MASK 0x07 /* RL12ROVOL - [5:3] */ +#define WM8991_RL12ROVOL_SHIFT 3 +#define WM8991_RR12ROVOL_MASK 0x07 /* RR12ROVOL - [2:0] */ +#define WM8991_RR12ROVOL_SHIFT 0 + +/* + * R49 (0x31) - Output Mixer5 + */ +#define WM8991_LRI3LOVOL_MASK 0x07 /* LRI3LOVOL - [8:6] */ +#define WM8991_LRI3LOVOL_SHIFT 6 +#define WM8991_LRBLOVOL_MASK 0x07 /* LRBLOVOL - [5:3] */ +#define WM8991_LRBLOVOL_SHIFT 3 +#define WM8991_LLBLOVOL_MASK 0x07 /* LLBLOVOL - [2:0] */ +#define WM8991_LLBLOVOL_SHIFT 0 + +/* + * R50 (0x32) - Output Mixer6 + */ +#define WM8991_RLI3ROVOL_MASK 0x07 /* RLI3ROVOL - [8:6] */ +#define WM8991_RLI3ROVOL_SHIFT 6 +#define WM8991_RLBROVOL_MASK 0x07 /* RLBROVOL - [5:3] */ +#define WM8991_RLBROVOL_SHIFT 3 +#define WM8991_RRBROVOL_MASK 0x07 /* RRBROVOL - [2:0] */ +#define WM8991_RRBROVOL_SHIFT 0 + +/* + * R51 (0x33) - Out3/4 Mixer + */ +#define WM8991_VSEL_MASK 0x0180 /* VSEL - [8:7] */ +#define WM8991_LI4O3 0x0020 /* LI4O3 */ +#define WM8991_LI4O3_BIT 5 +#define WM8991_LPGAO3 0x0010 /* LPGAO3 */ +#define WM8991_LPGAO3_BIT 4 +#define WM8991_RI4O4 0x0002 /* RI4O4 */ +#define WM8991_RI4O4_BIT 1 +#define WM8991_RPGAO4 0x0001 /* RPGAO4 */ +#define WM8991_RPGAO4_BIT 0 +/* + * R52 (0x34) - Line Mixer1 + */ +#define WM8991_LLOPGALON 0x0040 /* LLOPGALON */ +#define WM8991_LLOPGALON_BIT 6 +#define WM8991_LROPGALON 0x0020 /* LROPGALON */ +#define WM8991_LROPGALON_BIT 5 +#define WM8991_LOPLON 0x0010 /* LOPLON */ +#define WM8991_LOPLON_BIT 4 +#define WM8991_LR12LOP 0x0004 /* LR12LOP */ +#define WM8991_LR12LOP_BIT 2 +#define WM8991_LL12LOP 0x0002 /* LL12LOP */ +#define WM8991_LL12LOP_BIT 1 +#define WM8991_LLOPGALOP 0x0001 /* LLOPGALOP */ +#define WM8991_LLOPGALOP_BIT 0 +/* + * R53 (0x35) - Line Mixer2 + */ +#define WM8991_RROPGARON 0x0040 /* RROPGARON */ +#define WM8991_RROPGARON_BIT 6 +#define WM8991_RLOPGARON 0x0020 /* RLOPGARON */ +#define WM8991_RLOPGARON_BIT 5 +#define WM8991_ROPRON 0x0010 /* ROPRON */ +#define WM8991_ROPRON_BIT 4 +#define WM8991_RL12ROP 0x0004 /* RL12ROP */ +#define WM8991_RL12ROP_BIT 2 +#define WM8991_RR12ROP 0x0002 /* RR12ROP */ +#define WM8991_RR12ROP_BIT 1 +#define WM8991_RROPGAROP 0x0001 /* RROPGAROP */ +#define WM8991_RROPGAROP_BIT 0 + +/* + * R54 (0x36) - Speaker Mixer + */ +#define WM8991_LB2SPK 0x0080 /* LB2SPK */ +#define WM8991_LB2SPK_BIT 7 +#define WM8991_RB2SPK 0x0040 /* RB2SPK */ +#define WM8991_RB2SPK_BIT 6 +#define WM8991_LI2SPK 0x0020 /* LI2SPK */ +#define WM8991_LI2SPK_BIT 5 +#define WM8991_RI2SPK 0x0010 /* RI2SPK */ +#define WM8991_RI2SPK_BIT 4 +#define WM8991_LOPGASPK 0x0008 /* LOPGASPK */ +#define WM8991_LOPGASPK_BIT 3 +#define WM8991_ROPGASPK 0x0004 /* ROPGASPK */ +#define WM8991_ROPGASPK_BIT 2 +#define WM8991_LDSPK 0x0002 /* LDSPK */ +#define WM8991_LDSPK_BIT 1 +#define WM8991_RDSPK 0x0001 /* RDSPK */ +#define WM8991_RDSPK_BIT 0 + +/* + * R55 (0x37) - Additional Control + */ +#define WM8991_VROI 0x0001 /* VROI */ + +/* + * R56 (0x38) - AntiPOP1 + */ +#define WM8991_DIS_LLINE 0x0020 /* DIS_LLINE */ +#define WM8991_DIS_RLINE 0x0010 /* DIS_RLINE */ +#define WM8991_DIS_OUT3 0x0008 /* DIS_OUT3 */ +#define WM8991_DIS_OUT4 0x0004 /* DIS_OUT4 */ +#define WM8991_DIS_LOUT 0x0002 /* DIS_LOUT */ +#define WM8991_DIS_ROUT 0x0001 /* DIS_ROUT */ + +/* + * R57 (0x39) - AntiPOP2 + */ +#define WM8991_SOFTST 0x0040 /* SOFTST */ +#define WM8991_BUFIOEN 0x0008 /* BUFIOEN */ +#define WM8991_BUFDCOPEN 0x0004 /* BUFDCOPEN */ +#define WM8991_POBCTRL 0x0002 /* POBCTRL */ +#define WM8991_VMIDTOG 0x0001 /* VMIDTOG */ + +/* + * R58 (0x3A) - MICBIAS + */ +#define WM8991_MCDSCTH_MASK 0x00C0 /* MCDSCTH - [7:6] */ +#define WM8991_MCDTHR_MASK 0x0038 /* MCDTHR - [5:3] */ +#define WM8991_MCD 0x0004 /* MCD */ +#define WM8991_MBSEL 0x0001 /* MBSEL */ + +/* + * R60 (0x3C) - PLL1 + */ +#define WM8991_SDM 0x0080 /* SDM */ +#define WM8991_PRESCALE 0x0040 /* PRESCALE */ +#define WM8991_PLLN_MASK 0x000F /* PLLN - [3:0] */ + +/* + * R61 (0x3D) - PLL2 + */ +#define WM8991_PLLK1_MASK 0x00FF /* PLLK1 - [7:0] */ + +/* + * R62 (0x3E) - PLL3 + */ +#define WM8991_PLLK2_MASK 0x00FF /* PLLK2 - [7:0] */ + +/* + * R63 (0x3F) - Internal Driver Bits + */ +#define WM8991_INMIXL_PWR_BIT 0 +#define WM8991_AINLMUX_PWR_BIT 1 +#define WM8991_INMIXR_PWR_BIT 2 +#define WM8991_AINRMUX_PWR_BIT 3 + +#define WM8991_MCLK_DIV 0 +#define WM8991_DACCLK_DIV 1 +#define WM8991_ADCCLK_DIV 2 +#define WM8991_BCLK_DIV 3 + +#define SOC_WM899X_OUTPGA_SINGLE_R_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_get_volsw, .put = wm899x_outpga_put_volsw_vu, \ + .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } + +#endif /* _WM8991_H */ -- cgit v1.2.3-70-g09d2 From a1b3b5eeeebac8acfa7838ef90f5a00a6f9188a0 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 24 Dec 2010 16:59:30 +0000 Subject: ASoC: Avoid direct register cache access when setting defaults Directly accessing the register cache means that we can't use anything except a flat register cache so use snd_soc_update_bits(). Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8523.c | 6 +++--- sound/soc/codecs/wm8741.c | 13 ++++++++----- sound/soc/codecs/wm8904.c | 41 ++++++++++++++++++++++++++--------------- sound/soc/codecs/wm8955.c | 27 +++++++++++++++++++-------- sound/soc/codecs/wm8962.c | 30 ++++++++++++++++++++---------- sound/soc/codecs/wm8978.c | 2 +- sound/soc/codecs/wm9090.c | 41 ++++++++++++++++++++++++----------------- 7 files changed, 101 insertions(+), 59 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c index 83e86f077ee..4fd4d8dca0f 100644 --- a/sound/soc/codecs/wm8523.c +++ b/sound/soc/codecs/wm8523.c @@ -414,7 +414,6 @@ static int wm8523_resume(struct snd_soc_codec *codec) static int wm8523_probe(struct snd_soc_codec *codec) { struct wm8523_priv *wm8523 = snd_soc_codec_get_drvdata(codec); - u16 *reg_cache = codec->reg_cache; int ret, i; codec->hw_write = (hw_write_t)i2c_master_send; @@ -471,8 +470,9 @@ static int wm8523_probe(struct snd_soc_codec *codec) } /* Change some default settings - latch VU and enable ZC */ - reg_cache[WM8523_DAC_GAINR] |= WM8523_DACR_VU; - reg_cache[WM8523_DAC_CTRL3] |= WM8523_ZC; + snd_soc_update_bits(codec, WM8523_DAC_GAINR, + WM8523_DACR_VU, WM8523_DACR_VU); + snd_soc_update_bits(codec, WM8523_DAC_CTRL3, WM8523_ZC, WM8523_ZC); wm8523_set_bias_level(codec, SND_SOC_BIAS_STANDBY); diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index 494f2d31d75..25af901fe81 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -421,7 +421,6 @@ static int wm8741_resume(struct snd_soc_codec *codec) static int wm8741_probe(struct snd_soc_codec *codec) { struct wm8741_priv *wm8741 = snd_soc_codec_get_drvdata(codec); - u16 *reg_cache = codec->reg_cache; int ret = 0; ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8741->control_type); @@ -437,10 +436,14 @@ static int wm8741_probe(struct snd_soc_codec *codec) } /* Change some default settings - latch VU */ - reg_cache[WM8741_DACLLSB_ATTENUATION] |= WM8741_UPDATELL; - reg_cache[WM8741_DACLMSB_ATTENUATION] |= WM8741_UPDATELM; - reg_cache[WM8741_DACRLSB_ATTENUATION] |= WM8741_UPDATERL; - reg_cache[WM8741_DACRLSB_ATTENUATION] |= WM8741_UPDATERM; + snd_soc_update_bits(codec, WM8741_DACLLSB_ATTENUATION, + WM8741_UPDATELL, WM8741_UPDATELL); + snd_soc_update_bits(codec, WM8741_DACLMSB_ATTENUATION, + WM8741_UPDATELM, WM8741_UPDATELM); + snd_soc_update_bits(codec, WM8741_DACRLSB_ATTENUATION, + WM8741_UPDATERL, WM8741_UPDATERL); + snd_soc_update_bits(codec, WM8741_DACRLSB_ATTENUATION, + WM8741_UPDATERM, WM8741_UPDATERM); snd_soc_add_controls(codec, wm8741_snd_controls, ARRAY_SIZE(wm8741_snd_controls)); diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 17a8fe9b39b..443ae580445 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -2436,19 +2436,28 @@ static int wm8904_probe(struct snd_soc_codec *codec) } /* Change some default settings - latch VU and enable ZC */ - reg_cache[WM8904_ADC_DIGITAL_VOLUME_LEFT] |= WM8904_ADC_VU; - reg_cache[WM8904_ADC_DIGITAL_VOLUME_RIGHT] |= WM8904_ADC_VU; - reg_cache[WM8904_DAC_DIGITAL_VOLUME_LEFT] |= WM8904_DAC_VU; - reg_cache[WM8904_DAC_DIGITAL_VOLUME_RIGHT] |= WM8904_DAC_VU; - reg_cache[WM8904_ANALOGUE_OUT1_LEFT] |= WM8904_HPOUT_VU | - WM8904_HPOUTLZC; - reg_cache[WM8904_ANALOGUE_OUT1_RIGHT] |= WM8904_HPOUT_VU | - WM8904_HPOUTRZC; - reg_cache[WM8904_ANALOGUE_OUT2_LEFT] |= WM8904_LINEOUT_VU | - WM8904_LINEOUTLZC; - reg_cache[WM8904_ANALOGUE_OUT2_RIGHT] |= WM8904_LINEOUT_VU | - WM8904_LINEOUTRZC; - reg_cache[WM8904_CLOCK_RATES_0] &= ~WM8904_SR_MODE; + snd_soc_update_bits(codec, WM8904_ADC_DIGITAL_VOLUME_LEFT, + WM8904_ADC_VU, WM8904_ADC_VU); + snd_soc_update_bits(codec, WM8904_ADC_DIGITAL_VOLUME_RIGHT, + WM8904_ADC_VU, WM8904_ADC_VU); + snd_soc_update_bits(codec, WM8904_DAC_DIGITAL_VOLUME_LEFT, + WM8904_DAC_VU, WM8904_DAC_VU); + snd_soc_update_bits(codec, WM8904_DAC_DIGITAL_VOLUME_RIGHT, + WM8904_DAC_VU, WM8904_DAC_VU); + snd_soc_update_bits(codec, WM8904_ANALOGUE_OUT1_LEFT, + WM8904_HPOUT_VU | WM8904_HPOUTLZC, + WM8904_HPOUT_VU | WM8904_HPOUTLZC); + snd_soc_update_bits(codec, WM8904_ANALOGUE_OUT1_RIGHT, + WM8904_HPOUT_VU | WM8904_HPOUTRZC, + WM8904_HPOUT_VU | WM8904_HPOUTRZC); + snd_soc_update_bits(codec, WM8904_ANALOGUE_OUT2_LEFT, + WM8904_LINEOUT_VU | WM8904_LINEOUTLZC, + WM8904_LINEOUT_VU | WM8904_LINEOUTLZC); + snd_soc_update_bits(codec, WM8904_ANALOGUE_OUT2_RIGHT, + WM8904_LINEOUT_VU | WM8904_LINEOUTRZC, + WM8904_LINEOUT_VU | WM8904_LINEOUTRZC); + snd_soc_update_bits(codec, WM8904_CLOCK_RATES_0, + WM8904_SR_MODE, 0); /* Apply configuration from the platform data. */ if (wm8904->pdata) { @@ -2469,10 +2478,12 @@ static int wm8904_probe(struct snd_soc_codec *codec) /* Set Class W by default - this will be managed by the Class * G widget at runtime where bypass paths are available. */ - reg_cache[WM8904_CLASS_W_0] |= WM8904_CP_DYN_PWR; + snd_soc_update_bits(codec, WM8904_CLASS_W_0, + WM8904_CP_DYN_PWR, WM8904_CP_DYN_PWR); /* Use normal bias source */ - reg_cache[WM8904_BIAS_CONTROL_0] &= ~WM8904_POBCTRL; + snd_soc_update_bits(codec, WM8904_BIAS_CONTROL_0, + WM8904_POBCTRL, 0); wm8904_set_bias_level(codec, SND_SOC_BIAS_STANDBY); diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c index 7167dfc96aa..5e0214d6293 100644 --- a/sound/soc/codecs/wm8955.c +++ b/sound/soc/codecs/wm8955.c @@ -934,16 +934,27 @@ static int wm8955_probe(struct snd_soc_codec *codec) } /* Change some default settings - latch VU and enable ZC */ - reg_cache[WM8955_LEFT_DAC_VOLUME] |= WM8955_LDVU; - reg_cache[WM8955_RIGHT_DAC_VOLUME] |= WM8955_RDVU; - reg_cache[WM8955_LOUT1_VOLUME] |= WM8955_LO1VU | WM8955_LO1ZC; - reg_cache[WM8955_ROUT1_VOLUME] |= WM8955_RO1VU | WM8955_RO1ZC; - reg_cache[WM8955_LOUT2_VOLUME] |= WM8955_LO2VU | WM8955_LO2ZC; - reg_cache[WM8955_ROUT2_VOLUME] |= WM8955_RO2VU | WM8955_RO2ZC; - reg_cache[WM8955_MONOOUT_VOLUME] |= WM8955_MOZC; + snd_soc_update_bits(codec, WM8955_LEFT_DAC_VOLUME, + WM8955_LDVU, WM8955_LDVU); + snd_soc_update_bits(codec, WM8955_RIGHT_DAC_VOLUME, + WM8955_RDVU, WM8955_RDVU); + snd_soc_update_bits(codec, WM8955_LOUT1_VOLUME, + WM8955_LO1VU | WM8955_LO1ZC, + WM8955_LO1VU | WM8955_LO1ZC); + snd_soc_update_bits(codec, WM8955_ROUT1_VOLUME, + WM8955_RO1VU | WM8955_RO1ZC, + WM8955_RO1VU | WM8955_RO1ZC); + snd_soc_update_bits(codec, WM8955_LOUT2_VOLUME, + WM8955_LO2VU | WM8955_LO2ZC, + WM8955_LO2VU | WM8955_LO2ZC); + snd_soc_update_bits(codec, WM8955_ROUT2_VOLUME, + WM8955_RO2VU | WM8955_RO2ZC, + WM8955_RO2VU | WM8955_RO2ZC); + snd_soc_update_bits(codec, WM8955_MONOOUT_VOLUME, + WM8955_MOZC, WM8955_MOZC); /* Also enable adaptive bass boost by default */ - reg_cache[WM8955_BASS_CONTROL] |= WM8955_BB; + snd_soc_update_bits(codec, WM8955_BASS_CONTROL, WM8955_BB, WM8955_BB); /* Set platform data values */ if (pdata) { diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 7c02924bedd..5c7b730a864 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3822,16 +3822,26 @@ static int wm8962_probe(struct snd_soc_codec *codec) } /* Latch volume update bits */ - reg_cache[WM8962_LEFT_INPUT_VOLUME] |= WM8962_IN_VU; - reg_cache[WM8962_RIGHT_INPUT_VOLUME] |= WM8962_IN_VU; - reg_cache[WM8962_LEFT_ADC_VOLUME] |= WM8962_ADC_VU; - reg_cache[WM8962_RIGHT_ADC_VOLUME] |= WM8962_ADC_VU; - reg_cache[WM8962_LEFT_DAC_VOLUME] |= WM8962_DAC_VU; - reg_cache[WM8962_RIGHT_DAC_VOLUME] |= WM8962_DAC_VU; - reg_cache[WM8962_SPKOUTL_VOLUME] |= WM8962_SPKOUT_VU; - reg_cache[WM8962_SPKOUTR_VOLUME] |= WM8962_SPKOUT_VU; - reg_cache[WM8962_HPOUTL_VOLUME] |= WM8962_HPOUT_VU; - reg_cache[WM8962_HPOUTR_VOLUME] |= WM8962_HPOUT_VU; + snd_soc_update_bits(codec, WM8962_LEFT_INPUT_VOLUME, + WM8962_IN_VU, WM8962_IN_VU); + snd_soc_update_bits(codec, WM8962_RIGHT_INPUT_VOLUME, + WM8962_IN_VU, WM8962_IN_VU); + snd_soc_update_bits(codec, WM8962_LEFT_ADC_VOLUME, + WM8962_ADC_VU, WM8962_ADC_VU); + snd_soc_update_bits(codec, WM8962_RIGHT_ADC_VOLUME, + WM8962_ADC_VU, WM8962_ADC_VU); + snd_soc_update_bits(codec, WM8962_LEFT_DAC_VOLUME, + WM8962_DAC_VU, WM8962_DAC_VU); + snd_soc_update_bits(codec, WM8962_RIGHT_DAC_VOLUME, + WM8962_DAC_VU, WM8962_DAC_VU); + snd_soc_update_bits(codec, WM8962_SPKOUTL_VOLUME, + WM8962_SPKOUT_VU, WM8962_SPKOUT_VU); + snd_soc_update_bits(codec, WM8962_SPKOUTR_VOLUME, + WM8962_SPKOUT_VU, WM8962_SPKOUT_VU); + snd_soc_update_bits(codec, WM8962_HPOUTL_VOLUME, + WM8962_HPOUT_VU, WM8962_HPOUT_VU); + snd_soc_update_bits(codec, WM8962_HPOUTR_VOLUME, + WM8962_HPOUT_VU, WM8962_HPOUT_VU); wm8962_add_widgets(codec); diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index 4bbc3442703..30fb48ec279 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -965,7 +965,7 @@ static int wm8978_probe(struct snd_soc_codec *codec) * written. */ for (i = 0; i < ARRAY_SIZE(update_reg); i++) - ((u16 *)codec->reg_cache)[update_reg[i]] |= 0x100; + snd_soc_update_bits(codec, update_reg[i], 0x100, 0x100); /* Reset the codec */ ret = snd_soc_write(codec, WM8978_RESET, 0); diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c index d40bfc9f880..4de12203e61 100644 --- a/sound/soc/codecs/wm9090.c +++ b/sound/soc/codecs/wm9090.c @@ -551,7 +551,6 @@ static int wm9090_set_bias_level(struct snd_soc_codec *codec, static int wm9090_probe(struct snd_soc_codec *codec) { struct wm9090_priv *wm9090 = snd_soc_codec_get_drvdata(codec); - u16 *reg_cache = codec->reg_cache; int ret; codec->control_data = wm9090->control_data; @@ -576,22 +575,30 @@ static int wm9090_probe(struct snd_soc_codec *codec) /* Configure some defaults; they will be written out when we * bring the bias up. */ - reg_cache[WM9090_IN1_LINE_INPUT_A_VOLUME] |= WM9090_IN1_VU - | WM9090_IN1A_ZC; - reg_cache[WM9090_IN1_LINE_INPUT_B_VOLUME] |= WM9090_IN1_VU - | WM9090_IN1B_ZC; - reg_cache[WM9090_IN2_LINE_INPUT_A_VOLUME] |= WM9090_IN2_VU - | WM9090_IN2A_ZC; - reg_cache[WM9090_IN2_LINE_INPUT_B_VOLUME] |= WM9090_IN2_VU - | WM9090_IN2B_ZC; - reg_cache[WM9090_SPEAKER_VOLUME_LEFT] |= - WM9090_SPKOUT_VU | WM9090_SPKOUTL_ZC; - reg_cache[WM9090_LEFT_OUTPUT_VOLUME] |= - WM9090_HPOUT1_VU | WM9090_HPOUT1L_ZC; - reg_cache[WM9090_RIGHT_OUTPUT_VOLUME] |= - WM9090_HPOUT1_VU | WM9090_HPOUT1R_ZC; - - reg_cache[WM9090_CLOCKING_1] |= WM9090_TOCLK_ENA; + snd_soc_update_bits(codec, WM9090_IN1_LINE_INPUT_A_VOLUME, + WM9090_IN1_VU | WM9090_IN1A_ZC, + WM9090_IN1_VU | WM9090_IN1A_ZC); + snd_soc_update_bits(codec, WM9090_IN1_LINE_INPUT_B_VOLUME, + WM9090_IN1_VU | WM9090_IN1B_ZC, + WM9090_IN1_VU | WM9090_IN1B_ZC); + snd_soc_update_bits(codec, WM9090_IN2_LINE_INPUT_A_VOLUME, + WM9090_IN2_VU | WM9090_IN2A_ZC, + WM9090_IN2_VU | WM9090_IN2A_ZC); + snd_soc_update_bits(codec, WM9090_IN2_LINE_INPUT_B_VOLUME, + WM9090_IN2_VU | WM9090_IN2B_ZC, + WM9090_IN2_VU | WM9090_IN2B_ZC); + snd_soc_update_bits(codec, WM9090_SPEAKER_VOLUME_LEFT, + WM9090_SPKOUT_VU | WM9090_SPKOUTL_ZC, + WM9090_SPKOUT_VU | WM9090_SPKOUTL_ZC); + snd_soc_update_bits(codec, WM9090_LEFT_OUTPUT_VOLUME, + WM9090_HPOUT1_VU | WM9090_HPOUT1L_ZC, + WM9090_HPOUT1_VU | WM9090_HPOUT1L_ZC); + snd_soc_update_bits(codec, WM9090_RIGHT_OUTPUT_VOLUME, + WM9090_HPOUT1_VU | WM9090_HPOUT1R_ZC, + WM9090_HPOUT1_VU | WM9090_HPOUT1R_ZC); + + snd_soc_update_bits(codec, WM9090_CLOCKING_1, + WM9090_TOCLK_ENA, WM9090_TOCLK_ENA); wm9090_set_bias_level(codec, SND_SOC_BIAS_STANDBY); -- cgit v1.2.3-70-g09d2 From 219d8df86805b8bb20b375707e9be734100ce89d Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Mon, 17 Jan 2011 11:00:12 +0000 Subject: ASoC: WM8995: Add regulator handling code Signed-off-by: Dimitris Papastamos Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/wm8995.c | 102 +++++++++++++++++++++++++++++++++++++++++++--- 1 file changed, 97 insertions(+), 5 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index f0f678de489..7d563413df3 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -19,6 +19,7 @@ #include #include #include +#include #include #include #include @@ -30,6 +31,18 @@ #include "wm8995.h" +#define WM8995_NUM_SUPPLIES 8 +static const char *wm8995_supply_names[WM8995_NUM_SUPPLIES] = { + "DCVDD", + "DBVDD1", + "DBVDD2", + "DBVDD3", + "AVDD1", + "AVDD2", + "CPVDD", + "MICVDD" +}; + static const u16 wm8995_reg_defs[WM8995_MAX_REGISTER + 1] = { [0] = 0x8995, [5] = 0x0100, [16] = 0x000b, [17] = 0x000b, [24] = 0x02c0, [25] = 0x02c0, [26] = 0x02c0, [27] = 0x02c0, @@ -126,8 +139,37 @@ struct wm8995_priv { int mclk[2]; int aifclk[2]; struct fll_config fll[2], fll_suspend[2]; + struct regulator_bulk_data supplies[WM8995_NUM_SUPPLIES]; + struct notifier_block disable_nb[WM8995_NUM_SUPPLIES]; + struct snd_soc_codec *codec; }; +/* + * We can't use the same notifier block for more than one supply and + * there's no way I can see to get from a callback to the caller + * except container_of(). + */ +#define WM8995_REGULATOR_EVENT(n) \ +static int wm8995_regulator_event_##n(struct notifier_block *nb, \ + unsigned long event, void *data) \ +{ \ + struct wm8995_priv *wm8995 = container_of(nb, struct wm8995_priv, \ + disable_nb[n]); \ + if (event & REGULATOR_EVENT_DISABLE) { \ + wm8995->codec->cache_sync = 1; \ + } \ + return 0; \ +} + +WM8995_REGULATOR_EVENT(0) +WM8995_REGULATOR_EVENT(1) +WM8995_REGULATOR_EVENT(2) +WM8995_REGULATOR_EVENT(3) +WM8995_REGULATOR_EVENT(4) +WM8995_REGULATOR_EVENT(5) +WM8995_REGULATOR_EVENT(6) +WM8995_REGULATOR_EVENT(7) + static const DECLARE_TLV_DB_SCALE(digital_tlv, -7200, 75, 1); static const DECLARE_TLV_DB_SCALE(in1lr_pga_tlv, -1650, 150, 0); static const DECLARE_TLV_DB_SCALE(in1l_boost_tlv, 0, 600, 0); @@ -1483,6 +1525,11 @@ static int wm8995_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + ret = regulator_bulk_enable(ARRAY_SIZE(wm8995->supplies), + wm8995->supplies); + if (ret) + return ret; + ret = snd_soc_cache_sync(codec); if (ret) { dev_err(codec->dev, @@ -1492,13 +1539,13 @@ static int wm8995_set_bias_level(struct snd_soc_codec *codec, snd_soc_update_bits(codec, WM8995_POWER_MANAGEMENT_1, WM8995_BG_ENA_MASK, WM8995_BG_ENA); - } break; case SND_SOC_BIAS_OFF: snd_soc_update_bits(codec, WM8995_POWER_MANAGEMENT_1, WM8995_BG_ENA_MASK, 0); - codec->cache_sync = 1; + regulator_bulk_disable(ARRAY_SIZE(wm8995->supplies), + wm8995->supplies); break; } @@ -1537,10 +1584,12 @@ static int wm8995_remove(struct snd_soc_codec *codec) static int wm8995_probe(struct snd_soc_codec *codec) { struct wm8995_priv *wm8995; + int i; int ret; codec->dapm.idle_bias_off = 1; wm8995 = snd_soc_codec_get_drvdata(codec); + wm8995->codec = codec; ret = snd_soc_codec_set_cache_io(codec, 16, 16, wm8995->control_type); if (ret < 0) { @@ -1548,21 +1597,58 @@ static int wm8995_probe(struct snd_soc_codec *codec) return ret; } + for (i = 0; i < ARRAY_SIZE(wm8995->supplies); i++) + wm8995->supplies[i].supply = wm8995_supply_names[i]; + + ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(wm8995->supplies), + wm8995->supplies); + if (ret) { + dev_err(codec->dev, "Failed to request supplies: %d\n", ret); + return ret; + } + + wm8995->disable_nb[0].notifier_call = wm8995_regulator_event_0; + wm8995->disable_nb[1].notifier_call = wm8995_regulator_event_1; + wm8995->disable_nb[2].notifier_call = wm8995_regulator_event_2; + wm8995->disable_nb[3].notifier_call = wm8995_regulator_event_3; + wm8995->disable_nb[4].notifier_call = wm8995_regulator_event_4; + wm8995->disable_nb[5].notifier_call = wm8995_regulator_event_5; + wm8995->disable_nb[6].notifier_call = wm8995_regulator_event_6; + wm8995->disable_nb[7].notifier_call = wm8995_regulator_event_7; + + /* This should really be moved into the regulator core */ + for (i = 0; i < ARRAY_SIZE(wm8995->supplies); i++) { + ret = regulator_register_notifier(wm8995->supplies[i].consumer, + &wm8995->disable_nb[i]); + if (ret) { + dev_err(codec->dev, + "Failed to register regulator notifier: %d\n", + ret); + } + } + + ret = regulator_bulk_enable(ARRAY_SIZE(wm8995->supplies), + wm8995->supplies); + if (ret) { + dev_err(codec->dev, "Failed to enable supplies: %d\n", ret); + goto err_reg_get; + } + ret = snd_soc_read(codec, WM8995_SOFTWARE_RESET); if (ret < 0) { dev_err(codec->dev, "Failed to read device ID: %d\n", ret); - return ret; + goto err_reg_enable; } if (ret != 0x8995) { dev_err(codec->dev, "Invalid device ID: %#x\n", ret); - return -EINVAL; + goto err_reg_enable; } ret = snd_soc_write(codec, WM8995_SOFTWARE_RESET, 0); if (ret < 0) { dev_err(codec->dev, "Failed to issue reset: %d\n", ret); - return ret; + goto err_reg_enable; } wm8995_set_bias_level(codec, SND_SOC_BIAS_STANDBY); @@ -1597,6 +1683,12 @@ static int wm8995_probe(struct snd_soc_codec *codec) ARRAY_SIZE(wm8995_intercon)); return 0; + +err_reg_enable: + regulator_bulk_disable(ARRAY_SIZE(wm8995->supplies), wm8995->supplies); +err_reg_get: + regulator_bulk_free(ARRAY_SIZE(wm8995->supplies), wm8995->supplies); + return ret; } #define WM8995_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ -- cgit v1.2.3-70-g09d2 From 64ed98365062b748770b340fd27ae34564a5a322 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 20 Jan 2011 21:43:44 +0000 Subject: ASoC: Staticise twl6040_hs_jack_report() It's an internal function so shouldn't be exported (as sparse points out). Signed-off-by: Mark Brown Signed-off-by: Liam Girdwood --- sound/soc/codecs/twl6040.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 4bbf1b15a49..482fcdb59bf 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -724,8 +724,8 @@ static int twl6040_power_mode_event(struct snd_soc_dapm_widget *w, return 0; } -void twl6040_hs_jack_report(struct snd_soc_codec *codec, - struct snd_soc_jack *jack, int report) +static void twl6040_hs_jack_report(struct snd_soc_codec *codec, + struct snd_soc_jack *jack, int report) { struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); int status; -- cgit v1.2.3-70-g09d2 From 7cfe56172ac14d2031f1896ecb629033f71caafa Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Thu, 20 Jan 2011 13:52:08 -0700 Subject: ASoC: wm8903: Expose GPIOs through gpiolib Also, update platform_data GPIO handling to have an explicit "don't touch this pin" option. Add #defines for the GPIO pin functions. Signed-off-by: Stephen Warren Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- include/sound/wm8903.h | 20 +++++++- sound/soc/codecs/wm8903.c | 126 +++++++++++++++++++++++++++++++++++++++++++++- 2 files changed, 144 insertions(+), 2 deletions(-) (limited to 'sound/soc/codecs') diff --git a/include/sound/wm8903.h b/include/sound/wm8903.h index b4a0db2307e..86172cf4339 100644 --- a/include/sound/wm8903.h +++ b/include/sound/wm8903.h @@ -36,6 +36,21 @@ #define WM8903_MICBIAS_ENA_SHIFT 0 /* MICBIAS_ENA */ #define WM8903_MICBIAS_ENA_WIDTH 1 /* MICBIAS_ENA */ +/* + * WM8903_GPn_FN values + * + * See datasheets for list of valid values per pin + */ +#define WM8903_GPn_FN_GPIO_OUTPUT 0 +#define WM8903_GPn_FN_BCLK 1 +#define WM8903_GPn_FN_IRQ_OUTPT 2 +#define WM8903_GPn_FN_GPIO_INPUT 3 +#define WM8903_GPn_FN_MICBIAS_CURRENT_DETECT 4 +#define WM8903_GPn_FN_MICBIAS_SHORT_DETECT 5 +#define WM8903_GPn_FN_DMIC_LR_CLK_OUTPUT 6 +#define WM8903_GPn_FN_FLL_LOCK_OUTPUT 8 +#define WM8903_GPn_FN_FLL_CLOCK_OUTPUT 9 + /* * R116 (0x74) - GPIO Control 1 */ @@ -231,6 +246,8 @@ #define WM8903_GP5_DB_SHIFT 0 /* GP5_DB */ #define WM8903_GP5_DB_WIDTH 1 /* GP5_DB */ +#define WM8903_NUM_GPIO 5 + struct wm8903_platform_data { bool irq_active_low; /* Set if IRQ active low, default high */ @@ -243,7 +260,8 @@ struct wm8903_platform_data { int micdet_delay; /* Delay after microphone detection (ms) */ - u32 gpio_cfg[5]; /* Default register values for GPIO pin mux */ + int gpio_base; + u32 gpio_cfg[WM8903_NUM_GPIO]; /* Default register values for GPIO pin mux */ }; #endif diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index a2a446cb180..9c4f2c4febc 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -2,6 +2,7 @@ * wm8903.c -- WM8903 ALSA SoC Audio driver * * Copyright 2008 Wolfson Microelectronics + * Copyright 2011 NVIDIA, Inc. * * Author: Mark Brown * @@ -19,6 +20,7 @@ #include #include #include +#include #include #include #include @@ -213,6 +215,7 @@ static u16 wm8903_reg_defaults[] = { }; struct wm8903_priv { + struct snd_soc_codec *codec; int sysclk; int irq; @@ -230,6 +233,10 @@ struct wm8903_priv { int mic_short; int mic_last_report; int mic_delay; + +#ifdef CONFIG_GPIOLIB + struct gpio_chip gpio_chip; +#endif }; static int wm8903_volatile_register(struct snd_soc_codec *codec, unsigned int reg) @@ -1635,6 +1642,119 @@ static int wm8903_resume(struct snd_soc_codec *codec) return 0; } +#ifdef CONFIG_GPIOLIB +static inline struct wm8903_priv *gpio_to_wm8903(struct gpio_chip *chip) +{ + return container_of(chip, struct wm8903_priv, gpio_chip); +} + +static int wm8903_gpio_request(struct gpio_chip *chip, unsigned offset) +{ + if (offset >= WM8903_NUM_GPIO) + return -EINVAL; + + return 0; +} + +static int wm8903_gpio_direction_in(struct gpio_chip *chip, unsigned offset) +{ + struct wm8903_priv *wm8903 = gpio_to_wm8903(chip); + struct snd_soc_codec *codec = wm8903->codec; + unsigned int mask, val; + + mask = WM8903_GP1_FN_MASK | WM8903_GP1_DIR_MASK; + val = (WM8903_GPn_FN_GPIO_INPUT << WM8903_GP1_FN_SHIFT) | + WM8903_GP1_DIR; + + return snd_soc_update_bits(codec, WM8903_GPIO_CONTROL_1 + offset, + mask, val); +} + +static int wm8903_gpio_get(struct gpio_chip *chip, unsigned offset) +{ + struct wm8903_priv *wm8903 = gpio_to_wm8903(chip); + struct snd_soc_codec *codec = wm8903->codec; + int reg; + + reg = snd_soc_read(codec, WM8903_GPIO_CONTROL_1 + offset); + + return (reg & WM8903_GP1_LVL_MASK) >> WM8903_GP1_LVL_SHIFT; +} + +static int wm8903_gpio_direction_out(struct gpio_chip *chip, + unsigned offset, int value) +{ + struct wm8903_priv *wm8903 = gpio_to_wm8903(chip); + struct snd_soc_codec *codec = wm8903->codec; + unsigned int mask, val; + + mask = WM8903_GP1_FN_MASK | WM8903_GP1_DIR_MASK | WM8903_GP1_LVL_MASK; + val = (WM8903_GPn_FN_GPIO_OUTPUT << WM8903_GP1_FN_SHIFT) | + (value << WM8903_GP2_LVL_SHIFT); + + return snd_soc_update_bits(codec, WM8903_GPIO_CONTROL_1 + offset, + mask, val); +} + +static void wm8903_gpio_set(struct gpio_chip *chip, unsigned offset, int value) +{ + struct wm8903_priv *wm8903 = gpio_to_wm8903(chip); + struct snd_soc_codec *codec = wm8903->codec; + + snd_soc_update_bits(codec, WM8903_GPIO_CONTROL_1 + offset, + WM8903_GP1_LVL_MASK, value << WM8903_GP1_LVL_SHIFT); +} + +static struct gpio_chip wm8903_template_chip = { + .label = "wm8903", + .owner = THIS_MODULE, + .request = wm8903_gpio_request, + .direction_input = wm8903_gpio_direction_in, + .get = wm8903_gpio_get, + .direction_output = wm8903_gpio_direction_out, + .set = wm8903_gpio_set, + .can_sleep = 1, +}; + +static void wm8903_init_gpio(struct snd_soc_codec *codec) +{ + struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); + struct wm8903_platform_data *pdata = dev_get_platdata(codec->dev); + int ret; + + wm8903->gpio_chip = wm8903_template_chip; + wm8903->gpio_chip.ngpio = WM8903_NUM_GPIO; + wm8903->gpio_chip.dev = codec->dev; + + if (pdata && pdata->gpio_base) + wm8903->gpio_chip.base = pdata->gpio_base; + else + wm8903->gpio_chip.base = -1; + + ret = gpiochip_add(&wm8903->gpio_chip); + if (ret != 0) + dev_err(codec->dev, "Failed to add GPIOs: %d\n", ret); +} + +static void wm8903_free_gpio(struct snd_soc_codec *codec) +{ + struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); + int ret; + + ret = gpiochip_remove(&wm8903->gpio_chip); + if (ret != 0) + dev_err(codec->dev, "Failed to remove GPIOs: %d\n", ret); +} +#else +static void wm8903_init_gpio(struct snd_soc_codec *codec) +{ +} + +static void wm8903_free_gpio(struct snd_soc_codec *codec) +{ +} +#endif + static int wm8903_probe(struct snd_soc_codec *codec) { struct wm8903_platform_data *pdata = dev_get_platdata(codec->dev); @@ -1643,6 +1763,7 @@ static int wm8903_probe(struct snd_soc_codec *codec) int trigger, irq_pol; u16 val; + wm8903->codec = codec; init_completion(&wm8903->wseq); ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_I2C); @@ -1667,7 +1788,7 @@ static int wm8903_probe(struct snd_soc_codec *codec) /* Set up GPIOs and microphone detection */ if (pdata) { for (i = 0; i < ARRAY_SIZE(pdata->gpio_cfg); i++) { - if (!pdata->gpio_cfg[i]) + if (pdata->gpio_cfg[i] == WM8903_GPIO_NO_CONFIG) continue; snd_soc_write(codec, WM8903_GPIO_CONTROL_1 + i, @@ -1749,12 +1870,15 @@ static int wm8903_probe(struct snd_soc_codec *codec) ARRAY_SIZE(wm8903_snd_controls)); wm8903_add_widgets(codec); + wm8903_init_gpio(codec); + return ret; } /* power down chip */ static int wm8903_remove(struct snd_soc_codec *codec) { + wm8903_free_gpio(codec); wm8903_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; } -- cgit v1.2.3-70-g09d2 From 67b22517d8e48a97e1d2ab10d095c538bbb2374c Mon Sep 17 00:00:00 2001 From: Alexander Sverdlin Date: Wed, 19 Jan 2011 21:22:06 +0300 Subject: ASoC: CS4271 codec support Added support for CS4271 codec to ASoC. Signed-off-by: Alexander Sverdlin Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- include/sound/cs4271.h | 25 ++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs4271.c | 630 ++++++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 661 insertions(+) create mode 100644 include/sound/cs4271.h create mode 100644 sound/soc/codecs/cs4271.c (limited to 'sound/soc/codecs') diff --git a/include/sound/cs4271.h b/include/sound/cs4271.h new file mode 100644 index 00000000000..16f8d325d3d --- /dev/null +++ b/include/sound/cs4271.h @@ -0,0 +1,25 @@ +/* + * Definitions for CS4271 ASoC codec driver + * + * Copyright (c) 2010 Alexander Sverdlin + * + * 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 __CS4271_H +#define __CS4271_H + +struct cs4271_platform_data { + int gpio_nreset; /* GPIO driving Reset pin, if any */ + int gpio_disable; /* GPIO that disable serial bus, if any */ +}; + +#endif /* __CS4271_H */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index a9cb2a04ad5..e239345a4d5 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -26,6 +26,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC select SND_SOC_CS42L51 if I2C select SND_SOC_CS4270 if I2C + select SND_SOC_CS4271 if SND_SOC_I2C_AND_SPI select SND_SOC_CX20442 select SND_SOC_DA7210 if I2C select SND_SOC_JZ4740_CODEC if SOC_JZ4740 @@ -157,6 +158,9 @@ config SND_SOC_CS4270_VD33_ERRATA bool depends on SND_SOC_CS4270 +config SND_SOC_CS4271 + tristate + config SND_SOC_CX20442 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 68e76af894b..83b7accd703 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -12,6 +12,7 @@ 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-cs4271-objs := cs4271.o snd-soc-cx20442-objs := cx20442.o snd-soc-da7210-objs := da7210.o snd-soc-dmic-objs := dmic.o @@ -93,6 +94,7 @@ 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-$(CONFIG_SND_SOC_CS4271) += snd-soc-cs4271.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c new file mode 100644 index 00000000000..237ece3f104 --- /dev/null +++ b/sound/soc/codecs/cs4271.c @@ -0,0 +1,630 @@ +/* + * CS4271 ASoC codec driver + * + * Copyright (c) 2010 Alexander Sverdlin + * + * 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. + * + * This driver support CS4271 codec being master or slave, working + * in control port mode, connected either via SPI or I2C. + * The data format accepted is I2S or left-justified. + * DAPM support not implemented. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#define CS4271_PCM_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +/* + * CS4271 registers + * High byte represents SPI chip address (0x10) + write command (0) + * Low byte - codec register address + */ +#define CS4271_MODE1 0x2001 /* Mode Control 1 */ +#define CS4271_DACCTL 0x2002 /* DAC Control */ +#define CS4271_DACVOL 0x2003 /* DAC Volume & Mixing Control */ +#define CS4271_VOLA 0x2004 /* DAC Channel A Volume Control */ +#define CS4271_VOLB 0x2005 /* DAC Channel B Volume Control */ +#define CS4271_ADCCTL 0x2006 /* ADC Control */ +#define CS4271_MODE2 0x2007 /* Mode Control 2 */ +#define CS4271_CHIPID 0x2008 /* Chip ID */ + +#define CS4271_FIRSTREG CS4271_MODE1 +#define CS4271_LASTREG CS4271_MODE2 +#define CS4271_NR_REGS ((CS4271_LASTREG & 0xFF) + 1) + +/* Bit masks for the CS4271 registers */ +#define CS4271_MODE1_MODE_MASK 0xC0 +#define CS4271_MODE1_MODE_1X 0x00 +#define CS4271_MODE1_MODE_2X 0x80 +#define CS4271_MODE1_MODE_4X 0xC0 + +#define CS4271_MODE1_DIV_MASK 0x30 +#define CS4271_MODE1_DIV_1 0x00 +#define CS4271_MODE1_DIV_15 0x10 +#define CS4271_MODE1_DIV_2 0x20 +#define CS4271_MODE1_DIV_3 0x30 + +#define CS4271_MODE1_MASTER 0x08 + +#define CS4271_MODE1_DAC_DIF_MASK 0x07 +#define CS4271_MODE1_DAC_DIF_LJ 0x00 +#define CS4271_MODE1_DAC_DIF_I2S 0x01 +#define CS4271_MODE1_DAC_DIF_RJ16 0x02 +#define CS4271_MODE1_DAC_DIF_RJ24 0x03 +#define CS4271_MODE1_DAC_DIF_RJ20 0x04 +#define CS4271_MODE1_DAC_DIF_RJ18 0x05 + +#define CS4271_DACCTL_AMUTE 0x80 +#define CS4271_DACCTL_IF_SLOW 0x40 + +#define CS4271_DACCTL_DEM_MASK 0x30 +#define CS4271_DACCTL_DEM_DIS 0x00 +#define CS4271_DACCTL_DEM_441 0x10 +#define CS4271_DACCTL_DEM_48 0x20 +#define CS4271_DACCTL_DEM_32 0x30 + +#define CS4271_DACCTL_SVRU 0x08 +#define CS4271_DACCTL_SRD 0x04 +#define CS4271_DACCTL_INVA 0x02 +#define CS4271_DACCTL_INVB 0x01 + +#define CS4271_DACVOL_BEQUA 0x40 +#define CS4271_DACVOL_SOFT 0x20 +#define CS4271_DACVOL_ZEROC 0x10 + +#define CS4271_DACVOL_ATAPI_MASK 0x0F +#define CS4271_DACVOL_ATAPI_M_M 0x00 +#define CS4271_DACVOL_ATAPI_M_BR 0x01 +#define CS4271_DACVOL_ATAPI_M_BL 0x02 +#define CS4271_DACVOL_ATAPI_M_BLR2 0x03 +#define CS4271_DACVOL_ATAPI_AR_M 0x04 +#define CS4271_DACVOL_ATAPI_AR_BR 0x05 +#define CS4271_DACVOL_ATAPI_AR_BL 0x06 +#define CS4271_DACVOL_ATAPI_AR_BLR2 0x07 +#define CS4271_DACVOL_ATAPI_AL_M 0x08 +#define CS4271_DACVOL_ATAPI_AL_BR 0x09 +#define CS4271_DACVOL_ATAPI_AL_BL 0x0A +#define CS4271_DACVOL_ATAPI_AL_BLR2 0x0B +#define CS4271_DACVOL_ATAPI_ALR2_M 0x0C +#define CS4271_DACVOL_ATAPI_ALR2_BR 0x0D +#define CS4271_DACVOL_ATAPI_ALR2_BL 0x0E +#define CS4271_DACVOL_ATAPI_ALR2_BLR2 0x0F + +#define CS4271_VOLA_MUTE 0x80 +#define CS4271_VOLA_VOL_MASK 0x7F +#define CS4271_VOLB_MUTE 0x80 +#define CS4271_VOLB_VOL_MASK 0x7F + +#define CS4271_ADCCTL_DITHER16 0x20 + +#define CS4271_ADCCTL_ADC_DIF_MASK 0x10 +#define CS4271_ADCCTL_ADC_DIF_LJ 0x00 +#define CS4271_ADCCTL_ADC_DIF_I2S 0x10 + +#define CS4271_ADCCTL_MUTEA 0x08 +#define CS4271_ADCCTL_MUTEB 0x04 +#define CS4271_ADCCTL_HPFDA 0x02 +#define CS4271_ADCCTL_HPFDB 0x01 + +#define CS4271_MODE2_LOOP 0x10 +#define CS4271_MODE2_MUTECAEQUB 0x08 +#define CS4271_MODE2_FREEZE 0x04 +#define CS4271_MODE2_CPEN 0x02 +#define CS4271_MODE2_PDN 0x01 + +#define CS4271_CHIPID_PART_MASK 0xF0 +#define CS4271_CHIPID_REV_MASK 0x0F + +/* + * Default CS4271 power-up configuration + * Array contains non-existing in hw register at address 0 + * Array do not include Chip ID, as codec driver does not use + * registers read operations at all + */ +static const u8 cs4271_dflt_reg[CS4271_NR_REGS] = { + 0, + 0, + CS4271_DACCTL_AMUTE, + CS4271_DACVOL_SOFT | CS4271_DACVOL_ATAPI_AL_BR, + 0, + 0, + 0, + 0, +}; + +struct cs4271_private { + /* SND_SOC_I2C or SND_SOC_SPI */ + enum snd_soc_control_type bus_type; + void *control_data; + unsigned int mclk; + bool master; + bool deemph; + /* Current sample rate for de-emphasis control */ + int rate; + /* GPIO driving Reset pin, if any */ + int gpio_nreset; + /* GPIO that disable serial bus, if any */ + int gpio_disable; +}; + +struct cs4271_clk_cfg { + unsigned int ratio; /* MCLK / sample rate */ + u8 speed_mode; /* codec speed mode: 1x, 2x, 4x */ + u8 mclk_master; /* ratio bit mask for Master mode */ + u8 mclk_slave; /* ratio bit mask for Slave mode */ +}; + +static struct cs4271_clk_cfg cs4271_clk_tab[] = { + {64, CS4271_MODE1_MODE_4X, CS4271_MODE1_DIV_1, CS4271_MODE1_DIV_1}, + {96, CS4271_MODE1_MODE_4X, CS4271_MODE1_DIV_15, CS4271_MODE1_DIV_1}, + {128, CS4271_MODE1_MODE_2X, CS4271_MODE1_DIV_1, CS4271_MODE1_DIV_1}, + {192, CS4271_MODE1_MODE_2X, CS4271_MODE1_DIV_15, CS4271_MODE1_DIV_1}, + {256, CS4271_MODE1_MODE_1X, CS4271_MODE1_DIV_1, CS4271_MODE1_DIV_1}, + {384, CS4271_MODE1_MODE_1X, CS4271_MODE1_DIV_15, CS4271_MODE1_DIV_1}, + {512, CS4271_MODE1_MODE_1X, CS4271_MODE1_DIV_2, CS4271_MODE1_DIV_1}, + {768, CS4271_MODE1_MODE_1X, CS4271_MODE1_DIV_3, CS4271_MODE1_DIV_3}, + {1024, CS4271_MODE1_MODE_1X, CS4271_MODE1_DIV_3, CS4271_MODE1_DIV_3} +}; + +#define CS4171_NR_RATIOS ARRAY_SIZE(cs4271_clk_tab) + +/* + * @freq is the desired MCLK rate + * MCLK rate should (c) be the sample rate, multiplied by one of the + * ratios listed in cs4271_mclk_fs_ratios table + */ +static int cs4271_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 cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); + + cs4271->mclk = freq; + return 0; +} + +static int cs4271_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int format) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); + unsigned int val = 0; + + switch (format & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + cs4271->master = 0; + break; + case SND_SOC_DAIFMT_CBM_CFM: + cs4271->master = 1; + val |= CS4271_MODE1_MASTER; + break; + default: + dev_err(codec->dev, "Invalid DAI format\n"); + return -EINVAL; + } + + switch (format & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_LEFT_J: + val |= CS4271_MODE1_DAC_DIF_LJ; + snd_soc_update_bits(codec, CS4271_ADCCTL, + CS4271_ADCCTL_ADC_DIF_MASK, CS4271_ADCCTL_ADC_DIF_LJ); + break; + case SND_SOC_DAIFMT_I2S: + val |= CS4271_MODE1_DAC_DIF_I2S; + snd_soc_update_bits(codec, CS4271_ADCCTL, + CS4271_ADCCTL_ADC_DIF_MASK, CS4271_ADCCTL_ADC_DIF_I2S); + break; + default: + dev_err(codec->dev, "Invalid DAI format\n"); + return -EINVAL; + } + + snd_soc_update_bits(codec, CS4271_MODE1, + CS4271_MODE1_DAC_DIF_MASK | CS4271_MODE1_MASTER, val); + + return 0; +} + +static int cs4271_deemph[] = {0, 44100, 48000, 32000}; + +static int cs4271_set_deemph(struct snd_soc_codec *codec) +{ + struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); + int i; + int val = CS4271_DACCTL_DEM_DIS; + + if (cs4271->deemph) { + /* Find closest de-emphasis freq */ + val = 1; + for (i = 2; i < ARRAY_SIZE(cs4271_deemph); i++) + if (abs(cs4271_deemph[i] - cs4271->rate) < + abs(cs4271_deemph[val] - cs4271->rate)) + val = i; + val <<= 4; + } + + return snd_soc_update_bits(codec, CS4271_DACCTL, + CS4271_DACCTL_DEM_MASK, val); +} + +static int cs4271_get_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.enumerated.item[0] = cs4271->deemph; + return 0; +} + +static int cs4271_put_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); + + cs4271->deemph = ucontrol->value.enumerated.item[0]; + return cs4271_set_deemph(codec); +} + +static int cs4271_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); + unsigned int i, ratio, val; + + cs4271->rate = params_rate(params); + ratio = cs4271->mclk / cs4271->rate; + for (i = 0; i < CS4171_NR_RATIOS; i++) + if (cs4271_clk_tab[i].ratio == ratio) + break; + + if ((i == CS4171_NR_RATIOS) || ((ratio == 1024) && cs4271->master)) { + dev_err(codec->dev, "Invalid sample rate\n"); + return -EINVAL; + } + + /* Configure DAC */ + val = cs4271_clk_tab[i].speed_mode; + + if (cs4271->master) + val |= cs4271_clk_tab[i].mclk_master; + else + val |= cs4271_clk_tab[i].mclk_slave; + + snd_soc_update_bits(codec, CS4271_MODE1, + CS4271_MODE1_MODE_MASK | CS4271_MODE1_DIV_MASK, val); + + return cs4271_set_deemph(codec); +} + +static int cs4271_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + int val_a = 0; + int val_b = 0; + + if (mute) { + val_a = CS4271_VOLA_MUTE; + val_b = CS4271_VOLB_MUTE; + } + + snd_soc_update_bits(codec, CS4271_VOLA, CS4271_VOLA_MUTE, val_a); + snd_soc_update_bits(codec, CS4271_VOLB, CS4271_VOLB_MUTE, val_b); + + return 0; +} + +/* CS4271 controls */ +static DECLARE_TLV_DB_SCALE(cs4271_dac_tlv, -12700, 100, 0); + +static const struct snd_kcontrol_new cs4271_snd_controls[] = { + SOC_DOUBLE_R_TLV("Master Playback Volume", CS4271_VOLA, CS4271_VOLB, + 0, 0x7F, 1, cs4271_dac_tlv), + SOC_SINGLE("Digital Loopback Switch", CS4271_MODE2, 4, 1, 0), + SOC_SINGLE("Soft Ramp Switch", CS4271_DACVOL, 5, 1, 0), + SOC_SINGLE("Zero Cross Switch", CS4271_DACVOL, 4, 1, 0), + SOC_SINGLE_BOOL_EXT("De-emphasis Switch", 0, + cs4271_get_deemph, cs4271_put_deemph), + SOC_SINGLE("Auto-Mute Switch", CS4271_DACCTL, 7, 1, 0), + SOC_SINGLE("Slow Roll Off Filter Switch", CS4271_DACCTL, 6, 1, 0), + SOC_SINGLE("Soft Volume Ramp-Up Switch", CS4271_DACCTL, 3, 1, 0), + SOC_SINGLE("Soft Ramp-Down Switch", CS4271_DACCTL, 2, 1, 0), + SOC_SINGLE("Left Channel Inversion Switch", CS4271_DACCTL, 1, 1, 0), + SOC_SINGLE("Right Channel Inversion Switch", CS4271_DACCTL, 0, 1, 0), + SOC_DOUBLE("Master Capture Switch", CS4271_ADCCTL, 3, 2, 1, 1), + SOC_SINGLE("Dither 16-Bit Data Switch", CS4271_ADCCTL, 5, 1, 0), + SOC_DOUBLE("High Pass Filter Switch", CS4271_ADCCTL, 1, 0, 1, 1), + SOC_DOUBLE_R("Master Playback Switch", CS4271_VOLA, CS4271_VOLB, + 7, 1, 1), +}; + +static struct snd_soc_dai_ops cs4271_dai_ops = { + .hw_params = cs4271_hw_params, + .set_sysclk = cs4271_set_dai_sysclk, + .set_fmt = cs4271_set_dai_fmt, + .digital_mute = cs4271_digital_mute, +}; + +struct snd_soc_dai_driver cs4271_dai = { + .name = "cs4271-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = CS4271_PCM_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = CS4271_PCM_FORMATS, + }, + .ops = &cs4271_dai_ops, + .symmetric_rates = 1, +}; + +#ifdef CONFIG_PM +static int cs4271_soc_suspend(struct snd_soc_codec *codec, pm_message_t mesg) +{ + /* Set power-down bit */ + snd_soc_update_bits(codec, CS4271_MODE2, 0, CS4271_MODE2_PDN); + return 0; +} + +static int cs4271_soc_resume(struct snd_soc_codec *codec) +{ + /* Restore codec state */ + snd_soc_cache_sync(codec); + /* then disable the power-down bit */ + snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN, 0); + return 0; +} +#else +#define cs4271_soc_suspend NULL +#define cs4271_soc_resume NULL +#endif /* CONFIG_PM */ + +static int cs4271_probe(struct snd_soc_codec *codec) +{ + struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); + struct cs4271_platform_data *cs4271plat = codec->dev->platform_data; + int ret; + int gpio_nreset = -EINVAL; + int gpio_disable = -EINVAL; + + codec->control_data = cs4271->control_data; + + if (cs4271plat) { + if (gpio_is_valid(cs4271plat->gpio_nreset)) + gpio_nreset = cs4271plat->gpio_nreset; + if (gpio_is_valid(cs4271plat->gpio_disable)) + gpio_disable = cs4271plat->gpio_disable; + } + + if (gpio_disable >= 0) + if (gpio_request(gpio_disable, "CS4271 Disable")) + gpio_disable = -EINVAL; + if (gpio_disable >= 0) + gpio_direction_output(gpio_disable, 0); + + if (gpio_nreset >= 0) + if (gpio_request(gpio_nreset, "CS4271 Reset")) + gpio_nreset = -EINVAL; + if (gpio_nreset >= 0) { + /* Reset codec */ + gpio_direction_output(gpio_nreset, 0); + udelay(1); + gpio_set_value(gpio_nreset, 1); + /* Give the codec time to wake up */ + udelay(1); + } + + cs4271->gpio_nreset = gpio_nreset; + cs4271->gpio_disable = gpio_disable; + + /* + * In case of I2C, chip address specified in board data. + * So cache IO operations use 8 bit codec register address. + * In case of SPI, chip address and register address + * passed together as 16 bit value. + * Anyway, register address is masked with 0xFF inside + * soc-cache code. + */ + if (cs4271->bus_type == SND_SOC_SPI) + ret = snd_soc_codec_set_cache_io(codec, 16, 8, + cs4271->bus_type); + else + ret = snd_soc_codec_set_cache_io(codec, 8, 8, + cs4271->bus_type); + if (ret) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + snd_soc_update_bits(codec, CS4271_MODE2, 0, + CS4271_MODE2_PDN | CS4271_MODE2_CPEN); + snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN, 0); + /* Power-up sequence requires 85 uS */ + udelay(85); + + return snd_soc_add_controls(codec, cs4271_snd_controls, + ARRAY_SIZE(cs4271_snd_controls)); +} + +static int cs4271_remove(struct snd_soc_codec *codec) +{ + struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); + int gpio_nreset, gpio_disable; + + gpio_nreset = cs4271->gpio_nreset; + gpio_disable = cs4271->gpio_disable; + + if (gpio_is_valid(gpio_nreset)) { + /* Set codec to the reset state */ + gpio_set_value(gpio_nreset, 0); + gpio_free(gpio_nreset); + } + + if (gpio_is_valid(gpio_disable)) + gpio_free(gpio_disable); + + return 0; +}; + +struct snd_soc_codec_driver soc_codec_dev_cs4271 = { + .probe = cs4271_probe, + .remove = cs4271_remove, + .suspend = cs4271_soc_suspend, + .resume = cs4271_soc_resume, + .reg_cache_default = cs4271_dflt_reg, + .reg_cache_size = ARRAY_SIZE(cs4271_dflt_reg), + .reg_word_size = sizeof(cs4271_dflt_reg[0]), + .compress_type = SND_SOC_FLAT_COMPRESSION, +}; + +#if defined(CONFIG_SPI_MASTER) +static int __devinit cs4271_spi_probe(struct spi_device *spi) +{ + struct cs4271_private *cs4271; + + cs4271 = devm_kzalloc(&spi->dev, sizeof(*cs4271), GFP_KERNEL); + if (!cs4271) + return -ENOMEM; + + spi_set_drvdata(spi, cs4271); + cs4271->control_data = spi; + cs4271->bus_type = SND_SOC_SPI; + + return snd_soc_register_codec(&spi->dev, &soc_codec_dev_cs4271, + &cs4271_dai, 1); +} + +static int __devexit cs4271_spi_remove(struct spi_device *spi) +{ + snd_soc_unregister_codec(&spi->dev); + return 0; +} + +static struct spi_driver cs4271_spi_driver = { + .driver = { + .name = "cs4271", + .owner = THIS_MODULE, + }, + .probe = cs4271_spi_probe, + .remove = __devexit_p(cs4271_spi_remove), +}; +#endif /* defined(CONFIG_SPI_MASTER) */ + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +static struct i2c_device_id cs4271_i2c_id[] = { + {"cs4271", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, cs4271_i2c_id); + +static int __devinit cs4271_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct cs4271_private *cs4271; + + cs4271 = devm_kzalloc(&client->dev, sizeof(*cs4271), GFP_KERNEL); + if (!cs4271) + return -ENOMEM; + + i2c_set_clientdata(client, cs4271); + cs4271->control_data = client; + cs4271->bus_type = SND_SOC_I2C; + + return snd_soc_register_codec(&client->dev, &soc_codec_dev_cs4271, + &cs4271_dai, 1); +} + +static int __devexit cs4271_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static struct i2c_driver cs4271_i2c_driver = { + .driver = { + .name = "cs4271", + .owner = THIS_MODULE, + }, + .id_table = cs4271_i2c_id, + .probe = cs4271_i2c_probe, + .remove = __devexit_p(cs4271_i2c_remove), +}; +#endif /* defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) */ + +/* + * We only register our serial bus driver here without + * assignment to particular chip. So if any of the below + * fails, there is some problem with I2C or SPI subsystem. + * In most cases this module will be compiled with support + * of only one serial bus. + */ +static int __init cs4271_modinit(void) +{ + int ret; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + ret = i2c_add_driver(&cs4271_i2c_driver); + if (ret) { + pr_err("Failed to register CS4271 I2C driver: %d\n", ret); + return ret; + } +#endif + +#if defined(CONFIG_SPI_MASTER) + ret = spi_register_driver(&cs4271_spi_driver); + if (ret) { + pr_err("Failed to register CS4271 SPI driver: %d\n", ret); + return ret; + } +#endif + + return 0; +} +module_init(cs4271_modinit); + +static void __exit cs4271_modexit(void) +{ +#if defined(CONFIG_SPI_MASTER) + spi_unregister_driver(&cs4271_spi_driver); +#endif + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&cs4271_i2c_driver); +#endif +} +module_exit(cs4271_modexit); + +MODULE_AUTHOR("Alexander Sverdlin "); +MODULE_DESCRIPTION("Cirrus Logic CS4271 ALSA SoC Codec Driver"); +MODULE_LICENSE("GPL"); -- cgit v1.2.3-70-g09d2 From cb9c130aa97bd41887a0a391388ef4070caab4d9 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 20 Jan 2011 11:45:34 +0900 Subject: ASoC: ak4642: add SND_SOC_DAIFMT_FORMAT support This patch support LEFT_J / I2S only for now Signed-off-by: Kuninori Morimoto Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/ak4642.c | 24 ++++++++++++++++++++++++ sound/soc/sh/fsi-ak4642.c | 3 ++- 2 files changed, 26 insertions(+), 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index f00eba313df..4be0570e3f1 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -116,6 +116,12 @@ #define BCKO_MASK (1 << 3) #define BCKO_64 BCKO_MASK +#define DIF_MASK (3 << 0) +#define DSP (0 << 0) +#define RIGHT_J (1 << 0) +#define LEFT_J (2 << 0) +#define I2S (3 << 0) + /* MD_CTL2 */ #define FS0 (1 << 0) #define FS1 (1 << 1) @@ -354,6 +360,24 @@ static int ak4642_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) snd_soc_update_bits(codec, PW_MGMT2, MS, data); snd_soc_update_bits(codec, MD_CTL1, BCKO_MASK, bcko); + /* format type */ + data = 0; + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_LEFT_J: + data = LEFT_J; + break; + case SND_SOC_DAIFMT_I2S: + data = I2S; + break; + /* FIXME + * Please add RIGHT_J / DSP support here + */ + default: + return -EINVAL; + break; + } + snd_soc_update_bits(codec, MD_CTL1, DIF_MASK, data); + return 0; } diff --git a/sound/soc/sh/fsi-ak4642.c b/sound/soc/sh/fsi-ak4642.c index 56cd3422310..a722a4c661f 100644 --- a/sound/soc/sh/fsi-ak4642.c +++ b/sound/soc/sh/fsi-ak4642.c @@ -26,7 +26,8 @@ static int fsi_ak4642_dai_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_dai *dai = rtd->codec_dai; int ret; - ret = snd_soc_dai_set_fmt(dai, SND_SOC_DAIFMT_CBM_CFM); + ret = snd_soc_dai_set_fmt(dai, SND_SOC_DAIFMT_LEFT_J | + SND_SOC_DAIFMT_CBM_CFM); if (ret < 0) return ret; -- cgit v1.2.3-70-g09d2 From 0d42e6e77f8b872483833a7146286edaaaeb2f39 Mon Sep 17 00:00:00 2001 From: Alexander Sverdlin Date: Fri, 21 Jan 2011 22:22:07 +0300 Subject: ASoC: cs4271.c: improve error handling CS4271 CODEC driver adapted to recently introduced error handling in snd_soc_update_bits(). Added snd_soc_cache_sync() error handling. Signed-off-by: Alexander Sverdlin Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/cs4271.c | 59 +++++++++++++++++++++++++++++++++++------------ 1 file changed, 44 insertions(+), 15 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 237ece3f104..5357ec5f5d7 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -209,6 +209,7 @@ static int cs4271_set_dai_fmt(struct snd_soc_dai *codec_dai, struct snd_soc_codec *codec = codec_dai->codec; struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); unsigned int val = 0; + int ret; switch (format & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: @@ -226,22 +227,27 @@ static int cs4271_set_dai_fmt(struct snd_soc_dai *codec_dai, switch (format & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_LEFT_J: val |= CS4271_MODE1_DAC_DIF_LJ; - snd_soc_update_bits(codec, CS4271_ADCCTL, + ret = snd_soc_update_bits(codec, CS4271_ADCCTL, CS4271_ADCCTL_ADC_DIF_MASK, CS4271_ADCCTL_ADC_DIF_LJ); + if (ret < 0) + return ret; break; case SND_SOC_DAIFMT_I2S: val |= CS4271_MODE1_DAC_DIF_I2S; - snd_soc_update_bits(codec, CS4271_ADCCTL, + ret = snd_soc_update_bits(codec, CS4271_ADCCTL, CS4271_ADCCTL_ADC_DIF_MASK, CS4271_ADCCTL_ADC_DIF_I2S); + if (ret < 0) + return ret; break; default: dev_err(codec->dev, "Invalid DAI format\n"); return -EINVAL; } - snd_soc_update_bits(codec, CS4271_MODE1, + ret = snd_soc_update_bits(codec, CS4271_MODE1, CS4271_MODE1_DAC_DIF_MASK | CS4271_MODE1_MASTER, val); - + if (ret < 0) + return ret; return 0; } @@ -250,7 +256,7 @@ static int cs4271_deemph[] = {0, 44100, 48000, 32000}; static int cs4271_set_deemph(struct snd_soc_codec *codec) { struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); - int i; + int i, ret; int val = CS4271_DACCTL_DEM_DIS; if (cs4271->deemph) { @@ -263,8 +269,11 @@ static int cs4271_set_deemph(struct snd_soc_codec *codec) val <<= 4; } - return snd_soc_update_bits(codec, CS4271_DACCTL, + ret = snd_soc_update_bits(codec, CS4271_DACCTL, CS4271_DACCTL_DEM_MASK, val); + if (ret < 0) + return ret; + return 0; } static int cs4271_get_deemph(struct snd_kcontrol *kcontrol, @@ -294,7 +303,8 @@ static int cs4271_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->codec; struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); - unsigned int i, ratio, val; + int i, ret; + unsigned int ratio, val; cs4271->rate = params_rate(params); ratio = cs4271->mclk / cs4271->rate; @@ -315,8 +325,10 @@ static int cs4271_hw_params(struct snd_pcm_substream *substream, else val |= cs4271_clk_tab[i].mclk_slave; - snd_soc_update_bits(codec, CS4271_MODE1, + ret = snd_soc_update_bits(codec, CS4271_MODE1, CS4271_MODE1_MODE_MASK | CS4271_MODE1_DIV_MASK, val); + if (ret < 0) + return ret; return cs4271_set_deemph(codec); } @@ -324,6 +336,7 @@ static int cs4271_hw_params(struct snd_pcm_substream *substream, static int cs4271_digital_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; + int ret; int val_a = 0; int val_b = 0; @@ -332,8 +345,12 @@ static int cs4271_digital_mute(struct snd_soc_dai *dai, int mute) val_b = CS4271_VOLB_MUTE; } - snd_soc_update_bits(codec, CS4271_VOLA, CS4271_VOLA_MUTE, val_a); - snd_soc_update_bits(codec, CS4271_VOLB, CS4271_VOLB_MUTE, val_b); + ret = snd_soc_update_bits(codec, CS4271_VOLA, CS4271_VOLA_MUTE, val_a); + if (ret < 0) + return ret; + ret = snd_soc_update_bits(codec, CS4271_VOLB, CS4271_VOLB_MUTE, val_b); + if (ret < 0) + return ret; return 0; } @@ -392,17 +409,25 @@ struct snd_soc_dai_driver cs4271_dai = { #ifdef CONFIG_PM static int cs4271_soc_suspend(struct snd_soc_codec *codec, pm_message_t mesg) { + int ret; /* Set power-down bit */ - snd_soc_update_bits(codec, CS4271_MODE2, 0, CS4271_MODE2_PDN); + ret = snd_soc_update_bits(codec, CS4271_MODE2, 0, CS4271_MODE2_PDN); + if (ret < 0) + return ret; return 0; } static int cs4271_soc_resume(struct snd_soc_codec *codec) { + int ret; /* Restore codec state */ - snd_soc_cache_sync(codec); + ret = snd_soc_cache_sync(codec); + if (ret < 0) + return ret; /* then disable the power-down bit */ - snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN, 0); + ret = snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN, 0); + if (ret < 0) + return ret; return 0; } #else @@ -467,9 +492,13 @@ static int cs4271_probe(struct snd_soc_codec *codec) return ret; } - snd_soc_update_bits(codec, CS4271_MODE2, 0, + ret = snd_soc_update_bits(codec, CS4271_MODE2, 0, CS4271_MODE2_PDN | CS4271_MODE2_CPEN); - snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN, 0); + if (ret < 0) + return ret; + ret = snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN, 0); + if (ret < 0) + return ret; /* Power-up sequence requires 85 uS */ udelay(85); -- cgit v1.2.3-70-g09d2 From 16af7d60aa27d3fc39e46fd456b8e33d34d60437 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 26 Jan 2011 11:35:28 +0000 Subject: ASoC: Staticise non-exported symbols in cs4271 Signed-off-by: Mark Brown Acked-by: Alexander Sverdlin Acked-by: Liam Girdwood --- sound/soc/codecs/cs4271.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 5357ec5f5d7..9c5b7db0ce6 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -386,7 +386,7 @@ static struct snd_soc_dai_ops cs4271_dai_ops = { .digital_mute = cs4271_digital_mute, }; -struct snd_soc_dai_driver cs4271_dai = { +static struct snd_soc_dai_driver cs4271_dai = { .name = "cs4271-hifi", .playback = { .stream_name = "Playback", @@ -526,7 +526,7 @@ static int cs4271_remove(struct snd_soc_codec *codec) return 0; }; -struct snd_soc_codec_driver soc_codec_dev_cs4271 = { +static struct snd_soc_codec_driver soc_codec_dev_cs4271 = { .probe = cs4271_probe, .remove = cs4271_remove, .suspend = cs4271_soc_suspend, -- cgit v1.2.3-70-g09d2 From fd94eeef06ed4abc08f58e42f46341d0bc4f7793 Mon Sep 17 00:00:00 2001 From: Harsha Priya Date: Fri, 28 Jan 2011 22:26:53 +0530 Subject: ASoC: sn95031: add capture support This patch adds the support for capture path in sn95031 codec. This codec supports upto 6DMICs, 2 AMICs and Linein. The linein and AMICs are connected through a MUX to ADC. The TX paths can be assigned to any of the ADCs or DMICs. Signed-off-by: Vinod Koul Signed-off-by: Harsha Priya Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/sn95031.c | 281 +++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 281 insertions(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index 593632cf791..40e285df9ae 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -33,6 +33,7 @@ #include #include #include +#include #include "sn95031.h" #define SN95031_RATES (SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_44100) @@ -145,6 +146,129 @@ static int sn95031_vihf_event(struct snd_soc_dapm_widget *w, return 0; } +static int sn95031_dmic12_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + unsigned int ldo = 0, clk_dir = 0, data_dir = 0; + + if (SND_SOC_DAPM_EVENT_ON(event)) { + ldo = BIT(5)|BIT(4); + clk_dir = BIT(0); + data_dir = BIT(7); + } + /* program DMIC LDO, clock and set clock */ + snd_soc_update_bits(w->codec, SN95031_MICBIAS, BIT(5)|BIT(4), ldo); + snd_soc_update_bits(w->codec, SN95031_DMICBUF0123, BIT(0), clk_dir); + snd_soc_update_bits(w->codec, SN95031_DMICBUF0123, BIT(7), data_dir); + return 0; +} + +static int sn95031_dmic34_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + unsigned int ldo = 0, clk_dir = 0, data_dir = 0; + + if (SND_SOC_DAPM_EVENT_ON(event)) { + ldo = BIT(5)|BIT(4); + clk_dir = BIT(2); + data_dir = BIT(1); + } + /* program DMIC LDO, clock and set clock */ + snd_soc_update_bits(w->codec, SN95031_MICBIAS, BIT(5)|BIT(4), ldo); + snd_soc_update_bits(w->codec, SN95031_DMICBUF0123, BIT(2), clk_dir); + snd_soc_update_bits(w->codec, SN95031_DMICBUF45, BIT(1), data_dir); + return 0; +} + +static int sn95031_dmic56_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + unsigned int ldo = 0; + + if (SND_SOC_DAPM_EVENT_ON(event)) + ldo = BIT(7)|BIT(6); + + /* program DMIC LDO */ + snd_soc_update_bits(w->codec, SN95031_MICBIAS, BIT(7)|BIT(6), ldo); + return 0; +} + +/* mux controls */ +static const char *sn95031_mic_texts[] = { "AMIC", "LineIn" }; + +static const struct soc_enum sn95031_micl_enum = + SOC_ENUM_SINGLE(SN95031_ADCCONFIG, 1, 2, sn95031_mic_texts); + +static const struct snd_kcontrol_new sn95031_micl_mux_control = + SOC_DAPM_ENUM("Route", sn95031_micl_enum); + +static const struct soc_enum sn95031_micr_enum = + SOC_ENUM_SINGLE(SN95031_ADCCONFIG, 3, 2, sn95031_mic_texts); + +static const struct snd_kcontrol_new sn95031_micr_mux_control = + SOC_DAPM_ENUM("Route", sn95031_micr_enum); + +static const char *sn95031_input_texts[] = { "DMIC1", "DMIC2", "DMIC3", + "DMIC4", "DMIC5", "DMIC6", + "ADC Left", "ADC Right" }; + +static const struct soc_enum sn95031_input1_enum = + SOC_ENUM_SINGLE(SN95031_AUDIOMUX12, 0, 8, sn95031_input_texts); + +static const struct snd_kcontrol_new sn95031_input1_mux_control = + SOC_DAPM_ENUM("Route", sn95031_input1_enum); + +static const struct soc_enum sn95031_input2_enum = + SOC_ENUM_SINGLE(SN95031_AUDIOMUX12, 4, 8, sn95031_input_texts); + +static const struct snd_kcontrol_new sn95031_input2_mux_control = + SOC_DAPM_ENUM("Route", sn95031_input2_enum); + +static const struct soc_enum sn95031_input3_enum = + SOC_ENUM_SINGLE(SN95031_AUDIOMUX34, 0, 8, sn95031_input_texts); + +static const struct snd_kcontrol_new sn95031_input3_mux_control = + SOC_DAPM_ENUM("Route", sn95031_input3_enum); + +static const struct soc_enum sn95031_input4_enum = + SOC_ENUM_SINGLE(SN95031_AUDIOMUX34, 4, 8, sn95031_input_texts); + +static const struct snd_kcontrol_new sn95031_input4_mux_control = + SOC_DAPM_ENUM("Route", sn95031_input4_enum); + +/* capture path controls */ + +static const char *sn95031_micmode_text[] = {"Single Ended", "Differential"}; + +/* 0dB to 30dB in 10dB steps */ +static const DECLARE_TLV_DB_SCALE(mic_tlv, 0, 10, 30); + +static const struct soc_enum sn95031_micmode1_enum = + SOC_ENUM_SINGLE(SN95031_MICAMP1, 1, 2, sn95031_micmode_text); +static const struct soc_enum sn95031_micmode2_enum = + SOC_ENUM_SINGLE(SN95031_MICAMP2, 1, 2, sn95031_micmode_text); + +static const char *sn95031_dmic_cfg_text[] = {"GPO", "DMIC"}; + +static const struct soc_enum sn95031_dmic12_cfg_enum = + SOC_ENUM_SINGLE(SN95031_DMICMUX, 0, 2, sn95031_dmic_cfg_text); +static const struct soc_enum sn95031_dmic34_cfg_enum = + SOC_ENUM_SINGLE(SN95031_DMICMUX, 1, 2, sn95031_dmic_cfg_text); +static const struct soc_enum sn95031_dmic56_cfg_enum = + SOC_ENUM_SINGLE(SN95031_DMICMUX, 2, 2, sn95031_dmic_cfg_text); + +static const struct snd_kcontrol_new sn95031_snd_controls[] = { + SOC_ENUM("Mic1Mode Capture Route", sn95031_micmode1_enum), + SOC_ENUM("Mic2Mode Capture Route", sn95031_micmode2_enum), + SOC_ENUM("DMIC12 Capture Route", sn95031_dmic12_cfg_enum), + SOC_ENUM("DMIC34 Capture Route", sn95031_dmic34_cfg_enum), + SOC_ENUM("DMIC56 Capture Route", sn95031_dmic56_cfg_enum), + SOC_SINGLE_TLV("Mic1 Capture Volume", SN95031_MICAMP1, + 2, 4, 0, mic_tlv), + SOC_SINGLE_TLV("Mic2 Capture Volume", SN95031_MICAMP2, + 2, 4, 0, mic_tlv), +}; + /* DAPM widgets */ static const struct snd_soc_dapm_widget sn95031_dapm_widgets[] = { @@ -159,6 +283,36 @@ static const struct snd_soc_dapm_widget sn95031_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("VIB1OUT"), SND_SOC_DAPM_OUTPUT("VIB2OUT"), + SND_SOC_DAPM_INPUT("AMIC1"), /* headset mic */ + SND_SOC_DAPM_INPUT("AMIC2"), + SND_SOC_DAPM_INPUT("DMIC1"), + SND_SOC_DAPM_INPUT("DMIC2"), + SND_SOC_DAPM_INPUT("DMIC3"), + SND_SOC_DAPM_INPUT("DMIC4"), + SND_SOC_DAPM_INPUT("DMIC5"), + SND_SOC_DAPM_INPUT("DMIC6"), + SND_SOC_DAPM_INPUT("LINEINL"), + SND_SOC_DAPM_INPUT("LINEINR"), + + SND_SOC_DAPM_MICBIAS("AMIC1Bias", SN95031_MICBIAS, 2, 0), + SND_SOC_DAPM_MICBIAS("AMIC2Bias", SN95031_MICBIAS, 3, 0), + SND_SOC_DAPM_MICBIAS("DMIC12Bias", SN95031_DMICMUX, 3, 0), + SND_SOC_DAPM_MICBIAS("DMIC34Bias", SN95031_DMICMUX, 4, 0), + SND_SOC_DAPM_MICBIAS("DMIC56Bias", SN95031_DMICMUX, 5, 0), + + SND_SOC_DAPM_SUPPLY("DMIC12supply", SN95031_DMICLK, 0, 0, + sn95031_dmic12_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY("DMIC34supply", SN95031_DMICLK, 1, 0, + sn95031_dmic34_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY("DMIC56supply", SN95031_DMICLK, 2, 0, + sn95031_dmic56_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_AIF_OUT("PCM_Out", "Capture", 0, + SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_SUPPLY("Headset Rail", SND_SOC_NOPM, 0, 0, sn95031_vhs_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), @@ -209,6 +363,40 @@ static const struct snd_soc_dapm_widget sn95031_dapm_widgets[] = { SN95031_VIB1C5, 1, 0), SND_SOC_DAPM_DAC("Vibra2 DAC", "Vibra2", SN95031_VIB2C5, 1, 0), + + /* capture widgets */ + SND_SOC_DAPM_PGA("LineIn Enable Left", SN95031_MICAMP1, + 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("LineIn Enable Right", SN95031_MICAMP2, + 7, 0, NULL, 0), + + SND_SOC_DAPM_PGA("MIC1 Enable", SN95031_MICAMP1, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("MIC2 Enable", SN95031_MICAMP2, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("TX1 Enable", SN95031_AUDIOTXEN, 2, 0, NULL, 0), + SND_SOC_DAPM_PGA("TX2 Enable", SN95031_AUDIOTXEN, 3, 0, NULL, 0), + SND_SOC_DAPM_PGA("TX3 Enable", SN95031_AUDIOTXEN, 4, 0, NULL, 0), + SND_SOC_DAPM_PGA("TX4 Enable", SN95031_AUDIOTXEN, 5, 0, NULL, 0), + + /* ADC have null stream as they will be turned ON by TX path */ + SND_SOC_DAPM_ADC("ADC Left", NULL, + SN95031_ADCCONFIG, 0, 0), + SND_SOC_DAPM_ADC("ADC Right", NULL, + SN95031_ADCCONFIG, 2, 0), + + SND_SOC_DAPM_MUX("Mic_InputL Capture Route", + SND_SOC_NOPM, 0, 0, &sn95031_micl_mux_control), + SND_SOC_DAPM_MUX("Mic_InputR Capture Route", + SND_SOC_NOPM, 0, 0, &sn95031_micr_mux_control), + + SND_SOC_DAPM_MUX("Txpath1 Capture Route", + SND_SOC_NOPM, 0, 0, &sn95031_input1_mux_control), + SND_SOC_DAPM_MUX("Txpath2 Capture Route", + SND_SOC_NOPM, 0, 0, &sn95031_input2_mux_control), + SND_SOC_DAPM_MUX("Txpath3 Capture Route", + SND_SOC_NOPM, 0, 0, &sn95031_input3_mux_control), + SND_SOC_DAPM_MUX("Txpath4 Capture Route", + SND_SOC_NOPM, 0, 0, &sn95031_input4_mux_control), + }; static const struct snd_soc_dapm_route sn95031_audio_map[] = { @@ -250,6 +438,87 @@ static const struct snd_soc_dapm_route sn95031_audio_map[] = { { "Lineout Right Playback", NULL, "Headset Right Filter"}, { "Lineout Right Playback", NULL, "Speaker Right Filter"}, { "Lineout Right Playback", NULL, "Vibra2 DAC"}, + + /* Headset (AMIC1) mic */ + { "AMIC1Bias", NULL, "AMIC1"}, + { "MIC1 Enable", NULL, "AMIC1Bias"}, + { "Mic_InputL Capture Route", "AMIC", "MIC1 Enable"}, + + /* AMIC2 */ + { "AMIC2Bias", NULL, "AMIC2"}, + { "MIC2 Enable", NULL, "AMIC2Bias"}, + { "Mic_InputR Capture Route", "AMIC", "MIC2 Enable"}, + + + /* Linein */ + { "LineIn Enable Left", NULL, "LINEINL"}, + { "LineIn Enable Right", NULL, "LINEINR"}, + { "Mic_InputL Capture Route", "LineIn", "LineIn Enable Left"}, + { "Mic_InputR Capture Route", "LineIn", "LineIn Enable Right"}, + + /* ADC connection */ + { "ADC Left", NULL, "Mic_InputL Capture Route"}, + { "ADC Right", NULL, "Mic_InputR Capture Route"}, + + /*DMIC connections */ + { "DMIC1", NULL, "DMIC12supply"}, + { "DMIC2", NULL, "DMIC12supply"}, + { "DMIC3", NULL, "DMIC34supply"}, + { "DMIC4", NULL, "DMIC34supply"}, + { "DMIC5", NULL, "DMIC56supply"}, + { "DMIC6", NULL, "DMIC56supply"}, + + { "DMIC12Bias", NULL, "DMIC1"}, + { "DMIC12Bias", NULL, "DMIC2"}, + { "DMIC34Bias", NULL, "DMIC3"}, + { "DMIC34Bias", NULL, "DMIC4"}, + { "DMIC56Bias", NULL, "DMIC5"}, + { "DMIC56Bias", NULL, "DMIC6"}, + + /*TX path inputs*/ + { "Txpath1 Capture Route", "ADC Left", "ADC Left"}, + { "Txpath2 Capture Route", "ADC Left", "ADC Left"}, + { "Txpath3 Capture Route", "ADC Left", "ADC Left"}, + { "Txpath4 Capture Route", "ADC Left", "ADC Left"}, + { "Txpath1 Capture Route", "ADC Right", "ADC Right"}, + { "Txpath2 Capture Route", "ADC Right", "ADC Right"}, + { "Txpath3 Capture Route", "ADC Right", "ADC Right"}, + { "Txpath4 Capture Route", "ADC Right", "ADC Right"}, + { "Txpath1 Capture Route", NULL, "DMIC1"}, + { "Txpath2 Capture Route", NULL, "DMIC1"}, + { "Txpath3 Capture Route", NULL, "DMIC1"}, + { "Txpath4 Capture Route", NULL, "DMIC1"}, + { "Txpath1 Capture Route", NULL, "DMIC2"}, + { "Txpath2 Capture Route", NULL, "DMIC2"}, + { "Txpath3 Capture Route", NULL, "DMIC2"}, + { "Txpath4 Capture Route", NULL, "DMIC2"}, + { "Txpath1 Capture Route", NULL, "DMIC3"}, + { "Txpath2 Capture Route", NULL, "DMIC3"}, + { "Txpath3 Capture Route", NULL, "DMIC3"}, + { "Txpath4 Capture Route", NULL, "DMIC3"}, + { "Txpath1 Capture Route", NULL, "DMIC4"}, + { "Txpath2 Capture Route", NULL, "DMIC4"}, + { "Txpath3 Capture Route", NULL, "DMIC4"}, + { "Txpath4 Capture Route", NULL, "DMIC4"}, + { "Txpath1 Capture Route", NULL, "DMIC5"}, + { "Txpath2 Capture Route", NULL, "DMIC5"}, + { "Txpath3 Capture Route", NULL, "DMIC5"}, + { "Txpath4 Capture Route", NULL, "DMIC5"}, + { "Txpath1 Capture Route", NULL, "DMIC6"}, + { "Txpath2 Capture Route", NULL, "DMIC6"}, + { "Txpath3 Capture Route", NULL, "DMIC6"}, + { "Txpath4 Capture Route", NULL, "DMIC6"}, + + /* tx path */ + { "TX1 Enable", NULL, "Txpath1 Capture Route"}, + { "TX2 Enable", NULL, "Txpath2 Capture Route"}, + { "TX3 Enable", NULL, "Txpath3 Capture Route"}, + { "TX4 Enable", NULL, "Txpath4 Capture Route"}, + { "PCM_Out", NULL, "TX1 Enable"}, + { "PCM_Out", NULL, "TX2 Enable"}, + { "PCM_Out", NULL, "TX3 Enable"}, + { "PCM_Out", NULL, "TX4 Enable"}, + }; /* speaker and headset mutes, for audio pops and clicks */ @@ -339,6 +608,13 @@ struct snd_soc_dai_driver sn95031_dais[] = { .rates = SN95031_RATES, .formats = SN95031_FORMATS, }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 5, + .rates = SN95031_RATES, + .formats = SN95031_FORMATS, + }, .ops = &sn95031_headset_dai_ops, }, { .name = "SN95031 Speaker", @@ -390,6 +666,8 @@ static int sn95031_codec_probe(struct snd_soc_codec *codec) snd_soc_write(codec, SN95031_PCM2RXSLOT01, 0x10); snd_soc_write(codec, SN95031_PCM2RXSLOT23, 0x32); snd_soc_write(codec, SN95031_PCM2RXSLOT45, 0x54); + snd_soc_write(codec, SN95031_PCM2TXSLOT01, 0x10); + snd_soc_write(codec, SN95031_PCM2TXSLOT23, 0x32); /* pcm port setting * This sets the pcm port to slave and clock at 19.2Mhz which * can support 6slots, sampling rate set per stream in hw-params @@ -423,6 +701,9 @@ static int sn95031_codec_probe(struct snd_soc_codec *codec) snd_soc_write(codec, SN95031_SSR2, 0x10); snd_soc_write(codec, SN95031_SSR3, 0x40); + snd_soc_add_controls(codec, sn95031_snd_controls, + ARRAY_SIZE(sn95031_snd_controls)); + ret = snd_soc_dapm_new_controls(&codec->dapm, sn95031_dapm_widgets, ARRAY_SIZE(sn95031_dapm_widgets)); if (ret) -- cgit v1.2.3-70-g09d2 From d71bb810be602b7194d0dc336854e2f4a3d44492 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 31 Jan 2011 13:41:03 +0000 Subject: ASoC: Accept any logical value for WM8962 GPIO set() Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8962.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 5c7b730a864..3b71dd65c96 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3635,7 +3635,7 @@ static void wm8962_gpio_set(struct gpio_chip *chip, unsigned offset, int value) struct snd_soc_codec *codec = wm8962->codec; snd_soc_update_bits(codec, WM8962_GPIO_BASE + offset, - WM8962_GP2_LVL, value << WM8962_GP2_LVL_SHIFT); + WM8962_GP2_LVL, !!value << WM8962_GP2_LVL_SHIFT); } static int wm8962_gpio_direction_out(struct gpio_chip *chip, -- cgit v1.2.3-70-g09d2 From c8059930f0aafb77d94d268dfb4fceaa15cd2ad9 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 31 Jan 2011 13:41:17 +0000 Subject: ASoC: Accept any logical value WM8903 GPIO set() Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8903.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 9c4f2c4febc..3d4c55f3c7b 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1702,7 +1702,8 @@ static void wm8903_gpio_set(struct gpio_chip *chip, unsigned offset, int value) struct snd_soc_codec *codec = wm8903->codec; snd_soc_update_bits(codec, WM8903_GPIO_CONTROL_1 + offset, - WM8903_GP1_LVL_MASK, value << WM8903_GP1_LVL_SHIFT); + WM8903_GP1_LVL_MASK, + !!value << WM8903_GP1_LVL_SHIFT); } static struct gpio_chip wm8903_template_chip = { -- cgit v1.2.3-70-g09d2 From c45bfccfa299a9b3a20373963c286b4e9bd3c40f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 3 Feb 2011 14:07:15 +0000 Subject: ASoC: Sort ALC5623 in Kconfig and Makefile Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/Makefile | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 83b7accd703..ae10507dd2e 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -91,6 +91,7 @@ obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o obj-$(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-$(CONFIG_SND_SOC_ALC5623) += snd-soc-alc5623.o obj-$(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 @@ -102,7 +103,6 @@ obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o -obj-$(CONFIG_SND_SOC_ALC5623) += snd-soc-alc5623.o obj-$(CONFIG_SND_SOC_SN95031) +=snd-soc-sn95031.o obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o -- cgit v1.2.3-70-g09d2 From 338ee25393a5627e8ded5819147f98b919656ce9 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 6 Feb 2011 10:04:11 +0100 Subject: ASoC: codecs: wm8753: Fix DAI mode switching The wm8753 codec supports switching between different DAI modes. The current drivers tries to implement this by changing the DAI driver at runtime. But to properly work this would require support from the ASoC core. So this patch takes a different approch on how the DAI mode switching is implemented. The only difference, from a driver point of view, between the different DAI modes is how to program the DAI format to the hardware. So what this patch is, it stores the current format for each DAI in the drivers private struct and when the DAI mode is changed the format gets simply reprogrammed according to the new DAI mode. Futhermore this patch restricts the changing of the DAI format to when the codec is inactive. Signed-off-by: Lars-Peter Clausen Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/wm8753.c | 296 +++++++++++++++++++--------------------------- 1 file changed, 121 insertions(+), 175 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 79b02ae125c..3f09deea8d9 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -55,8 +55,10 @@ static int caps_charge = 2000; module_param(caps_charge, int, 0); MODULE_PARM_DESC(caps_charge, "WM8753 cap charge time (msecs)"); -static void wm8753_set_dai_mode(struct snd_soc_codec *codec, - struct snd_soc_dai *dai, unsigned int hifi); +static int wm8753_hifi_write_dai_fmt(struct snd_soc_codec *codec, + unsigned int fmt); +static int wm8753_voice_write_dai_fmt(struct snd_soc_codec *codec, + unsigned int fmt); /* * wm8753 register cache @@ -87,6 +89,10 @@ struct wm8753_priv { enum snd_soc_control_type control_type; unsigned int sysclk; unsigned int pcmclk; + + unsigned int voice_fmt; + unsigned int hifi_fmt; + int dai_func; }; @@ -170,9 +176,9 @@ static int wm8753_get_dai(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - int mode = snd_soc_read(codec, WM8753_IOCTL); + struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec); - ucontrol->value.integer.value[0] = (mode & 0xc) >> 2; + ucontrol->value.integer.value[0] = wm8753->dai_func; return 0; } @@ -180,16 +186,26 @@ static int wm8753_set_dai(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - int mode = snd_soc_read(codec, WM8753_IOCTL); struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec); + u16 ioctl; + + if (codec->active) + return -EBUSY; + + ioctl = snd_soc_read(codec, WM8753_IOCTL); + + wm8753->dai_func = ucontrol->value.integer.value[0]; + + if (((ioctl >> 2) & 0x3) == wm8753->dai_func) + return 1; + + ioctl = (ioctl & 0x1f3) | (wm8753->dai_func << 2); + snd_soc_write(codec, WM8753_IOCTL, ioctl); - if (((mode & 0xc) >> 2) == ucontrol->value.integer.value[0]) - return 0; - mode &= 0xfff3; - mode |= (ucontrol->value.integer.value[0] << 2); + wm8753_hifi_write_dai_fmt(codec, wm8753->hifi_fmt); + wm8753_voice_write_dai_fmt(codec, wm8753->voice_fmt); - wm8753->dai_func = ucontrol->value.integer.value[0]; return 1; } @@ -828,10 +844,9 @@ static int wm8753_set_dai_sysclk(struct snd_soc_dai *codec_dai, /* * Set's ADC and Voice DAC format. */ -static int wm8753_vdac_adc_set_dai_fmt(struct snd_soc_dai *codec_dai, +static int wm8753_vdac_adc_set_dai_fmt(struct snd_soc_codec *codec, unsigned int fmt) { - struct snd_soc_codec *codec = codec_dai->codec; u16 voice = snd_soc_read(codec, WM8753_PCM) & 0x01ec; /* interface format */ @@ -858,13 +873,6 @@ static int wm8753_vdac_adc_set_dai_fmt(struct snd_soc_dai *codec_dai, return 0; } -static int wm8753_pcm_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - wm8753_set_dai_mode(dai->codec, dai, 0); - return 0; -} - /* * Set PCM DAI bit size and sample rate. */ @@ -905,10 +913,9 @@ static int wm8753_pcm_hw_params(struct snd_pcm_substream *substream, /* * Set's PCM dai fmt and BCLK. */ -static int wm8753_pcm_set_dai_fmt(struct snd_soc_dai *codec_dai, +static int wm8753_pcm_set_dai_fmt(struct snd_soc_codec *codec, unsigned int fmt) { - struct snd_soc_codec *codec = codec_dai->codec; u16 voice, ioctl; voice = snd_soc_read(codec, WM8753_PCM) & 0x011f; @@ -999,10 +1006,9 @@ static int wm8753_set_dai_clkdiv(struct snd_soc_dai *codec_dai, /* * Set's HiFi DAC format. */ -static int wm8753_hdac_set_dai_fmt(struct snd_soc_dai *codec_dai, +static int wm8753_hdac_set_dai_fmt(struct snd_soc_codec *codec, unsigned int fmt) { - struct snd_soc_codec *codec = codec_dai->codec; u16 hifi = snd_soc_read(codec, WM8753_HIFI) & 0x01e0; /* interface format */ @@ -1032,10 +1038,9 @@ static int wm8753_hdac_set_dai_fmt(struct snd_soc_dai *codec_dai, /* * Set's I2S DAI format. */ -static int wm8753_i2s_set_dai_fmt(struct snd_soc_dai *codec_dai, +static int wm8753_i2s_set_dai_fmt(struct snd_soc_codec *codec, unsigned int fmt) { - struct snd_soc_codec *codec = codec_dai->codec; u16 ioctl, hifi; hifi = snd_soc_read(codec, WM8753_HIFI) & 0x011f; @@ -1098,13 +1103,6 @@ static int wm8753_i2s_set_dai_fmt(struct snd_soc_dai *codec_dai, return 0; } -static int wm8753_i2s_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - wm8753_set_dai_mode(dai->codec, dai, 1); - return 0; -} - /* * Set PCM DAI bit size and sample rate. */ @@ -1147,61 +1145,117 @@ static int wm8753_i2s_hw_params(struct snd_pcm_substream *substream, return 0; } -static int wm8753_mode1v_set_dai_fmt(struct snd_soc_dai *codec_dai, +static int wm8753_mode1v_set_dai_fmt(struct snd_soc_codec *codec, unsigned int fmt) { - struct snd_soc_codec *codec = codec_dai->codec; u16 clock; /* set clk source as pcmclk */ clock = snd_soc_read(codec, WM8753_CLOCK) & 0xfffb; snd_soc_write(codec, WM8753_CLOCK, clock); - if (wm8753_vdac_adc_set_dai_fmt(codec_dai, fmt) < 0) - return -EINVAL; - return wm8753_pcm_set_dai_fmt(codec_dai, fmt); + return wm8753_vdac_adc_set_dai_fmt(codec, fmt); } -static int wm8753_mode1h_set_dai_fmt(struct snd_soc_dai *codec_dai, +static int wm8753_mode1h_set_dai_fmt(struct snd_soc_codec *codec, unsigned int fmt) { - if (wm8753_hdac_set_dai_fmt(codec_dai, fmt) < 0) - return -EINVAL; - return wm8753_i2s_set_dai_fmt(codec_dai, fmt); + return wm8753_hdac_set_dai_fmt(codec, fmt); } -static int wm8753_mode2_set_dai_fmt(struct snd_soc_dai *codec_dai, +static int wm8753_mode2_set_dai_fmt(struct snd_soc_codec *codec, unsigned int fmt) { - struct snd_soc_codec *codec = codec_dai->codec; u16 clock; /* set clk source as pcmclk */ clock = snd_soc_read(codec, WM8753_CLOCK) & 0xfffb; snd_soc_write(codec, WM8753_CLOCK, clock); - if (wm8753_vdac_adc_set_dai_fmt(codec_dai, fmt) < 0) - return -EINVAL; - return wm8753_i2s_set_dai_fmt(codec_dai, fmt); + return wm8753_vdac_adc_set_dai_fmt(codec, fmt); } -static int wm8753_mode3_4_set_dai_fmt(struct snd_soc_dai *codec_dai, +static int wm8753_mode3_4_set_dai_fmt(struct snd_soc_codec *codec, unsigned int fmt) { - struct snd_soc_codec *codec = codec_dai->codec; u16 clock; /* set clk source as mclk */ clock = snd_soc_read(codec, WM8753_CLOCK) & 0xfffb; snd_soc_write(codec, WM8753_CLOCK, clock | 0x4); - if (wm8753_hdac_set_dai_fmt(codec_dai, fmt) < 0) + if (wm8753_hdac_set_dai_fmt(codec, fmt) < 0) return -EINVAL; - if (wm8753_vdac_adc_set_dai_fmt(codec_dai, fmt) < 0) - return -EINVAL; - return wm8753_i2s_set_dai_fmt(codec_dai, fmt); + return wm8753_vdac_adc_set_dai_fmt(codec, fmt); } +static int wm8753_hifi_write_dai_fmt(struct snd_soc_codec *codec, + unsigned int fmt) +{ + struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + switch (wm8753->dai_func) { + case 0: + ret = wm8753_mode1h_set_dai_fmt(codec, fmt); + break; + case 1: + ret = wm8753_mode2_set_dai_fmt(codec, fmt); + break; + case 2: + case 3: + ret = wm8753_mode3_4_set_dai_fmt(codec, fmt); + break; + default: + break; + } + if (ret) + return ret; + + return wm8753_i2s_set_dai_fmt(codec, fmt); +} + +static int wm8753_hifi_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec); + + wm8753->hifi_fmt = fmt; + + return wm8753_hifi_write_dai_fmt(codec, fmt); +}; + +static int wm8753_voice_write_dai_fmt(struct snd_soc_codec *codec, + unsigned int fmt) +{ + struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + if (wm8753->dai_func != 0) + return 0; + + ret = wm8753_mode1v_set_dai_fmt(codec, fmt); + if (ret) + return ret; + ret = wm8753_pcm_set_dai_fmt(codec, fmt); + if (ret) + return ret; + + return 0; +}; + +static int wm8753_voice_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec); + + wm8753->voice_fmt = fmt; + + return wm8753_voice_write_dai_fmt(codec, fmt); +}; + static int wm8753_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; @@ -1268,57 +1322,25 @@ static int wm8753_set_bias_level(struct snd_soc_codec *codec, * 3. Voice disabled - HIFI over HIFI * 4. Voice disabled - HIFI over HIFI, uses voice DAI LRC for capture */ -static struct snd_soc_dai_ops wm8753_dai_ops_hifi_mode1 = { - .startup = wm8753_i2s_startup, +static struct snd_soc_dai_ops wm8753_dai_ops_hifi_mode = { .hw_params = wm8753_i2s_hw_params, .digital_mute = wm8753_mute, - .set_fmt = wm8753_mode1h_set_dai_fmt, - .set_clkdiv = wm8753_set_dai_clkdiv, - .set_pll = wm8753_set_dai_pll, - .set_sysclk = wm8753_set_dai_sysclk, -}; - -static struct snd_soc_dai_ops wm8753_dai_ops_voice_mode1 = { - .startup = wm8753_pcm_startup, - .hw_params = wm8753_pcm_hw_params, - .digital_mute = wm8753_mute, - .set_fmt = wm8753_mode1v_set_dai_fmt, + .set_fmt = wm8753_hifi_set_dai_fmt, .set_clkdiv = wm8753_set_dai_clkdiv, .set_pll = wm8753_set_dai_pll, .set_sysclk = wm8753_set_dai_sysclk, }; -static struct snd_soc_dai_ops wm8753_dai_ops_voice_mode2 = { - .startup = wm8753_pcm_startup, +static struct snd_soc_dai_ops wm8753_dai_ops_voice_mode = { .hw_params = wm8753_pcm_hw_params, .digital_mute = wm8753_mute, - .set_fmt = wm8753_mode2_set_dai_fmt, - .set_clkdiv = wm8753_set_dai_clkdiv, - .set_pll = wm8753_set_dai_pll, - .set_sysclk = wm8753_set_dai_sysclk, -}; - -static struct snd_soc_dai_ops wm8753_dai_ops_hifi_mode3 = { - .startup = wm8753_i2s_startup, - .hw_params = wm8753_i2s_hw_params, - .digital_mute = wm8753_mute, - .set_fmt = wm8753_mode3_4_set_dai_fmt, - .set_clkdiv = wm8753_set_dai_clkdiv, - .set_pll = wm8753_set_dai_pll, - .set_sysclk = wm8753_set_dai_sysclk, -}; - -static struct snd_soc_dai_ops wm8753_dai_ops_hifi_mode4 = { - .startup = wm8753_i2s_startup, - .hw_params = wm8753_i2s_hw_params, - .digital_mute = wm8753_mute, - .set_fmt = wm8753_mode3_4_set_dai_fmt, + .set_fmt = wm8753_voice_set_dai_fmt, .set_clkdiv = wm8753_set_dai_clkdiv, .set_pll = wm8753_set_dai_pll, .set_sysclk = wm8753_set_dai_sysclk, }; -static struct snd_soc_dai_driver wm8753_all_dai[] = { +static struct snd_soc_dai_driver wm8753_dai[] = { /* DAI HiFi mode 1 */ { .name = "wm8753-hifi", .playback = { @@ -1326,14 +1348,16 @@ static struct snd_soc_dai_driver wm8753_all_dai[] = { .channels_min = 1, .channels_max = 2, .rates = WM8753_RATES, - .formats = WM8753_FORMATS}, + .formats = WM8753_FORMATS + }, .capture = { /* dummy for fast DAI switching */ .stream_name = "Capture", .channels_min = 1, .channels_max = 2, .rates = WM8753_RATES, - .formats = WM8753_FORMATS}, - .ops = &wm8753_dai_ops_hifi_mode1, + .formats = WM8753_FORMATS + }, + .ops = &wm8753_dai_ops_hifi_mode, }, /* DAI Voice mode 1 */ { .name = "wm8753-voice", @@ -1342,97 +1366,19 @@ static struct snd_soc_dai_driver wm8753_all_dai[] = { .channels_min = 1, .channels_max = 1, .rates = WM8753_RATES, - .formats = WM8753_FORMATS,}, - .capture = { - .stream_name = "Capture", - .channels_min = 1, - .channels_max = 2, - .rates = WM8753_RATES, - .formats = WM8753_FORMATS,}, - .ops = &wm8753_dai_ops_voice_mode1, -}, -/* DAI HiFi mode 2 - dummy */ -{ .name = "wm8753-hifi", -}, -/* DAI Voice mode 2 */ -{ .name = "wm8753-voice", - .playback = { - .stream_name = "Voice Playback", - .channels_min = 1, - .channels_max = 1, - .rates = WM8753_RATES, - .formats = WM8753_FORMATS,}, - .capture = { - .stream_name = "Capture", - .channels_min = 1, - .channels_max = 2, - .rates = WM8753_RATES, - .formats = WM8753_FORMATS,}, - .ops = &wm8753_dai_ops_voice_mode2, -}, -/* DAI HiFi mode 3 */ -{ .name = "wm8753-hifi", - .playback = { - .stream_name = "HiFi Playback", - .channels_min = 1, - .channels_max = 2, - .rates = WM8753_RATES, - .formats = WM8753_FORMATS,}, - .capture = { - .stream_name = "Capture", - .channels_min = 1, - .channels_max = 2, - .rates = WM8753_RATES, - .formats = WM8753_FORMATS,}, - .ops = &wm8753_dai_ops_hifi_mode3, -}, -/* DAI Voice mode 3 - dummy */ -{ .name = "wm8753-voice", -}, -/* DAI HiFi mode 4 */ -{ .name = "wm8753-hifi", - .playback = { - .stream_name = "HiFi Playback", - .channels_min = 1, - .channels_max = 2, - .rates = WM8753_RATES, - .formats = WM8753_FORMATS,}, + .formats = WM8753_FORMATS, + }, .capture = { .stream_name = "Capture", .channels_min = 1, .channels_max = 2, .rates = WM8753_RATES, - .formats = WM8753_FORMATS,}, - .ops = &wm8753_dai_ops_hifi_mode4, -}, -/* DAI Voice mode 4 - dummy */ -{ .name = "wm8753-voice", -}, -}; - -static struct snd_soc_dai_driver wm8753_dai[] = { - { - .name = "wm8753-aif0", - }, - { - .name = "wm8753-aif1", + .formats = WM8753_FORMATS, }, + .ops = &wm8753_dai_ops_voice_mode, +}, }; -static void wm8753_set_dai_mode(struct snd_soc_codec *codec, - struct snd_soc_dai *dai, unsigned int hifi) -{ - struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec); - - if (wm8753->dai_func < 4) { - if (hifi) - dai->driver = &wm8753_all_dai[wm8753->dai_func << 1]; - else - dai->driver = &wm8753_all_dai[(wm8753->dai_func << 1) + 1]; - } - snd_soc_write(codec, WM8753_IOCTL, wm8753->dai_func); -} - static void wm8753_work(struct work_struct *work) { struct snd_soc_dapm_context *dapm = -- cgit v1.2.3-70-g09d2 From a98a0bc6c92eacd181417a9c0ccd2e8028066622 Mon Sep 17 00:00:00 2001 From: Alexander Sverdlin Date: Thu, 3 Feb 2011 03:11:45 +0300 Subject: ASoC: CS4271: Move Chip Select control out of the CODEC code. Move Chip Select control out of the CODEC code for CS4271. Signed-off-by: Alexander Sverdlin Reviewed-by: H Hartley Sweeten Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- include/sound/cs4271.h | 1 - sound/soc/codecs/cs4271.c | 22 +++------------------- 2 files changed, 3 insertions(+), 20 deletions(-) (limited to 'sound/soc/codecs') diff --git a/include/sound/cs4271.h b/include/sound/cs4271.h index 16f8d325d3d..50a059e7d11 100644 --- a/include/sound/cs4271.h +++ b/include/sound/cs4271.h @@ -19,7 +19,6 @@ struct cs4271_platform_data { int gpio_nreset; /* GPIO driving Reset pin, if any */ - int gpio_disable; /* GPIO that disable serial bus, if any */ }; #endif /* __CS4271_H */ diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 9c5b7db0ce6..1791796216c 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -441,22 +441,11 @@ static int cs4271_probe(struct snd_soc_codec *codec) struct cs4271_platform_data *cs4271plat = codec->dev->platform_data; int ret; int gpio_nreset = -EINVAL; - int gpio_disable = -EINVAL; codec->control_data = cs4271->control_data; - if (cs4271plat) { - if (gpio_is_valid(cs4271plat->gpio_nreset)) - gpio_nreset = cs4271plat->gpio_nreset; - if (gpio_is_valid(cs4271plat->gpio_disable)) - gpio_disable = cs4271plat->gpio_disable; - } - - if (gpio_disable >= 0) - if (gpio_request(gpio_disable, "CS4271 Disable")) - gpio_disable = -EINVAL; - if (gpio_disable >= 0) - gpio_direction_output(gpio_disable, 0); + if (cs4271plat && gpio_is_valid(cs4271plat->gpio_nreset)) + gpio_nreset = cs4271plat->gpio_nreset; if (gpio_nreset >= 0) if (gpio_request(gpio_nreset, "CS4271 Reset")) @@ -471,7 +460,6 @@ static int cs4271_probe(struct snd_soc_codec *codec) } cs4271->gpio_nreset = gpio_nreset; - cs4271->gpio_disable = gpio_disable; /* * In case of I2C, chip address specified in board data. @@ -509,10 +497,9 @@ static int cs4271_probe(struct snd_soc_codec *codec) static int cs4271_remove(struct snd_soc_codec *codec) { struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); - int gpio_nreset, gpio_disable; + int gpio_nreset; gpio_nreset = cs4271->gpio_nreset; - gpio_disable = cs4271->gpio_disable; if (gpio_is_valid(gpio_nreset)) { /* Set codec to the reset state */ @@ -520,9 +507,6 @@ static int cs4271_remove(struct snd_soc_codec *codec) gpio_free(gpio_nreset); } - if (gpio_is_valid(gpio_disable)) - gpio_free(gpio_disable); - return 0; }; -- cgit v1.2.3-70-g09d2 From 1e2f5932e472f11b987e8339ffc855aa00ecebf5 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Wed, 9 Feb 2011 21:44:32 +0530 Subject: ASoC: sn95031: Add jack support in the codec This patch adds support for jack detection and reporting in the codec It however is not fully functional as it doesn't measure adc to figure out what got inserted which will be added later Signed-off-by: Vinod Koul Signed-off-by: Harsha Priya Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/sn95031.c | 56 ++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/sn95031.h | 9 ++++++++ 2 files changed, 65 insertions(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index 40e285df9ae..b49d79017d3 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -34,6 +34,7 @@ #include #include #include +#include #include "sn95031.h" #define SN95031_RATES (SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_44100) @@ -649,6 +650,61 @@ struct snd_soc_dai_driver sn95031_dais[] = { }, }; +static inline void sn95031_disable_jack_btn(struct snd_soc_codec *codec) +{ + snd_soc_write(codec, SN95031_BTNCTRL2, 0x00); +} + +static inline void sn95031_enable_jack_btn(struct snd_soc_codec *codec) +{ + snd_soc_write(codec, SN95031_BTNCTRL1, 0x77); + snd_soc_write(codec, SN95031_BTNCTRL2, 0x01); +} + +static int sn95031_get_headset_state(struct snd_soc_jack *mfld_jack) +{ + /* Defaulting to HEADSET for now. + * will change after adding soc-jack detection apis */ + int jack_type = SND_JACK_HEADSET; + + pr_debug("jack type detected = %d\n", jack_type); + if (jack_type == SND_JACK_HEADSET) + sn95031_enable_jack_btn(mfld_jack->codec); + return jack_type; +} + +void sn95031_jack_detection(struct mfld_jack_data *jack_data) +{ + unsigned int status; + unsigned int mask = SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_HEADSET; + + pr_debug("interrupt id read in sram = 0x%x\n", jack_data->intr_id); + if (jack_data->intr_id & 0x1) { + pr_debug("short_push detected\n"); + status = SND_JACK_HEADSET | SND_JACK_BTN_0; + } else if (jack_data->intr_id & 0x2) { + pr_debug("long_push detected\n"); + status = SND_JACK_HEADSET | SND_JACK_BTN_1; + } else if (jack_data->intr_id & 0x4) { + pr_debug("headset or headphones inserted\n"); + status = sn95031_get_headset_state(jack_data->mfld_jack); + } else if (jack_data->intr_id & 0x8) { + pr_debug("headset or headphones removed\n"); + status = 0; + sn95031_disable_jack_btn(jack_data->mfld_jack->codec); + } else { + pr_err("unidentified interrupt\n"); + return; + } + + snd_soc_jack_report(jack_data->mfld_jack, status, mask); + /*button pressed and released so we send explicit button release */ + if ((status & SND_JACK_BTN_0) | (status & SND_JACK_BTN_1)) + snd_soc_jack_report(jack_data->mfld_jack, + SND_JACK_HEADSET, mask); +} +EXPORT_SYMBOL_GPL(sn95031_jack_detection); + /* codec registration */ static int sn95031_codec_probe(struct snd_soc_codec *codec) { diff --git a/sound/soc/codecs/sn95031.h b/sound/soc/codecs/sn95031.h index e2b17d908ae..2dbae614bac 100644 --- a/sound/soc/codecs/sn95031.h +++ b/sound/soc/codecs/sn95031.h @@ -96,4 +96,13 @@ #define SN95031_SSR5 0x384 #define SN95031_SSR6 0x385 +#define SN95031_AUDIO_GPIO_CTRL 0x070 +struct mfld_jack_data { + int intr_id; + int micbias_vol; + struct snd_soc_jack *mfld_jack; +}; + +extern void sn95031_jack_detection(struct mfld_jack_data *jack_data); + #endif -- cgit v1.2.3-70-g09d2 From 36633237be60c0ec88b11e00d5fa22a305563d03 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Wed, 9 Feb 2011 21:44:34 +0530 Subject: ASoC: sn95031: Add support for reading mic bias This patch adds support to read the mic bias voltage when a jack is inserted. It uses ADC to measure. Signed-off-by: Vinod Koul Signed-off-by: Harsha Priya Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/sn95031.c | 127 +++++++++++++++++++++++++++++++++++++++++++-- sound/soc/codecs/sn95031.h | 24 +++++++++ 2 files changed, 147 insertions(+), 4 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index b49d79017d3..4cc00177ee3 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -40,12 +40,129 @@ #define SN95031_RATES (SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_44100) #define SN95031_FORMATS (SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE) +/* adc helper functions */ + +/* enables mic bias voltage */ +static void sn95031_enable_mic_bias(struct snd_soc_codec *codec) +{ + snd_soc_write(codec, SN95031_VAUD, BIT(2)|BIT(1)|BIT(0)); + snd_soc_update_bits(codec, SN95031_MICBIAS, BIT(2), BIT(2)); +} + +/* Enable/Disable the ADC depending on the argument */ +static void configure_adc(struct snd_soc_codec *sn95031_codec, int val) +{ + int value = snd_soc_read(sn95031_codec, SN95031_ADC1CNTL1); + + if (val) { + /* Enable and start the ADC */ + value |= (SN95031_ADC_ENBL | SN95031_ADC_START); + value &= (~SN95031_ADC_NO_LOOP); + } else { + /* Just stop the ADC */ + value &= (~SN95031_ADC_START); + } + snd_soc_write(sn95031_codec, SN95031_ADC1CNTL1, value); +} + /* - * todo: - * capture paths - * jack detection - * PM functions + * finds an empty channel for conversion + * If the ADC is not enabled then start using 0th channel + * itself. Otherwise find an empty channel by looking for a + * channel in which the stopbit is set to 1. returns the index + * of the first free channel if succeeds or an error code. + * + * Context: can sleep + * */ +static int find_free_channel(struct snd_soc_codec *sn95031_codec) +{ + int ret = 0, i, value; + + /* check whether ADC is enabled */ + value = snd_soc_read(sn95031_codec, SN95031_ADC1CNTL1); + + if ((value & SN95031_ADC_ENBL) == 0) + return 0; + + /* ADC is already enabled; Looking for an empty channel */ + for (i = 0; i < SN95031_ADC_CHANLS_MAX; i++) { + value = snd_soc_read(sn95031_codec, + SN95031_ADC_CHNL_START_ADDR + i); + if (value & SN95031_STOPBIT_MASK) { + ret = i; + break; + } + } + return (ret > SN95031_ADC_LOOP_MAX) ? (-EINVAL) : ret; +} + +/* Initialize the ADC for reading micbias values. Can sleep. */ +static int sn95031_initialize_adc(struct snd_soc_codec *sn95031_codec) +{ + int base_addr, chnl_addr; + int value; + static int channel_index; + + /* Index of the first channel in which the stop bit is set */ + channel_index = find_free_channel(sn95031_codec); + if (channel_index < 0) { + pr_err("No free ADC channels"); + return channel_index; + } + + base_addr = SN95031_ADC_CHNL_START_ADDR + channel_index; + + if (!(channel_index == 0 || channel_index == SN95031_ADC_LOOP_MAX)) { + /* Reset stop bit for channels other than 0 and 12 */ + value = snd_soc_read(sn95031_codec, base_addr); + /* Set the stop bit to zero */ + snd_soc_write(sn95031_codec, base_addr, value & 0xEF); + /* Index of the first free channel */ + base_addr++; + channel_index++; + } + + /* Since this is the last channel, set the stop bit + to 1 by ORing the DIE_SENSOR_CODE with 0x10 */ + snd_soc_write(sn95031_codec, base_addr, + SN95031_AUDIO_DETECT_CODE | 0x10); + + chnl_addr = SN95031_ADC_DATA_START_ADDR + 2 * channel_index; + pr_debug("mid_initialize : %x", chnl_addr); + configure_adc(sn95031_codec, 1); + return chnl_addr; +} + + +/* reads the ADC registers and gets the mic bias value in mV. */ +static unsigned int sn95031_get_mic_bias(struct snd_soc_codec *codec) +{ + u16 adc_adr = sn95031_initialize_adc(codec); + u16 adc_val1, adc_val2; + unsigned int mic_bias; + + sn95031_enable_mic_bias(codec); + + /* Enable the sound card for conversion before reading */ + snd_soc_write(codec, SN95031_ADC1CNTL3, 0x05); + /* Re-toggle the RRDATARD bit */ + snd_soc_write(codec, SN95031_ADC1CNTL3, 0x04); + + /* Read the higher bits of data */ + msleep(1000); + adc_val1 = snd_soc_read(codec, adc_adr); + adc_adr++; + adc_val2 = snd_soc_read(codec, adc_adr); + + /* Adding lower two bits to the higher bits */ + mic_bias = (adc_val1 << 2) + (adc_val2 & 3); + mic_bias = (mic_bias * SN95031_ADC_ONE_LSB_MULTIPLIER) / 1000; + pr_debug("mic bias = %dmV\n", mic_bias); + return mic_bias; +} +EXPORT_SYMBOL_GPL(sn95031_get_mic_bias); +/*end - adc helper functions */ static inline unsigned int sn95031_read(struct snd_soc_codec *codec, unsigned int reg) @@ -663,6 +780,8 @@ static inline void sn95031_enable_jack_btn(struct snd_soc_codec *codec) static int sn95031_get_headset_state(struct snd_soc_jack *mfld_jack) { + int micbias = sn95031_get_mic_bias(mfld_jack->codec); + /* Defaulting to HEADSET for now. * will change after adding soc-jack detection apis */ int jack_type = SND_JACK_HEADSET; diff --git a/sound/soc/codecs/sn95031.h b/sound/soc/codecs/sn95031.h index 2dbae614bac..20376d234fb 100644 --- a/sound/soc/codecs/sn95031.h +++ b/sound/soc/codecs/sn95031.h @@ -96,7 +96,31 @@ #define SN95031_SSR5 0x384 #define SN95031_SSR6 0x385 +/* ADC registers */ + +#define SN95031_ADC1CNTL1 0x1C0 +#define SN95031_ADC_ENBL 0x10 +#define SN95031_ADC_START 0x08 +#define SN95031_ADC1CNTL3 0x1C2 +#define SN95031_ADCTHERM_ENBL 0x04 +#define SN95031_ADCRRDATA_ENBL 0x05 +#define SN95031_STOPBIT_MASK 16 +#define SN95031_ADCTHERM_MASK 4 +#define SN95031_ADC_CHANLS_MAX 15 /* Number of ADC channels */ +#define SN95031_ADC_LOOP_MAX (SN95031_ADC_CHANLS_MAX - 1) +#define SN95031_ADC_NO_LOOP 0x07 #define SN95031_AUDIO_GPIO_CTRL 0x070 + +/* ADC channel code values */ +#define SN95031_AUDIO_DETECT_CODE 0x06 + +/* ADC base addresses */ +#define SN95031_ADC_CHNL_START_ADDR 0x1C5 /* increments by 1 */ +#define SN95031_ADC_DATA_START_ADDR 0x1D4 /* increments by 2 */ +/* multipier to convert to mV */ +#define SN95031_ADC_ONE_LSB_MULTIPLIER 2346 + + struct mfld_jack_data { int intr_id; int micbias_vol; -- cgit v1.2.3-70-g09d2 From 4b592c919c694de79c31d5fde59c169fc79595a9 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 9 Feb 2011 13:47:06 +0000 Subject: ASoC: Remove redundant -codec from WM8903 driver name It causes noisy -codecs to appear in things like .codec_name. Signed-off-by: Mark Brown Acked-by: Stephen Warren Acked-by: Liam Girdwood --- sound/soc/codecs/wm8903.c | 2 +- sound/soc/tegra/harmony.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 3d4c55f3c7b..7c84ebf3420 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1932,7 +1932,7 @@ MODULE_DEVICE_TABLE(i2c, wm8903_i2c_id); static struct i2c_driver wm8903_i2c_driver = { .driver = { - .name = "wm8903-codec", + .name = "wm8903", .owner = THIS_MODULE, }, .probe = wm8903_i2c_probe, diff --git a/sound/soc/tegra/harmony.c b/sound/soc/tegra/harmony.c index be95405df3f..61befcc281a 100644 --- a/sound/soc/tegra/harmony.c +++ b/sound/soc/tegra/harmony.c @@ -225,7 +225,7 @@ static int harmony_asoc_init(struct snd_soc_pcm_runtime *rtd) static struct snd_soc_dai_link harmony_wm8903_dai = { .name = "WM8903", .stream_name = "WM8903 PCM", - .codec_name = "wm8903-codec.0-001a", + .codec_name = "wm8903.0-001a", .platform_name = "tegra-pcm-audio", .cpu_dai_name = "tegra-i2s.0", .codec_dai_name = "wm8903-hifi", -- cgit v1.2.3-70-g09d2 From 1d8d62d637577afc568da32e3ecc7bddc11db56d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 9 Feb 2011 13:47:07 +0000 Subject: ASoC: Display WM8903 chip revision alphabetically Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8903.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 7c84ebf3420..e6ba4d05bf2 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1781,8 +1781,8 @@ static int wm8903_probe(struct snd_soc_codec *codec) } val = snd_soc_read(codec, WM8903_REVISION_NUMBER); - dev_info(codec->dev, "WM8903 revision %d\n", - val & WM8903_CHIP_REV_MASK); + dev_info(codec->dev, "WM8903 revision %c\n", + (val & WM8903_CHIP_REV_MASK) + 'A'); wm8903_reset(codec); -- cgit v1.2.3-70-g09d2 From 1e113bf9e088f1a6f4a1cdadce598ccc340f8fc1 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 9 Feb 2011 13:47:08 +0000 Subject: ASoC: Add support for AIF channel muxing on WM8903 Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8903.c | 66 ++++++++++++++++++++++++++++++++++++++++++++--- 1 file changed, 62 insertions(+), 4 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index e6ba4d05bf2..0190c5aa44f 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -674,6 +674,22 @@ static const struct soc_enum lsidetone_enum = static const struct soc_enum rsidetone_enum = SOC_ENUM_SINGLE(WM8903_DAC_DIGITAL_0, 0, 3, sidetone_text); +static const char *aif_text[] = { + "Left", "Right" +}; + +static const struct soc_enum lcapture_enum = + SOC_ENUM_SINGLE(WM8903_AUDIO_INTERFACE_0, 7, 2, aif_text); + +static const struct soc_enum rcapture_enum = + SOC_ENUM_SINGLE(WM8903_AUDIO_INTERFACE_0, 6, 2, aif_text); + +static const struct soc_enum lplay_enum = + SOC_ENUM_SINGLE(WM8903_AUDIO_INTERFACE_0, 5, 2, aif_text); + +static const struct soc_enum rplay_enum = + SOC_ENUM_SINGLE(WM8903_AUDIO_INTERFACE_0, 4, 2, aif_text); + static const struct snd_kcontrol_new wm8903_snd_controls[] = { /* Input PGAs - No TLV since the scale depends on PGA mode */ @@ -791,6 +807,18 @@ static const struct snd_kcontrol_new lsidetone_mux = static const struct snd_kcontrol_new rsidetone_mux = SOC_DAPM_ENUM("DACR Sidetone Mux", rsidetone_enum); +static const struct snd_kcontrol_new lcapture_mux = + SOC_DAPM_ENUM("Left Capture Mux", lcapture_enum); + +static const struct snd_kcontrol_new rcapture_mux = + SOC_DAPM_ENUM("Right Capture Mux", rcapture_enum); + +static const struct snd_kcontrol_new lplay_mux = + SOC_DAPM_ENUM("Left Playback Mux", lplay_enum); + +static const struct snd_kcontrol_new rplay_mux = + SOC_DAPM_ENUM("Right Playback Mux", rplay_enum); + static const struct snd_kcontrol_new left_output_mixer[] = { SOC_DAPM_SINGLE("DACL Switch", WM8903_ANALOGUE_LEFT_MIX_0, 3, 1, 0), SOC_DAPM_SINGLE("DACR Switch", WM8903_ANALOGUE_LEFT_MIX_0, 2, 1, 0), @@ -854,14 +882,26 @@ SND_SOC_DAPM_MUX("Right Input Mode Mux", SND_SOC_NOPM, 0, 0, &rinput_mode_mux), SND_SOC_DAPM_PGA("Left Input PGA", WM8903_POWER_MANAGEMENT_0, 1, 0, NULL, 0), SND_SOC_DAPM_PGA("Right Input PGA", WM8903_POWER_MANAGEMENT_0, 0, 0, NULL, 0), -SND_SOC_DAPM_ADC("ADCL", "Left HiFi Capture", WM8903_POWER_MANAGEMENT_6, 1, 0), -SND_SOC_DAPM_ADC("ADCR", "Right HiFi Capture", WM8903_POWER_MANAGEMENT_6, 0, 0), +SND_SOC_DAPM_ADC("ADCL", NULL, WM8903_POWER_MANAGEMENT_6, 1, 0), +SND_SOC_DAPM_ADC("ADCR", NULL, WM8903_POWER_MANAGEMENT_6, 0, 0), + +SND_SOC_DAPM_MUX("Left Capture Mux", SND_SOC_NOPM, 0, 0, &lcapture_mux), +SND_SOC_DAPM_MUX("Right Capture Mux", SND_SOC_NOPM, 0, 0, &rcapture_mux), + +SND_SOC_DAPM_AIF_OUT("AIFTXL", "Left HiFi Capture", 0, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_AIF_OUT("AIFTXR", "Right HiFi Capture", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_MUX("DACL Sidetone", SND_SOC_NOPM, 0, 0, &lsidetone_mux), SND_SOC_DAPM_MUX("DACR Sidetone", SND_SOC_NOPM, 0, 0, &rsidetone_mux), -SND_SOC_DAPM_DAC("DACL", "Left Playback", WM8903_POWER_MANAGEMENT_6, 3, 0), -SND_SOC_DAPM_DAC("DACR", "Right Playback", WM8903_POWER_MANAGEMENT_6, 2, 0), +SND_SOC_DAPM_AIF_IN("AIFRXL", "Left Playback", 0, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_AIF_IN("AIFRXR", "Right Playback", 0, SND_SOC_NOPM, 0, 0), + +SND_SOC_DAPM_MUX("Left Playback Mux", SND_SOC_NOPM, 0, 0, &lplay_mux), +SND_SOC_DAPM_MUX("Right Playback Mux", SND_SOC_NOPM, 0, 0, &rplay_mux), + +SND_SOC_DAPM_DAC("DACL", NULL, WM8903_POWER_MANAGEMENT_6, 3, 0), +SND_SOC_DAPM_DAC("DACR", NULL, WM8903_POWER_MANAGEMENT_6, 2, 0), SND_SOC_DAPM_MIXER("Left Output Mixer", WM8903_POWER_MANAGEMENT_1, 1, 0, left_output_mixer, ARRAY_SIZE(left_output_mixer)), @@ -943,18 +983,36 @@ static const struct snd_soc_dapm_route intercon[] = { { "Left Input PGA", NULL, "Left Input Mode Mux" }, { "Right Input PGA", NULL, "Right Input Mode Mux" }, + { "Left Capture Mux", "Left", "ADCL" }, + { "Left Capture Mux", "Right", "ADCR" }, + + { "Right Capture Mux", "Left", "ADCL" }, + { "Right Capture Mux", "Right", "ADCR" }, + + { "AIFTXL", NULL, "Left Capture Mux" }, + { "AIFTXR", NULL, "Right Capture Mux" }, + { "ADCL", NULL, "Left Input PGA" }, { "ADCL", NULL, "CLK_DSP" }, { "ADCR", NULL, "Right Input PGA" }, { "ADCR", NULL, "CLK_DSP" }, + { "Left Playback Mux", "Left", "AIFRXL" }, + { "Left Playback Mux", "Right", "AIFRXR" }, + + { "Right Playback Mux", "Left", "AIFRXL" }, + { "Right Playback Mux", "Right", "AIFRXR" }, + { "DACL Sidetone", "Left", "ADCL" }, { "DACL Sidetone", "Right", "ADCR" }, { "DACR Sidetone", "Left", "ADCL" }, { "DACR Sidetone", "Right", "ADCR" }, + { "DACL", NULL, "Left Playback Mux" }, { "DACL", NULL, "DACL Sidetone" }, { "DACL", NULL, "CLK_DSP" }, + + { "DACR", NULL, "Right Playback Mux" }, { "DACR", NULL, "DACR Sidetone" }, { "DACR", NULL, "CLK_DSP" }, -- cgit v1.2.3-70-g09d2 From 13a9983eb197254dffd9ea63a2d5f12c54eb651c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 9 Feb 2011 17:42:55 +0000 Subject: ASoC: Convert WM8903 to use PGA_S for output stage enables This simplfies the code and slightly reduces the startup time. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8903.c | 180 ++++++++++++++++------------------------------ 1 file changed, 60 insertions(+), 120 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 0190c5aa44f..9793775d579 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -246,6 +246,8 @@ static int wm8903_volatile_register(struct snd_soc_codec *codec, unsigned int re case WM8903_REVISION_NUMBER: case WM8903_INTERRUPT_STATUS_1: case WM8903_WRITE_SEQUENCER_4: + case WM8903_POWER_MANAGEMENT_3: + case WM8903_POWER_MANAGEMENT_2: return 1; default: @@ -304,11 +306,6 @@ static void wm8903_reset(struct snd_soc_codec *codec) sizeof(wm8903_reg_defaults)); } -#define WM8903_OUTPUT_SHORT 0x8 -#define WM8903_OUTPUT_OUT 0x4 -#define WM8903_OUTPUT_INT 0x2 -#define WM8903_OUTPUT_IN 0x1 - static int wm8903_cp_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -318,99 +315,6 @@ static int wm8903_cp_event(struct snd_soc_dapm_widget *w, return 0; } -/* - * Event for headphone and line out amplifier power changes. Special - * power up/down sequences are required in order to maximise pop/click - * performance. - */ -static int wm8903_output_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) -{ - struct snd_soc_codec *codec = w->codec; - u16 val; - u16 reg; - u16 dcs_reg; - u16 dcs_bit; - int shift; - - switch (w->reg) { - case WM8903_POWER_MANAGEMENT_2: - reg = WM8903_ANALOGUE_HP_0; - dcs_bit = 0 + w->shift; - break; - case WM8903_POWER_MANAGEMENT_3: - reg = WM8903_ANALOGUE_LINEOUT_0; - dcs_bit = 2 + w->shift; - break; - default: - BUG(); - return -EINVAL; /* Spurious warning from some compilers */ - } - - switch (w->shift) { - case 0: - shift = 0; - break; - case 1: - shift = 4; - break; - default: - BUG(); - return -EINVAL; /* Spurious warning from some compilers */ - } - - if (event & SND_SOC_DAPM_PRE_PMU) { - val = snd_soc_read(codec, reg); - - /* Short the output */ - val &= ~(WM8903_OUTPUT_SHORT << shift); - snd_soc_write(codec, reg, val); - } - - if (event & SND_SOC_DAPM_POST_PMU) { - val = snd_soc_read(codec, reg); - - val |= (WM8903_OUTPUT_IN << shift); - snd_soc_write(codec, reg, val); - - val |= (WM8903_OUTPUT_INT << shift); - snd_soc_write(codec, reg, val); - - /* Turn on the output ENA_OUTP */ - val |= (WM8903_OUTPUT_OUT << shift); - snd_soc_write(codec, reg, val); - - /* Enable the DC servo */ - dcs_reg = snd_soc_read(codec, WM8903_DC_SERVO_0); - dcs_reg |= dcs_bit; - snd_soc_write(codec, WM8903_DC_SERVO_0, dcs_reg); - - /* Remove the short */ - val |= (WM8903_OUTPUT_SHORT << shift); - snd_soc_write(codec, reg, val); - } - - if (event & SND_SOC_DAPM_PRE_PMD) { - val = snd_soc_read(codec, reg); - - /* Short the output */ - val &= ~(WM8903_OUTPUT_SHORT << shift); - snd_soc_write(codec, reg, val); - - /* Disable the DC servo */ - dcs_reg = snd_soc_read(codec, WM8903_DC_SERVO_0); - dcs_reg &= ~dcs_bit; - snd_soc_write(codec, WM8903_DC_SERVO_0, dcs_reg); - - /* Then disable the intermediate and output stages */ - val &= ~((WM8903_OUTPUT_OUT | WM8903_OUTPUT_INT | - WM8903_OUTPUT_IN) << shift); - snd_soc_write(codec, reg, val); - } - - return 0; -} - /* * When used with DAC outputs only the WM8903 charge pump supports * operation in class W mode, providing very low power consumption @@ -913,23 +817,40 @@ SND_SOC_DAPM_MIXER("Left Speaker Mixer", WM8903_POWER_MANAGEMENT_4, 1, 0, SND_SOC_DAPM_MIXER("Right Speaker Mixer", WM8903_POWER_MANAGEMENT_4, 0, 0, right_speaker_mixer, ARRAY_SIZE(right_speaker_mixer)), -SND_SOC_DAPM_PGA_E("Left Headphone Output PGA", WM8903_POWER_MANAGEMENT_2, - 1, 0, NULL, 0, wm8903_output_event, - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | - SND_SOC_DAPM_PRE_PMD), -SND_SOC_DAPM_PGA_E("Right Headphone Output PGA", WM8903_POWER_MANAGEMENT_2, - 0, 0, NULL, 0, wm8903_output_event, - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | - SND_SOC_DAPM_PRE_PMD), - -SND_SOC_DAPM_PGA_E("Left Line Output PGA", WM8903_POWER_MANAGEMENT_3, 1, 0, - NULL, 0, wm8903_output_event, - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | - SND_SOC_DAPM_PRE_PMD), -SND_SOC_DAPM_PGA_E("Right Line Output PGA", WM8903_POWER_MANAGEMENT_3, 0, 0, - NULL, 0, wm8903_output_event, - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | - SND_SOC_DAPM_PRE_PMD), +SND_SOC_DAPM_PGA_S("Left Headphone Output PGA", 0, WM8903_ANALOGUE_HP_0, + 4, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("Right Headphone Output PGA", 0, WM8903_ANALOGUE_HP_0, + 0, 0, NULL, 0), + +SND_SOC_DAPM_PGA_S("Left Line Output PGA", 0, WM8903_ANALOGUE_LINEOUT_0, 4, 0, + NULL, 0), +SND_SOC_DAPM_PGA_S("Right Line Output PGA", 0, WM8903_ANALOGUE_LINEOUT_0, 0, 0, + NULL, 0), + +SND_SOC_DAPM_PGA_S("HPL_RMV_SHORT", 4, WM8903_ANALOGUE_HP_0, 7, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("HPL_ENA_OUTP", 3, WM8903_ANALOGUE_HP_0, 6, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("HPL_ENA_DLY", 1, WM8903_ANALOGUE_HP_0, 5, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("HPR_RMV_SHORT", 4, WM8903_ANALOGUE_HP_0, 3, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("HPR_ENA_OUTP", 3, WM8903_ANALOGUE_HP_0, 2, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("HPR_ENA_DLY", 1, WM8903_ANALOGUE_HP_0, 1, 0, NULL, 0), + +SND_SOC_DAPM_PGA_S("LINEOUTL_RMV_SHORT", 4, WM8903_ANALOGUE_LINEOUT_0, 7, 0, + NULL, 0), +SND_SOC_DAPM_PGA_S("LINEOUTL_ENA_OUTP", 3, WM8903_ANALOGUE_LINEOUT_0, 6, 0, + NULL, 0), +SND_SOC_DAPM_PGA_S("LINEOUTL_ENA_DLY", 1, WM8903_ANALOGUE_LINEOUT_0, 5, 0, + NULL, 0), +SND_SOC_DAPM_PGA_S("LINEOUTR_RMV_SHORT", 4, WM8903_ANALOGUE_LINEOUT_0, 3, 0, + NULL, 0), +SND_SOC_DAPM_PGA_S("LINEOUTR_ENA_OUTP", 3, WM8903_ANALOGUE_LINEOUT_0, 2, 0, + NULL, 0), +SND_SOC_DAPM_PGA_S("LINEOUTR_ENA_DLY", 1, WM8903_ANALOGUE_LINEOUT_0, 1, 0, + NULL, 0), + +SND_SOC_DAPM_PGA_S("HPL_DCS", 3, WM8903_DC_SERVO_0, 3, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("HPR_DCS", 3, WM8903_DC_SERVO_0, 2, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("LINEOUTL_DCS", 3, WM8903_DC_SERVO_0, 1, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("LINEOUTR_DCS", 3, WM8903_DC_SERVO_0, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("Left Speaker PGA", WM8903_POWER_MANAGEMENT_5, 1, 0, NULL, 0), @@ -1045,11 +966,30 @@ static const struct snd_soc_dapm_route intercon[] = { { "Left Speaker PGA", NULL, "Left Speaker Mixer" }, { "Right Speaker PGA", NULL, "Right Speaker Mixer" }, - { "HPOUTL", NULL, "Left Headphone Output PGA" }, - { "HPOUTR", NULL, "Right Headphone Output PGA" }, - - { "LINEOUTL", NULL, "Left Line Output PGA" }, - { "LINEOUTR", NULL, "Right Line Output PGA" }, + { "HPL_ENA_DLY", NULL, "Left Headphone Output PGA" }, + { "HPR_ENA_DLY", NULL, "Right Headphone Output PGA" }, + { "LINEOUTL_ENA_DLY", NULL, "Left Line Output PGA" }, + { "LINEOUTR_ENA_DLY", NULL, "Right Line Output PGA" }, + + { "HPL_DCS", NULL, "HPL_ENA_DLY" }, + { "HPR_DCS", NULL, "HPR_ENA_DLY" }, + { "LINEOUTL_DCS", NULL, "LINEOUTL_ENA_DLY" }, + { "LINEOUTR_DCS", NULL, "LINEOUTR_ENA_DLY" }, + + { "HPL_ENA_OUTP", NULL, "HPL_DCS" }, + { "HPR_ENA_OUTP", NULL, "HPR_DCS" }, + { "LINEOUTL_ENA_OUTP", NULL, "LINEOUTL_DCS" }, + { "LINEOUTR_ENA_OUTP", NULL, "LINEOUTR_DCS" }, + + { "HPL_RMV_SHORT", NULL, "HPL_ENA_OUTP" }, + { "HPR_RMV_SHORT", NULL, "HPR_ENA_OUTP" }, + { "LINEOUTL_RMV_SHORT", NULL, "LINEOUTL_ENA_OUTP" }, + { "LINEOUTR_RMV_SHORT", NULL, "LINEOUTR_ENA_OUTP" }, + + { "HPOUTL", NULL, "HPL_RMV_SHORT" }, + { "HPOUTR", NULL, "HPR_RMV_SHORT" }, + { "LINEOUTL", NULL, "LINEOUTL_RMV_SHORT" }, + { "LINEOUTR", NULL, "LINEOUTR_RMV_SHORT" }, { "LOP", NULL, "Left Speaker PGA" }, { "LON", NULL, "Left Speaker PGA" }, -- cgit v1.2.3-70-g09d2 From 2c8be5a26e42cfc4906c4daa8a5a5c82610ddb3d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 9 Feb 2011 17:42:56 +0000 Subject: ASoC: Dynamically manage CLK_SYS in WM8903 Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8903.c | 15 ++++++++++++--- 1 file changed, 12 insertions(+), 3 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 9793775d579..389d2e8088f 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -860,10 +860,18 @@ SND_SOC_DAPM_PGA("Right Speaker PGA", WM8903_POWER_MANAGEMENT_5, 0, 0, SND_SOC_DAPM_SUPPLY("Charge Pump", WM8903_CHARGE_PUMP_0, 0, 0, wm8903_cp_event, SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_SUPPLY("CLK_DSP", WM8903_CLOCK_RATES_2, 1, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("CLK_SYS", WM8903_CLOCK_RATES_2, 2, 0, NULL, 0), }; static const struct snd_soc_dapm_route intercon[] = { + { "CLK_DSP", NULL, "CLK_SYS" }, + { "Mic Bias", NULL, "CLK_SYS" }, + { "HPL_DCS", NULL, "CLK_SYS" }, + { "HPR_DCS", NULL, "CLK_SYS" }, + { "LINEOUTL_DCS", NULL, "CLK_SYS" }, + { "LINEOUTR_DCS", NULL, "CLK_SYS" }, + { "Left Input Mux", "IN1L", "IN1L" }, { "Left Input Mux", "IN2L", "IN2L" }, { "Left Input Mux", "IN3L", "IN3L" }, @@ -1059,10 +1067,11 @@ static int wm8903_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_OFF: + snd_soc_update_bits(codec, WM8903_CLOCK_RATES_2, + WM8903_CLK_SYS_ENA, WM8903_CLK_SYS_ENA); wm8903_run_sequence(codec, 32); - reg = snd_soc_read(codec, WM8903_CLOCK_RATES_2); - reg &= ~WM8903_CLK_SYS_ENA; - snd_soc_write(codec, WM8903_CLOCK_RATES_2, reg); + snd_soc_update_bits(codec, WM8903_CLOCK_RATES_2, + WM8903_CLK_SYS_ENA, 0); break; } -- cgit v1.2.3-70-g09d2 From e12adab00222817213fcdc68c5fd6ee2e5dfb247 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 9 Feb 2011 17:42:57 +0000 Subject: ASoC: Fix WM8903 DAC mute default The WM8903 register map does not mute the DAC by default at startup so we need to explicitly do so. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8903.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 389d2e8088f..119abceb1d8 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1870,9 +1870,9 @@ static int wm8903_probe(struct snd_soc_codec *codec) snd_soc_write(codec, WM8903_ANALOGUE_OUT3_RIGHT, val); /* Enable DAC soft mute by default */ - val = snd_soc_read(codec, WM8903_DAC_DIGITAL_1); - val |= WM8903_DAC_MUTEMODE; - snd_soc_write(codec, WM8903_DAC_DIGITAL_1, val); + snd_soc_update_bits(codec, WM8903_DAC_DIGITAL_1, + WM8903_DAC_MUTEMODE | WM8903_DAC_MUTE, + WM8903_DAC_MUTEMODE | WM8903_DAC_MUTE); snd_soc_add_controls(codec, wm8903_snd_controls, ARRAY_SIZE(wm8903_snd_controls)); -- cgit v1.2.3-70-g09d2 From c5b6a9feaeb0fa0e39e3fc10f9bf8cc8de498739 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 9 Feb 2011 20:14:42 +0000 Subject: ASoC: Actively manage WM8903 DC servo configuration Explicitly cache the DC servo offsets for digital paths in the driver, allowing them to be preserved over suspend and resume, and ensure that we recalibrate analogue outputs paths when they are in use so that we cover any changes in the input offset. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8903.c | 123 ++++++++++++++++++++++++++++++++++++++++++++-- sound/soc/codecs/wm8903.h | 8 +++ 2 files changed, 127 insertions(+), 4 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 119abceb1d8..2f912276a8f 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -223,6 +223,9 @@ struct wm8903_priv { int fs; int deemph; + int dcs_pending; + int dcs_cache[4]; + /* Reference count */ int class_w_users; @@ -248,6 +251,10 @@ static int wm8903_volatile_register(struct snd_soc_codec *codec, unsigned int re case WM8903_WRITE_SEQUENCER_4: case WM8903_POWER_MANAGEMENT_3: case WM8903_POWER_MANAGEMENT_2: + case WM8903_DC_SERVO_READBACK_1: + case WM8903_DC_SERVO_READBACK_2: + case WM8903_DC_SERVO_READBACK_3: + case WM8903_DC_SERVO_READBACK_4: return 1; default: @@ -315,6 +322,103 @@ static int wm8903_cp_event(struct snd_soc_dapm_widget *w, return 0; } +static int wm8903_dcs_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + wm8903->dcs_pending |= 1 << w->shift; + break; + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, WM8903_DC_SERVO_0, + 1 << w->shift, 0); + break; + } + + return 0; +} + +#define WM8903_DCS_MODE_WRITE_STOP 0 +#define WM8903_DCS_MODE_START_STOP 2 + +static void wm8903_seq_notifier(struct snd_soc_dapm_context *dapm, + enum snd_soc_dapm_type event, int subseq) +{ + struct snd_soc_codec *codec = container_of(dapm, + struct snd_soc_codec, dapm); + struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); + int dcs_mode = WM8903_DCS_MODE_WRITE_STOP; + int i, val; + + /* Complete any pending DC servo starts */ + if (wm8903->dcs_pending) { + dev_dbg(codec->dev, "Starting DC servo for %x\n", + wm8903->dcs_pending); + + /* If we've no cached values then we need to do startup */ + for (i = 0; i < ARRAY_SIZE(wm8903->dcs_cache); i++) { + if (!(wm8903->dcs_pending & (1 << i))) + continue; + + if (wm8903->dcs_cache[i]) { + dev_dbg(codec->dev, + "Restore DC servo %d value %x\n", + 3 - i, wm8903->dcs_cache[i]); + + snd_soc_write(codec, WM8903_DC_SERVO_4 + i, + wm8903->dcs_cache[i] & 0xff); + } else { + dev_dbg(codec->dev, + "Calibrate DC servo %d\n", 3 - i); + dcs_mode = WM8903_DCS_MODE_START_STOP; + } + } + + /* Don't trust the cache for analogue */ + if (wm8903->class_w_users) + dcs_mode = WM8903_DCS_MODE_START_STOP; + + snd_soc_update_bits(codec, WM8903_DC_SERVO_2, + WM8903_DCS_MODE_MASK, dcs_mode); + + snd_soc_update_bits(codec, WM8903_DC_SERVO_0, + WM8903_DCS_ENA_MASK, wm8903->dcs_pending); + + switch (dcs_mode) { + case WM8903_DCS_MODE_WRITE_STOP: + break; + + case WM8903_DCS_MODE_START_STOP: + msleep(270); + + /* Cache the measured offsets for digital */ + if (wm8903->class_w_users) + break; + + for (i = 0; i < ARRAY_SIZE(wm8903->dcs_cache); i++) { + if (!(wm8903->dcs_pending & (1 << i))) + continue; + + val = snd_soc_read(codec, + WM8903_DC_SERVO_READBACK_1 + i); + dev_dbg(codec->dev, "DC servo %d: %x\n", + 3 - i, val); + wm8903->dcs_cache[i] = val; + } + break; + + default: + pr_warn("DCS mode %d delay not set\n", dcs_mode); + break; + } + + wm8903->dcs_pending = 0; + } +} + /* * When used with DAC outputs only the WM8903 charge pump supports * operation in class W mode, providing very low power consumption @@ -847,10 +951,15 @@ SND_SOC_DAPM_PGA_S("LINEOUTR_ENA_OUTP", 3, WM8903_ANALOGUE_LINEOUT_0, 2, 0, SND_SOC_DAPM_PGA_S("LINEOUTR_ENA_DLY", 1, WM8903_ANALOGUE_LINEOUT_0, 1, 0, NULL, 0), -SND_SOC_DAPM_PGA_S("HPL_DCS", 3, WM8903_DC_SERVO_0, 3, 0, NULL, 0), -SND_SOC_DAPM_PGA_S("HPR_DCS", 3, WM8903_DC_SERVO_0, 2, 0, NULL, 0), -SND_SOC_DAPM_PGA_S("LINEOUTL_DCS", 3, WM8903_DC_SERVO_0, 1, 0, NULL, 0), -SND_SOC_DAPM_PGA_S("LINEOUTR_DCS", 3, WM8903_DC_SERVO_0, 0, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("DCS Master", WM8903_DC_SERVO_0, 4, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("HPL_DCS", 3, SND_SOC_NOPM, 3, 0, wm8903_dcs_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), +SND_SOC_DAPM_PGA_S("HPR_DCS", 3, SND_SOC_NOPM, 2, 0, wm8903_dcs_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), +SND_SOC_DAPM_PGA_S("LINEOUTL_DCS", 3, SND_SOC_NOPM, 1, 0, wm8903_dcs_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), +SND_SOC_DAPM_PGA_S("LINEOUTR_DCS", 3, SND_SOC_NOPM, 0, 0, wm8903_dcs_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_PGA("Left Speaker PGA", WM8903_POWER_MANAGEMENT_5, 1, 0, NULL, 0), @@ -979,6 +1088,11 @@ static const struct snd_soc_dapm_route intercon[] = { { "LINEOUTL_ENA_DLY", NULL, "Left Line Output PGA" }, { "LINEOUTR_ENA_DLY", NULL, "Right Line Output PGA" }, + { "HPL_DCS", NULL, "DCS Master" }, + { "HPR_DCS", NULL, "DCS Master" }, + { "LINEOUTL_DCS", NULL, "DCS Master" }, + { "LINEOUTR_DCS", NULL, "DCS Master" }, + { "HPL_DCS", NULL, "HPL_ENA_DLY" }, { "HPR_DCS", NULL, "HPR_ENA_DLY" }, { "LINEOUTL_DCS", NULL, "LINEOUTL_ENA_DLY" }, @@ -1901,6 +2015,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8903 = { .reg_word_size = sizeof(u16), .reg_cache_default = wm8903_reg_defaults, .volatile_register = wm8903_volatile_register, + .seq_notifier = wm8903_seq_notifier, }; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) diff --git a/sound/soc/codecs/wm8903.h b/sound/soc/codecs/wm8903.h index e8490f3edd0..9ef394ae0dd 100644 --- a/sound/soc/codecs/wm8903.h +++ b/sound/soc/codecs/wm8903.h @@ -75,6 +75,14 @@ extern int wm8903_mic_detect(struct snd_soc_codec *codec, #define WM8903_ANALOGUE_SPK_OUTPUT_CONTROL_0 0x41 #define WM8903_DC_SERVO_0 0x43 #define WM8903_DC_SERVO_2 0x45 +#define WM8903_DC_SERVO_4 0x47 +#define WM8903_DC_SERVO_5 0x48 +#define WM8903_DC_SERVO_6 0x49 +#define WM8903_DC_SERVO_7 0x4A +#define WM8903_DC_SERVO_READBACK_1 0x51 +#define WM8903_DC_SERVO_READBACK_2 0x52 +#define WM8903_DC_SERVO_READBACK_3 0x53 +#define WM8903_DC_SERVO_READBACK_4 0x54 #define WM8903_ANALOGUE_HP_0 0x5A #define WM8903_ANALOGUE_LINEOUT_0 0x5E #define WM8903_CHARGE_PUMP_0 0x62 -- cgit v1.2.3-70-g09d2 From 66daaa59d5f0310238de183918e13062428fb59f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 10 Feb 2011 13:32:58 +0000 Subject: ASoC: Convert WM8903 bias management to use snd_soc_update_bits() Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8903.c | 17 +++++++---------- 1 file changed, 7 insertions(+), 10 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 2f912276a8f..b88c6165dd2 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1139,15 +1139,13 @@ static int wm8903_add_widgets(struct snd_soc_codec *codec) static int wm8903_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u16 reg; - switch (level) { case SND_SOC_BIAS_ON: + break; case SND_SOC_BIAS_PREPARE: - reg = snd_soc_read(codec, WM8903_VMID_CONTROL_0); - reg &= ~(WM8903_VMID_RES_MASK); - reg |= WM8903_VMID_RES_50K; - snd_soc_write(codec, WM8903_VMID_CONTROL_0, reg); + snd_soc_update_bits(codec, WM8903_VMID_CONTROL_0, + WM8903_VMID_RES_MASK, + WM8903_VMID_RES_50K); break; case SND_SOC_BIAS_STANDBY: @@ -1174,10 +1172,9 @@ static int wm8903_set_bias_level(struct snd_soc_codec *codec, WM8903_CP_DYN_V); } - reg = snd_soc_read(codec, WM8903_VMID_CONTROL_0); - reg &= ~(WM8903_VMID_RES_MASK); - reg |= WM8903_VMID_RES_250K; - snd_soc_write(codec, WM8903_VMID_CONTROL_0, reg); + snd_soc_update_bits(codec, WM8903_VMID_CONTROL_0, + WM8903_VMID_RES_MASK, + WM8903_VMID_RES_250K); break; case SND_SOC_BIAS_OFF: -- cgit v1.2.3-70-g09d2 From 22f226dd1496a0fa470e64a66e2da474f34eebf8 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 10 Feb 2011 14:01:38 +0000 Subject: ASoC: Don't use write sequencer to power up WM8903 The write sequencer sequencer sequence takes longer than is desirable as it brings up a full playback path which is not required at this point. Open coding the sequence cuts the startup time by two thirds. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8903.c | 73 +++++++++++++++++++++++++++++++++++------------ 1 file changed, 54 insertions(+), 19 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index b88c6165dd2..e203a3ec655 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -297,15 +297,6 @@ static int wm8903_run_sequence(struct snd_soc_codec *codec, unsigned int start) return 0; } -static void wm8903_sync_reg_cache(struct snd_soc_codec *codec, u16 *cache) -{ - int i; - - /* There really ought to be something better we can do here :/ */ - for (i = 0; i < ARRAY_SIZE(wm8903_reg_defaults); i++) - cache[i] = codec->hw_read(codec, i); -} - static void wm8903_reset(struct snd_soc_codec *codec) { snd_soc_write(codec, WM8903_SW_RESET_AND_ID, 0); @@ -1142,6 +1133,7 @@ static int wm8903_set_bias_level(struct snd_soc_codec *codec, switch (level) { case SND_SOC_BIAS_ON: break; + case SND_SOC_BIAS_PREPARE: snd_soc_update_bits(codec, WM8903_VMID_CONTROL_0, WM8903_VMID_RES_MASK, @@ -1150,16 +1142,59 @@ static int wm8903_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_STANDBY: if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { - snd_soc_write(codec, WM8903_CLOCK_RATES_2, - WM8903_CLK_SYS_ENA); - - /* Change DC servo dither level in startup sequence */ - snd_soc_write(codec, WM8903_WRITE_SEQUENCER_0, 0x11); - snd_soc_write(codec, WM8903_WRITE_SEQUENCER_1, 0x1257); - snd_soc_write(codec, WM8903_WRITE_SEQUENCER_2, 0x2); - - wm8903_run_sequence(codec, 0); - wm8903_sync_reg_cache(codec, codec->reg_cache); + snd_soc_update_bits(codec, WM8903_BIAS_CONTROL_0, + WM8903_POBCTRL | WM8903_ISEL_MASK | + WM8903_STARTUP_BIAS_ENA | + WM8903_BIAS_ENA, + WM8903_POBCTRL | + (2 << WM8903_ISEL_SHIFT) | + WM8903_STARTUP_BIAS_ENA); + + snd_soc_update_bits(codec, + WM8903_ANALOGUE_SPK_OUTPUT_CONTROL_0, + WM8903_SPK_DISCHARGE, + WM8903_SPK_DISCHARGE); + + msleep(33); + + snd_soc_update_bits(codec, WM8903_POWER_MANAGEMENT_5, + WM8903_SPKL_ENA | WM8903_SPKR_ENA, + WM8903_SPKL_ENA | WM8903_SPKR_ENA); + + snd_soc_update_bits(codec, + WM8903_ANALOGUE_SPK_OUTPUT_CONTROL_0, + WM8903_SPK_DISCHARGE, 0); + + snd_soc_update_bits(codec, WM8903_VMID_CONTROL_0, + WM8903_VMID_TIE_ENA | + WM8903_BUFIO_ENA | + WM8903_VMID_IO_ENA | + WM8903_VMID_SOFT_MASK | + WM8903_VMID_RES_MASK | + WM8903_VMID_BUF_ENA, + WM8903_VMID_TIE_ENA | + WM8903_BUFIO_ENA | + WM8903_VMID_IO_ENA | + (2 << WM8903_VMID_SOFT_SHIFT) | + WM8903_VMID_RES_250K | + WM8903_VMID_BUF_ENA); + + msleep(129); + + snd_soc_update_bits(codec, WM8903_POWER_MANAGEMENT_5, + WM8903_SPKL_ENA | WM8903_SPKR_ENA, + 0); + + snd_soc_update_bits(codec, WM8903_VMID_CONTROL_0, + WM8903_VMID_SOFT_MASK, 0); + + snd_soc_update_bits(codec, WM8903_VMID_CONTROL_0, + WM8903_VMID_RES_MASK, + WM8903_VMID_RES_50K); + + snd_soc_update_bits(codec, WM8903_BIAS_CONTROL_0, + WM8903_BIAS_ENA | WM8903_POBCTRL, + WM8903_BIAS_ENA); /* By default no bypass paths are enabled so * enable Class W support. -- cgit v1.2.3-70-g09d2 From b4d06f456dcf956761e2c927e62b03861f07dbbf Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 10 Feb 2011 14:20:49 +0000 Subject: ASoC: Use explicit sequence for WM8903 bias off This makes no real difference compared to the write sequencer sequence that was previously used but can run without a clock being provided. Also remove the write sequencer support code as this was the last use of it. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8903.c | 66 +++++++++++++++-------------------------------- 1 file changed, 21 insertions(+), 45 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index e203a3ec655..c7b52a04fe1 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -229,8 +229,6 @@ struct wm8903_priv { /* Reference count */ int class_w_users; - struct completion wseq; - struct snd_soc_jack *mic_jack; int mic_det; int mic_short; @@ -262,41 +260,6 @@ static int wm8903_volatile_register(struct snd_soc_codec *codec, unsigned int re } } -static int wm8903_run_sequence(struct snd_soc_codec *codec, unsigned int start) -{ - u16 reg[5]; - struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); - - BUG_ON(start > 48); - - /* Enable the sequencer if it's not already on */ - reg[0] = snd_soc_read(codec, WM8903_WRITE_SEQUENCER_0); - snd_soc_write(codec, WM8903_WRITE_SEQUENCER_0, - reg[0] | WM8903_WSEQ_ENA); - - dev_dbg(codec->dev, "Starting sequence at %d\n", start); - - snd_soc_write(codec, WM8903_WRITE_SEQUENCER_3, - start | WM8903_WSEQ_START); - - /* Wait for it to complete. If we have the interrupt wired up then - * that will break us out of the poll early. - */ - do { - wait_for_completion_timeout(&wm8903->wseq, - msecs_to_jiffies(10)); - - reg[4] = snd_soc_read(codec, WM8903_WRITE_SEQUENCER_4); - } while (reg[4] & WM8903_WSEQ_BUSY); - - dev_dbg(codec->dev, "Sequence complete\n"); - - /* Disable the sequencer again if we enabled it */ - snd_soc_write(codec, WM8903_WRITE_SEQUENCER_0, reg[0]); - - return 0; -} - static void wm8903_reset(struct snd_soc_codec *codec) { snd_soc_write(codec, WM8903_SW_RESET_AND_ID, 0); @@ -1213,11 +1176,26 @@ static int wm8903_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_OFF: - snd_soc_update_bits(codec, WM8903_CLOCK_RATES_2, - WM8903_CLK_SYS_ENA, WM8903_CLK_SYS_ENA); - wm8903_run_sequence(codec, 32); - snd_soc_update_bits(codec, WM8903_CLOCK_RATES_2, - WM8903_CLK_SYS_ENA, 0); + snd_soc_update_bits(codec, WM8903_BIAS_CONTROL_0, + WM8903_BIAS_ENA, 0); + + snd_soc_update_bits(codec, WM8903_VMID_CONTROL_0, + WM8903_VMID_SOFT_MASK, + 2 << WM8903_VMID_SOFT_SHIFT); + + snd_soc_update_bits(codec, WM8903_VMID_CONTROL_0, + WM8903_VMID_BUF_ENA, 0); + + msleep(290); + + snd_soc_update_bits(codec, WM8903_VMID_CONTROL_0, + WM8903_VMID_TIE_ENA | WM8903_BUFIO_ENA | + WM8903_VMID_IO_ENA | WM8903_VMID_RES_MASK | + WM8903_VMID_SOFT_MASK | + WM8903_VMID_BUF_ENA, 0); + + snd_soc_update_bits(codec, WM8903_BIAS_CONTROL_0, + WM8903_STARTUP_BIAS_ENA, 0); break; } @@ -1670,8 +1648,7 @@ static irqreturn_t wm8903_irq(int irq, void *data) int_val = snd_soc_read(codec, WM8903_INTERRUPT_STATUS_1) & mask; if (int_val & WM8903_WSEQ_BUSY_EINT) { - dev_dbg(codec->dev, "Write sequencer done\n"); - complete(&wm8903->wseq); + dev_warn(codec->dev, "Write sequencer done\n"); } /* @@ -1918,7 +1895,6 @@ static int wm8903_probe(struct snd_soc_codec *codec) u16 val; wm8903->codec = codec; - init_completion(&wm8903->wseq); ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_I2C); if (ret != 0) { -- cgit v1.2.3-70-g09d2 From 7ae7434086f5b106021276e88b8ef49debf30aa8 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Thu, 10 Feb 2011 12:58:01 +0530 Subject: ASoC: mid-x86: Use the soc-jack apis for jack type detection This patch modifies the mfld_machine to use the new jack apis for adding the voltage zones for jack type detection. It also modifed TI sn95031 codec driver to use these new apis Signed-off-by: Vinod Koul Signed-off-by: Harsha Priya Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/sn95031.c | 4 +--- sound/soc/mid-x86/mfld_machine.c | 12 ++++++++++++ 2 files changed, 13 insertions(+), 3 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index 4cc00177ee3..d0b78020671 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -782,9 +782,7 @@ static int sn95031_get_headset_state(struct snd_soc_jack *mfld_jack) { int micbias = sn95031_get_mic_bias(mfld_jack->codec); - /* Defaulting to HEADSET for now. - * will change after adding soc-jack detection apis */ - int jack_type = SND_JACK_HEADSET; + int jack_type = snd_soc_jack_get_type(mfld_jack, micbias); pr_debug("jack type detected = %d\n", jack_type); if (jack_type == SND_JACK_HEADSET) diff --git a/sound/soc/mid-x86/mfld_machine.c b/sound/soc/mid-x86/mfld_machine.c index 45a00670802..96487fb8d26 100644 --- a/sound/soc/mid-x86/mfld_machine.c +++ b/sound/soc/mid-x86/mfld_machine.c @@ -74,6 +74,12 @@ static struct snd_soc_jack_pin mfld_jack_pins[] = { }, }; +/* jack detection voltage zones */ +static struct snd_soc_jack_zone mfld_zones[] = { + {MFLD_MV_START, MFLD_MV_AM_HS, SND_JACK_HEADPHONE}, + {MFLD_MV_AM_HS, MFLD_MV_HS, SND_JACK_HEADSET}, +}; + /* sound card controls */ static const char *headset_switch_text[] = {"Earpiece", "Headset"}; @@ -264,6 +270,12 @@ static int mfld_init(struct snd_soc_pcm_runtime *runtime) pr_err("adding jack pins failed\n"); return ret_val; } + ret_val = snd_soc_jack_add_zones(&mfld_jack, + ARRAY_SIZE(mfld_zones), mfld_zones); + if (ret_val) { + pr_err("adding jack zones failed\n"); + return ret_val; + } /* we want to check if anything is inserted at boot, * so send a fake event to codec and it will read adc -- cgit v1.2.3-70-g09d2 From 905f6952c5bc8126f1d82c2eb8a699271080b57e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 11 Feb 2011 14:39:13 +0000 Subject: ASoC: Warn if WM8903 platform data is used to enable microphone IRQ The WM8903 interrupts are clear on read so if the WM8903 detection is enabled from platform data when the IRQ is in use (rather than using a direct signal from a GPIO) status may be lost during startup. Help users spot this misconfiguration by adding a WARN_ON(). Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8903.c | 22 ++++++++++++++++++++++ 1 file changed, 22 insertions(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index c7b52a04fe1..f656a000b36 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1917,12 +1917,26 @@ static int wm8903_probe(struct snd_soc_codec *codec) /* Set up GPIOs and microphone detection */ if (pdata) { + bool mic_gpio = false; + for (i = 0; i < ARRAY_SIZE(pdata->gpio_cfg); i++) { if (pdata->gpio_cfg[i] == WM8903_GPIO_NO_CONFIG) continue; snd_soc_write(codec, WM8903_GPIO_CONTROL_1 + i, pdata->gpio_cfg[i] & 0xffff); + + val = (pdata->gpio_cfg[i] & WM8903_GP1_FN_MASK) + >> WM8903_GP1_FN_SHIFT; + + switch (val) { + case WM8903_GPn_FN_MICBIAS_CURRENT_DETECT: + case WM8903_GPn_FN_MICBIAS_SHORT_DETECT: + mic_gpio = true; + break; + default: + break; + } } snd_soc_write(codec, WM8903_MIC_BIAS_CONTROL_0, @@ -1933,6 +1947,14 @@ static int wm8903_probe(struct snd_soc_codec *codec) snd_soc_update_bits(codec, WM8903_WRITE_SEQUENCER_0, WM8903_WSEQ_ENA, WM8903_WSEQ_ENA); + /* If microphone detection is enabled by pdata but + * detected via IRQ then interrupts can be lost before + * the machine driver has set up microphone detection + * IRQs as the IRQs are clear on read. The detection + * will be enabled when the machine driver configures. + */ + WARN_ON(!mic_gpio && (pdata->micdet_cfg & WM8903_MICDET_ENA)); + wm8903->mic_delay = pdata->micdet_delay; } -- cgit v1.2.3-70-g09d2 From 1461d0630ef74073e308a79834ba09e1bd2df08e Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Tue, 15 Feb 2011 18:28:51 +0530 Subject: ASoC: sn95031: make playback rails depend on actual pins they control This patch makes the codec playback rails (headset and speaker) depend on actual pins they control. This enables better power management of the codec Signed-off-by: Vinod Koul Signed-off-by: Harsha Priya Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/sn95031.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index d0b78020671..9c3db0d7b75 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -519,6 +519,8 @@ static const struct snd_soc_dapm_widget sn95031_dapm_widgets[] = { static const struct snd_soc_dapm_route sn95031_audio_map[] = { /* headset and earpiece map */ + { "HPOUTL", NULL, "Headset Rail"}, + { "HPOUTR", NULL, "Headset Rail"}, { "HPOUTL", NULL, "Headset Left Playback" }, { "HPOUTR", NULL, "Headset Right Playback" }, { "EPOUT", NULL, "Earpiece Playback" }, @@ -527,18 +529,16 @@ static const struct snd_soc_dapm_route sn95031_audio_map[] = { { "Earpiece Playback", NULL, "Headset Left Filter"}, { "Headset Left Filter", NULL, "HSDAC Left"}, { "Headset Right Filter", NULL, "HSDAC Right"}, - { "HSDAC Left", NULL, "Headset Rail"}, - { "HSDAC Right", NULL, "Headset Rail"}, /* speaker map */ + { "IHFOUTL", NULL, "Speaker Rail"}, + { "IHFOUTR", NULL, "Speaker Rail"}, { "IHFOUTL", "NULL", "Speaker Left Playback"}, { "IHFOUTR", "NULL", "Speaker Right Playback"}, { "Speaker Left Playback", NULL, "Speaker Left Filter"}, { "Speaker Right Playback", NULL, "Speaker Right Filter"}, { "Speaker Left Filter", NULL, "IHFDAC Left"}, { "Speaker Right Filter", NULL, "IHFDAC Right"}, - { "IHFDAC Left", NULL, "Speaker Rail"}, - { "IHFDAC Right", NULL, "Speaker Rail"}, /* vibra map */ { "VIB1OUT", NULL, "Vibra1 Playback"}, -- cgit v1.2.3-70-g09d2 From a62ffc92e8332135c3dbc3351c7e90031830e2f7 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Tue, 15 Feb 2011 18:28:52 +0530 Subject: ASoC: sn95031: fix the DMIC path routing This patch makes the DMIC dynamically connect to TX Mux, earlier code had erroneously made this as static path Signed-off-by: Vinod Koul Signed-off-by: Harsha Priya Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/sn95031.c | 48 +++++++++++++++++++++++----------------------- 1 file changed, 24 insertions(+), 24 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index 9c3db0d7b75..5eb39c7ed96 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -602,30 +602,30 @@ static const struct snd_soc_dapm_route sn95031_audio_map[] = { { "Txpath2 Capture Route", "ADC Right", "ADC Right"}, { "Txpath3 Capture Route", "ADC Right", "ADC Right"}, { "Txpath4 Capture Route", "ADC Right", "ADC Right"}, - { "Txpath1 Capture Route", NULL, "DMIC1"}, - { "Txpath2 Capture Route", NULL, "DMIC1"}, - { "Txpath3 Capture Route", NULL, "DMIC1"}, - { "Txpath4 Capture Route", NULL, "DMIC1"}, - { "Txpath1 Capture Route", NULL, "DMIC2"}, - { "Txpath2 Capture Route", NULL, "DMIC2"}, - { "Txpath3 Capture Route", NULL, "DMIC2"}, - { "Txpath4 Capture Route", NULL, "DMIC2"}, - { "Txpath1 Capture Route", NULL, "DMIC3"}, - { "Txpath2 Capture Route", NULL, "DMIC3"}, - { "Txpath3 Capture Route", NULL, "DMIC3"}, - { "Txpath4 Capture Route", NULL, "DMIC3"}, - { "Txpath1 Capture Route", NULL, "DMIC4"}, - { "Txpath2 Capture Route", NULL, "DMIC4"}, - { "Txpath3 Capture Route", NULL, "DMIC4"}, - { "Txpath4 Capture Route", NULL, "DMIC4"}, - { "Txpath1 Capture Route", NULL, "DMIC5"}, - { "Txpath2 Capture Route", NULL, "DMIC5"}, - { "Txpath3 Capture Route", NULL, "DMIC5"}, - { "Txpath4 Capture Route", NULL, "DMIC5"}, - { "Txpath1 Capture Route", NULL, "DMIC6"}, - { "Txpath2 Capture Route", NULL, "DMIC6"}, - { "Txpath3 Capture Route", NULL, "DMIC6"}, - { "Txpath4 Capture Route", NULL, "DMIC6"}, + { "Txpath1 Capture Route", "DMIC1", "DMIC1"}, + { "Txpath2 Capture Route", "DMIC1", "DMIC1"}, + { "Txpath3 Capture Route", "DMIC1", "DMIC1"}, + { "Txpath4 Capture Route", "DMIC1", "DMIC1"}, + { "Txpath1 Capture Route", "DMIC2", "DMIC2"}, + { "Txpath2 Capture Route", "DMIC2", "DMIC2"}, + { "Txpath3 Capture Route", "DMIC2", "DMIC2"}, + { "Txpath4 Capture Route", "DMIC2", "DMIC2"}, + { "Txpath1 Capture Route", "DMIC3", "DMIC3"}, + { "Txpath2 Capture Route", "DMIC3", "DMIC3"}, + { "Txpath3 Capture Route", "DMIC3", "DMIC3"}, + { "Txpath4 Capture Route", "DMIC3", "DMIC3"}, + { "Txpath1 Capture Route", "DMIC4", "DMIC4"}, + { "Txpath2 Capture Route", "DMIC4", "DMIC4"}, + { "Txpath3 Capture Route", "DMIC4", "DMIC4"}, + { "Txpath4 Capture Route", "DMIC4", "DMIC4"}, + { "Txpath1 Capture Route", "DMIC5", "DMIC5"}, + { "Txpath2 Capture Route", "DMIC5", "DMIC5"}, + { "Txpath3 Capture Route", "DMIC5", "DMIC5"}, + { "Txpath4 Capture Route", "DMIC5", "DMIC5"}, + { "Txpath1 Capture Route", "DMIC6", "DMIC6"}, + { "Txpath2 Capture Route", "DMIC6", "DMIC6"}, + { "Txpath3 Capture Route", "DMIC6", "DMIC6"}, + { "Txpath4 Capture Route", "DMIC6", "DMIC6"}, /* tx path */ { "TX1 Enable", NULL, "Txpath1 Capture Route"}, -- cgit v1.2.3-70-g09d2 From 65e9625e1f86658ee869420713be3afb9a75debd Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Tue, 15 Feb 2011 18:28:53 +0530 Subject: ASoC: sn95031: fix the amic tlv scale The tlv scale is defined as (min, step, mute). The mute is not supported here so put the value to 0 Signed-off-by: Vinod Koul Signed-off-by: Harsha Priya Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/sn95031.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index 5eb39c7ed96..2a30eae1881 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -359,7 +359,7 @@ static const struct snd_kcontrol_new sn95031_input4_mux_control = static const char *sn95031_micmode_text[] = {"Single Ended", "Differential"}; /* 0dB to 30dB in 10dB steps */ -static const DECLARE_TLV_DB_SCALE(mic_tlv, 0, 10, 30); +static const DECLARE_TLV_DB_SCALE(mic_tlv, 0, 10, 0); static const struct soc_enum sn95031_micmode1_enum = SOC_ENUM_SINGLE(SN95031_MICAMP1, 1, 2, sn95031_micmode_text); -- cgit v1.2.3-70-g09d2 From 4a8d929d142ae594d8a5f0e1efba1d278d07bd8b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 16 Feb 2011 14:57:17 -0800 Subject: ASoC: Fix missing space in WM8994 Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8994.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index d78abeceaed..c0adbf8098e 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -526,7 +526,7 @@ static int wm8994_get_retune_mobile_enum(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - struct wm8994_priv *wm8994 =snd_soc_codec_get_drvdata(codec); + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); int block = wm8994_get_retune_mobile_block(kcontrol->id.name); ucontrol->value.enumerated.item[0] = wm8994->retune_mobile_cfg[block]; -- cgit v1.2.3-70-g09d2 From 5a9f91ca7994bd6a7c696fd397716da3bb440921 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 17 Feb 2011 12:05:46 -0800 Subject: ASoC: Log wm_hubs DC servo operation code when reporting a timeout Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm_hubs.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 613df5db0b3..a5c1556e32e 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -82,7 +82,8 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec, unsigned int op) } while (reg & op && count < 400); if (reg & op) - dev_err(codec->dev, "Timed out waiting for DC Servo\n"); + dev_err(codec->dev, "Timed out waiting for DC Servo %x\n", + op); } /* -- cgit v1.2.3-70-g09d2 From 40d2f1592ac262f13487b12cb47057e65b947bce Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 18 Feb 2011 14:47:02 -0800 Subject: ASoC: Mark WM8958 microphone detection registers readable So they show up in codec_reg. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994-tables.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8994-tables.c b/sound/soc/codecs/wm8994-tables.c index 68e9b024dd4..42b3248ba2f 100644 --- a/sound/soc/codecs/wm8994-tables.c +++ b/sound/soc/codecs/wm8994-tables.c @@ -209,9 +209,9 @@ const struct wm8994_access_mask wm8994_access_masks[WM8994_CACHE_SIZE] = { { 0x0000, 0x0000 }, /* R205 */ { 0x0000, 0x0000 }, /* R206 */ { 0x0000, 0x0000 }, /* R207 */ - { 0x0000, 0x0000 }, /* R208 */ - { 0x0000, 0x0000 }, /* R209 */ - { 0x0000, 0x0000 }, /* R210 */ + { 0xFFFF, 0xFFFF }, /* R208 */ + { 0xFFFF, 0xFFFF }, /* R209 */ + { 0xFFFF, 0xFFFF }, /* R210 */ { 0x0000, 0x0000 }, /* R211 */ { 0x0000, 0x0000 }, /* R212 */ { 0x0000, 0x0000 }, /* R213 */ -- cgit v1.2.3-70-g09d2 From 4baafdd76bafc51699a924d2bc1317f50b4ea75f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 18 Feb 2011 15:05:53 -0800 Subject: ASoC: Hook wm_hubs micbiases up to CLK_SYS The microphone detection functionality requires a clock to work. In any non-detection case where the MICBIAS is enabled CLK_SYS will be needed anyway so there is no negative impact on power consumption. Signed-off-by: Mark Brown --- sound/soc/codecs/wm_hubs.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index a5c1556e32e..7b6b3c18e29 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -675,6 +675,9 @@ SND_SOC_DAPM_OUTPUT("LINEOUT2N"), }; static const struct snd_soc_dapm_route analogue_routes[] = { + { "MICBIAS1", NULL, "CLK_SYS" }, + { "MICBIAS2", NULL, "CLK_SYS" }, + { "IN1L PGA", "IN1LP Switch", "IN1LP" }, { "IN1L PGA", "IN1LN Switch", "IN1LN" }, -- cgit v1.2.3-70-g09d2 From 7d700ac8d91f63f25cb58edeba06caddc65d85b0 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 18 Feb 2011 16:57:22 -0800 Subject: ASoC: Mark WM8958 microphone bias registers as readable Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8994-tables.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8994-tables.c b/sound/soc/codecs/wm8994-tables.c index 42b3248ba2f..61dfd91c6c7 100644 --- a/sound/soc/codecs/wm8994-tables.c +++ b/sound/soc/codecs/wm8994-tables.c @@ -62,8 +62,8 @@ const struct wm8994_access_mask wm8994_access_masks[WM8994_CACHE_SIZE] = { { 0x00FF, 0x00FF }, /* R58 - MICBIAS */ { 0x000F, 0x000F }, /* R59 - LDO 1 */ { 0x0007, 0x0007 }, /* R60 - LDO 2 */ - { 0x0000, 0x0000 }, /* R61 */ - { 0x0000, 0x0000 }, /* R62 */ + { 0xFFFF, 0xFFFF }, /* R61 */ + { 0xFFFF, 0xFFFF }, /* R62 */ { 0x0000, 0x0000 }, /* R63 */ { 0x0000, 0x0000 }, /* R64 */ { 0x0000, 0x0000 }, /* R65 */ -- cgit v1.2.3-70-g09d2 From 9b7c525dfaa9a1b5f01db1f3a1edc50bbb6eb739 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 17 Feb 2011 20:05:44 -0800 Subject: ASoC: Support WM8958 direct microphone detection IRQ Allow direct routing of the WM8958 microphone detection signal to a GPIO to be used, saving the need to demux the interrupt. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- include/linux/mfd/wm8994/pdata.h | 5 ++++ sound/soc/codecs/wm8994.c | 57 ++++++++++++++++++++++++---------------- 2 files changed, 40 insertions(+), 22 deletions(-) (limited to 'sound/soc/codecs') diff --git a/include/linux/mfd/wm8994/pdata.h b/include/linux/mfd/wm8994/pdata.h index 9eab263658b..06869466b7f 100644 --- a/include/linux/mfd/wm8994/pdata.h +++ b/include/linux/mfd/wm8994/pdata.h @@ -103,6 +103,11 @@ struct wm8994_pdata { unsigned int lineout1fb:1; unsigned int lineout2fb:1; + /* IRQ for microphone detection if brought out directly as a + * signal. + */ + int micdet_irq; + /* Microphone biases: 0=0.9*AVDD1 1=0.65*AVVD1 */ unsigned int micbias1_lvl:1; unsigned int micbias2_lvl:1; diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index b23e91027d6..1ad6e3db780 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -104,6 +104,7 @@ struct wm8994_priv { void *jack_cb_data; bool jack_is_mic; bool jack_is_video; + int micdet_irq; int revision; struct wm8994_pdata *pdata; @@ -3102,6 +3103,12 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) wm8994->pdata = dev_get_platdata(codec->dev->parent); wm8994->codec = codec; + if (wm8994->pdata && wm8994->pdata->micdet_irq) + wm8994->micdet_irq = wm8994->pdata->micdet_irq; + else if (wm8994->pdata && wm8994->pdata->irq_base) + wm8994->micdet_irq = wm8994->pdata->irq_base + + WM8994_IRQ_MIC1_DET; + pm_runtime_enable(codec->dev); pm_runtime_resume(codec->dev); @@ -3150,14 +3157,17 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) switch (control->type) { case WM8994: - ret = wm8994_request_irq(codec->control_data, - WM8994_IRQ_MIC1_DET, - wm8994_mic_irq, "Mic 1 detect", - wm8994); - if (ret != 0) - dev_warn(codec->dev, - "Failed to request Mic1 detect IRQ: %d\n", - ret); + if (wm8994->micdet_irq) { + ret = request_threaded_irq(wm8994->micdet_irq, NULL, + wm8994_mic_irq, + IRQF_TRIGGER_RISING, + "Mic1 detect", + wm8994); + if (ret != 0) + dev_warn(codec->dev, + "Failed to request Mic1 detect IRQ: %d\n", + ret); + } ret = wm8994_request_irq(codec->control_data, WM8994_IRQ_MIC1_SHRT, @@ -3188,15 +3198,17 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) break; case WM8958: - ret = wm8994_request_irq(codec->control_data, - WM8994_IRQ_MIC1_DET, - wm8958_mic_irq, "Mic detect", - wm8994); - if (ret != 0) - dev_warn(codec->dev, - "Failed to request Mic detect IRQ: %d\n", - ret); - break; + if (wm8994->micdet_irq) { + ret = request_threaded_irq(wm8994->micdet_irq, NULL, + wm8958_mic_irq, + IRQF_TRIGGER_RISING, + "Mic detect", + wm8994); + if (ret != 0) + dev_warn(codec->dev, + "Failed to request Mic detect IRQ: %d\n", + ret); + } } /* Remember if AIFnLRCLK is configured as a GPIO. This should be @@ -3328,7 +3340,8 @@ err_irq: wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC2_SHRT, wm8994); wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC2_DET, wm8994); wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC1_SHRT, wm8994); - wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC1_DET, wm8994); + if (wm8994->micdet_irq) + free_irq(wm8994->micdet_irq, wm8994); err: kfree(wm8994); return ret; @@ -3345,8 +3358,8 @@ static int wm8994_codec_remove(struct snd_soc_codec *codec) switch (control->type) { case WM8994: - wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC2_SHRT, - wm8994); + if (wm8994->micdet_irq) + free_irq(wm8994->micdet_irq, wm8994); wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC2_DET, wm8994); wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC1_SHRT, @@ -3356,8 +3369,8 @@ static int wm8994_codec_remove(struct snd_soc_codec *codec) break; case WM8958: - wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC1_DET, - wm8994); + if (wm8994->micdet_irq) + free_irq(wm8994->micdet_irq, wm8994); break; } kfree(wm8994->retune_mobile_texts); -- cgit v1.2.3-70-g09d2 From 48e028eccabc9c246bfad175262582a1ce34a316 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 21 Feb 2011 17:11:59 -0800 Subject: ASoC: Support configuration of WM8958 microphone bias analogue parameters The WM8958 has a different microphone bias architecture to WM8994 so needs different configuration to WM8994. Support this in platform data. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- include/linux/mfd/wm8994/pdata.h | 7 +++++-- include/linux/mfd/wm8994/registers.h | 2 ++ sound/soc/codecs/wm8994.c | 7 +++++++ 3 files changed, 14 insertions(+), 2 deletions(-) (limited to 'sound/soc/codecs') diff --git a/include/linux/mfd/wm8994/pdata.h b/include/linux/mfd/wm8994/pdata.h index 06869466b7f..466b1c777af 100644 --- a/include/linux/mfd/wm8994/pdata.h +++ b/include/linux/mfd/wm8994/pdata.h @@ -108,13 +108,16 @@ struct wm8994_pdata { */ int micdet_irq; - /* Microphone biases: 0=0.9*AVDD1 1=0.65*AVVD1 */ + /* WM8994 microphone biases: 0=0.9*AVDD1 1=0.65*AVVD1 */ unsigned int micbias1_lvl:1; unsigned int micbias2_lvl:1; - /* Jack detect threashold levels, see datasheet for values */ + /* WM8994 jack detect threashold levels, see datasheet for values */ unsigned int jd_scthr:2; unsigned int jd_thr:2; + + /* WM8958 microphone bias configuration */ + int micbias[2]; }; #endif diff --git a/include/linux/mfd/wm8994/registers.h b/include/linux/mfd/wm8994/registers.h index be072faec6f..f3ee8428467 100644 --- a/include/linux/mfd/wm8994/registers.h +++ b/include/linux/mfd/wm8994/registers.h @@ -63,6 +63,8 @@ #define WM8994_MICBIAS 0x3A #define WM8994_LDO_1 0x3B #define WM8994_LDO_2 0x3C +#define WM8958_MICBIAS1 0x3D +#define WM8958_MICBIAS2 0x3E #define WM8994_CHARGE_PUMP_1 0x4C #define WM8958_CHARGE_PUMP_2 0x4D #define WM8994_CLASS_W_1 0x51 diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 1ad6e3db780..9b9c15ffb7d 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2855,6 +2855,13 @@ static void wm8994_handle_pdata(struct wm8994_priv *wm8994) else snd_soc_add_controls(wm8994->codec, wm8994_eq_controls, ARRAY_SIZE(wm8994_eq_controls)); + + for (i = 0; i < ARRAY_SIZE(pdata->micbias); i++) { + if (pdata->micbias[i]) { + snd_soc_write(codec, WM8958_MICBIAS1 + i, + pdata->micbias[i] & 0xffff); + } + } } /** -- cgit v1.2.3-70-g09d2 From 864c4bd2487619564acd75fdcf1a4349992e9090 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 21 Feb 2011 20:51:13 -0800 Subject: ASoC: Simplify default WM8958 jack detection code The default WM8958 jack detection handler implements a full set of buttons and also support for video detection. Support for multi-button jacks is fairly system specific and will usually require some tuning for headsets so simplify the implementation to only report a simple short to ground button, leaving multi-button headsets to be handled by system specific code. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8994.c | 38 ++++---------------------------------- 1 file changed, 4 insertions(+), 34 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 9b9c15ffb7d..0dc14115f10 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -102,8 +102,6 @@ struct wm8994_priv { wm8958_micdet_cb jack_cb; void *jack_cb_data; - bool jack_is_mic; - bool jack_is_video; int micdet_irq; int revision; @@ -2972,46 +2970,18 @@ static void wm8958_default_micdet(u16 status, void *data) int report = 0; /* If nothing present then clear our statuses */ - if (!(status & WM8958_MICD_STS)) { - wm8994->jack_is_video = false; - wm8994->jack_is_mic = false; + if (!(status & WM8958_MICD_STS)) goto done; - } - - /* Assume anything over 475 ohms is a microphone and remember - * that we've seen one (since buttons override it) */ - if (status & 0x600) - wm8994->jack_is_mic = true; - if (wm8994->jack_is_mic) - report |= SND_JACK_MICROPHONE; - /* Video has an impedence of approximately 75 ohms; assume - * this isn't used as a button and remember it since buttons - * override it. */ - if (status & 0x40) - wm8994->jack_is_video = true; - if (wm8994->jack_is_video) - report |= SND_JACK_VIDEOOUT; + report = SND_JACK_MICROPHONE; /* Everything else is buttons; just assign slots */ - if (status & 0x4) + if (status & 0x1c0) report |= SND_JACK_BTN_0; - if (status & 0x8) - report |= SND_JACK_BTN_1; - if (status & 0x10) - report |= SND_JACK_BTN_2; - if (status & 0x20) - report |= SND_JACK_BTN_3; - if (status & 0x80) - report |= SND_JACK_BTN_4; - if (status & 0x100) - report |= SND_JACK_BTN_5; done: snd_soc_jack_report(wm8994->micdet[0].jack, report, - SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 | - SND_JACK_BTN_3 | SND_JACK_BTN_4 | SND_JACK_BTN_5 | - SND_JACK_MICROPHONE | SND_JACK_VIDEOOUT); + SND_JACK_BTN_0 | SND_JACK_MICROPHONE); } /** -- cgit v1.2.3-70-g09d2 From 2031c0645c8e2f3935254bf201b4f6f00dd0790d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 24 Feb 2011 20:26:02 +0000 Subject: ASoC: Remove -codec suffix from WM9081 driver Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm9081.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 5c224dd917d..b62e98942cf 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -1363,7 +1363,7 @@ MODULE_DEVICE_TABLE(i2c, wm9081_i2c_id); static struct i2c_driver wm9081_i2c_driver = { .driver = { - .name = "wm9081-codec", + .name = "wm9081", .owner = THIS_MODULE, }, .probe = wm9081_i2c_probe, -- cgit v1.2.3-70-g09d2 From 9b34e6cc3bc2bf826c078c93c81b46f6c08da25f Mon Sep 17 00:00:00 2001 From: Zeng Zhaoming Date: Thu, 24 Feb 2011 02:08:21 +0800 Subject: ASoC: Add Freescale SGTL5000 codec support Add Freescale SGTL5000 codec support. Supported features: - line-in and mic input - headphone and line-out output - line-in bypass ADC and DAC to headphone - 16, 20, 24, 32 bit audio - 8 ~ 96k sample rates Signed-off-by: Zeng Zhaoming Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/sgtl5000.c | 1512 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/sgtl5000.h | 402 ++++++++++++ 4 files changed, 1921 insertions(+) create mode 100644 sound/soc/codecs/sgtl5000.c create mode 100644 sound/soc/codecs/sgtl5000.h (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index e239345a4d5..c04da187129 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -33,6 +33,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX98088 if I2C select SND_SOC_MAX9877 if I2C select SND_SOC_PCM3008 + select SND_SOC_SGTL5000 if I2C select SND_SOC_SN95031 if INTEL_SCU_IPC select SND_SOC_SPDIF select SND_SOC_SSM2602 if I2C @@ -182,6 +183,10 @@ config SND_SOC_MAX98088 config SND_SOC_PCM3008 tristate +#Freescale sgtl5000 codec +config SND_SOC_SGTL5000 + tristate + config SND_SOC_SN95031 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index ae10507dd2e..3bbb08c512d 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -19,6 +19,7 @@ snd-soc-dmic-objs := dmic.o snd-soc-l3-objs := l3.o snd-soc-max98088-objs := max98088.o snd-soc-pcm3008-objs := pcm3008.o +snd-soc-sgtl5000-objs := sgtl5000.o snd-soc-alc5623-objs := alc5623.o snd-soc-sn95031-objs := sn95031.o snd-soc-spdif-objs := spdif_transciever.o @@ -103,6 +104,7 @@ obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o +obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o obj-$(CONFIG_SND_SOC_SN95031) +=snd-soc-sn95031.o obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c new file mode 100644 index 00000000000..9eb206e852d --- /dev/null +++ b/sound/soc/codecs/sgtl5000.c @@ -0,0 +1,1512 @@ +/* + * sgtl5000.c -- SGTL5000 ALSA SoC Audio driver + * + * Copyright 2010-2011 Freescale Semiconductor, Inc. All Rights Reserved. + * + * 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 +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "sgtl5000.h" + +#define SGTL5000_DAP_REG_OFFSET 0x0100 +#define SGTL5000_MAX_REG_OFFSET 0x013A + +/* default value of sgtl5000 registers except DAP */ +static const u16 sgtl5000_regs[SGTL5000_MAX_REG_OFFSET >> 1] = { + 0xa011, /* 0x0000, CHIP_ID. 11 stand for revison 17 */ + 0x0000, /* 0x0002, CHIP_DIG_POWER. */ + 0x0008, /* 0x0004, CHIP_CKL_CTRL */ + 0x0010, /* 0x0006, CHIP_I2S_CTRL */ + 0x0000, /* 0x0008, reserved */ + 0x0008, /* 0x000A, CHIP_SSS_CTRL */ + 0x0000, /* 0x000C, reserved */ + 0x020c, /* 0x000E, CHIP_ADCDAC_CTRL */ + 0x3c3c, /* 0x0010, CHIP_DAC_VOL */ + 0x0000, /* 0x0012, reserved */ + 0x015f, /* 0x0014, CHIP_PAD_STRENGTH */ + 0x0000, /* 0x0016, reserved */ + 0x0000, /* 0x0018, reserved */ + 0x0000, /* 0x001A, reserved */ + 0x0000, /* 0x001E, reserved */ + 0x0000, /* 0x0020, CHIP_ANA_ADC_CTRL */ + 0x1818, /* 0x0022, CHIP_ANA_HP_CTRL */ + 0x0111, /* 0x0024, CHIP_ANN_CTRL */ + 0x0000, /* 0x0026, CHIP_LINREG_CTRL */ + 0x0000, /* 0x0028, CHIP_REF_CTRL */ + 0x0000, /* 0x002A, CHIP_MIC_CTRL */ + 0x0000, /* 0x002C, CHIP_LINE_OUT_CTRL */ + 0x0404, /* 0x002E, CHIP_LINE_OUT_VOL */ + 0x7060, /* 0x0030, CHIP_ANA_POWER */ + 0x5000, /* 0x0032, CHIP_PLL_CTRL */ + 0x0000, /* 0x0034, CHIP_CLK_TOP_CTRL */ + 0x0000, /* 0x0036, CHIP_ANA_STATUS */ + 0x0000, /* 0x0038, reserved */ + 0x0000, /* 0x003A, CHIP_ANA_TEST2 */ + 0x0000, /* 0x003C, CHIP_SHORT_CTRL */ + 0x0000, /* reserved */ +}; + +/* default value of dap registers */ +static const u16 sgtl5000_dap_regs[] = { + 0x0000, /* 0x0100, DAP_CONTROL */ + 0x0000, /* 0x0102, DAP_PEQ */ + 0x0040, /* 0x0104, DAP_BASS_ENHANCE */ + 0x051f, /* 0x0106, DAP_BASS_ENHANCE_CTRL */ + 0x0000, /* 0x0108, DAP_AUDIO_EQ */ + 0x0040, /* 0x010A, DAP_SGTL_SURROUND */ + 0x0000, /* 0x010C, DAP_FILTER_COEF_ACCESS */ + 0x0000, /* 0x010E, DAP_COEF_WR_B0_MSB */ + 0x0000, /* 0x0110, DAP_COEF_WR_B0_LSB */ + 0x0000, /* 0x0112, reserved */ + 0x0000, /* 0x0114, reserved */ + 0x002f, /* 0x0116, DAP_AUDIO_EQ_BASS_BAND0 */ + 0x002f, /* 0x0118, DAP_AUDIO_EQ_BAND0 */ + 0x002f, /* 0x011A, DAP_AUDIO_EQ_BAND2 */ + 0x002f, /* 0x011C, DAP_AUDIO_EQ_BAND3 */ + 0x002f, /* 0x011E, DAP_AUDIO_EQ_TREBLE_BAND4 */ + 0x8000, /* 0x0120, DAP_MAIN_CHAN */ + 0x0000, /* 0x0122, DAP_MIX_CHAN */ + 0x0510, /* 0x0124, DAP_AVC_CTRL */ + 0x1473, /* 0x0126, DAP_AVC_THRESHOLD */ + 0x0028, /* 0x0128, DAP_AVC_ATTACK */ + 0x0050, /* 0x012A, DAP_AVC_DECAY */ + 0x0000, /* 0x012C, DAP_COEF_WR_B1_MSB */ + 0x0000, /* 0x012E, DAP_COEF_WR_B1_LSB */ + 0x0000, /* 0x0130, DAP_COEF_WR_B2_MSB */ + 0x0000, /* 0x0132, DAP_COEF_WR_B2_LSB */ + 0x0000, /* 0x0134, DAP_COEF_WR_A1_MSB */ + 0x0000, /* 0x0136, DAP_COEF_WR_A1_LSB */ + 0x0000, /* 0x0138, DAP_COEF_WR_A2_MSB */ + 0x0000, /* 0x013A, DAP_COEF_WR_A2_LSB */ +}; + +/* regulator supplies for sgtl5000, VDDD is an optional external supply */ +enum sgtl5000_regulator_supplies { + VDDA, + VDDIO, + VDDD, + SGTL5000_SUPPLY_NUM +}; + +/* vddd is optional supply */ +static const char *supply_names[SGTL5000_SUPPLY_NUM] = { + "VDDA", + "VDDIO", + "VDDD" +}; + +#define LDO_CONSUMER_NAME "VDDD_LDO" +#define LDO_VOLTAGE 1200000 + +static struct regulator_consumer_supply ldo_consumer[] = { + REGULATOR_SUPPLY(LDO_CONSUMER_NAME, NULL), +}; + +struct regulator_init_data ldo_init_data = { + .constraints = { + .min_uV = 850000, + .max_uV = 1600000, + .valid_modes_mask = REGULATOR_MODE_NORMAL, + .valid_ops_mask = REGULATOR_CHANGE_STATUS, + }, + .num_consumer_supplies = 1, + .consumer_supplies = &ldo_consumer[0], +}; + +/* + * sgtl5000 internal ldo regulator, + * enabled when VDDD not provided + */ +struct ldo_regulator { + struct regulator_desc desc; + struct regulator_dev *dev; + int voltage; + void *codec_data; + bool enabled; +}; + +/* sgtl5000 private structure in codec */ +struct sgtl5000_priv { + int sysclk; /* sysclk rate */ + int master; /* i2s master or not */ + int fmt; /* i2s data format */ + struct regulator_bulk_data supplies[SGTL5000_SUPPLY_NUM]; + struct ldo_regulator *ldo; +}; + +/* + * mic_bias power on/off share the same register bits with + * output impedance of mic bias, when power on mic bias, we + * need reclaim it to impedance value. + * 0x0 = Powered off + * 0x1 = 2Kohm + * 0x2 = 4Kohm + * 0x3 = 8Kohm + */ +static int mic_bias_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + switch (event) { + case SND_SOC_DAPM_POST_PMU: + /* change mic bias resistor to 4Kohm */ + snd_soc_update_bits(w->codec, SGTL5000_CHIP_MIC_CTRL, + SGTL5000_BIAS_R_4k, SGTL5000_BIAS_R_4k); + break; + + case SND_SOC_DAPM_PRE_PMD: + /* + * SGTL5000_BIAS_R_8k as mask to clean the two bits + * of mic bias and output impedance + */ + snd_soc_update_bits(w->codec, SGTL5000_CHIP_MIC_CTRL, + SGTL5000_BIAS_R_8k, 0); + break; + } + return 0; +} + +/* + * using codec assist to small pop, hp_powerup or lineout_powerup + * should stay setting until vag_powerup is fully ramped down, + * vag fully ramped down require 400ms. + */ +static int small_pop_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER, + SGTL5000_VAG_POWERUP, SGTL5000_VAG_POWERUP); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER, + SGTL5000_VAG_POWERUP, 0); + msleep(400); + break; + default: + break; + } + + return 0; +} + +/* input sources for ADC */ +static const char *adc_mux_text[] = { + "MIC_IN", "LINE_IN" +}; + +static const struct soc_enum adc_enum = +SOC_ENUM_SINGLE(SGTL5000_CHIP_ANA_CTRL, 2, 2, adc_mux_text); + +static const struct snd_kcontrol_new adc_mux = +SOC_DAPM_ENUM("Capture Mux", adc_enum); + +/* input sources for DAC */ +static const char *dac_mux_text[] = { + "DAC", "LINE_IN" +}; + +static const struct soc_enum dac_enum = +SOC_ENUM_SINGLE(SGTL5000_CHIP_ANA_CTRL, 6, 2, dac_mux_text); + +static const struct snd_kcontrol_new dac_mux = +SOC_DAPM_ENUM("Headphone Mux", dac_enum); + +static const struct snd_soc_dapm_widget sgtl5000_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("LINE_IN"), + SND_SOC_DAPM_INPUT("MIC_IN"), + + SND_SOC_DAPM_OUTPUT("HP_OUT"), + SND_SOC_DAPM_OUTPUT("LINE_OUT"), + + SND_SOC_DAPM_MICBIAS_E("Mic Bias", SGTL5000_CHIP_MIC_CTRL, 8, 0, + mic_bias_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + + SND_SOC_DAPM_PGA_E("HP", SGTL5000_CHIP_ANA_POWER, 4, 0, NULL, 0, + small_pop_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_PGA_E("LO", SGTL5000_CHIP_ANA_POWER, 0, 0, NULL, 0, + small_pop_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD), + + SND_SOC_DAPM_MUX("Capture Mux", SND_SOC_NOPM, 0, 0, &adc_mux), + SND_SOC_DAPM_MUX("Headphone Mux", SND_SOC_NOPM, 0, 0, &dac_mux), + + /* aif for i2s input */ + SND_SOC_DAPM_AIF_IN("AIFIN", "Playback", + 0, SGTL5000_CHIP_DIG_POWER, + 0, 0), + + /* aif for i2s output */ + SND_SOC_DAPM_AIF_OUT("AIFOUT", "Capture", + 0, SGTL5000_CHIP_DIG_POWER, + 1, 0), + + SND_SOC_DAPM_ADC("ADC", "Capture", SGTL5000_CHIP_ANA_POWER, 1, 0), + + SND_SOC_DAPM_DAC("DAC", "Playback", SGTL5000_CHIP_ANA_POWER, 3, 0), +}; + +/* routes for sgtl5000 */ +static const struct snd_soc_dapm_route audio_map[] = { + {"Capture Mux", "LINE_IN", "LINE_IN"}, /* line_in --> adc_mux */ + {"Capture Mux", "MIC_IN", "MIC_IN"}, /* mic_in --> adc_mux */ + + {"ADC", NULL, "Capture Mux"}, /* adc_mux --> adc */ + {"AIFOUT", NULL, "ADC"}, /* adc --> i2s_out */ + + {"DAC", NULL, "AIFIN"}, /* i2s-->dac,skip audio mux */ + {"Headphone Mux", "DAC", "DAC"}, /* dac --> hp_mux */ + {"LO", NULL, "DAC"}, /* dac --> line_out */ + + {"Headphone Mux", "LINE_IN", "LINE_IN"},/* line_in --> hp_mux */ + {"HP", NULL, "Headphone Mux"}, /* hp_mux --> hp */ + + {"LINE_OUT", NULL, "LO"}, + {"HP_OUT", NULL, "HP"}, +}; + +/* custom function to fetch info of PCM playback volume */ +static int dac_info_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 2; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 0xfc - 0x3c; + return 0; +} + +/* + * custom function to get of PCM playback volume + * + * dac volume register + * 15-------------8-7--------------0 + * | R channel vol | L channel vol | + * ------------------------------- + * + * PCM volume with 0.5017 dB steps from 0 to -90 dB + * + * register values map to dB + * 0x3B and less = Reserved + * 0x3C = 0 dB + * 0x3D = -0.5 dB + * 0xF0 = -90 dB + * 0xFC and greater = Muted + * + * register value map to userspace value + * + * register value 0x3c(0dB) 0xf0(-90dB)0xfc + * ------------------------------ + * userspace value 0xc0 0 + */ +static int dac_get_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int reg; + int l; + int r; + + reg = snd_soc_read(codec, SGTL5000_CHIP_DAC_VOL); + + /* get left channel volume */ + l = (reg & SGTL5000_DAC_VOL_LEFT_MASK) >> SGTL5000_DAC_VOL_LEFT_SHIFT; + + /* get right channel volume */ + r = (reg & SGTL5000_DAC_VOL_RIGHT_MASK) >> SGTL5000_DAC_VOL_RIGHT_SHIFT; + + /* make sure value fall in (0x3c,0xfc) */ + l = clamp(l, 0x3c, 0xfc); + r = clamp(r, 0x3c, 0xfc); + + /* invert it and map to userspace value */ + l = 0xfc - l; + r = 0xfc - r; + + ucontrol->value.integer.value[0] = l; + ucontrol->value.integer.value[1] = r; + + return 0; +} + +/* + * custom function to put of PCM playback volume + * + * dac volume register + * 15-------------8-7--------------0 + * | R channel vol | L channel vol | + * ------------------------------- + * + * PCM volume with 0.5017 dB steps from 0 to -90 dB + * + * register values map to dB + * 0x3B and less = Reserved + * 0x3C = 0 dB + * 0x3D = -0.5 dB + * 0xF0 = -90 dB + * 0xFC and greater = Muted + * + * userspace value map to register value + * + * userspace value 0xc0 0 + * ------------------------------ + * register value 0x3c(0dB) 0xf0(-90dB)0xfc + */ +static int dac_put_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int reg; + int l; + int r; + + l = ucontrol->value.integer.value[0]; + r = ucontrol->value.integer.value[1]; + + /* make sure userspace volume fall in (0, 0xfc-0x3c) */ + l = clamp(l, 0, 0xfc - 0x3c); + r = clamp(r, 0, 0xfc - 0x3c); + + /* invert it, get the value can be set to register */ + l = 0xfc - l; + r = 0xfc - r; + + /* shift to get the register value */ + reg = l << SGTL5000_DAC_VOL_LEFT_SHIFT | + r << SGTL5000_DAC_VOL_RIGHT_SHIFT; + + snd_soc_write(codec, SGTL5000_CHIP_DAC_VOL, reg); + + return 0; +} + +static const DECLARE_TLV_DB_SCALE(capture_6db_attenuate, -600, 600, 0); + +/* tlv for mic gain, 0db 20db 30db 40db */ +static const unsigned int mic_gain_tlv[] = { + TLV_DB_RANGE_HEAD(4), + 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), + 1, 3, TLV_DB_SCALE_ITEM(2000, 1000, 0), +}; + +/* tlv for hp volume, -51.5db to 12.0db, step .5db */ +static const DECLARE_TLV_DB_SCALE(headphone_volume, -5150, 50, 0); + +static const struct snd_kcontrol_new sgtl5000_snd_controls[] = { + /* SOC_DOUBLE_S8_TLV with invert */ + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "PCM Playback Volume", + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | + SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = dac_info_volsw, + .get = dac_get_volsw, + .put = dac_put_volsw, + }, + + SOC_DOUBLE("Capture Volume", SGTL5000_CHIP_ANA_ADC_CTRL, 0, 4, 0xf, 0), + SOC_SINGLE_TLV("Capture Attenuate Switch (-6dB)", + SGTL5000_CHIP_ANA_ADC_CTRL, + 8, 2, 0, capture_6db_attenuate), + SOC_SINGLE("Capture ZC Switch", SGTL5000_CHIP_ANA_CTRL, 1, 1, 0), + + SOC_DOUBLE_TLV("Headphone Playback Volume", + SGTL5000_CHIP_ANA_HP_CTRL, + 0, 8, + 0x7f, 1, + headphone_volume), + SOC_SINGLE("Headphone Playback ZC Switch", SGTL5000_CHIP_ANA_CTRL, + 5, 1, 0), + + SOC_SINGLE_TLV("Mic Volume", SGTL5000_CHIP_MIC_CTRL, + 0, 4, 0, mic_gain_tlv), +}; + +/* mute the codec used by alsa core */ +static int sgtl5000_digital_mute(struct snd_soc_dai *codec_dai, int mute) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 adcdac_ctrl = SGTL5000_DAC_MUTE_LEFT | SGTL5000_DAC_MUTE_RIGHT; + + snd_soc_update_bits(codec, SGTL5000_CHIP_ADCDAC_CTRL, + adcdac_ctrl, mute ? adcdac_ctrl : 0); + + return 0; +} + +/* set codec format */ +static int sgtl5000_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + u16 i2sctl = 0; + + sgtl5000->master = 0; + /* + * i2s clock and frame master setting. + * ONLY support: + * - clock and frame slave, + * - clock and frame master + */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + case SND_SOC_DAIFMT_CBM_CFM: + i2sctl |= SGTL5000_I2S_MASTER; + sgtl5000->master = 1; + break; + default: + return -EINVAL; + } + + /* setting i2s data format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + i2sctl |= SGTL5000_I2S_MODE_PCM; + break; + case SND_SOC_DAIFMT_DSP_B: + i2sctl |= SGTL5000_I2S_MODE_PCM; + i2sctl |= SGTL5000_I2S_LRALIGN; + break; + case SND_SOC_DAIFMT_I2S: + i2sctl |= SGTL5000_I2S_MODE_I2S_LJ; + break; + case SND_SOC_DAIFMT_RIGHT_J: + i2sctl |= SGTL5000_I2S_MODE_RJ; + i2sctl |= SGTL5000_I2S_LRPOL; + break; + case SND_SOC_DAIFMT_LEFT_J: + i2sctl |= SGTL5000_I2S_MODE_I2S_LJ; + i2sctl |= SGTL5000_I2S_LRALIGN; + break; + default: + return -EINVAL; + } + + sgtl5000->fmt = fmt & SND_SOC_DAIFMT_FORMAT_MASK; + + /* Clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + i2sctl |= SGTL5000_I2S_SCLK_INV; + break; + default: + return -EINVAL; + } + + snd_soc_write(codec, SGTL5000_CHIP_I2S_CTRL, i2sctl); + + return 0; +} + +/* set codec sysclk */ +static int sgtl5000_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 sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + + switch (clk_id) { + case SGTL5000_SYSCLK: + sgtl5000->sysclk = freq; + break; + default: + return -EINVAL; + } + + return 0; +} + +/* + * set clock according to i2s frame clock, + * sgtl5000 provide 2 clock sources. + * 1. sys_mclk. sample freq can only configure to + * 1/256, 1/384, 1/512 of sys_mclk. + * 2. pll. can derive any audio clocks. + * + * clock setting rules: + * 1. in slave mode, only sys_mclk can use. + * 2. as constraint by sys_mclk, sample freq should + * set to 32k, 44.1k and above. + * 3. using sys_mclk prefer to pll to save power. + */ +static int sgtl5000_set_clock(struct snd_soc_codec *codec, int frame_rate) +{ + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + int clk_ctl = 0; + int sys_fs; /* sample freq */ + + /* + * sample freq should be divided by frame clock, + * if frame clock lower than 44.1khz, sample feq should set to + * 32khz or 44.1khz. + */ + switch (frame_rate) { + case 8000: + case 16000: + sys_fs = 32000; + break; + case 11025: + case 22050: + sys_fs = 44100; + break; + default: + sys_fs = frame_rate; + break; + } + + /* set divided factor of frame clock */ + switch (sys_fs / frame_rate) { + case 4: + clk_ctl |= SGTL5000_RATE_MODE_DIV_4 << SGTL5000_RATE_MODE_SHIFT; + break; + case 2: + clk_ctl |= SGTL5000_RATE_MODE_DIV_2 << SGTL5000_RATE_MODE_SHIFT; + break; + case 1: + clk_ctl |= SGTL5000_RATE_MODE_DIV_1 << SGTL5000_RATE_MODE_SHIFT; + break; + default: + return -EINVAL; + } + + /* set the sys_fs according to frame rate */ + switch (sys_fs) { + case 32000: + clk_ctl |= SGTL5000_SYS_FS_32k << SGTL5000_SYS_FS_SHIFT; + break; + case 44100: + clk_ctl |= SGTL5000_SYS_FS_44_1k << SGTL5000_SYS_FS_SHIFT; + break; + case 48000: + clk_ctl |= SGTL5000_SYS_FS_48k << SGTL5000_SYS_FS_SHIFT; + break; + case 96000: + clk_ctl |= SGTL5000_SYS_FS_96k << SGTL5000_SYS_FS_SHIFT; + break; + default: + dev_err(codec->dev, "frame rate %d not supported\n", + frame_rate); + return -EINVAL; + } + + /* + * calculate the divider of mclk/sample_freq, + * factor of freq =96k can only be 256, since mclk in range (12m,27m) + */ + switch (sgtl5000->sysclk / sys_fs) { + case 256: + clk_ctl |= SGTL5000_MCLK_FREQ_256FS << + SGTL5000_MCLK_FREQ_SHIFT; + break; + case 384: + clk_ctl |= SGTL5000_MCLK_FREQ_384FS << + SGTL5000_MCLK_FREQ_SHIFT; + break; + case 512: + clk_ctl |= SGTL5000_MCLK_FREQ_512FS << + SGTL5000_MCLK_FREQ_SHIFT; + break; + default: + /* if mclk not satisify the divider, use pll */ + if (sgtl5000->master) { + clk_ctl |= SGTL5000_MCLK_FREQ_PLL << + SGTL5000_MCLK_FREQ_SHIFT; + } else { + dev_err(codec->dev, + "PLL not supported in slave mode\n"); + return -EINVAL; + } + } + + /* if using pll, please check manual 6.4.2 for detail */ + if ((clk_ctl & SGTL5000_MCLK_FREQ_MASK) == SGTL5000_MCLK_FREQ_PLL) { + u64 out, t; + int div2; + int pll_ctl; + unsigned int in, int_div, frac_div; + + if (sgtl5000->sysclk > 17000000) { + div2 = 1; + in = sgtl5000->sysclk / 2; + } else { + div2 = 0; + in = sgtl5000->sysclk; + } + if (sys_fs == 44100) + out = 180633600; + else + out = 196608000; + t = do_div(out, in); + int_div = out; + t *= 2048; + do_div(t, in); + frac_div = t; + pll_ctl = int_div << SGTL5000_PLL_INT_DIV_SHIFT | + frac_div << SGTL5000_PLL_FRAC_DIV_SHIFT; + + snd_soc_write(codec, SGTL5000_CHIP_PLL_CTRL, pll_ctl); + if (div2) + snd_soc_update_bits(codec, + SGTL5000_CHIP_CLK_TOP_CTRL, + SGTL5000_INPUT_FREQ_DIV2, + SGTL5000_INPUT_FREQ_DIV2); + else + snd_soc_update_bits(codec, + SGTL5000_CHIP_CLK_TOP_CTRL, + SGTL5000_INPUT_FREQ_DIV2, + 0); + + /* power up pll */ + snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, + SGTL5000_PLL_POWERUP | SGTL5000_VCOAMP_POWERUP, + SGTL5000_PLL_POWERUP | SGTL5000_VCOAMP_POWERUP); + } else { + /* power down pll */ + snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, + SGTL5000_PLL_POWERUP | SGTL5000_VCOAMP_POWERUP, + 0); + } + + /* if using pll, clk_ctrl must be set after pll power up */ + snd_soc_write(codec, SGTL5000_CHIP_CLK_CTRL, clk_ctl); + + return 0; +} + +/* + * Set PCM DAI bit size and sample rate. + * input: params_rate, params_fmt + */ +static int sgtl5000_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + int channels = params_channels(params); + int i2s_ctl = 0; + int stereo; + int ret; + + /* sysclk should already set */ + if (!sgtl5000->sysclk) { + dev_err(codec->dev, "%s: set sysclk first!\n", __func__); + return -EFAULT; + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + stereo = SGTL5000_DAC_STEREO; + else + stereo = SGTL5000_ADC_STEREO; + + /* set mono to save power */ + snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, stereo, + channels == 1 ? 0 : stereo); + + /* set codec clock base on lrclk */ + ret = sgtl5000_set_clock(codec, params_rate(params)); + if (ret) + return ret; + + /* set i2s data format */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + if (sgtl5000->fmt == SND_SOC_DAIFMT_RIGHT_J) + return -EINVAL; + i2s_ctl |= SGTL5000_I2S_DLEN_16 << SGTL5000_I2S_DLEN_SHIFT; + i2s_ctl |= SGTL5000_I2S_SCLKFREQ_32FS << + SGTL5000_I2S_SCLKFREQ_SHIFT; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + i2s_ctl |= SGTL5000_I2S_DLEN_20 << SGTL5000_I2S_DLEN_SHIFT; + i2s_ctl |= SGTL5000_I2S_SCLKFREQ_64FS << + SGTL5000_I2S_SCLKFREQ_SHIFT; + break; + case SNDRV_PCM_FORMAT_S24_LE: + i2s_ctl |= SGTL5000_I2S_DLEN_24 << SGTL5000_I2S_DLEN_SHIFT; + i2s_ctl |= SGTL5000_I2S_SCLKFREQ_64FS << + SGTL5000_I2S_SCLKFREQ_SHIFT; + break; + case SNDRV_PCM_FORMAT_S32_LE: + if (sgtl5000->fmt == SND_SOC_DAIFMT_RIGHT_J) + return -EINVAL; + i2s_ctl |= SGTL5000_I2S_DLEN_32 << SGTL5000_I2S_DLEN_SHIFT; + i2s_ctl |= SGTL5000_I2S_SCLKFREQ_64FS << + SGTL5000_I2S_SCLKFREQ_SHIFT; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, SGTL5000_CHIP_I2S_CTRL, i2s_ctl, i2s_ctl); + + return 0; +} + +static int ldo_regulator_is_enabled(struct regulator_dev *dev) +{ + struct ldo_regulator *ldo = rdev_get_drvdata(dev); + + return ldo->enabled; +} + +static int ldo_regulator_enable(struct regulator_dev *dev) +{ + struct ldo_regulator *ldo = rdev_get_drvdata(dev); + struct snd_soc_codec *codec = (struct snd_soc_codec *)ldo->codec_data; + int reg; + + if (ldo_regulator_is_enabled(dev)) + return 0; + + /* set regulator value firstly */ + reg = (1600 - ldo->voltage / 1000) / 50; + reg = clamp(reg, 0x0, 0xf); + + /* amend the voltage value, unit: uV */ + ldo->voltage = (1600 - reg * 50) * 1000; + + /* set voltage to register */ + snd_soc_update_bits(codec, SGTL5000_CHIP_LINREG_CTRL, + (0x1 << 4) - 1, reg); + + snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, + SGTL5000_LINEREG_D_POWERUP, + SGTL5000_LINEREG_D_POWERUP); + + /* when internal ldo enabled, simple digital power can be disabled */ + snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, + SGTL5000_LINREG_SIMPLE_POWERUP, + 0); + + ldo->enabled = 1; + return 0; +} + +static int ldo_regulator_disable(struct regulator_dev *dev) +{ + struct ldo_regulator *ldo = rdev_get_drvdata(dev); + struct snd_soc_codec *codec = (struct snd_soc_codec *)ldo->codec_data; + + snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, + SGTL5000_LINEREG_D_POWERUP, + 0); + + /* clear voltage info */ + snd_soc_update_bits(codec, SGTL5000_CHIP_LINREG_CTRL, + (0x1 << 4) - 1, 0); + + ldo->enabled = 0; + + return 0; +} + +static int ldo_regulator_get_voltage(struct regulator_dev *dev) +{ + struct ldo_regulator *ldo = rdev_get_drvdata(dev); + + return ldo->voltage; +} + +static struct regulator_ops ldo_regulator_ops = { + .is_enabled = ldo_regulator_is_enabled, + .enable = ldo_regulator_enable, + .disable = ldo_regulator_disable, + .get_voltage = ldo_regulator_get_voltage, +}; + +static int ldo_regulator_register(struct snd_soc_codec *codec, + struct regulator_init_data *init_data, + int voltage) +{ + struct ldo_regulator *ldo; + + ldo = kzalloc(sizeof(struct ldo_regulator), GFP_KERNEL); + + if (!ldo) { + dev_err(codec->dev, "failed to allocate ldo_regulator\n"); + return -ENOMEM; + } + + ldo->desc.name = kstrdup(dev_name(codec->dev), GFP_KERNEL); + if (!ldo->desc.name) { + kfree(ldo); + dev_err(codec->dev, "failed to allocate decs name memory\n"); + return -ENOMEM; + } + + ldo->desc.type = REGULATOR_VOLTAGE; + ldo->desc.owner = THIS_MODULE; + ldo->desc.ops = &ldo_regulator_ops; + ldo->desc.n_voltages = 1; + + ldo->codec_data = codec; + ldo->voltage = voltage; + + ldo->dev = regulator_register(&ldo->desc, codec->dev, + init_data, ldo); + if (IS_ERR(ldo->dev)) { + dev_err(codec->dev, "failed to register regulator\n"); + kfree(ldo->desc.name); + kfree(ldo); + + return PTR_ERR(ldo->dev); + } + + return 0; +} + +static int ldo_regulator_remove(struct snd_soc_codec *codec) +{ + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + struct ldo_regulator *ldo = sgtl5000->ldo; + + if (!ldo) + return 0; + + regulator_unregister(ldo->dev); + kfree(ldo->desc.name); + kfree(ldo); + + return 0; +} + +/* + * set dac bias + * common state changes: + * startup: + * off --> standby --> prepare --> on + * standby --> prepare --> on + * + * stop: + * on --> prepare --> standby + */ +static int sgtl5000_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + int ret; + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + ret = regulator_bulk_enable( + ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); + if (ret) + return ret; + udelay(10); + } + + break; + case SND_SOC_BIAS_OFF: + regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); + break; + } + + codec->dapm.bias_level = level; + return 0; +} + +#define SGTL5000_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE |\ + SNDRV_PCM_FMTBIT_S32_LE) + +struct snd_soc_dai_ops sgtl5000_ops = { + .hw_params = sgtl5000_pcm_hw_params, + .digital_mute = sgtl5000_digital_mute, + .set_fmt = sgtl5000_set_dai_fmt, + .set_sysclk = sgtl5000_set_dai_sysclk, +}; + +static struct snd_soc_dai_driver sgtl5000_dai = { + .name = "sgtl5000", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + /* + * only support 8~48K + 96K, + * TODO modify hw_param to support more + */ + .rates = SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_96000, + .formats = SGTL5000_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_96000, + .formats = SGTL5000_FORMATS, + }, + .ops = &sgtl5000_ops, + .symmetric_rates = 1, +}; + +static int sgtl5000_volatile_register(struct snd_soc_codec *codec, + unsigned int reg) +{ + switch (reg) { + case SGTL5000_CHIP_ID: + case SGTL5000_CHIP_ADCDAC_CTRL: + case SGTL5000_CHIP_ANA_STATUS: + return 1; + } + + return 0; +} + +#ifdef CONFIG_SUSPEND +static int sgtl5000_suspend(struct snd_soc_codec *codec, pm_message_t state) +{ + sgtl5000_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +/* + * restore all sgtl5000 registers, + * since a big hole between dap and regular registers, + * we will restore them respectively. + */ +static int sgtl5000_restore_regs(struct snd_soc_codec *codec) +{ + u16 *cache = codec->reg_cache; + int i; + int regular_regs = SGTL5000_CHIP_SHORT_CTRL >> 1; + + /* restore regular registers */ + for (i = 0; i < regular_regs; i++) { + int reg = i << 1; + + /* this regs depends on the others */ + if (reg == SGTL5000_CHIP_ANA_POWER || + reg == SGTL5000_CHIP_CLK_CTRL || + reg == SGTL5000_CHIP_LINREG_CTRL || + reg == SGTL5000_CHIP_LINE_OUT_CTRL || + reg == SGTL5000_CHIP_CLK_CTRL) + continue; + + snd_soc_write(codec, reg, cache[i]); + } + + /* restore dap registers */ + for (i = SGTL5000_DAP_REG_OFFSET >> 1; + i < SGTL5000_MAX_REG_OFFSET >> 1; i++) { + int reg = i << 1; + + snd_soc_write(codec, reg, cache[i]); + } + + /* + * restore power and other regs according + * to set_power() and set_clock() + */ + snd_soc_write(codec, SGTL5000_CHIP_LINREG_CTRL, + cache[SGTL5000_CHIP_LINREG_CTRL >> 1]); + + snd_soc_write(codec, SGTL5000_CHIP_ANA_POWER, + cache[SGTL5000_CHIP_ANA_POWER >> 1]); + + snd_soc_write(codec, SGTL5000_CHIP_CLK_CTRL, + cache[SGTL5000_CHIP_CLK_CTRL >> 1]); + + snd_soc_write(codec, SGTL5000_CHIP_REF_CTRL, + cache[SGTL5000_CHIP_REF_CTRL >> 1]); + + snd_soc_write(codec, SGTL5000_CHIP_LINE_OUT_CTRL, + cache[SGTL5000_CHIP_LINE_OUT_CTRL >> 1]); + return 0; +} + +static int sgtl5000_resume(struct snd_soc_codec *codec) +{ + /* Bring the codec back up to standby to enable regulators */ + sgtl5000_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + /* Restore registers by cached in memory */ + sgtl5000_restore_regs(codec); + return 0; +} +#else +#define sgtl5000_suspend NULL +#define sgtl5000_resume NULL +#endif /* CONFIG_SUSPEND */ + +/* + * sgtl5000 has 3 internal power supplies: + * 1. VAG, normally set to vdda/2 + * 2. chargepump, set to different value + * according to voltage of vdda and vddio + * 3. line out VAG, normally set to vddio/2 + * + * and should be set according to: + * 1. vddd provided by external or not + * 2. vdda and vddio voltage value. > 3.1v or not + * 3. chip revision >=0x11 or not. If >=0x11, not use external vddd. + */ +static int sgtl5000_set_power_regs(struct snd_soc_codec *codec) +{ + int vddd; + int vdda; + int vddio; + u16 ana_pwr; + u16 lreg_ctrl; + int vag; + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + + vdda = regulator_get_voltage(sgtl5000->supplies[VDDA].consumer); + vddio = regulator_get_voltage(sgtl5000->supplies[VDDIO].consumer); + vddd = regulator_get_voltage(sgtl5000->supplies[VDDD].consumer); + + vdda = vdda / 1000; + vddio = vddio / 1000; + vddd = vddd / 1000; + + if (vdda <= 0 || vddio <= 0 || vddd < 0) { + dev_err(codec->dev, "regulator voltage not set correctly\n"); + + return -EINVAL; + } + + /* according to datasheet, maximum voltage of supplies */ + if (vdda > 3600 || vddio > 3600 || vddd > 1980) { + dev_err(codec->dev, + "exceed max voltage vdda %dmv vddio %dma vddd %dma\n", + vdda, vddio, vddd); + + return -EINVAL; + } + + /* reset value */ + ana_pwr = snd_soc_read(codec, SGTL5000_CHIP_ANA_POWER); + ana_pwr |= SGTL5000_DAC_STEREO | + SGTL5000_ADC_STEREO | + SGTL5000_REFTOP_POWERUP; + lreg_ctrl = snd_soc_read(codec, SGTL5000_CHIP_LINREG_CTRL); + + if (vddio < 3100 && vdda < 3100) { + /* enable internal oscillator used for charge pump */ + snd_soc_update_bits(codec, SGTL5000_CHIP_CLK_TOP_CTRL, + SGTL5000_INT_OSC_EN, + SGTL5000_INT_OSC_EN); + /* Enable VDDC charge pump */ + ana_pwr |= SGTL5000_VDDC_CHRGPMP_POWERUP; + } else if (vddio >= 3100 && vdda >= 3100) { + /* + * if vddio and vddd > 3.1v, + * charge pump should be clean before set ana_pwr + */ + snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, + SGTL5000_VDDC_CHRGPMP_POWERUP, 0); + + /* VDDC use VDDIO rail */ + lreg_ctrl |= SGTL5000_VDDC_ASSN_OVRD; + lreg_ctrl |= SGTL5000_VDDC_MAN_ASSN_VDDIO << + SGTL5000_VDDC_MAN_ASSN_SHIFT; + } + + snd_soc_write(codec, SGTL5000_CHIP_LINREG_CTRL, lreg_ctrl); + + snd_soc_write(codec, SGTL5000_CHIP_ANA_POWER, ana_pwr); + + /* set voltage to register */ + snd_soc_update_bits(codec, SGTL5000_CHIP_LINREG_CTRL, + (0x1 << 4) - 1, 0x8); + + /* + * if vddd linear reg has been enabled, + * simple digital supply should be clear to get + * proper VDDD voltage. + */ + if (ana_pwr & SGTL5000_LINEREG_D_POWERUP) + snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, + SGTL5000_LINREG_SIMPLE_POWERUP, + 0); + else + snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, + SGTL5000_LINREG_SIMPLE_POWERUP | + SGTL5000_STARTUP_POWERUP, + 0); + + /* + * set ADC/DAC VAG to vdda / 2, + * should stay in range (0.8v, 1.575v) + */ + vag = vdda / 2; + if (vag <= SGTL5000_ANA_GND_BASE) + vag = 0; + else if (vag >= SGTL5000_ANA_GND_BASE + SGTL5000_ANA_GND_STP * + (SGTL5000_ANA_GND_MASK >> SGTL5000_ANA_GND_SHIFT)) + vag = SGTL5000_ANA_GND_MASK >> SGTL5000_ANA_GND_SHIFT; + else + vag = (vag - SGTL5000_ANA_GND_BASE) / SGTL5000_ANA_GND_STP; + + snd_soc_update_bits(codec, SGTL5000_CHIP_REF_CTRL, + vag << SGTL5000_ANA_GND_SHIFT, + vag << SGTL5000_ANA_GND_SHIFT); + + /* set line out VAG to vddio / 2, in range (0.8v, 1.675v) */ + vag = vddio / 2; + if (vag <= SGTL5000_LINE_OUT_GND_BASE) + vag = 0; + else if (vag >= SGTL5000_LINE_OUT_GND_BASE + + SGTL5000_LINE_OUT_GND_STP * SGTL5000_LINE_OUT_GND_MAX) + vag = SGTL5000_LINE_OUT_GND_MAX; + else + vag = (vag - SGTL5000_LINE_OUT_GND_BASE) / + SGTL5000_LINE_OUT_GND_STP; + + snd_soc_update_bits(codec, SGTL5000_CHIP_LINE_OUT_CTRL, + vag << SGTL5000_LINE_OUT_GND_SHIFT | + SGTL5000_LINE_OUT_CURRENT_360u << + SGTL5000_LINE_OUT_CURRENT_SHIFT, + vag << SGTL5000_LINE_OUT_GND_SHIFT | + SGTL5000_LINE_OUT_CURRENT_360u << + SGTL5000_LINE_OUT_CURRENT_SHIFT); + + return 0; +} + +static int sgtl5000_enable_regulators(struct snd_soc_codec *codec) +{ + u16 reg; + int ret; + int rev; + int i; + int external_vddd = 0; + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + + for (i = 0; i < ARRAY_SIZE(sgtl5000->supplies); i++) + sgtl5000->supplies[i].supply = supply_names[i]; + + ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); + if (!ret) + external_vddd = 1; + else { + /* set internal ldo to 1.2v */ + int voltage = LDO_VOLTAGE; + + ret = ldo_regulator_register(codec, &ldo_init_data, voltage); + if (ret) { + dev_err(codec->dev, + "Failed to register vddd internal supplies: %d\n", + ret); + return ret; + } + + sgtl5000->supplies[VDDD].supply = LDO_CONSUMER_NAME; + + ret = regulator_bulk_get(codec->dev, + ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); + + if (ret) { + ldo_regulator_remove(codec); + dev_err(codec->dev, + "Failed to request supplies: %d\n", ret); + + return ret; + } + } + + ret = regulator_bulk_enable(ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); + if (ret) + goto err_regulator_free; + + /* wait for all power rails bring up */ + udelay(10); + + /* read chip information */ + reg = snd_soc_read(codec, SGTL5000_CHIP_ID); + if (((reg & SGTL5000_PARTID_MASK) >> SGTL5000_PARTID_SHIFT) != + SGTL5000_PARTID_PART_ID) { + dev_err(codec->dev, + "Device with ID register %x is not a sgtl5000\n", reg); + ret = -ENODEV; + goto err_regulator_disable; + } + + rev = (reg & SGTL5000_REVID_MASK) >> SGTL5000_REVID_SHIFT; + dev_info(codec->dev, "sgtl5000 revision %d\n", rev); + + /* + * workaround for revision 0x11 and later, + * roll back to use internal LDO + */ + if (external_vddd && rev >= 0x11) { + int voltage = LDO_VOLTAGE; + /* disable all regulator first */ + regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); + /* free VDDD regulator */ + regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); + + ret = ldo_regulator_register(codec, &ldo_init_data, voltage); + if (ret) + return ret; + + sgtl5000->supplies[VDDD].supply = LDO_CONSUMER_NAME; + + ret = regulator_bulk_get(codec->dev, + ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); + if (ret) { + ldo_regulator_remove(codec); + dev_err(codec->dev, + "Failed to request supplies: %d\n", ret); + + return ret; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); + if (ret) + goto err_regulator_free; + + /* wait for all power rails bring up */ + udelay(10); + } + + return 0; + +err_regulator_disable: + regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); +err_regulator_free: + regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); + if (external_vddd) + ldo_regulator_remove(codec); + return ret; + +} + +static int sgtl5000_probe(struct snd_soc_codec *codec) +{ + int ret; + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + + /* setup i2c data ops */ + ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_I2C); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + ret = sgtl5000_enable_regulators(codec); + if (ret) + return ret; + + /* power up sgtl5000 */ + ret = sgtl5000_set_power_regs(codec); + if (ret) + goto err; + + /* enable small pop, introduce 400ms delay in turning off */ + snd_soc_update_bits(codec, SGTL5000_CHIP_REF_CTRL, + SGTL5000_SMALL_POP, + SGTL5000_SMALL_POP); + + /* disable short cut detector */ + snd_soc_write(codec, SGTL5000_CHIP_SHORT_CTRL, 0); + + /* + * set i2s as default input of sound switch + * TODO: add sound switch to control and dapm widge. + */ + snd_soc_write(codec, SGTL5000_CHIP_SSS_CTRL, + SGTL5000_DAC_SEL_I2S_IN << SGTL5000_DAC_SEL_SHIFT); + snd_soc_write(codec, SGTL5000_CHIP_DIG_POWER, + SGTL5000_ADC_EN | SGTL5000_DAC_EN); + + /* enable dac volume ramp by default */ + snd_soc_write(codec, SGTL5000_CHIP_ADCDAC_CTRL, + SGTL5000_DAC_VOL_RAMP_EN | + SGTL5000_DAC_MUTE_RIGHT | + SGTL5000_DAC_MUTE_LEFT); + + snd_soc_write(codec, SGTL5000_CHIP_PAD_STRENGTH, 0x015f); + + snd_soc_write(codec, SGTL5000_CHIP_ANA_CTRL, + SGTL5000_HP_ZCD_EN | + SGTL5000_ADC_ZCD_EN); + + snd_soc_write(codec, SGTL5000_CHIP_MIC_CTRL, 0); + + /* + * disable DAP + * TODO: + * Enable DAP in kcontrol and dapm. + */ + snd_soc_write(codec, SGTL5000_DAP_CTRL, 0); + + /* leading to standby state */ + ret = sgtl5000_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + if (ret) + goto err; + + snd_soc_add_controls(codec, sgtl5000_snd_controls, + ARRAY_SIZE(sgtl5000_snd_controls)); + + snd_soc_dapm_new_controls(&codec->dapm, sgtl5000_dapm_widgets, + ARRAY_SIZE(sgtl5000_dapm_widgets)); + + snd_soc_dapm_add_routes(&codec->dapm, audio_map, + ARRAY_SIZE(audio_map)); + + snd_soc_dapm_new_widgets(&codec->dapm); + + return 0; + +err: + regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); + regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); + ldo_regulator_remove(codec); + + return ret; +} + +static int sgtl5000_remove(struct snd_soc_codec *codec) +{ + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + + sgtl5000_set_bias_level(codec, SND_SOC_BIAS_OFF); + + regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); + regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); + ldo_regulator_remove(codec); + + return 0; +} + +struct snd_soc_codec_driver sgtl5000_driver = { + .probe = sgtl5000_probe, + .remove = sgtl5000_remove, + .suspend = sgtl5000_suspend, + .resume = sgtl5000_resume, + .set_bias_level = sgtl5000_set_bias_level, + .reg_cache_size = ARRAY_SIZE(sgtl5000_regs), + .reg_word_size = sizeof(u16), + .reg_cache_step = 2, + .reg_cache_default = sgtl5000_regs, + .volatile_register = sgtl5000_volatile_register, +}; + +static __devinit int sgtl5000_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct sgtl5000_priv *sgtl5000; + int ret; + + sgtl5000 = kzalloc(sizeof(struct sgtl5000_priv), GFP_KERNEL); + if (!sgtl5000) + return -ENOMEM; + + /* + * copy DAP default values to default value array. + * sgtl5000 register space has a big hole, merge it + * at init phase makes life easy. + * FIXME: should we drop 'const' of sgtl5000_regs? + */ + memcpy((void *)(&sgtl5000_regs[0] + (SGTL5000_DAP_REG_OFFSET >> 1)), + sgtl5000_dap_regs, + SGTL5000_MAX_REG_OFFSET - SGTL5000_DAP_REG_OFFSET); + + i2c_set_clientdata(client, sgtl5000); + + ret = snd_soc_register_codec(&client->dev, + &sgtl5000_driver, &sgtl5000_dai, 1); + if (ret) { + dev_err(&client->dev, "Failed to register codec: %d\n", ret); + kfree(sgtl5000); + return ret; + } + + return 0; +} + +static __devexit int sgtl5000_i2c_remove(struct i2c_client *client) +{ + struct sgtl5000_priv *sgtl5000 = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + + kfree(sgtl5000); + return 0; +} + +static const struct i2c_device_id sgtl5000_id[] = { + {"sgtl5000", 0}, + {}, +}; + +MODULE_DEVICE_TABLE(i2c, sgtl5000_id); + +static struct i2c_driver sgtl5000_i2c_driver = { + .driver = { + .name = "sgtl5000", + .owner = THIS_MODULE, + }, + .probe = sgtl5000_i2c_probe, + .remove = __devexit_p(sgtl5000_i2c_remove), + .id_table = sgtl5000_id, +}; + +static int __init sgtl5000_modinit(void) +{ + return i2c_add_driver(&sgtl5000_i2c_driver); +} +module_init(sgtl5000_modinit); + +static void __exit sgtl5000_exit(void) +{ + i2c_del_driver(&sgtl5000_i2c_driver); +} +module_exit(sgtl5000_exit); + +MODULE_DESCRIPTION("Freescale SGTL5000 ALSA SoC Codec Driver"); +MODULE_AUTHOR("Zeng Zhaoming "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/sgtl5000.h b/sound/soc/codecs/sgtl5000.h new file mode 100644 index 00000000000..c29312ab801 --- /dev/null +++ b/sound/soc/codecs/sgtl5000.h @@ -0,0 +1,402 @@ +/* + * sgtl5000.h - SGTL5000 audio codec interface + * + * Copyright 2010-2011 Freescale Semiconductor, Inc. + * + * 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 _SGTL5000_H +#define _SGTL5000_H + +#include + +/* + * Register values. + */ +#define SGTL5000_CHIP_ID 0x0000 +#define SGTL5000_CHIP_DIG_POWER 0x0002 +#define SGTL5000_CHIP_CLK_CTRL 0x0004 +#define SGTL5000_CHIP_I2S_CTRL 0x0006 +#define SGTL5000_CHIP_SSS_CTRL 0x000a +#define SGTL5000_CHIP_ADCDAC_CTRL 0x000e +#define SGTL5000_CHIP_DAC_VOL 0x0010 +#define SGTL5000_CHIP_PAD_STRENGTH 0x0014 +#define SGTL5000_CHIP_ANA_ADC_CTRL 0x0020 +#define SGTL5000_CHIP_ANA_HP_CTRL 0x0022 +#define SGTL5000_CHIP_ANA_CTRL 0x0024 +#define SGTL5000_CHIP_LINREG_CTRL 0x0026 +#define SGTL5000_CHIP_REF_CTRL 0x0028 +#define SGTL5000_CHIP_MIC_CTRL 0x002a +#define SGTL5000_CHIP_LINE_OUT_CTRL 0x002c +#define SGTL5000_CHIP_LINE_OUT_VOL 0x002e +#define SGTL5000_CHIP_ANA_POWER 0x0030 +#define SGTL5000_CHIP_PLL_CTRL 0x0032 +#define SGTL5000_CHIP_CLK_TOP_CTRL 0x0034 +#define SGTL5000_CHIP_ANA_STATUS 0x0036 +#define SGTL5000_CHIP_SHORT_CTRL 0x003c +#define SGTL5000_CHIP_ANA_TEST2 0x003a +#define SGTL5000_DAP_CTRL 0x0100 +#define SGTL5000_DAP_PEQ 0x0102 +#define SGTL5000_DAP_BASS_ENHANCE 0x0104 +#define SGTL5000_DAP_BASS_ENHANCE_CTRL 0x0106 +#define SGTL5000_DAP_AUDIO_EQ 0x0108 +#define SGTL5000_DAP_SURROUND 0x010a +#define SGTL5000_DAP_FLT_COEF_ACCESS 0x010c +#define SGTL5000_DAP_COEF_WR_B0_MSB 0x010e +#define SGTL5000_DAP_COEF_WR_B0_LSB 0x0110 +#define SGTL5000_DAP_EQ_BASS_BAND0 0x0116 +#define SGTL5000_DAP_EQ_BASS_BAND1 0x0118 +#define SGTL5000_DAP_EQ_BASS_BAND2 0x011a +#define SGTL5000_DAP_EQ_BASS_BAND3 0x011c +#define SGTL5000_DAP_EQ_BASS_BAND4 0x011e +#define SGTL5000_DAP_MAIN_CHAN 0x0120 +#define SGTL5000_DAP_MIX_CHAN 0x0122 +#define SGTL5000_DAP_AVC_CTRL 0x0124 +#define SGTL5000_DAP_AVC_THRESHOLD 0x0126 +#define SGTL5000_DAP_AVC_ATTACK 0x0128 +#define SGTL5000_DAP_AVC_DECAY 0x012a +#define SGTL5000_DAP_COEF_WR_B1_MSB 0x012c +#define SGTL5000_DAP_COEF_WR_B1_LSB 0x012e +#define SGTL5000_DAP_COEF_WR_B2_MSB 0x0130 +#define SGTL5000_DAP_COEF_WR_B2_LSB 0x0132 +#define SGTL5000_DAP_COEF_WR_A1_MSB 0x0134 +#define SGTL5000_DAP_COEF_WR_A1_LSB 0x0136 +#define SGTL5000_DAP_COEF_WR_A2_MSB 0x0138 +#define SGTL5000_DAP_COEF_WR_A2_LSB 0x013a + +/* + * Field Definitions. + */ + +/* + * SGTL5000_CHIP_ID + */ +#define SGTL5000_PARTID_MASK 0xff00 +#define SGTL5000_PARTID_SHIFT 8 +#define SGTL5000_PARTID_WIDTH 8 +#define SGTL5000_PARTID_PART_ID 0xa0 +#define SGTL5000_REVID_MASK 0x00ff +#define SGTL5000_REVID_SHIFT 0 +#define SGTL5000_REVID_WIDTH 8 + +/* + * SGTL5000_CHIP_DIG_POWER + */ +#define SGTL5000_ADC_EN 0x0040 +#define SGTL5000_DAC_EN 0x0020 +#define SGTL5000_DAP_POWERUP 0x0010 +#define SGTL5000_I2S_OUT_POWERUP 0x0002 +#define SGTL5000_I2S_IN_POWERUP 0x0001 + +/* + * SGTL5000_CHIP_CLK_CTRL + */ +#define SGTL5000_RATE_MODE_MASK 0x0030 +#define SGTL5000_RATE_MODE_SHIFT 4 +#define SGTL5000_RATE_MODE_WIDTH 2 +#define SGTL5000_RATE_MODE_DIV_1 0 +#define SGTL5000_RATE_MODE_DIV_2 1 +#define SGTL5000_RATE_MODE_DIV_4 2 +#define SGTL5000_RATE_MODE_DIV_6 3 +#define SGTL5000_SYS_FS_MASK 0x000c +#define SGTL5000_SYS_FS_SHIFT 2 +#define SGTL5000_SYS_FS_WIDTH 2 +#define SGTL5000_SYS_FS_32k 0x0 +#define SGTL5000_SYS_FS_44_1k 0x1 +#define SGTL5000_SYS_FS_48k 0x2 +#define SGTL5000_SYS_FS_96k 0x3 +#define SGTL5000_MCLK_FREQ_MASK 0x0003 +#define SGTL5000_MCLK_FREQ_SHIFT 0 +#define SGTL5000_MCLK_FREQ_WIDTH 2 +#define SGTL5000_MCLK_FREQ_256FS 0x0 +#define SGTL5000_MCLK_FREQ_384FS 0x1 +#define SGTL5000_MCLK_FREQ_512FS 0x2 +#define SGTL5000_MCLK_FREQ_PLL 0x3 + +/* + * SGTL5000_CHIP_I2S_CTRL + */ +#define SGTL5000_I2S_SCLKFREQ_MASK 0x0100 +#define SGTL5000_I2S_SCLKFREQ_SHIFT 8 +#define SGTL5000_I2S_SCLKFREQ_WIDTH 1 +#define SGTL5000_I2S_SCLKFREQ_64FS 0x0 +#define SGTL5000_I2S_SCLKFREQ_32FS 0x1 /* Not for RJ mode */ +#define SGTL5000_I2S_MASTER 0x0080 +#define SGTL5000_I2S_SCLK_INV 0x0040 +#define SGTL5000_I2S_DLEN_MASK 0x0030 +#define SGTL5000_I2S_DLEN_SHIFT 4 +#define SGTL5000_I2S_DLEN_WIDTH 2 +#define SGTL5000_I2S_DLEN_32 0x0 +#define SGTL5000_I2S_DLEN_24 0x1 +#define SGTL5000_I2S_DLEN_20 0x2 +#define SGTL5000_I2S_DLEN_16 0x3 +#define SGTL5000_I2S_MODE_MASK 0x000c +#define SGTL5000_I2S_MODE_SHIFT 2 +#define SGTL5000_I2S_MODE_WIDTH 2 +#define SGTL5000_I2S_MODE_I2S_LJ 0x0 +#define SGTL5000_I2S_MODE_RJ 0x1 +#define SGTL5000_I2S_MODE_PCM 0x2 +#define SGTL5000_I2S_LRALIGN 0x0002 +#define SGTL5000_I2S_LRPOL 0x0001 /* set for which mode */ + +/* + * SGTL5000_CHIP_SSS_CTRL + */ +#define SGTL5000_DAP_MIX_LRSWAP 0x4000 +#define SGTL5000_DAP_LRSWAP 0x2000 +#define SGTL5000_DAC_LRSWAP 0x1000 +#define SGTL5000_I2S_OUT_LRSWAP 0x0400 +#define SGTL5000_DAP_MIX_SEL_MASK 0x0300 +#define SGTL5000_DAP_MIX_SEL_SHIFT 8 +#define SGTL5000_DAP_MIX_SEL_WIDTH 2 +#define SGTL5000_DAP_MIX_SEL_ADC 0x0 +#define SGTL5000_DAP_MIX_SEL_I2S_IN 0x1 +#define SGTL5000_DAP_SEL_MASK 0x00c0 +#define SGTL5000_DAP_SEL_SHIFT 6 +#define SGTL5000_DAP_SEL_WIDTH 2 +#define SGTL5000_DAP_SEL_ADC 0x0 +#define SGTL5000_DAP_SEL_I2S_IN 0x1 +#define SGTL5000_DAC_SEL_MASK 0x0030 +#define SGTL5000_DAC_SEL_SHIFT 4 +#define SGTL5000_DAC_SEL_WIDTH 2 +#define SGTL5000_DAC_SEL_ADC 0x0 +#define SGTL5000_DAC_SEL_I2S_IN 0x1 +#define SGTL5000_DAC_SEL_DAP 0x3 +#define SGTL5000_I2S_OUT_SEL_MASK 0x0003 +#define SGTL5000_I2S_OUT_SEL_SHIFT 0 +#define SGTL5000_I2S_OUT_SEL_WIDTH 2 +#define SGTL5000_I2S_OUT_SEL_ADC 0x0 +#define SGTL5000_I2S_OUT_SEL_I2S_IN 0x1 +#define SGTL5000_I2S_OUT_SEL_DAP 0x3 + +/* + * SGTL5000_CHIP_ADCDAC_CTRL + */ +#define SGTL5000_VOL_BUSY_DAC_RIGHT 0x2000 +#define SGTL5000_VOL_BUSY_DAC_LEFT 0x1000 +#define SGTL5000_DAC_VOL_RAMP_EN 0x0200 +#define SGTL5000_DAC_VOL_RAMP_EXPO 0x0100 +#define SGTL5000_DAC_MUTE_RIGHT 0x0008 +#define SGTL5000_DAC_MUTE_LEFT 0x0004 +#define SGTL5000_ADC_HPF_FREEZE 0x0002 +#define SGTL5000_ADC_HPF_BYPASS 0x0001 + +/* + * SGTL5000_CHIP_DAC_VOL + */ +#define SGTL5000_DAC_VOL_RIGHT_MASK 0xff00 +#define SGTL5000_DAC_VOL_RIGHT_SHIFT 8 +#define SGTL5000_DAC_VOL_RIGHT_WIDTH 8 +#define SGTL5000_DAC_VOL_LEFT_MASK 0x00ff +#define SGTL5000_DAC_VOL_LEFT_SHIFT 0 +#define SGTL5000_DAC_VOL_LEFT_WIDTH 8 + +/* + * SGTL5000_CHIP_PAD_STRENGTH + */ +#define SGTL5000_PAD_I2S_LRCLK_MASK 0x0300 +#define SGTL5000_PAD_I2S_LRCLK_SHIFT 8 +#define SGTL5000_PAD_I2S_LRCLK_WIDTH 2 +#define SGTL5000_PAD_I2S_SCLK_MASK 0x00c0 +#define SGTL5000_PAD_I2S_SCLK_SHIFT 6 +#define SGTL5000_PAD_I2S_SCLK_WIDTH 2 +#define SGTL5000_PAD_I2S_DOUT_MASK 0x0030 +#define SGTL5000_PAD_I2S_DOUT_SHIFT 4 +#define SGTL5000_PAD_I2S_DOUT_WIDTH 2 +#define SGTL5000_PAD_I2C_SDA_MASK 0x000c +#define SGTL5000_PAD_I2C_SDA_SHIFT 2 +#define SGTL5000_PAD_I2C_SDA_WIDTH 2 +#define SGTL5000_PAD_I2C_SCL_MASK 0x0003 +#define SGTL5000_PAD_I2C_SCL_SHIFT 0 +#define SGTL5000_PAD_I2C_SCL_WIDTH 2 + +/* + * SGTL5000_CHIP_ANA_ADC_CTRL + */ +#define SGTL5000_ADC_VOL_M6DB 0x0100 +#define SGTL5000_ADC_VOL_RIGHT_MASK 0x00f0 +#define SGTL5000_ADC_VOL_RIGHT_SHIFT 4 +#define SGTL5000_ADC_VOL_RIGHT_WIDTH 4 +#define SGTL5000_ADC_VOL_LEFT_MASK 0x000f +#define SGTL5000_ADC_VOL_LEFT_SHIFT 0 +#define SGTL5000_ADC_VOL_LEFT_WIDTH 4 + +/* + * SGTL5000_CHIP_ANA_HP_CTRL + */ +#define SGTL5000_HP_VOL_RIGHT_MASK 0x7f00 +#define SGTL5000_HP_VOL_RIGHT_SHIFT 8 +#define SGTL5000_HP_VOL_RIGHT_WIDTH 7 +#define SGTL5000_HP_VOL_LEFT_MASK 0x007f +#define SGTL5000_HP_VOL_LEFT_SHIFT 0 +#define SGTL5000_HP_VOL_LEFT_WIDTH 7 + +/* + * SGTL5000_CHIP_ANA_CTRL + */ +#define SGTL5000_LINE_OUT_MUTE 0x0100 +#define SGTL5000_HP_SEL_MASK 0x0040 +#define SGTL5000_HP_SEL_SHIFT 6 +#define SGTL5000_HP_SEL_WIDTH 1 +#define SGTL5000_HP_SEL_DAC 0x0 +#define SGTL5000_HP_SEL_LINE_IN 0x1 +#define SGTL5000_HP_ZCD_EN 0x0020 +#define SGTL5000_HP_MUTE 0x0010 +#define SGTL5000_ADC_SEL_MASK 0x0004 +#define SGTL5000_ADC_SEL_SHIFT 2 +#define SGTL5000_ADC_SEL_WIDTH 1 +#define SGTL5000_ADC_SEL_MIC 0x0 +#define SGTL5000_ADC_SEL_LINE_IN 0x1 +#define SGTL5000_ADC_ZCD_EN 0x0002 +#define SGTL5000_ADC_MUTE 0x0001 + +/* + * SGTL5000_CHIP_LINREG_CTRL + */ +#define SGTL5000_VDDC_MAN_ASSN_MASK 0x0040 +#define SGTL5000_VDDC_MAN_ASSN_SHIFT 6 +#define SGTL5000_VDDC_MAN_ASSN_WIDTH 1 +#define SGTL5000_VDDC_MAN_ASSN_VDDA 0x0 +#define SGTL5000_VDDC_MAN_ASSN_VDDIO 0x1 +#define SGTL5000_VDDC_ASSN_OVRD 0x0020 +#define SGTL5000_LINREG_VDDD_MASK 0x000f +#define SGTL5000_LINREG_VDDD_SHIFT 0 +#define SGTL5000_LINREG_VDDD_WIDTH 4 + +/* + * SGTL5000_CHIP_REF_CTRL + */ +#define SGTL5000_ANA_GND_MASK 0x01f0 +#define SGTL5000_ANA_GND_SHIFT 4 +#define SGTL5000_ANA_GND_WIDTH 5 +#define SGTL5000_ANA_GND_BASE 800 /* mv */ +#define SGTL5000_ANA_GND_STP 25 /*mv */ +#define SGTL5000_BIAS_CTRL_MASK 0x000e +#define SGTL5000_BIAS_CTRL_SHIFT 1 +#define SGTL5000_BIAS_CTRL_WIDTH 3 +#define SGTL5000_SMALL_POP 0x0001 + +/* + * SGTL5000_CHIP_MIC_CTRL + */ +#define SGTL5000_BIAS_R_MASK 0x0200 +#define SGTL5000_BIAS_R_SHIFT 8 +#define SGTL5000_BIAS_R_WIDTH 2 +#define SGTL5000_BIAS_R_off 0x0 +#define SGTL5000_BIAS_R_2K 0x1 +#define SGTL5000_BIAS_R_4k 0x2 +#define SGTL5000_BIAS_R_8k 0x3 +#define SGTL5000_BIAS_VOLT_MASK 0x0070 +#define SGTL5000_BIAS_VOLT_SHIFT 4 +#define SGTL5000_BIAS_VOLT_WIDTH 3 +#define SGTL5000_MIC_GAIN_MASK 0x0003 +#define SGTL5000_MIC_GAIN_SHIFT 0 +#define SGTL5000_MIC_GAIN_WIDTH 2 + +/* + * SGTL5000_CHIP_LINE_OUT_CTRL + */ +#define SGTL5000_LINE_OUT_CURRENT_MASK 0x0f00 +#define SGTL5000_LINE_OUT_CURRENT_SHIFT 8 +#define SGTL5000_LINE_OUT_CURRENT_WIDTH 4 +#define SGTL5000_LINE_OUT_CURRENT_180u 0x0 +#define SGTL5000_LINE_OUT_CURRENT_270u 0x1 +#define SGTL5000_LINE_OUT_CURRENT_360u 0x3 +#define SGTL5000_LINE_OUT_CURRENT_450u 0x7 +#define SGTL5000_LINE_OUT_CURRENT_540u 0xf +#define SGTL5000_LINE_OUT_GND_MASK 0x003f +#define SGTL5000_LINE_OUT_GND_SHIFT 0 +#define SGTL5000_LINE_OUT_GND_WIDTH 6 +#define SGTL5000_LINE_OUT_GND_BASE 800 /* mv */ +#define SGTL5000_LINE_OUT_GND_STP 25 +#define SGTL5000_LINE_OUT_GND_MAX 0x23 + +/* + * SGTL5000_CHIP_LINE_OUT_VOL + */ +#define SGTL5000_LINE_OUT_VOL_RIGHT_MASK 0x1f00 +#define SGTL5000_LINE_OUT_VOL_RIGHT_SHIFT 8 +#define SGTL5000_LINE_OUT_VOL_RIGHT_WIDTH 5 +#define SGTL5000_LINE_OUT_VOL_LEFT_MASK 0x001f +#define SGTL5000_LINE_OUT_VOL_LEFT_SHIFT 0 +#define SGTL5000_LINE_OUT_VOL_LEFT_WIDTH 5 + +/* + * SGTL5000_CHIP_ANA_POWER + */ +#define SGTL5000_DAC_STEREO 0x4000 +#define SGTL5000_LINREG_SIMPLE_POWERUP 0x2000 +#define SGTL5000_STARTUP_POWERUP 0x1000 +#define SGTL5000_VDDC_CHRGPMP_POWERUP 0x0800 +#define SGTL5000_PLL_POWERUP 0x0400 +#define SGTL5000_LINEREG_D_POWERUP 0x0200 +#define SGTL5000_VCOAMP_POWERUP 0x0100 +#define SGTL5000_VAG_POWERUP 0x0080 +#define SGTL5000_ADC_STEREO 0x0040 +#define SGTL5000_REFTOP_POWERUP 0x0020 +#define SGTL5000_HP_POWERUP 0x0010 +#define SGTL5000_DAC_POWERUP 0x0008 +#define SGTL5000_CAPLESS_HP_POWERUP 0x0004 +#define SGTL5000_ADC_POWERUP 0x0002 +#define SGTL5000_LINE_OUT_POWERUP 0x0001 + +/* + * SGTL5000_CHIP_PLL_CTRL + */ +#define SGTL5000_PLL_INT_DIV_MASK 0xf800 +#define SGTL5000_PLL_INT_DIV_SHIFT 11 +#define SGTL5000_PLL_INT_DIV_WIDTH 5 +#define SGTL5000_PLL_FRAC_DIV_MASK 0x0700 +#define SGTL5000_PLL_FRAC_DIV_SHIFT 0 +#define SGTL5000_PLL_FRAC_DIV_WIDTH 11 + +/* + * SGTL5000_CHIP_CLK_TOP_CTRL + */ +#define SGTL5000_INT_OSC_EN 0x0800 +#define SGTL5000_INPUT_FREQ_DIV2 0x0008 + +/* + * SGTL5000_CHIP_ANA_STATUS + */ +#define SGTL5000_HP_LRSHORT 0x0200 +#define SGTL5000_CAPLESS_SHORT 0x0100 +#define SGTL5000_PLL_LOCKED 0x0010 + +/* + * SGTL5000_CHIP_SHORT_CTRL + */ +#define SGTL5000_LVLADJR_MASK 0x7000 +#define SGTL5000_LVLADJR_SHIFT 12 +#define SGTL5000_LVLADJR_WIDTH 3 +#define SGTL5000_LVLADJL_MASK 0x0700 +#define SGTL5000_LVLADJL_SHIFT 8 +#define SGTL5000_LVLADJL_WIDTH 3 +#define SGTL5000_LVLADJC_MASK 0x0070 +#define SGTL5000_LVLADJC_SHIFT 4 +#define SGTL5000_LVLADJC_WIDTH 3 +#define SGTL5000_LR_SHORT_MOD_MASK 0x000c +#define SGTL5000_LR_SHORT_MOD_SHIFT 2 +#define SGTL5000_LR_SHORT_MOD_WIDTH 2 +#define SGTL5000_CM_SHORT_MOD_MASK 0x0003 +#define SGTL5000_CM_SHORT_MOD_SHIFT 0 +#define SGTL5000_CM_SHORT_MOD_WIDTH 2 + +/* + *SGTL5000_CHIP_ANA_TEST2 + */ +#define SGTL5000_MONO_DAC 0x1000 + +/* + * SGTL5000_DAP_CTRL + */ +#define SGTL5000_DAP_MIX_EN 0x0010 +#define SGTL5000_DAP_EN 0x0001 + +#define SGTL5000_SYSCLK 0x00 +#define SGTL5000_LRCLK 0x01 + +#endif -- cgit v1.2.3-70-g09d2 From b3111a9aa8823e360f20e3ed9fb106d757b89704 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 25 Feb 2011 12:25:18 +0000 Subject: ASoC: Move WM2000 to dev_pm_ops There's a general move to use dev_pm_ops rather than bus specific functions in order to facilitate work on the PM core. Do this conversion to WM2000. The driver ought to be updated to work better in a multi-component model but the mechanical conversion ensures that we avoid blocking PM core work until that happens. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm2000.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index 80ddf4fd23d..a3b9cbb20ee 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -836,24 +836,25 @@ static void wm2000_i2c_shutdown(struct i2c_client *i2c) } #ifdef CONFIG_PM -static int wm2000_i2c_suspend(struct i2c_client *i2c, pm_message_t mesg) +static int wm2000_i2c_suspend(struct device *dev) { + struct i2c_client *i2c = to_i2c_client(dev); struct wm2000_priv *wm2000 = dev_get_drvdata(&i2c->dev); return wm2000_anc_transition(wm2000, ANC_OFF); } -static int wm2000_i2c_resume(struct i2c_client *i2c) +static int wm2000_i2c_resume(struct device *dev) { + struct i2c_client *i2c = to_i2c_client(dev); struct wm2000_priv *wm2000 = dev_get_drvdata(&i2c->dev); return wm2000_anc_set_mode(wm2000); } -#else -#define wm2000_i2c_suspend NULL -#define wm2000_i2c_resume NULL #endif +static SIMPLE_DEV_PM_OPS(wm2000_pm, wm2000_i2c_suspend, wm2000_i2c_resume); + static const struct i2c_device_id wm2000_i2c_id[] = { { "wm2000", 0 }, { } @@ -864,11 +865,10 @@ static struct i2c_driver wm2000_i2c_driver = { .driver = { .name = "wm2000", .owner = THIS_MODULE, + .pm = &wm2000_pm, }, .probe = wm2000_i2c_probe, .remove = __devexit_p(wm2000_i2c_remove), - .suspend = wm2000_i2c_suspend, - .resume = wm2000_i2c_resume, .shutdown = wm2000_i2c_shutdown, .id_table = wm2000_i2c_id, }; -- cgit v1.2.3-70-g09d2 From 5f83df9a6192b3197a58beb66908b3732cb5a670 Mon Sep 17 00:00:00 2001 From: Zeng Zhaoming Date: Mon, 28 Feb 2011 03:45:21 +0800 Subject: ASoC: remove unnecessary header including in SGTL5000 codec driver Remove unnecessary headers: - mach/hardware.h in sgtl5000.c - linux/i2c.h in sgtl5000.h Signed-off-by: Zeng Zhaoming Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 1 - sound/soc/codecs/sgtl5000.h | 2 -- 2 files changed, 3 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 9eb206e852d..b528f971f3f 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -27,7 +27,6 @@ #include #include #include -#include #include "sgtl5000.h" diff --git a/sound/soc/codecs/sgtl5000.h b/sound/soc/codecs/sgtl5000.h index c29312ab801..eec3ab368f3 100644 --- a/sound/soc/codecs/sgtl5000.h +++ b/sound/soc/codecs/sgtl5000.h @@ -11,8 +11,6 @@ #ifndef _SGTL5000_H #define _SGTL5000_H -#include - /* * Register values. */ -- cgit v1.2.3-70-g09d2 From 61a142b7e4b5c4cce1b4ea52a829984959120089 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 28 Feb 2011 14:33:01 +0000 Subject: ASoC: Staticise non-exported symbols in SGTL5000 Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index b528f971f3f..b7e97c02689 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -124,7 +124,7 @@ static struct regulator_consumer_supply ldo_consumer[] = { REGULATOR_SUPPLY(LDO_CONSUMER_NAME, NULL), }; -struct regulator_init_data ldo_init_data = { +static struct regulator_init_data ldo_init_data = { .constraints = { .min_uV = 850000, .max_uV = 1600000, @@ -946,7 +946,7 @@ static int sgtl5000_set_bias_level(struct snd_soc_codec *codec, SNDRV_PCM_FMTBIT_S24_LE |\ SNDRV_PCM_FMTBIT_S32_LE) -struct snd_soc_dai_ops sgtl5000_ops = { +static struct snd_soc_dai_ops sgtl5000_ops = { .hw_params = sgtl5000_pcm_hw_params, .digital_mute = sgtl5000_digital_mute, .set_fmt = sgtl5000_set_dai_fmt, @@ -1421,7 +1421,7 @@ static int sgtl5000_remove(struct snd_soc_codec *codec) return 0; } -struct snd_soc_codec_driver sgtl5000_driver = { +static struct snd_soc_codec_driver sgtl5000_driver = { .probe = sgtl5000_probe, .remove = sgtl5000_remove, .suspend = sgtl5000_suspend, -- cgit v1.2.3-70-g09d2 From b462c6e69a26dd534d6372ed65a6fc7c01073883 Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Tue, 1 Mar 2011 12:54:39 +0000 Subject: ASoC: WM8994: Ensure MICBIAS is provided with a clock The patch 'ASoC: WM8994: Improve Playback Robustness' did not handle this case properly. Signed-off-by: Dimitris Papastamos Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index b17ac1971b0..125bfb6eb24 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1102,6 +1102,13 @@ static int adc_mux_ev(struct snd_soc_dapm_widget *w, return 0; } +static int micbias_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + late_enable_ev(w, kcontrol, event); + return 0; +} + static int dac_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -1439,6 +1446,10 @@ SND_SOC_DAPM_INPUT("DMIC1DAT"), SND_SOC_DAPM_INPUT("DMIC2DAT"), SND_SOC_DAPM_INPUT("Clock"), +SND_SOC_DAPM_MICBIAS("MICBIAS", WM8994_MICBIAS, 2, 0), +SND_SOC_DAPM_SUPPLY_S("MICBIAS Supply", 1, SND_SOC_NOPM, 0, 0, micbias_ev, + SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_SUPPLY("CLK_SYS", SND_SOC_NOPM, 0, 0, clk_sys_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), @@ -1754,6 +1765,8 @@ static const struct snd_soc_dapm_route wm8994_revd_intercon[] = { { "AIF2DACDAT", NULL, "AIF1DACDAT" }, { "AIF1ADCDAT", NULL, "AIF2ADCDAT" }, { "AIF2ADCDAT", NULL, "AIF1ADCDAT" }, + { "MICBIAS", NULL, "CLK_SYS" }, + { "MICBIAS", NULL, "MICBIAS Supply" }, }; static const struct snd_soc_dapm_route wm8994_intercon[] = { -- cgit v1.2.3-70-g09d2 From 49542656ade68b4d4952feec6a4d508fd32be6f1 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 24 Feb 2011 20:25:45 +0000 Subject: ASoC: Remove module probe announcements from CODEC drivers Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/ak4104.c | 1 - sound/soc/codecs/cs4270.c | 2 -- 2 files changed, 3 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c index c27f8f59dc6..cbf0b6d400b 100644 --- a/sound/soc/codecs/ak4104.c +++ b/sound/soc/codecs/ak4104.c @@ -294,7 +294,6 @@ static struct spi_driver ak4104_spi_driver = { static int __init ak4104_init(void) { - pr_info("Asahi Kasei AK4104 ALSA SoC Codec Driver\n"); return spi_register_driver(&ak4104_spi_driver); } module_init(ak4104_init); diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index c0fccadaea9..65f578ff611 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -743,8 +743,6 @@ static struct i2c_driver cs4270_i2c_driver = { static int __init cs4270_init(void) { - pr_info("Cirrus Logic CS4270 ALSA SoC Codec Driver\n"); - return i2c_add_driver(&cs4270_i2c_driver); } module_init(cs4270_init); -- cgit v1.2.3-70-g09d2 From 4a5f7bda8fe9d0ed08ed4c5beb5dc3fa62f09d05 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 1 Mar 2011 20:10:46 +0000 Subject: ASoC: Add platform data for WM9081 IRQ pin configuration The WM9081 IRQ output can be either active high or active low and can support either CMOS or open drain modes. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- include/sound/wm9081.h | 9 ++++++--- sound/soc/codecs/wm9081.c | 29 +++++++++++++++++++---------- 2 files changed, 25 insertions(+), 13 deletions(-) (limited to 'sound/soc/codecs') diff --git a/include/sound/wm9081.h b/include/sound/wm9081.h index e173ddbf6bd..f34b0b1716d 100644 --- a/include/sound/wm9081.h +++ b/include/sound/wm9081.h @@ -17,9 +17,12 @@ struct wm9081_retune_mobile_setting { u16 config[20]; }; -struct wm9081_retune_mobile_config { - struct wm9081_retune_mobile_setting *configs; - int num_configs; +struct wm9081_pdata { + bool irq_high; /* IRQ is active high */ + bool irq_cmos; /* IRQ is in CMOS mode */ + + struct wm9081_retune_mobile_setting *retune_configs; + int num_retune_configs; }; #endif diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 2103623a077..7883f3ed797 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -167,7 +167,7 @@ struct wm9081_priv { int fll_fref; int fll_fout; int tdm_width; - struct wm9081_retune_mobile_config *retune; + struct wm9081_pdata pdata; }; static int wm9081_volatile_register(struct snd_soc_codec *codec, unsigned int reg) @@ -1082,21 +1082,22 @@ static int wm9081_hw_params(struct snd_pcm_substream *substream, aif4 |= wm9081->bclk / wm9081->fs; /* Apply a ReTune Mobile configuration if it's in use */ - if (wm9081->retune) { - struct wm9081_retune_mobile_config *retune = wm9081->retune; + if (wm9081->pdata.num_retune_configs) { + struct wm9081_pdata *pdata = &wm9081->pdata; struct wm9081_retune_mobile_setting *s; int eq1; best = 0; - best_val = abs(retune->configs[0].rate - wm9081->fs); - for (i = 0; i < retune->num_configs; i++) { - cur_val = abs(retune->configs[i].rate - wm9081->fs); + best_val = abs(pdata->retune_configs[0].rate - wm9081->fs); + for (i = 0; i < pdata->num_retune_configs; i++) { + cur_val = abs(pdata->retune_configs[i].rate - + wm9081->fs); if (cur_val < best_val) { best_val = cur_val; best = i; } } - s = &retune->configs[best]; + s = &pdata->retune_configs[best]; dev_dbg(codec->dev, "ReTune Mobile %s tuned for %dHz\n", s->name, s->rate); @@ -1255,6 +1256,14 @@ static int wm9081_probe(struct snd_soc_codec *codec) return ret; } + reg = 0; + if (wm9081->pdata.irq_high) + reg |= WM9081_IRQ_POL; + if (!wm9081->pdata.irq_cmos) + reg |= WM9081_IRQ_OP_CTRL; + snd_soc_update_bits(codec, WM9081_INTERRUPT_CONTROL, + WM9081_IRQ_POL | WM9081_IRQ_OP_CTRL, reg); + wm9081_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* Enable zero cross by default */ @@ -1266,7 +1275,7 @@ static int wm9081_probe(struct snd_soc_codec *codec) snd_soc_add_controls(codec, wm9081_snd_controls, ARRAY_SIZE(wm9081_snd_controls)); - if (!wm9081->retune) { + if (!wm9081->pdata.num_retune_configs) { dev_dbg(codec->dev, "No ReTune Mobile data, using normal EQ\n"); snd_soc_add_controls(codec, wm9081_eq_controls, @@ -1343,8 +1352,8 @@ static __devinit int wm9081_i2c_probe(struct i2c_client *i2c, wm9081->control_data = i2c; if (dev_get_platdata(&i2c->dev)) - memcpy(&wm9081->retune, dev_get_platdata(&i2c->dev), - sizeof(wm9081->retune)); + memcpy(&wm9081->pdata, dev_get_platdata(&i2c->dev), + sizeof(wm9081->pdata)); ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm9081, &wm9081_dai, 1); -- cgit v1.2.3-70-g09d2 From 8959c910884e8faf7987391d194d508e74904c16 Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Tue, 1 Mar 2011 12:54:39 +0000 Subject: ASoC: WM8994: Ensure MICBIAS is provided with a clock The patch 'ASoC: WM8994: Improve Playback Robustness' did not handle this case properly. Signed-off-by: Dimitris Papastamos Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 4afbe3b2e44..38bfff7d209 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1103,6 +1103,13 @@ static int adc_mux_ev(struct snd_soc_dapm_widget *w, return 0; } +static int micbias_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + late_enable_ev(w, kcontrol, event); + return 0; +} + static int dac_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -1440,6 +1447,10 @@ SND_SOC_DAPM_INPUT("DMIC1DAT"), SND_SOC_DAPM_INPUT("DMIC2DAT"), SND_SOC_DAPM_INPUT("Clock"), +SND_SOC_DAPM_MICBIAS("MICBIAS", WM8994_MICBIAS, 2, 0), +SND_SOC_DAPM_SUPPLY_S("MICBIAS Supply", 1, SND_SOC_NOPM, 0, 0, micbias_ev, + SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_SUPPLY("CLK_SYS", SND_SOC_NOPM, 0, 0, clk_sys_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), @@ -1755,6 +1766,8 @@ static const struct snd_soc_dapm_route wm8994_revd_intercon[] = { { "AIF2DACDAT", NULL, "AIF1DACDAT" }, { "AIF1ADCDAT", NULL, "AIF2ADCDAT" }, { "AIF2ADCDAT", NULL, "AIF1ADCDAT" }, + { "MICBIAS", NULL, "CLK_SYS" }, + { "MICBIAS", NULL, "MICBIAS Supply" }, }; static const struct snd_soc_dapm_route wm8994_intercon[] = { -- cgit v1.2.3-70-g09d2 From c8fb034ccd38ecce61564119bcd56ce6e8e97a80 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 2 Mar 2011 11:01:18 +0000 Subject: ASoC: Fix broken bitfield definitions in WM8978 Signed-off-by: Mark Brown Acked-by: Liam Girdwood Cc: stable@kernel.org --- sound/soc/codecs/wm8978.c | 14 ++++++++------ 1 file changed, 8 insertions(+), 6 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index 4bbc3442703..8dfb0a0da67 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -145,18 +145,18 @@ static const struct snd_kcontrol_new wm8978_snd_controls[] = { SOC_SINGLE("DAC Playback Limiter Threshold", WM8978_DAC_LIMITER_2, 4, 7, 0), SOC_SINGLE("DAC Playback Limiter Boost", - WM8978_DAC_LIMITER_2, 0, 15, 0), + WM8978_DAC_LIMITER_2, 0, 12, 0), SOC_ENUM("ALC Enable Switch", alc1), SOC_SINGLE("ALC Capture Min Gain", WM8978_ALC_CONTROL_1, 0, 7, 0), SOC_SINGLE("ALC Capture Max Gain", WM8978_ALC_CONTROL_1, 3, 7, 0), - SOC_SINGLE("ALC Capture Hold", WM8978_ALC_CONTROL_2, 4, 7, 0), + SOC_SINGLE("ALC Capture Hold", WM8978_ALC_CONTROL_2, 4, 10, 0), SOC_SINGLE("ALC Capture Target", WM8978_ALC_CONTROL_2, 0, 15, 0), SOC_ENUM("ALC Capture Mode", alc3), - SOC_SINGLE("ALC Capture Decay", WM8978_ALC_CONTROL_3, 4, 15, 0), - SOC_SINGLE("ALC Capture Attack", WM8978_ALC_CONTROL_3, 0, 15, 0), + SOC_SINGLE("ALC Capture Decay", WM8978_ALC_CONTROL_3, 4, 10, 0), + SOC_SINGLE("ALC Capture Attack", WM8978_ALC_CONTROL_3, 0, 10, 0), SOC_SINGLE("ALC Capture Noise Gate Switch", WM8978_NOISE_GATE, 3, 1, 0), SOC_SINGLE("ALC Capture Noise Gate Threshold", @@ -211,8 +211,10 @@ static const struct snd_kcontrol_new wm8978_snd_controls[] = { WM8978_LOUT2_SPK_CONTROL, WM8978_ROUT2_SPK_CONTROL, 6, 1, 1), /* DAC / ADC oversampling */ - SOC_SINGLE("DAC 128x Oversampling Switch", WM8978_DAC_CONTROL, 8, 1, 0), - SOC_SINGLE("ADC 128x Oversampling Switch", WM8978_ADC_CONTROL, 8, 1, 0), + SOC_SINGLE("DAC 128x Oversampling Switch", WM8978_DAC_CONTROL, + 5, 1, 0), + SOC_SINGLE("ADC 128x Oversampling Switch", WM8978_ADC_CONTROL, + 5, 1, 0), }; /* Mixer #1: Output (OUT1, OUT2) Mixer: mix AUX, Input mixer output and DAC */ -- cgit v1.2.3-70-g09d2 From 1916a2aae52b8cb8f992599204ce06c0accd08e2 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 2 Mar 2011 11:04:10 +0000 Subject: ASoC: Add TLV information for WM8978 DAC limiter Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8978.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index 7ce4f49a67c..85e3e630e76 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -93,6 +93,7 @@ static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); static const DECLARE_TLV_DB_SCALE(inpga_tlv, -1200, 75, 0); static const DECLARE_TLV_DB_SCALE(spk_tlv, -5700, 100, 0); static const DECLARE_TLV_DB_SCALE(boost_tlv, -1500, 300, 1); +static const DECLARE_TLV_DB_SCALE(limiter_tlv, 0, 100, 0); static const struct snd_kcontrol_new wm8978_snd_controls[] = { @@ -144,8 +145,8 @@ static const struct snd_kcontrol_new wm8978_snd_controls[] = { SOC_SINGLE("DAC Playback Limiter Threshold", WM8978_DAC_LIMITER_2, 4, 7, 0), - SOC_SINGLE("DAC Playback Limiter Boost", - WM8978_DAC_LIMITER_2, 0, 12, 0), + SOC_SINGLE_TLV("DAC Playback Limiter Volume", + WM8978_DAC_LIMITER_2, 0, 12, 0, limiter_tlv), SOC_ENUM("ALC Enable Switch", alc1), SOC_SINGLE("ALC Capture Min Gain", WM8978_ALC_CONTROL_1, 0, 7, 0), -- cgit v1.2.3-70-g09d2 From 1d471cd1261a44a3b28350bef7e5113a4609c106 Mon Sep 17 00:00:00 2001 From: Javier Martin Date: Wed, 2 Mar 2011 14:52:32 +0100 Subject: ASoC: Add TI tlv320aic32x4 codec support. This patch adds support for tlv320aic3205 and tlv320aic3254 codecs. It doesn't include miniDSP support for aic3254. Signed-off-by: Javier Martin Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- include/sound/tlv320aic32x4.h | 31 ++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tlv320aic32x4.c | 794 +++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/tlv320aic32x4.h | 143 +++++++ 5 files changed, 974 insertions(+) create mode 100644 include/sound/tlv320aic32x4.h create mode 100644 sound/soc/codecs/tlv320aic32x4.c create mode 100644 sound/soc/codecs/tlv320aic32x4.h (limited to 'sound/soc/codecs') diff --git a/include/sound/tlv320aic32x4.h b/include/sound/tlv320aic32x4.h new file mode 100644 index 00000000000..c009f70b402 --- /dev/null +++ b/include/sound/tlv320aic32x4.h @@ -0,0 +1,31 @@ +/* + * tlv320aic32x4.h -- TLV320AIC32X4 Soc Audio driver platform data + * + * Copyright 2011 Vista Silicon S.L. + * + * Author: Javier Martin + * + * 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 _AIC32X4_PDATA_H +#define _AIC32X4_PDATA_H + +#define AIC32X4_PWR_MICBIAS_2075_LDOIN 0x00000001 +#define AIC32X4_PWR_AVDD_DVDD_WEAK_DISABLE 0x00000002 +#define AIC32X4_PWR_AIC32X4_LDO_ENABLE 0x00000004 +#define AIC32X4_PWR_CMMODE_LDOIN_RANGE_18_36 0x00000008 +#define AIC32X4_PWR_CMMODE_HP_LDOIN_POWERED 0x00000010 + +#define AIC32X4_MICPGA_ROUTE_LMIC_IN2R_10K 0x00000001 +#define AIC32X4_MICPGA_ROUTE_RMIC_IN1L_10K 0x00000002 + +struct aic32x4_pdata { + u32 power_cfg; + u32 micpga_routing; + bool swapdacs; +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index c04da187129..82a46309ded 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -40,6 +40,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_STAC9766 if SND_SOC_AC97_BUS select SND_SOC_TLV320AIC23 if I2C select SND_SOC_TLV320AIC26 if SPI_MASTER + select SND_SOC_TVL320AIC32X4 if I2C select SND_SOC_TLV320AIC3X if I2C select SND_SOC_TPA6130A2 if I2C select SND_SOC_TLV320DAC33 if I2C @@ -206,6 +207,9 @@ config SND_SOC_TLV320AIC26 tristate "TI TLV320AIC26 Codec support" if SND_SOC_OF_SIMPLE depends on SPI +config SND_SOC_TVL320AIC32X4 + tristate + config SND_SOC_TLV320AIC3X tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 3bbb08c512d..b43f9d418c9 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -28,6 +28,7 @@ snd-soc-stac9766-objs := stac9766.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o +snd-soc-tlv320aic32x4-objs := tlv320aic32x4.o snd-soc-tlv320dac33-objs := tlv320dac33.o snd-soc-twl4030-objs := twl4030.o snd-soc-twl6040-objs := twl6040.o @@ -112,6 +113,7 @@ obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o obj-$(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-$(CONFIG_SND_SOC_TVL320AIC32X4) += snd-soc-tlv320aic32x4.o obj-$(CONFIG_SND_SOC_TLV320DAC33) += snd-soc-tlv320dac33.o obj-$(CONFIG_SND_SOC_TWL4030) += snd-soc-twl4030.o obj-$(CONFIG_SND_SOC_TWL6040) += snd-soc-twl6040.o diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c new file mode 100644 index 00000000000..ee82e389603 --- /dev/null +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -0,0 +1,794 @@ +/* + * linux/sound/soc/codecs/tlv320aic32x4.c + * + * Copyright 2011 Vista Silicon S.L. + * + * Author: Javier Martin + * + * Based on sound/soc/codecs/wm8974 and TI driver for kernel 2.6.27. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, + * MA 02110-1301, USA. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include +#include +#include +#include + +#include "tlv320aic32x4.h" + +struct aic32x4_rate_divs { + u32 mclk; + u32 rate; + u8 p_val; + u8 pll_j; + u16 pll_d; + u16 dosr; + u8 ndac; + u8 mdac; + u8 aosr; + u8 nadc; + u8 madc; + u8 blck_N; +}; + +struct aic32x4_priv { + u32 sysclk; + s32 master; + u8 page_no; + void *control_data; + u32 power_cfg; + u32 micpga_routing; + bool swapdacs; +}; + +/* 0dB min, 1dB steps */ +static DECLARE_TLV_DB_SCALE(tlv_step_1, 0, 100, 0); +/* 0dB min, 0.5dB steps */ +static DECLARE_TLV_DB_SCALE(tlv_step_0_5, 0, 50, 0); + +static const struct snd_kcontrol_new aic32x4_snd_controls[] = { + SOC_DOUBLE_R_TLV("PCM Playback Volume", AIC32X4_LDACVOL, + AIC32X4_RDACVOL, 0, 0x30, 0, tlv_step_0_5), + SOC_DOUBLE_R_TLV("HP Driver Gain Volume", AIC32X4_HPLGAIN, + AIC32X4_HPRGAIN, 0, 0x1D, 0, tlv_step_1), + SOC_DOUBLE_R_TLV("LO Driver Gain Volume", AIC32X4_LOLGAIN, + AIC32X4_LORGAIN, 0, 0x1D, 0, tlv_step_1), + SOC_DOUBLE_R("HP DAC Playback Switch", AIC32X4_HPLGAIN, + AIC32X4_HPRGAIN, 6, 0x01, 1), + SOC_DOUBLE_R("LO DAC Playback Switch", AIC32X4_LOLGAIN, + AIC32X4_LORGAIN, 6, 0x01, 1), + SOC_DOUBLE_R("Mic PGA Switch", AIC32X4_LMICPGAVOL, + AIC32X4_RMICPGAVOL, 7, 0x01, 1), + + SOC_SINGLE("ADCFGA Left Mute Switch", AIC32X4_ADCFGA, 7, 1, 0), + SOC_SINGLE("ADCFGA Right Mute Switch", AIC32X4_ADCFGA, 3, 1, 0), + + SOC_DOUBLE_R_TLV("ADC Level Volume", AIC32X4_LADCVOL, + AIC32X4_RADCVOL, 0, 0x28, 0, tlv_step_0_5), + SOC_DOUBLE_R_TLV("PGA Level Volume", AIC32X4_LMICPGAVOL, + AIC32X4_RMICPGAVOL, 0, 0x5f, 0, tlv_step_0_5), + + SOC_SINGLE("Auto-mute Switch", AIC32X4_DACMUTE, 4, 7, 0), + + SOC_SINGLE("AGC Left Switch", AIC32X4_LAGC1, 7, 1, 0), + SOC_SINGLE("AGC Right Switch", AIC32X4_RAGC1, 7, 1, 0), + SOC_DOUBLE_R("AGC Target Level", AIC32X4_LAGC1, AIC32X4_RAGC1, + 4, 0x07, 0), + SOC_DOUBLE_R("AGC Gain Hysteresis", AIC32X4_LAGC1, AIC32X4_RAGC1, + 0, 0x03, 0), + SOC_DOUBLE_R("AGC Hysteresis", AIC32X4_LAGC2, AIC32X4_RAGC2, + 6, 0x03, 0), + SOC_DOUBLE_R("AGC Noise Threshold", AIC32X4_LAGC2, AIC32X4_RAGC2, + 1, 0x1F, 0), + SOC_DOUBLE_R("AGC Max PGA", AIC32X4_LAGC3, AIC32X4_RAGC3, + 0, 0x7F, 0), + SOC_DOUBLE_R("AGC Attack Time", AIC32X4_LAGC4, AIC32X4_RAGC4, + 3, 0x1F, 0), + SOC_DOUBLE_R("AGC Decay Time", AIC32X4_LAGC5, AIC32X4_RAGC5, + 3, 0x1F, 0), + SOC_DOUBLE_R("AGC Noise Debounce", AIC32X4_LAGC6, AIC32X4_RAGC6, + 0, 0x1F, 0), + SOC_DOUBLE_R("AGC Signal Debounce", AIC32X4_LAGC7, AIC32X4_RAGC7, + 0, 0x0F, 0), +}; + +static const struct aic32x4_rate_divs aic32x4_divs[] = { + /* 8k rate */ + {AIC32X4_FREQ_12000000, 8000, 1, 7, 6800, 768, 5, 3, 128, 5, 18, 24}, + {AIC32X4_FREQ_24000000, 8000, 2, 7, 6800, 768, 15, 1, 64, 45, 4, 24}, + {AIC32X4_FREQ_25000000, 8000, 2, 7, 3728, 768, 15, 1, 64, 45, 4, 24}, + /* 11.025k rate */ + {AIC32X4_FREQ_12000000, 11025, 1, 7, 5264, 512, 8, 2, 128, 8, 8, 16}, + {AIC32X4_FREQ_24000000, 11025, 2, 7, 5264, 512, 16, 1, 64, 32, 4, 16}, + /* 16k rate */ + {AIC32X4_FREQ_12000000, 16000, 1, 7, 6800, 384, 5, 3, 128, 5, 9, 12}, + {AIC32X4_FREQ_24000000, 16000, 2, 7, 6800, 384, 15, 1, 64, 18, 5, 12}, + {AIC32X4_FREQ_25000000, 16000, 2, 7, 3728, 384, 15, 1, 64, 18, 5, 12}, + /* 22.05k rate */ + {AIC32X4_FREQ_12000000, 22050, 1, 7, 5264, 256, 4, 4, 128, 4, 8, 8}, + {AIC32X4_FREQ_24000000, 22050, 2, 7, 5264, 256, 16, 1, 64, 16, 4, 8}, + {AIC32X4_FREQ_25000000, 22050, 2, 7, 2253, 256, 16, 1, 64, 16, 4, 8}, + /* 32k rate */ + {AIC32X4_FREQ_12000000, 32000, 1, 7, 1680, 192, 2, 7, 64, 2, 21, 6}, + {AIC32X4_FREQ_24000000, 32000, 2, 7, 1680, 192, 7, 2, 64, 7, 6, 6}, + /* 44.1k rate */ + {AIC32X4_FREQ_12000000, 44100, 1, 7, 5264, 128, 2, 8, 128, 2, 8, 4}, + {AIC32X4_FREQ_24000000, 44100, 2, 7, 5264, 128, 8, 2, 64, 8, 4, 4}, + {AIC32X4_FREQ_25000000, 44100, 2, 7, 2253, 128, 8, 2, 64, 8, 4, 4}, + /* 48k rate */ + {AIC32X4_FREQ_12000000, 48000, 1, 8, 1920, 128, 2, 8, 128, 2, 8, 4}, + {AIC32X4_FREQ_24000000, 48000, 2, 8, 1920, 128, 8, 2, 64, 8, 4, 4}, + {AIC32X4_FREQ_25000000, 48000, 2, 7, 8643, 128, 8, 2, 64, 8, 4, 4} +}; + +static const struct snd_kcontrol_new hpl_output_mixer_controls[] = { + SOC_DAPM_SINGLE("L_DAC Switch", AIC32X4_HPLROUTE, 3, 1, 0), + SOC_DAPM_SINGLE("IN1_L Switch", AIC32X4_HPLROUTE, 2, 1, 0), +}; + +static const struct snd_kcontrol_new hpr_output_mixer_controls[] = { + SOC_DAPM_SINGLE("R_DAC Switch", AIC32X4_HPRROUTE, 3, 1, 0), + SOC_DAPM_SINGLE("IN1_R Switch", AIC32X4_HPRROUTE, 2, 1, 0), +}; + +static const struct snd_kcontrol_new lol_output_mixer_controls[] = { + SOC_DAPM_SINGLE("L_DAC Switch", AIC32X4_LOLROUTE, 3, 1, 0), +}; + +static const struct snd_kcontrol_new lor_output_mixer_controls[] = { + SOC_DAPM_SINGLE("R_DAC Switch", AIC32X4_LORROUTE, 3, 1, 0), +}; + +static const struct snd_kcontrol_new left_input_mixer_controls[] = { + SOC_DAPM_SINGLE("IN1_L P Switch", AIC32X4_LMICPGAPIN, 6, 1, 0), + SOC_DAPM_SINGLE("IN2_L P Switch", AIC32X4_LMICPGAPIN, 4, 1, 0), + SOC_DAPM_SINGLE("IN3_L P Switch", AIC32X4_LMICPGAPIN, 2, 1, 0), +}; + +static const struct snd_kcontrol_new right_input_mixer_controls[] = { + SOC_DAPM_SINGLE("IN1_R P Switch", AIC32X4_RMICPGAPIN, 6, 1, 0), + SOC_DAPM_SINGLE("IN2_R P Switch", AIC32X4_RMICPGAPIN, 4, 1, 0), + SOC_DAPM_SINGLE("IN3_R P Switch", AIC32X4_RMICPGAPIN, 2, 1, 0), +}; + +static const struct snd_soc_dapm_widget aic32x4_dapm_widgets[] = { + SND_SOC_DAPM_DAC("Left DAC", "Left Playback", AIC32X4_DACSETUP, 7, 0), + SND_SOC_DAPM_MIXER("HPL Output Mixer", SND_SOC_NOPM, 0, 0, + &hpl_output_mixer_controls[0], + ARRAY_SIZE(hpl_output_mixer_controls)), + SND_SOC_DAPM_PGA("HPL Power", AIC32X4_OUTPWRCTL, 5, 0, NULL, 0), + + SND_SOC_DAPM_MIXER("LOL Output Mixer", SND_SOC_NOPM, 0, 0, + &lol_output_mixer_controls[0], + ARRAY_SIZE(lol_output_mixer_controls)), + SND_SOC_DAPM_PGA("LOL Power", AIC32X4_OUTPWRCTL, 3, 0, NULL, 0), + + SND_SOC_DAPM_DAC("Right DAC", "Right Playback", AIC32X4_DACSETUP, 6, 0), + SND_SOC_DAPM_MIXER("HPR Output Mixer", SND_SOC_NOPM, 0, 0, + &hpr_output_mixer_controls[0], + ARRAY_SIZE(hpr_output_mixer_controls)), + SND_SOC_DAPM_PGA("HPR Power", AIC32X4_OUTPWRCTL, 4, 0, NULL, 0), + SND_SOC_DAPM_MIXER("LOR Output Mixer", SND_SOC_NOPM, 0, 0, + &lor_output_mixer_controls[0], + ARRAY_SIZE(lor_output_mixer_controls)), + SND_SOC_DAPM_PGA("LOR Power", AIC32X4_OUTPWRCTL, 2, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Left Input Mixer", SND_SOC_NOPM, 0, 0, + &left_input_mixer_controls[0], + ARRAY_SIZE(left_input_mixer_controls)), + SND_SOC_DAPM_MIXER("Right Input Mixer", SND_SOC_NOPM, 0, 0, + &right_input_mixer_controls[0], + ARRAY_SIZE(right_input_mixer_controls)), + SND_SOC_DAPM_ADC("Left ADC", "Left Capture", AIC32X4_ADCSETUP, 7, 0), + SND_SOC_DAPM_ADC("Right ADC", "Right Capture", AIC32X4_ADCSETUP, 6, 0), + SND_SOC_DAPM_MICBIAS("Mic Bias", AIC32X4_MICBIAS, 6, 0), + + SND_SOC_DAPM_OUTPUT("HPL"), + SND_SOC_DAPM_OUTPUT("HPR"), + SND_SOC_DAPM_OUTPUT("LOL"), + SND_SOC_DAPM_OUTPUT("LOR"), + SND_SOC_DAPM_INPUT("IN1_L"), + SND_SOC_DAPM_INPUT("IN1_R"), + SND_SOC_DAPM_INPUT("IN2_L"), + SND_SOC_DAPM_INPUT("IN2_R"), + SND_SOC_DAPM_INPUT("IN3_L"), + SND_SOC_DAPM_INPUT("IN3_R"), +}; + +static const struct snd_soc_dapm_route aic32x4_dapm_routes[] = { + /* Left Output */ + {"HPL Output Mixer", "L_DAC Switch", "Left DAC"}, + {"HPL Output Mixer", "IN1_L Switch", "IN1_L"}, + + {"HPL Power", NULL, "HPL Output Mixer"}, + {"HPL", NULL, "HPL Power"}, + + {"LOL Output Mixer", "L_DAC Switch", "Left DAC"}, + + {"LOL Power", NULL, "LOL Output Mixer"}, + {"LOL", NULL, "LOL Power"}, + + /* Right Output */ + {"HPR Output Mixer", "R_DAC Switch", "Right DAC"}, + {"HPR Output Mixer", "IN1_R Switch", "IN1_R"}, + + {"HPR Power", NULL, "HPR Output Mixer"}, + {"HPR", NULL, "HPR Power"}, + + {"LOR Output Mixer", "R_DAC Switch", "Right DAC"}, + + {"LOR Power", NULL, "LOR Output Mixer"}, + {"LOR", NULL, "LOR Power"}, + + /* Left input */ + {"Left Input Mixer", "IN1_L P Switch", "IN1_L"}, + {"Left Input Mixer", "IN2_L P Switch", "IN2_L"}, + {"Left Input Mixer", "IN3_L P Switch", "IN3_L"}, + + {"Left ADC", NULL, "Left Input Mixer"}, + + /* Right Input */ + {"Right Input Mixer", "IN1_R P Switch", "IN1_R"}, + {"Right Input Mixer", "IN2_R P Switch", "IN2_R"}, + {"Right Input Mixer", "IN3_R P Switch", "IN3_R"}, + + {"Right ADC", NULL, "Right Input Mixer"}, +}; + +static inline int aic32x4_change_page(struct snd_soc_codec *codec, + unsigned int new_page) +{ + struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); + u8 data[2]; + int ret; + + data[0] = 0x00; + data[1] = new_page & 0xff; + + ret = codec->hw_write(codec->control_data, data, 2); + if (ret == 2) { + aic32x4->page_no = new_page; + return 0; + } else { + return ret; + } +} + +static int aic32x4_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int val) +{ + struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); + unsigned int page = reg / 128; + unsigned int fixed_reg = reg % 128; + u8 data[2]; + int ret; + + /* A write to AIC32X4_PSEL is really a non-explicit page change */ + if (reg == AIC32X4_PSEL) + return aic32x4_change_page(codec, val); + + if (aic32x4->page_no != page) { + ret = aic32x4_change_page(codec, page); + if (ret != 0) + return ret; + } + + data[0] = fixed_reg & 0xff; + data[1] = val & 0xff; + + if (codec->hw_write(codec->control_data, data, 2) == 2) + return 0; + else + return -EIO; +} + +static unsigned int aic32x4_read(struct snd_soc_codec *codec, unsigned int reg) +{ + struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); + unsigned int page = reg / 128; + unsigned int fixed_reg = reg % 128; + int ret; + + if (aic32x4->page_no != page) { + ret = aic32x4_change_page(codec, page); + if (ret != 0) + return ret; + } + return i2c_smbus_read_byte_data(codec->control_data, fixed_reg & 0xff); +} + +static inline int aic32x4_get_divs(int mclk, int rate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(aic32x4_divs); i++) { + if ((aic32x4_divs[i].rate == rate) + && (aic32x4_divs[i].mclk == mclk)) { + return i; + } + } + printk(KERN_ERR "aic32x4: master clock and sample rate is not supported\n"); + return -EINVAL; +} + +static int aic32x4_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, aic32x4_dapm_widgets, + ARRAY_SIZE(aic32x4_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, aic32x4_dapm_routes, + ARRAY_SIZE(aic32x4_dapm_routes)); + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +static int aic32x4_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 aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); + + switch (freq) { + case AIC32X4_FREQ_12000000: + case AIC32X4_FREQ_24000000: + case AIC32X4_FREQ_25000000: + aic32x4->sysclk = freq; + return 0; + } + printk(KERN_ERR "aic32x4: invalid frequency to set DAI system clock\n"); + return -EINVAL; +} + +static int aic32x4_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); + u8 iface_reg_1; + u8 iface_reg_2; + u8 iface_reg_3; + + iface_reg_1 = snd_soc_read(codec, AIC32X4_IFACE1); + iface_reg_1 = iface_reg_1 & ~(3 << 6 | 3 << 2); + iface_reg_2 = snd_soc_read(codec, AIC32X4_IFACE2); + iface_reg_2 = 0; + iface_reg_3 = snd_soc_read(codec, AIC32X4_IFACE3); + iface_reg_3 = iface_reg_3 & ~(1 << 3); + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + aic32x4->master = 1; + iface_reg_1 |= AIC32X4_BCLKMASTER | AIC32X4_WCLKMASTER; + break; + case SND_SOC_DAIFMT_CBS_CFS: + aic32x4->master = 0; + break; + default: + printk(KERN_ERR "aic32x4: invalid DAI master/slave interface\n"); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_DSP_A: + iface_reg_1 |= (AIC32X4_DSP_MODE << AIC32X4_PLLJ_SHIFT); + iface_reg_3 |= (1 << 3); /* invert bit clock */ + iface_reg_2 = 0x01; /* add offset 1 */ + break; + case SND_SOC_DAIFMT_DSP_B: + iface_reg_1 |= (AIC32X4_DSP_MODE << AIC32X4_PLLJ_SHIFT); + iface_reg_3 |= (1 << 3); /* invert bit clock */ + break; + case SND_SOC_DAIFMT_RIGHT_J: + iface_reg_1 |= + (AIC32X4_RIGHT_JUSTIFIED_MODE << AIC32X4_PLLJ_SHIFT); + break; + case SND_SOC_DAIFMT_LEFT_J: + iface_reg_1 |= + (AIC32X4_LEFT_JUSTIFIED_MODE << AIC32X4_PLLJ_SHIFT); + break; + default: + printk(KERN_ERR "aic32x4: invalid DAI interface format\n"); + return -EINVAL; + } + + snd_soc_write(codec, AIC32X4_IFACE1, iface_reg_1); + snd_soc_write(codec, AIC32X4_IFACE2, iface_reg_2); + snd_soc_write(codec, AIC32X4_IFACE3, iface_reg_3); + return 0; +} + +static int aic32x4_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); + u8 data; + int i; + + i = aic32x4_get_divs(aic32x4->sysclk, params_rate(params)); + if (i < 0) { + printk(KERN_ERR "aic32x4: sampling rate not supported\n"); + return i; + } + + /* Use PLL as CODEC_CLKIN and DAC_MOD_CLK as BDIV_CLKIN */ + snd_soc_write(codec, AIC32X4_CLKMUX, AIC32X4_PLLCLKIN); + snd_soc_write(codec, AIC32X4_IFACE3, AIC32X4_DACMOD2BCLK); + + /* We will fix R value to 1 and will make P & J=K.D as varialble */ + data = snd_soc_read(codec, AIC32X4_PLLPR); + data &= ~(7 << 4); + snd_soc_write(codec, AIC32X4_PLLPR, + (data | (aic32x4_divs[i].p_val << 4) | 0x01)); + + snd_soc_write(codec, AIC32X4_PLLJ, aic32x4_divs[i].pll_j); + + snd_soc_write(codec, AIC32X4_PLLDMSB, (aic32x4_divs[i].pll_d >> 8)); + snd_soc_write(codec, AIC32X4_PLLDLSB, + (aic32x4_divs[i].pll_d & 0xff)); + + /* NDAC divider value */ + data = snd_soc_read(codec, AIC32X4_NDAC); + data &= ~(0x7f); + snd_soc_write(codec, AIC32X4_NDAC, data | aic32x4_divs[i].ndac); + + /* MDAC divider value */ + data = snd_soc_read(codec, AIC32X4_MDAC); + data &= ~(0x7f); + snd_soc_write(codec, AIC32X4_MDAC, data | aic32x4_divs[i].mdac); + + /* DOSR MSB & LSB values */ + snd_soc_write(codec, AIC32X4_DOSRMSB, aic32x4_divs[i].dosr >> 8); + snd_soc_write(codec, AIC32X4_DOSRLSB, + (aic32x4_divs[i].dosr & 0xff)); + + /* NADC divider value */ + data = snd_soc_read(codec, AIC32X4_NADC); + data &= ~(0x7f); + snd_soc_write(codec, AIC32X4_NADC, data | aic32x4_divs[i].nadc); + + /* MADC divider value */ + data = snd_soc_read(codec, AIC32X4_MADC); + data &= ~(0x7f); + snd_soc_write(codec, AIC32X4_MADC, data | aic32x4_divs[i].madc); + + /* AOSR value */ + snd_soc_write(codec, AIC32X4_AOSR, aic32x4_divs[i].aosr); + + /* BCLK N divider */ + data = snd_soc_read(codec, AIC32X4_BCLKN); + data &= ~(0x7f); + snd_soc_write(codec, AIC32X4_BCLKN, data | aic32x4_divs[i].blck_N); + + data = snd_soc_read(codec, AIC32X4_IFACE1); + data = data & ~(3 << 4); + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + data |= (AIC32X4_WORD_LEN_20BITS << AIC32X4_DOSRMSB_SHIFT); + break; + case SNDRV_PCM_FORMAT_S24_LE: + data |= (AIC32X4_WORD_LEN_24BITS << AIC32X4_DOSRMSB_SHIFT); + break; + case SNDRV_PCM_FORMAT_S32_LE: + data |= (AIC32X4_WORD_LEN_32BITS << AIC32X4_DOSRMSB_SHIFT); + break; + } + snd_soc_write(codec, AIC32X4_IFACE1, data); + + return 0; +} + +static int aic32x4_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u8 dac_reg; + + dac_reg = snd_soc_read(codec, AIC32X4_DACMUTE) & ~AIC32X4_MUTEON; + if (mute) + snd_soc_write(codec, AIC32X4_DACMUTE, dac_reg | AIC32X4_MUTEON); + else + snd_soc_write(codec, AIC32X4_DACMUTE, dac_reg); + return 0; +} + +static int aic32x4_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); + u8 value; + + switch (level) { + case SND_SOC_BIAS_ON: + if (aic32x4->master) { + /* Switch on PLL */ + value = snd_soc_read(codec, AIC32X4_PLLPR); + snd_soc_write(codec, AIC32X4_PLLPR, + (value | AIC32X4_PLLEN)); + + /* Switch on NDAC Divider */ + value = snd_soc_read(codec, AIC32X4_NDAC); + snd_soc_write(codec, AIC32X4_NDAC, + value | AIC32X4_NDACEN); + + /* Switch on MDAC Divider */ + value = snd_soc_read(codec, AIC32X4_MDAC); + snd_soc_write(codec, AIC32X4_MDAC, + value | AIC32X4_MDACEN); + + /* Switch on NADC Divider */ + value = snd_soc_read(codec, AIC32X4_NADC); + snd_soc_write(codec, AIC32X4_NADC, + value | AIC32X4_MDACEN); + + /* Switch on MADC Divider */ + value = snd_soc_read(codec, AIC32X4_MADC); + snd_soc_write(codec, AIC32X4_MADC, + value | AIC32X4_MDACEN); + + /* Switch on BCLK_N Divider */ + value = snd_soc_read(codec, AIC32X4_BCLKN); + snd_soc_write(codec, AIC32X4_BCLKN, + value | AIC32X4_BCLKEN); + } + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + if (aic32x4->master) { + /* Switch off PLL */ + value = snd_soc_read(codec, AIC32X4_PLLPR); + snd_soc_write(codec, AIC32X4_PLLPR, + (value & ~AIC32X4_PLLEN)); + + /* Switch off NDAC Divider */ + value = snd_soc_read(codec, AIC32X4_NDAC); + snd_soc_write(codec, AIC32X4_NDAC, + value & ~AIC32X4_NDACEN); + + /* Switch off MDAC Divider */ + value = snd_soc_read(codec, AIC32X4_MDAC); + snd_soc_write(codec, AIC32X4_MDAC, + value & ~AIC32X4_MDACEN); + + /* Switch off NADC Divider */ + value = snd_soc_read(codec, AIC32X4_NADC); + snd_soc_write(codec, AIC32X4_NADC, + value & ~AIC32X4_NDACEN); + + /* Switch off MADC Divider */ + value = snd_soc_read(codec, AIC32X4_MADC); + snd_soc_write(codec, AIC32X4_MADC, + value & ~AIC32X4_MDACEN); + value = snd_soc_read(codec, AIC32X4_BCLKN); + + /* Switch off BCLK_N Divider */ + snd_soc_write(codec, AIC32X4_BCLKN, + value & ~AIC32X4_BCLKEN); + } + break; + case SND_SOC_BIAS_OFF: + break; + } + codec->bias_level = level; + return 0; +} + +#define AIC32X4_RATES SNDRV_PCM_RATE_8000_48000 +#define AIC32X4_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE \ + | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops aic32x4_ops = { + .hw_params = aic32x4_hw_params, + .digital_mute = aic32x4_mute, + .set_fmt = aic32x4_set_dai_fmt, + .set_sysclk = aic32x4_set_dai_sysclk, +}; + +static struct snd_soc_dai_driver aic32x4_dai = { + .name = "tlv320aic32x4-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = AIC32X4_RATES, + .formats = AIC32X4_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = AIC32X4_RATES, + .formats = AIC32X4_FORMATS,}, + .ops = &aic32x4_ops, + .symmetric_rates = 1, +}; + +static int aic32x4_suspend(struct snd_soc_codec *codec, pm_message_t state) +{ + aic32x4_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int aic32x4_resume(struct snd_soc_codec *codec) +{ + aic32x4_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + return 0; +} + +static int aic32x4_probe(struct snd_soc_codec *codec) +{ + struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); + u32 tmp_reg; + + codec->hw_write = (hw_write_t) i2c_master_send; + codec->control_data = aic32x4->control_data; + + snd_soc_write(codec, AIC32X4_RESET, 0x01); + + /* Power platform configuration */ + if (aic32x4->power_cfg & AIC32X4_PWR_MICBIAS_2075_LDOIN) { + snd_soc_write(codec, AIC32X4_MICBIAS, AIC32X4_MICBIAS_LDOIN | + AIC32X4_MICBIAS_2075V); + } + if (aic32x4->power_cfg & AIC32X4_PWR_AVDD_DVDD_WEAK_DISABLE) { + snd_soc_write(codec, AIC32X4_PWRCFG, AIC32X4_AVDDWEAKDISABLE); + } + if (aic32x4->power_cfg & AIC32X4_PWR_AIC32X4_LDO_ENABLE) { + snd_soc_write(codec, AIC32X4_LDOCTL, AIC32X4_LDOCTLEN); + } + tmp_reg = snd_soc_read(codec, AIC32X4_CMMODE); + if (aic32x4->power_cfg & AIC32X4_PWR_CMMODE_LDOIN_RANGE_18_36) { + tmp_reg |= AIC32X4_LDOIN_18_36; + } + if (aic32x4->power_cfg & AIC32X4_PWR_CMMODE_HP_LDOIN_POWERED) { + tmp_reg |= AIC32X4_LDOIN2HP; + } + snd_soc_write(codec, AIC32X4_CMMODE, tmp_reg); + + /* Do DACs need to be swapped? */ + if (aic32x4->swapdacs) { + snd_soc_write(codec, AIC32X4_DACSETUP, AIC32X4_LDAC2RCHN | AIC32X4_RDAC2LCHN); + } else { + snd_soc_write(codec, AIC32X4_DACSETUP, AIC32X4_LDAC2LCHN | AIC32X4_RDAC2RCHN); + } + + /* Mic PGA routing */ + if (aic32x4->micpga_routing | AIC32X4_MICPGA_ROUTE_LMIC_IN2R_10K) { + snd_soc_write(codec, AIC32X4_LMICPGANIN, AIC32X4_LMICPGANIN_IN2R_10K); + } + if (aic32x4->micpga_routing | AIC32X4_MICPGA_ROUTE_RMIC_IN1L_10K) { + snd_soc_write(codec, AIC32X4_RMICPGANIN, AIC32X4_RMICPGANIN_IN1L_10K); + } + + aic32x4_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + snd_soc_add_controls(codec, aic32x4_snd_controls, + ARRAY_SIZE(aic32x4_snd_controls)); + aic32x4_add_widgets(codec); + + return 0; +} + +static int aic32x4_remove(struct snd_soc_codec *codec) +{ + aic32x4_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_aic32x4 = { + .read = aic32x4_read, + .write = aic32x4_write, + .probe = aic32x4_probe, + .remove = aic32x4_remove, + .suspend = aic32x4_suspend, + .resume = aic32x4_resume, + .set_bias_level = aic32x4_set_bias_level, +}; + +static __devinit int aic32x4_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct aic32x4_pdata *pdata = i2c->dev.platform_data; + struct aic32x4_priv *aic32x4; + int ret; + + aic32x4 = kzalloc(sizeof(struct aic32x4_priv), GFP_KERNEL); + if (aic32x4 == NULL) + return -ENOMEM; + + aic32x4->control_data = i2c; + i2c_set_clientdata(i2c, aic32x4); + + if (pdata) { + aic32x4->power_cfg = pdata->power_cfg; + aic32x4->swapdacs = pdata->swapdacs; + aic32x4->micpga_routing = pdata->micpga_routing; + } else { + aic32x4->power_cfg = 0; + aic32x4->swapdacs = false; + aic32x4->micpga_routing = 0; + } + + ret = snd_soc_register_codec(&i2c->dev, + &soc_codec_dev_aic32x4, &aic32x4_dai, 1); + if (ret < 0) + kfree(aic32x4); + return ret; +} + +static __devexit int aic32x4_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + kfree(i2c_get_clientdata(client)); + return 0; +} + +static const struct i2c_device_id aic32x4_i2c_id[] = { + { "tlv320aic32x4", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, aic32x4_i2c_id); + +static struct i2c_driver aic32x4_i2c_driver = { + .driver = { + .name = "tlv320aic32x4", + .owner = THIS_MODULE, + }, + .probe = aic32x4_i2c_probe, + .remove = __devexit_p(aic32x4_i2c_remove), + .id_table = aic32x4_i2c_id, +}; + +static int __init aic32x4_modinit(void) +{ + int ret = 0; + + ret = i2c_add_driver(&aic32x4_i2c_driver); + if (ret != 0) { + printk(KERN_ERR "Failed to register aic32x4 I2C driver: %d\n", + ret); + } + return ret; +} +module_init(aic32x4_modinit); + +static void __exit aic32x4_exit(void) +{ + i2c_del_driver(&aic32x4_i2c_driver); +} +module_exit(aic32x4_exit); + +MODULE_DESCRIPTION("ASoC tlv320aic32x4 codec driver"); +MODULE_AUTHOR("Javier Martin "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tlv320aic32x4.h b/sound/soc/codecs/tlv320aic32x4.h new file mode 100644 index 00000000000..aae2b244039 --- /dev/null +++ b/sound/soc/codecs/tlv320aic32x4.h @@ -0,0 +1,143 @@ +/* + * tlv320aic32x4.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 _TLV320AIC32X4_H +#define _TLV320AIC32X4_H + +/* tlv320aic32x4 register space (in decimal to match datasheet) */ + +#define AIC32X4_PAGE1 128 + +#define AIC32X4_PSEL 0 +#define AIC32X4_RESET 1 +#define AIC32X4_CLKMUX 4 +#define AIC32X4_PLLPR 5 +#define AIC32X4_PLLJ 6 +#define AIC32X4_PLLDMSB 7 +#define AIC32X4_PLLDLSB 8 +#define AIC32X4_NDAC 11 +#define AIC32X4_MDAC 12 +#define AIC32X4_DOSRMSB 13 +#define AIC32X4_DOSRLSB 14 +#define AIC32X4_NADC 18 +#define AIC32X4_MADC 19 +#define AIC32X4_AOSR 20 +#define AIC32X4_CLKMUX2 25 +#define AIC32X4_CLKOUTM 26 +#define AIC32X4_IFACE1 27 +#define AIC32X4_IFACE2 28 +#define AIC32X4_IFACE3 29 +#define AIC32X4_BCLKN 30 +#define AIC32X4_IFACE4 31 +#define AIC32X4_IFACE5 32 +#define AIC32X4_IFACE6 33 +#define AIC32X4_DOUTCTL 53 +#define AIC32X4_DINCTL 54 +#define AIC32X4_DACSPB 60 +#define AIC32X4_ADCSPB 61 +#define AIC32X4_DACSETUP 63 +#define AIC32X4_DACMUTE 64 +#define AIC32X4_LDACVOL 65 +#define AIC32X4_RDACVOL 66 +#define AIC32X4_ADCSETUP 81 +#define AIC32X4_ADCFGA 82 +#define AIC32X4_LADCVOL 83 +#define AIC32X4_RADCVOL 84 +#define AIC32X4_LAGC1 86 +#define AIC32X4_LAGC2 87 +#define AIC32X4_LAGC3 88 +#define AIC32X4_LAGC4 89 +#define AIC32X4_LAGC5 90 +#define AIC32X4_LAGC6 91 +#define AIC32X4_LAGC7 92 +#define AIC32X4_RAGC1 94 +#define AIC32X4_RAGC2 95 +#define AIC32X4_RAGC3 96 +#define AIC32X4_RAGC4 97 +#define AIC32X4_RAGC5 98 +#define AIC32X4_RAGC6 99 +#define AIC32X4_RAGC7 100 +#define AIC32X4_PWRCFG (AIC32X4_PAGE1 + 1) +#define AIC32X4_LDOCTL (AIC32X4_PAGE1 + 2) +#define AIC32X4_OUTPWRCTL (AIC32X4_PAGE1 + 9) +#define AIC32X4_CMMODE (AIC32X4_PAGE1 + 10) +#define AIC32X4_HPLROUTE (AIC32X4_PAGE1 + 12) +#define AIC32X4_HPRROUTE (AIC32X4_PAGE1 + 13) +#define AIC32X4_LOLROUTE (AIC32X4_PAGE1 + 14) +#define AIC32X4_LORROUTE (AIC32X4_PAGE1 + 15) +#define AIC32X4_HPLGAIN (AIC32X4_PAGE1 + 16) +#define AIC32X4_HPRGAIN (AIC32X4_PAGE1 + 17) +#define AIC32X4_LOLGAIN (AIC32X4_PAGE1 + 18) +#define AIC32X4_LORGAIN (AIC32X4_PAGE1 + 19) +#define AIC32X4_HEADSTART (AIC32X4_PAGE1 + 20) +#define AIC32X4_MICBIAS (AIC32X4_PAGE1 + 51) +#define AIC32X4_LMICPGAPIN (AIC32X4_PAGE1 + 52) +#define AIC32X4_LMICPGANIN (AIC32X4_PAGE1 + 54) +#define AIC32X4_RMICPGAPIN (AIC32X4_PAGE1 + 55) +#define AIC32X4_RMICPGANIN (AIC32X4_PAGE1 + 57) +#define AIC32X4_FLOATINGINPUT (AIC32X4_PAGE1 + 58) +#define AIC32X4_LMICPGAVOL (AIC32X4_PAGE1 + 59) +#define AIC32X4_RMICPGAVOL (AIC32X4_PAGE1 + 60) + +#define AIC32X4_FREQ_12000000 12000000 +#define AIC32X4_FREQ_24000000 24000000 +#define AIC32X4_FREQ_25000000 25000000 + +#define AIC32X4_WORD_LEN_16BITS 0x00 +#define AIC32X4_WORD_LEN_20BITS 0x01 +#define AIC32X4_WORD_LEN_24BITS 0x02 +#define AIC32X4_WORD_LEN_32BITS 0x03 + +#define AIC32X4_I2S_MODE 0x00 +#define AIC32X4_DSP_MODE 0x01 +#define AIC32X4_RIGHT_JUSTIFIED_MODE 0x02 +#define AIC32X4_LEFT_JUSTIFIED_MODE 0x03 + +#define AIC32X4_AVDDWEAKDISABLE 0x08 +#define AIC32X4_LDOCTLEN 0x01 + +#define AIC32X4_LDOIN_18_36 0x01 +#define AIC32X4_LDOIN2HP 0x02 + +#define AIC32X4_DACSPBLOCK_MASK 0x1f +#define AIC32X4_ADCSPBLOCK_MASK 0x1f + +#define AIC32X4_PLLJ_SHIFT 6 +#define AIC32X4_DOSRMSB_SHIFT 4 + +#define AIC32X4_PLLCLKIN 0x03 + +#define AIC32X4_MICBIAS_LDOIN 0x08 +#define AIC32X4_MICBIAS_2075V 0x60 + +#define AIC32X4_LMICPGANIN_IN2R_10K 0x10 +#define AIC32X4_RMICPGANIN_IN1L_10K 0x10 + +#define AIC32X4_LMICPGAVOL_NOGAIN 0x80 +#define AIC32X4_RMICPGAVOL_NOGAIN 0x80 + +#define AIC32X4_BCLKMASTER 0x08 +#define AIC32X4_WCLKMASTER 0x04 +#define AIC32X4_PLLEN (0x01 << 7) +#define AIC32X4_NDACEN (0x01 << 7) +#define AIC32X4_MDACEN (0x01 << 7) +#define AIC32X4_NADCEN (0x01 << 7) +#define AIC32X4_MADCEN (0x01 << 7) +#define AIC32X4_BCLKEN (0x01 << 7) +#define AIC32X4_DACEN (0x03 << 6) +#define AIC32X4_RDAC2LCHN (0x02 << 2) +#define AIC32X4_LDAC2RCHN (0x02 << 4) +#define AIC32X4_LDAC2LCHN (0x01 << 4) +#define AIC32X4_RDAC2RCHN (0x01 << 2) + +#define AIC32X4_SSTEP2WCLK 0x01 +#define AIC32X4_MUTEON 0x0C +#define AIC32X4_DACMOD2BCLK 0x01 + +#endif /* _TLV320AIC32X4_H */ -- cgit v1.2.3-70-g09d2 From 20d660653a488c1c88db6fe51c2459e00cb79230 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 7 Mar 2011 11:07:24 +0000 Subject: ASoC: Fix outdated API usage in tlv320aic32x4 Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/tlv320aic32x4.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index ee82e389603..e93b9d1ae1d 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -340,13 +340,13 @@ static inline int aic32x4_get_divs(int mclk, int rate) static int aic32x4_add_widgets(struct snd_soc_codec *codec) { - snd_soc_dapm_new_controls(codec, aic32x4_dapm_widgets, - ARRAY_SIZE(aic32x4_dapm_widgets)); + snd_soc_dapm_new_controls(&codec->dapm, aic32x4_dapm_widgets, + ARRAY_SIZE(aic32x4_dapm_widgets)); - snd_soc_dapm_add_routes(codec, aic32x4_dapm_routes, + snd_soc_dapm_add_routes(&codec->dapm, aic32x4_dapm_routes, ARRAY_SIZE(aic32x4_dapm_routes)); - snd_soc_dapm_new_widgets(codec); + snd_soc_dapm_new_widgets(&codec->dapm); return 0; } @@ -602,7 +602,7 @@ static int aic32x4_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_OFF: break; } - codec->bias_level = level; + codec->dapm.bias_level = level; return 0; } -- cgit v1.2.3-70-g09d2 From 573f26e3c36ca7036d117bc89d498856073e7284 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 4 Mar 2011 15:18:18 +0800 Subject: ASoC: tlv320dac33: add MODULE_DEVICE_TABLE The device table is required to load modules based on modaliases. Signed-off-by: Axel Lin Acked-by: Peter Ujfalusi Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320dac33.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 71d7be8ac48..00b6d87e7bd 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -1615,6 +1615,7 @@ static const struct i2c_device_id tlv320dac33_i2c_id[] = { }, { }, }; +MODULE_DEVICE_TABLE(i2c, tlv320dac33_i2c_id); static struct i2c_driver tlv320dac33_i2c_driver = { .driver = { -- cgit v1.2.3-70-g09d2 From 79a54ea1ede0f028e5b0be1016bff8f49326a265 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 4 Mar 2011 15:22:03 +0800 Subject: ASoC: Constify i2c_device_id table Signed-off-by: Axel Lin Acked-by: Alexander Sverdlin Acked-by: Timur Tabi Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/cs4270.c | 2 +- sound/soc/codecs/cs4271.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 65f578ff611..0206a17d728 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -719,7 +719,7 @@ static int cs4270_i2c_remove(struct i2c_client *i2c_client) /* * cs4270_id - I2C device IDs supported by this driver */ -static struct i2c_device_id cs4270_id[] = { +static const struct i2c_device_id cs4270_id[] = { {"cs4270", 0}, {} }; diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 1791796216c..41a3ee6f614 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -555,7 +555,7 @@ static struct spi_driver cs4271_spi_driver = { #endif /* defined(CONFIG_SPI_MASTER) */ #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) -static struct i2c_device_id cs4271_i2c_id[] = { +static const struct i2c_device_id cs4271_i2c_id[] = { {"cs4271", 0}, {} }; -- cgit v1.2.3-70-g09d2 From 9b0a25f0386f9775b69c46ced9b5632b649f00ba Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 7 Mar 2011 08:04:55 +0100 Subject: ASoC: neo1973_wm8753: Move lm4857 specefic code to its own module This patch moves the code for the lm4857 AMP from the neo1973_wm8753 sound board driver to its own module. The lm4857 is a generic AMP IC and not specific to the neo1973. Signed-off-by: Lars-Peter Clausen Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 3 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/lm4857.c | 276 +++++++++++++++++++++++++++++++++++++ sound/soc/samsung/Kconfig | 1 + sound/soc/samsung/lm4857.h | 32 ----- sound/soc/samsung/neo1973_wm8753.c | 269 +++++++++--------------------------- 6 files changed, 347 insertions(+), 236 deletions(-) create mode 100644 sound/soc/codecs/lm4857.c delete mode 100644 sound/soc/samsung/lm4857.h (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 82a46309ded..37035e6984b 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -347,6 +347,9 @@ config SND_SOC_WM9713 tristate # Amp +config SND_SOC_LM4857 + tristate + config SND_SOC_MAX9877 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index b43f9d418c9..0663d22e86b 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -77,6 +77,7 @@ snd-soc-wm-hubs-objs := wm_hubs.o snd-soc-jz4740-codec-objs := jz4740.o # Amp +snd-soc-lm4857-objs := lm4857.o snd-soc-max9877-objs := max9877.o snd-soc-tpa6130a2-objs := tpa6130a2.o snd-soc-wm2000-objs := wm2000.o @@ -161,6 +162,7 @@ obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o obj-$(CONFIG_SND_SOC_WM_HUBS) += snd-soc-wm-hubs.o # Amp +obj-$(CONFIG_SND_SOC_LM4857) += snd-soc-lm4857.o obj-$(CONFIG_SND_SOC_MAX9877) += snd-soc-max9877.o obj-$(CONFIG_SND_SOC_TPA6130A2) += snd-soc-tpa6130a2.o obj-$(CONFIG_SND_SOC_WM2000) += snd-soc-wm2000.o diff --git a/sound/soc/codecs/lm4857.c b/sound/soc/codecs/lm4857.c new file mode 100644 index 00000000000..72de47e5d04 --- /dev/null +++ b/sound/soc/codecs/lm4857.c @@ -0,0 +1,276 @@ +/* + * LM4857 AMP driver + * + * Copyright 2007 Wolfson Microelectronics PLC. + * Author: Graeme Gregory + * graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com + * Copyright 2011 Lars-Peter Clausen + * + * 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. + * + */ + +#include +#include +#include +#include + +#include +#include +#include + +struct lm4857 { + struct i2c_client *i2c; + uint8_t mode; +}; + +static const uint8_t lm4857_default_regs[] = { + 0x00, 0x00, 0x00, 0x00, +}; + +/* The register offsets in the cache array */ +#define LM4857_MVOL 0 +#define LM4857_LVOL 1 +#define LM4857_RVOL 2 +#define LM4857_CTRL 3 + +/* the shifts required to set these bits */ +#define LM4857_3D 5 +#define LM4857_WAKEUP 5 +#define LM4857_EPGAIN 4 + +static int lm4857_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + uint8_t data; + int ret; + + ret = snd_soc_cache_write(codec, reg, value); + if (ret < 0) + return ret; + + data = (reg << 6) | value; + ret = i2c_master_send(codec->control_data, &data, 1); + if (ret != 1) { + dev_err(codec->dev, "Failed to write register: %d\n", ret); + return ret; + } + + return 0; +} + +static unsigned int lm4857_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + unsigned int val; + int ret; + + ret = snd_soc_cache_read(codec, reg, &val); + if (ret) + return -1; + + return val; +} + +static int lm4857_get_mode(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct lm4857 *lm4857 = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.integer.value[0] = lm4857->mode; + + return 0; +} + +static int lm4857_set_mode(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct lm4857 *lm4857 = snd_soc_codec_get_drvdata(codec); + uint8_t value = ucontrol->value.integer.value[0]; + + lm4857->mode = value; + + if (codec->dapm.bias_level == SND_SOC_BIAS_ON) + snd_soc_update_bits(codec, LM4857_CTRL, 0x0F, value + 6); + + return 1; +} + +static int lm4857_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct lm4857 *lm4857 = snd_soc_codec_get_drvdata(codec); + + switch (level) { + case SND_SOC_BIAS_ON: + snd_soc_update_bits(codec, LM4857_CTRL, 0x0F, lm4857->mode + 6); + break; + case SND_SOC_BIAS_STANDBY: + snd_soc_update_bits(codec, LM4857_CTRL, 0x0F, 0); + break; + default: + break; + } + + codec->dapm.bias_level = level; + + return 0; +} + +static const char *lm4857_mode[] = { + "Earpiece", + "Loudspeaker", + "Loudspeaker + Headphone", + "Headphone", +}; + +static const struct soc_enum lm4857_mode_enum = + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(lm4857_mode), lm4857_mode); + +static const struct snd_soc_dapm_widget lm4857_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("IN"), + + SND_SOC_DAPM_OUTPUT("LS"), + SND_SOC_DAPM_OUTPUT("HP"), + SND_SOC_DAPM_OUTPUT("EP"), +}; + +static const DECLARE_TLV_DB_SCALE(stereo_tlv, -4050, 150, 0); +static const DECLARE_TLV_DB_SCALE(mono_tlv, -3450, 150, 0); + +static const struct snd_kcontrol_new lm4857_controls[] = { + SOC_SINGLE_TLV("Left Playback Volume", LM4857_LVOL, 0, 31, 0, + stereo_tlv), + SOC_SINGLE_TLV("Right Playback Volume", LM4857_RVOL, 0, 31, 0, + stereo_tlv), + SOC_SINGLE_TLV("Mono Playback Volume", LM4857_MVOL, 0, 31, 0, + mono_tlv), + SOC_SINGLE("Spk 3D Playback Switch", LM4857_LVOL, LM4857_3D, 1, 0), + SOC_SINGLE("HP 3D Playback Switch", LM4857_RVOL, LM4857_3D, 1, 0), + SOC_SINGLE("Fast Wakeup Playback Switch", LM4857_CTRL, + LM4857_WAKEUP, 1, 0), + SOC_SINGLE("Earpiece 6dB Playback Switch", LM4857_CTRL, + LM4857_EPGAIN, 1, 0), + + SOC_ENUM_EXT("Mode", lm4857_mode_enum, + lm4857_get_mode, lm4857_set_mode), +}; + +/* There is a demux inbetween the the input signal and the output signals. + * Currently there is no easy way to model it in ASoC and since it does not make + * much of a difference in practice simply connect the input direclty to the + * outputs. */ +static const struct snd_soc_dapm_route lm4857_routes[] = { + {"LS", NULL, "IN"}, + {"HP", NULL, "IN"}, + {"EP", NULL, "IN"}, +}; + +static int lm4857_probe(struct snd_soc_codec *codec) +{ + struct lm4857 *lm4857 = snd_soc_codec_get_drvdata(codec); + struct snd_soc_dapm_context *dapm = &codec->dapm; + int ret; + + codec->control_data = lm4857->i2c; + + ret = snd_soc_add_controls(codec, lm4857_controls, + ARRAY_SIZE(lm4857_controls)); + if (ret) + return ret; + + ret = snd_soc_dapm_new_controls(dapm, lm4857_dapm_widgets, + ARRAY_SIZE(lm4857_dapm_widgets)); + if (ret) + return ret; + + ret = snd_soc_dapm_add_routes(dapm, lm4857_routes, + ARRAY_SIZE(lm4857_routes)); + if (ret) + return ret; + + snd_soc_dapm_new_widgets(dapm); + + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_lm4857 = { + .write = lm4857_write, + .read = lm4857_read, + .probe = lm4857_probe, + .reg_cache_size = ARRAY_SIZE(lm4857_default_regs), + .reg_word_size = sizeof(uint8_t), + .reg_cache_default = lm4857_default_regs, + .set_bias_level = lm4857_set_bias_level, +}; + +static int __devinit lm4857_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct lm4857 *lm4857; + int ret; + + lm4857 = kzalloc(sizeof(*lm4857), GFP_KERNEL); + if (!lm4857) + return -ENOMEM; + + i2c_set_clientdata(i2c, lm4857); + + lm4857->i2c = i2c; + + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_lm4857, NULL, 0); + + if (ret) { + kfree(lm4857); + return ret; + } + + return 0; +} + +static int __devexit lm4857_i2c_remove(struct i2c_client *i2c) +{ + struct lm4857 *lm4857 = i2c_get_clientdata(i2c); + + snd_soc_unregister_codec(&i2c->dev); + kfree(lm4857); + + return 0; +} + +static const struct i2c_device_id lm4857_i2c_id[] = { + { "lm4857", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, lm4857_i2c_id); + +static struct i2c_driver lm4857_i2c_driver = { + .driver = { + .name = "lm4857", + .owner = THIS_MODULE, + }, + .probe = lm4857_i2c_probe, + .remove = __devexit_p(lm4857_i2c_remove), + .id_table = lm4857_i2c_id, +}; + +static int __init lm4857_init(void) +{ + return i2c_add_driver(&lm4857_i2c_driver); +} +module_init(lm4857_init); + +static void __exit lm4857_exit(void) +{ + i2c_del_driver(&lm4857_i2c_driver); +} +module_exit(lm4857_exit); + +MODULE_AUTHOR("Lars-Peter Clausen "); +MODULE_DESCRIPTION("LM4857 amplifier driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index a6a6b5fa2f2..ba78e26e20f 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -39,6 +39,7 @@ config SND_SOC_SAMSUNG_NEO1973_WM8753 depends on SND_SOC_SAMSUNG && MACH_NEO1973_GTA01 select SND_S3C24XX_I2S select SND_SOC_WM8753 + select SND_SOC_LM4857 help Say Y if you want to add support for SoC audio on smdk2440 with the WM8753. diff --git a/sound/soc/samsung/lm4857.h b/sound/soc/samsung/lm4857.h deleted file mode 100644 index 0cf5b7011d6..00000000000 --- a/sound/soc/samsung/lm4857.h +++ /dev/null @@ -1,32 +0,0 @@ -/* - * lm4857.h -- ALSA Soc Audio Layer - * - * Copyright 2007 Wolfson Microelectronics PLC. - * Author: Graeme Gregory - * graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.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 - * 18th Jun 2007 Initial version. - */ - -#ifndef LM4857_H_ -#define LM4857_H_ - -/* The register offsets in the cache array */ -#define LM4857_MVOL 0 -#define LM4857_LVOL 1 -#define LM4857_RVOL 2 -#define LM4857_CTRL 3 - -/* the shifts required to set these bits */ -#define LM4857_3D 5 -#define LM4857_WAKEUP 5 -#define LM4857_EPGAIN 4 - -#endif /*LM4857_H_*/ - diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index fe929467f93..7761827314b 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -17,11 +17,9 @@ #include #include #include -#include #include #include #include -#include #include #include @@ -33,12 +31,10 @@ #include #include "../codecs/wm8753.h" -#include "lm4857.h" #include "dma.h" #include "s3c24xx-i2s.h" static struct snd_soc_card neo1973; -static struct i2c_client *i2c; static int neo1973_hifi_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) @@ -213,85 +209,7 @@ static struct snd_soc_ops neo1973_voice_ops = { .hw_free = neo1973_voice_hw_free, }; -static u8 lm4857_regs[4] = {0x00, 0x40, 0x80, 0xC0}; - -static void lm4857_write_regs(void) -{ - pr_debug("Entered %s\n", __func__); - - if (i2c_master_send(i2c, lm4857_regs, 4) != 4) - printk(KERN_ERR "lm4857: i2c write failed\n"); -} - -static int lm4857_get_reg(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - int reg = mc->reg; - int shift = mc->shift; - int mask = mc->max; - - pr_debug("Entered %s\n", __func__); - - ucontrol->value.integer.value[0] = (lm4857_regs[reg] >> shift) & mask; - return 0; -} - -static int lm4857_set_reg(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - int reg = mc->reg; - int shift = mc->shift; - int mask = mc->max; - - if (((lm4857_regs[reg] >> shift) & mask) == - ucontrol->value.integer.value[0]) - return 0; - - lm4857_regs[reg] &= ~(mask << shift); - lm4857_regs[reg] |= ucontrol->value.integer.value[0] << shift; - lm4857_write_regs(); - return 1; -} - -static int lm4857_get_mode(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - u8 value = lm4857_regs[LM4857_CTRL] & 0x0F; - - pr_debug("Entered %s\n", __func__); - - if (value) - value -= 5; - - ucontrol->value.integer.value[0] = value; - return 0; -} - -static int lm4857_set_mode(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - u8 value = ucontrol->value.integer.value[0]; - - pr_debug("Entered %s\n", __func__); - - if (value) - value += 5; - - if ((lm4857_regs[LM4857_CTRL] & 0x0F) == value) - return 0; - - lm4857_regs[LM4857_CTRL] &= 0xF0; - lm4857_regs[LM4857_CTRL] |= value; - lm4857_write_regs(); - return 1; -} - static const struct snd_soc_dapm_widget wm8753_dapm_widgets[] = { - SND_SOC_DAPM_LINE("Audio Out", NULL), SND_SOC_DAPM_LINE("GSM Line Out", NULL), SND_SOC_DAPM_LINE("GSM Line In", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), @@ -299,12 +217,7 @@ static const struct snd_soc_dapm_widget wm8753_dapm_widgets[] = { }; -static const struct snd_soc_dapm_route dapm_routes[] = { - - /* Connections to the lm4857 amp */ - {"Audio Out", NULL, "LOUT1"}, - {"Audio Out", NULL, "ROUT1"}, - +static const struct snd_soc_dapm_route neo1973_wm8753_routes[] = { /* Connections to the GSM Module */ {"GSM Line Out", NULL, "MONO1"}, {"GSM Line Out", NULL, "MONO2"}, @@ -324,46 +237,14 @@ static const struct snd_soc_dapm_route dapm_routes[] = { {"ACIN", NULL, "ACOP"}, }; -static const char *lm4857_mode[] = { - "Off", - "Call Speaker", - "Stereo Speakers", - "Stereo Speakers + Headphones", - "Headphones" -}; - -static const struct soc_enum lm4857_mode_enum[] = { - SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(lm4857_mode), lm4857_mode), -}; - -static const DECLARE_TLV_DB_SCALE(stereo_tlv, -4050, 150, 0); -static const DECLARE_TLV_DB_SCALE(mono_tlv, -3450, 150, 0); - -static const struct snd_kcontrol_new wm8753_neo1973_controls[] = { - SOC_DAPM_PIN_SWITCH("Audio Out"), +static const struct snd_kcontrol_new neo1973_wm8753_controls[] = { SOC_DAPM_PIN_SWITCH("GSM Line Out"), SOC_DAPM_PIN_SWITCH("GSM Line In"), SOC_DAPM_PIN_SWITCH("Headset Mic"), SOC_DAPM_PIN_SWITCH("Call Mic"), - - SOC_SINGLE_EXT_TLV("Amp Left Playback Volume", LM4857_LVOL, 0, 31, 0, - lm4857_get_reg, lm4857_set_reg, stereo_tlv), - SOC_SINGLE_EXT_TLV("Amp Right Playback Volume", LM4857_RVOL, 0, 31, 0, - lm4857_get_reg, lm4857_set_reg, stereo_tlv), - SOC_SINGLE_EXT_TLV("Amp Mono Playback Volume", LM4857_MVOL, 0, 31, 0, - lm4857_get_reg, lm4857_set_reg, mono_tlv), - SOC_ENUM_EXT("Amp Mode", lm4857_mode_enum[0], - lm4857_get_mode, lm4857_set_mode), - SOC_SINGLE_EXT("Amp Spk 3D Playback Switch", LM4857_LVOL, 5, 1, 0, - lm4857_get_reg, lm4857_set_reg), - SOC_SINGLE_EXT("Amp HP 3d Playback Switch", LM4857_RVOL, 5, 1, 0, - lm4857_get_reg, lm4857_set_reg), - SOC_SINGLE_EXT("Amp Fast Wakeup Playback Switch", LM4857_CTRL, 5, 1, 0, - lm4857_get_reg, lm4857_set_reg), - SOC_SINGLE_EXT("Amp Earpiece 6dB Playback Switch", LM4857_CTRL, 4, 1, 0, - lm4857_get_reg, lm4857_set_reg), }; + /* * This is an example machine initialisation for a wm8753 connected to a * neo1973 II. It is missing logic to detect hp/mic insertions and logic @@ -387,29 +268,66 @@ static int neo1973_wm8753_init(struct snd_soc_pcm_runtime *rtd) /* Add neo1973 specific widgets */ snd_soc_dapm_new_controls(dapm, wm8753_dapm_widgets, - ARRAY_SIZE(wm8753_dapm_widgets)); + ARRAY_SIZE(wm8753_dapm_widgets)); /* set endpoints to default mode */ - snd_soc_dapm_disable_pin(dapm, "Audio Out"); snd_soc_dapm_disable_pin(dapm, "GSM Line Out"); snd_soc_dapm_disable_pin(dapm, "GSM Line In"); snd_soc_dapm_disable_pin(dapm, "Headset Mic"); snd_soc_dapm_disable_pin(dapm, "Call Mic"); /* add neo1973 specific controls */ - err = snd_soc_add_controls(codec, wm8753_neo1973_controls, - ARRAY_SIZE(8753_neo1973_controls)); + err = snd_soc_add_controls(codec, neo1973_wm8753_controls, + ARRAY_SIZE(neo1973_wm8753_controls)); if (err < 0) return err; /* set up neo1973 specific audio routes */ - err = snd_soc_dapm_add_routes(dapm, dapm_routes, - ARRAY_SIZE(dapm_routes)); + err = snd_soc_dapm_add_routes(dapm, neo1973_wm8753_routes, + ARRAY_SIZE(neo1973_wm8753_routes)); snd_soc_dapm_sync(dapm); return 0; } +static const struct snd_soc_dapm_route neo1973_lm4857_routes[] = { + {"Amp IN", NULL, "ROUT1"}, + {"Amp IN", NULL, "LOUT1"}, + + {"Handset Spk", NULL, "Amp EP"}, + {"Stereo Out", NULL, "Amp LS"}, + {"Headphone", NULL, "Amp HP"}, +}; + +static const struct snd_soc_dapm_widget neo1973_lm4857_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Handset Spk", NULL), + SND_SOC_DAPM_SPK("Stereo Out", NULL), + SND_SOC_DAPM_HP("Headphone", NULL), +}; + +static int neo1973_lm4857_init(struct snd_soc_dapm_context *dapm) +{ + int ret; + + ret = snd_soc_dapm_new_controls(dapm, neo1973_lm4857_dapm_widgets, + ARRAY_SIZE(neo1973_lm4857_dapm_widgets)); + if (ret) + return ret; + + ret = snd_soc_dapm_add_routes(dapm, neo1973_lm4857_routes, + ARRAY_SIZE(neo1973_lm4857_routes)); + if (ret) + return ret; + + snd_soc_dapm_ignore_suspend(dapm, "Stereo Out"); + snd_soc_dapm_ignore_suspend(dapm, "Handset Spk"); + snd_soc_dapm_ignore_suspend(dapm, "Headphone"); + + snd_soc_dapm_sync(dapm); + + return 0; +} + /* * BT Codec DAI */ @@ -449,83 +367,29 @@ static struct snd_soc_dai_link neo1973_dai[] = { }, }; -static struct snd_soc_card neo1973 = { - .name = "neo1973", - .dai_link = neo1973_dai, - .num_links = ARRAY_SIZE(neo1973_dai), +static struct snd_soc_aux_dev neo1973_aux_devs[] = { + { + .name = "lm4857", + .codec_name = "lm4857.0-007c", + .init = neo1973_lm4857_init, + }, }; -static int lm4857_i2c_probe(struct i2c_client *client, - const struct i2c_device_id *id) -{ - pr_debug("Entered %s\n", __func__); - - i2c = client; - - lm4857_write_regs(); - return 0; -} - -static int lm4857_i2c_remove(struct i2c_client *client) -{ - pr_debug("Entered %s\n", __func__); - - i2c = NULL; - - return 0; -} - -static u8 lm4857_state; - -static int lm4857_suspend(struct i2c_client *dev, pm_message_t state) -{ - pr_debug("Entered %s\n", __func__); - - dev_dbg(&dev->dev, "lm4857_suspend\n"); - lm4857_state = lm4857_regs[LM4857_CTRL] & 0xf; - if (lm4857_state) { - lm4857_regs[LM4857_CTRL] &= 0xf0; - lm4857_write_regs(); - } - return 0; -} - -static int lm4857_resume(struct i2c_client *dev) -{ - pr_debug("Entered %s\n", __func__); - - if (lm4857_state) { - lm4857_regs[LM4857_CTRL] |= (lm4857_state & 0x0f); - lm4857_write_regs(); - } - return 0; -} - -static void lm4857_shutdown(struct i2c_client *dev) -{ - pr_debug("Entered %s\n", __func__); - - dev_dbg(&dev->dev, "lm4857_shutdown\n"); - lm4857_regs[LM4857_CTRL] &= 0xf0; - lm4857_write_regs(); -} - -static const struct i2c_device_id lm4857_i2c_id[] = { - { "neo1973_lm4857", 0 }, - { } +static struct snd_soc_codec_conf neo1973_codec_conf[] = { + { + .dev_name = "lm4857.0-007c", + .name_prefix = "Amp", + }, }; -static struct i2c_driver lm4857_i2c_driver = { - .driver = { - .name = "LM4857 I2C Amp", - .owner = THIS_MODULE, - }, - .suspend = lm4857_suspend, - .resume = lm4857_resume, - .shutdown = lm4857_shutdown, - .probe = lm4857_i2c_probe, - .remove = lm4857_i2c_remove, - .id_table = lm4857_i2c_id, +static struct snd_soc_card neo1973 = { + .name = "neo1973", + .dai_link = neo1973_dai, + .num_links = ARRAY_SIZE(neo1973_dai), + .aux_dev = neo1973_aux_devs, + .num_aux_devs = ARRAY_SIZE(neo1973_aux_devs), + .codec_conf = neo1973_codec_conf, + .num_configs = ARRAY_SIZE(neo1973_codec_conf), }; static struct platform_device *neo1973_snd_device; @@ -554,8 +418,6 @@ static int __init neo1973_init(void) return ret; } - ret = i2c_add_driver(&lm4857_i2c_driver); - if (ret != 0) platform_device_unregister(neo1973_snd_device); @@ -566,7 +428,6 @@ static void __exit neo1973_exit(void) { pr_debug("Entered %s\n", __func__); - i2c_del_driver(&lm4857_i2c_driver); platform_device_unregister(neo1973_snd_device); } -- cgit v1.2.3-70-g09d2 From a077ff9034897232ab4208f55880221390bd6877 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 7 Mar 2011 08:04:59 +0100 Subject: ASoC: Add driver for the dfbmcs320 bluetooth module This patch adds a codec driver for the dfbmcs320 bluetooth module, which is used on the neo1973 boards. The patch also modifies the neo1937_wm8753 sound board driver to use the new driver instead of registering the bluetooth DAI manually. Previously there was a name mismatch between the bluetooth DAI and the bluetooth DAI link and the sound card was not instantiated, with this patch the issue is no longer present and sound support works again. Signed-off-by: Lars-Peter Clausen Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 5 ++- sound/soc/codecs/Makefile | 2 ++ sound/soc/codecs/dfbmcs320.c | 72 ++++++++++++++++++++++++++++++++++++++ sound/soc/samsung/Kconfig | 1 + sound/soc/samsung/neo1973_wm8753.c | 35 ++++-------------- 5 files changed, 86 insertions(+), 29 deletions(-) create mode 100644 sound/soc/codecs/dfbmcs320.c (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 37035e6984b..212859cf19a 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -29,6 +29,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS4271 if SND_SOC_I2C_AND_SPI select SND_SOC_CX20442 select SND_SOC_DA7210 if I2C + select SND_SOC_DFBMCS320 select SND_SOC_JZ4740_CODEC if SOC_JZ4740 select SND_SOC_MAX98088 if I2C select SND_SOC_MAX9877 if I2C @@ -175,6 +176,9 @@ config SND_SOC_L3 config SND_SOC_DA7210 tristate +config SND_SOC_DFBMCS320 + tristate + config SND_SOC_DMIC tristate @@ -361,4 +365,3 @@ config SND_SOC_WM2000 config SND_SOC_WM9090 tristate - diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 0663d22e86b..ebb059c0eef 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -15,6 +15,7 @@ snd-soc-cs4270-objs := cs4270.o snd-soc-cs4271-objs := cs4271.o snd-soc-cx20442-objs := cx20442.o snd-soc-da7210-objs := da7210.o +snd-soc-dfbmcs320-objs := dfbmcs320.o snd-soc-dmic-objs := dmic.o snd-soc-l3-objs := l3.o snd-soc-max98088-objs := max98088.o @@ -101,6 +102,7 @@ obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_CS4271) += snd-soc-cs4271.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o +obj-$(CONFIG_SND_SOC_DFBMCS320) += snd-soc-dfbmcs320.o obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o diff --git a/sound/soc/codecs/dfbmcs320.c b/sound/soc/codecs/dfbmcs320.c new file mode 100644 index 00000000000..704bbde6573 --- /dev/null +++ b/sound/soc/codecs/dfbmcs320.c @@ -0,0 +1,72 @@ +/* + * Driver for the DFBM-CS320 bluetooth module + * Copyright 2011 Lars-Peter Clausen + * + * 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. + * + */ + +#include +#include +#include + +#include + +static struct snd_soc_dai_driver dfbmcs320_dai = { + .name = "dfbmcs320-pcm", + .playback = { + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, +}; + +static struct snd_soc_codec_driver soc_codec_dev_dfbmcs320; + +static int __devinit dfbmcs320_probe(struct platform_device *pdev) +{ + return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_dfbmcs320, + &dfbmcs320_dai, 1); +} + +static int __devexit dfbmcs320_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + + return 0; +} + +static struct platform_driver dfmcs320_driver = { + .driver = { + .name = "dfbmcs320", + .owner = THIS_MODULE, + }, + .probe = dfbmcs320_probe, + .remove = __devexit_p(dfbmcs320_remove), +}; + +static int __init dfbmcs320_init(void) +{ + return platform_driver_register(&dfmcs320_driver); +} +module_init(dfbmcs320_init); + +static void __exit dfbmcs320_exit(void) +{ + platform_driver_unregister(&dfmcs320_driver); +} +module_exit(dfbmcs320_exit); + +MODULE_AUTHOR("Lars-Peter Clausen "); +MODULE_DESCRIPTION("ASoC DFBM-CS320 bluethooth module driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index c3014e82157..a08237acc53 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -40,6 +40,7 @@ config SND_SOC_SAMSUNG_NEO1973_WM8753 select SND_S3C24XX_I2S select SND_SOC_WM8753 select SND_SOC_LM4857 if MACH_NEO1973_GTA01 + select SND_SOC_DFBMCS320 help Say Y here to enable audio support for the Openmoko Neo1973 Smartphones. diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index 37cfbb8ca39..78bfdb3f5d7 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -418,23 +418,6 @@ static int neo1973_lm4857_init(struct snd_soc_dapm_context *dapm) static int neo1973_lm4857_init(struct snd_soc_dapm_context *dapm) { return 0; }; #endif -/* - * BT Codec DAI - */ -static struct snd_soc_dai_driver bt_dai = { - .name = "bluetooth-dai", - .playback = { - .channels_min = 1, - .channels_max = 1, - .rates = SNDRV_PCM_RATE_8000, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .capture = { - .channels_min = 1, - .channels_max = 1, - .rates = SNDRV_PCM_RATE_8000, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, -}; - static struct snd_soc_dai_link neo1973_dai[] = { { /* Hifi Playback - for similatious use with voice below */ .name = "WM8753", @@ -450,7 +433,7 @@ static struct snd_soc_dai_link neo1973_dai[] = { .name = "Bluetooth", .stream_name = "Voice", .platform_name = "samsung-audio", - .cpu_dai_name = "bluetooth-dai", + .cpu_dai_name = "dfbmcs320-pcm", .codec_dai_name = "wm8753-voice", .codec_name = "wm8753-codec.0-001a", .ops = &neo1973_voice_ops, @@ -458,6 +441,10 @@ static struct snd_soc_dai_link neo1973_dai[] = { }; static struct snd_soc_aux_dev neo1973_aux_devs[] = { + { + .name = "dfbmcs320", + .codec_name = "dfbmcs320.0", + }, { .name = "lm4857", .codec_name = "lm4857.0-007c", @@ -502,7 +489,7 @@ static int __init neo1973_init(void) if (machine_is_neo1973_gta02()) { neo1973.name = "neo1973gta02"; - neo1973.num_aux_devs = 0; + neo1973.num_aux_devs = 1; ret = gpio_request_array(neo1973_gta02_gpios, ARRAY_SIZE(neo1973_gta02_gpios)); @@ -516,21 +503,14 @@ static int __init neo1973_init(void) goto err_gpio_free; } - /* register bluetooth DAI here */ - ret = snd_soc_register_dai(&neo1973_snd_device->dev, &bt_dai); - if (ret) - goto err_put_device; - platform_set_drvdata(neo1973_snd_device, &neo1973); ret = platform_device_add(neo1973_snd_device); if (ret) - goto err_unregister_dai; + goto err_put_device; return 0; -err_unregister_dai: - snd_soc_unregister_dai(&neo1973_snd_device->dev); err_put_device: platform_device_put(neo1973_snd_device); err_gpio_free: @@ -544,7 +524,6 @@ module_init(neo1973_init); static void __exit neo1973_exit(void) { - snd_soc_unregister_dai(&neo1973_snd_device->dev); platform_device_unregister(neo1973_snd_device); if (machine_is_neo1973_gta02()) { -- cgit v1.2.3-70-g09d2 From 9b74c7d6ba8b95b4feb8c17b2b38d72eb4240ea4 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 7 Mar 2011 12:23:10 +0000 Subject: ASoC: Add LM4857 to SND_SOC_ALL_CODECS Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 212859cf19a..10f11dd98c2 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -31,6 +31,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_DA7210 if I2C select SND_SOC_DFBMCS320 select SND_SOC_JZ4740_CODEC if SOC_JZ4740 + select SND_SOC_LM4857 if I2C select SND_SOC_MAX98088 if I2C select SND_SOC_MAX9877 if I2C select SND_SOC_PCM3008 -- cgit v1.2.3-70-g09d2 From 383f8465d1579635d7b7df5d7a1d1966b6ffb868 Mon Sep 17 00:00:00 2001 From: Alexander Sverdlin Date: Mon, 7 Mar 2011 20:29:36 +0300 Subject: ASoC: Extend range of supported sample rates for CS4271 CODEC. Extend range of supported sample rates for CS4271 CODEC. Signed-off-by: Alexander Sverdlin Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/cs4271.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 41a3ee6f614..538e814136d 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -33,6 +33,7 @@ #define CS4271_PCM_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S24_LE | \ SNDRV_PCM_FMTBIT_S32_LE) +#define CS4271_PCM_RATES SNDRV_PCM_RATE_8000_192000 /* * CS4271 registers @@ -392,14 +393,14 @@ static struct snd_soc_dai_driver cs4271_dai = { .stream_name = "Playback", .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_96000, + .rates = CS4271_PCM_RATES, .formats = CS4271_PCM_FORMATS, }, .capture = { .stream_name = "Capture", .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_96000, + .rates = CS4271_PCM_RATES, .formats = CS4271_PCM_FORMATS, }, .ops = &cs4271_dai_ops, -- cgit v1.2.3-70-g09d2 From 149c7b441bc66b37325f0c494bfe9f303484bd88 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 7 Mar 2011 16:39:50 +0000 Subject: ASoC: Use data based init for WM9081 DAPM Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm9081.c | 11 +++++------ 1 file changed, 5 insertions(+), 6 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 7883f3ed797..8b1b2c9ab59 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -762,7 +762,7 @@ SND_SOC_DAPM_SUPPLY("TOCLK", WM9081_CLOCK_CONTROL_3, 2, 0, NULL, 0), }; -static const struct snd_soc_dapm_route audio_paths[] = { +static const struct snd_soc_dapm_route wm9081_audio_paths[] = { { "DAC", NULL, "CLK_SYS" }, { "DAC", NULL, "CLK_DSP" }, @@ -1232,7 +1232,6 @@ static struct snd_soc_dai_driver wm9081_dai = { static int wm9081_probe(struct snd_soc_codec *codec) { struct wm9081_priv *wm9081 = snd_soc_codec_get_drvdata(codec); - struct snd_soc_dapm_context *dapm = &codec->dapm; int ret; u16 reg; @@ -1282,10 +1281,6 @@ static int wm9081_probe(struct snd_soc_codec *codec) ARRAY_SIZE(wm9081_eq_controls)); } - snd_soc_dapm_new_controls(dapm, wm9081_dapm_widgets, - ARRAY_SIZE(wm9081_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, audio_paths, ARRAY_SIZE(audio_paths)); - return ret; } @@ -1334,6 +1329,10 @@ static struct snd_soc_codec_driver soc_codec_dev_wm9081 = { .reg_word_size = sizeof(u16), .reg_cache_default = wm9081_reg_defaults, .volatile_register = wm9081_volatile_register, + .dapm_widgets = wm9081_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm9081_dapm_widgets), + .dapm_routes = wm9081_audio_paths, + .num_dapm_routes = ARRAY_SIZE(wm9081_audio_paths), }; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) -- cgit v1.2.3-70-g09d2 From b993f92b99288d4b3a1a1237f3e40fa6460e4b47 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 7 Mar 2011 16:42:20 +0000 Subject: ASoC: Fix section mismatch warnings in WM8994 Annoying as the __devinitdata is actually correct. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8994-tables.c | 2 +- sound/soc/codecs/wm8994.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8994-tables.c b/sound/soc/codecs/wm8994-tables.c index 61dfd91c6c7..a87adbd05ee 100644 --- a/sound/soc/codecs/wm8994-tables.c +++ b/sound/soc/codecs/wm8994-tables.c @@ -1573,7 +1573,7 @@ const struct wm8994_access_mask wm8994_access_masks[WM8994_CACHE_SIZE] = { { 0x03C3, 0x03C3 }, /* R1569 - Sidetone */ }; -const __devinitdata u16 wm8994_reg_defaults[WM8994_CACHE_SIZE] = { +const u16 wm8994_reg_defaults[WM8994_CACHE_SIZE] = { 0x8994, /* R0 - Software Reset */ 0x0000, /* R1 - Power Management (1) */ 0x6000, /* R2 - Power Management (2) */ diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h index 0c355bfc88f..999b8851226 100644 --- a/sound/soc/codecs/wm8994.h +++ b/sound/soc/codecs/wm8994.h @@ -43,6 +43,6 @@ struct wm8994_access_mask { }; extern const struct wm8994_access_mask wm8994_access_masks[WM8994_CACHE_SIZE]; -extern const __devinitdata u16 wm8994_reg_defaults[WM8994_CACHE_SIZE]; +extern const u16 wm8994_reg_defaults[WM8994_CACHE_SIZE]; #endif -- cgit v1.2.3-70-g09d2 From 63d24b79b66ea5d5b04dadb9e92f31c0141948fb Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 7 Mar 2011 20:59:45 +0000 Subject: ASoC: Convert WM9081 SYSCLK configuration to be device wide Also respace the CODEC ops a bit for legibility. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm9081.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 8b1b2c9ab59..effaf75cacc 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -1140,10 +1140,9 @@ static int wm9081_digital_mute(struct snd_soc_dai *codec_dai, int mute) return 0; } -static int wm9081_set_sysclk(struct snd_soc_dai *codec_dai, +static int wm9081_set_sysclk(struct snd_soc_codec *codec, int clk_id, unsigned int freq, int dir) { - struct snd_soc_codec *codec = codec_dai->codec; struct wm9081_priv *wm9081 = snd_soc_codec_get_drvdata(codec); switch (clk_id) { @@ -1208,7 +1207,6 @@ static int wm9081_set_tdm_slot(struct snd_soc_dai *dai, static struct snd_soc_dai_ops wm9081_dai_ops = { .hw_params = wm9081_hw_params, - .set_sysclk = wm9081_set_sysclk, .set_fmt = wm9081_set_dai_fmt, .digital_mute = wm9081_digital_mute, .set_tdm_slot = wm9081_set_tdm_slot, @@ -1324,11 +1322,15 @@ static struct snd_soc_codec_driver soc_codec_dev_wm9081 = { .remove = wm9081_remove, .suspend = wm9081_suspend, .resume = wm9081_resume, + + .set_sysclk = wm9081_set_sysclk, .set_bias_level = wm9081_set_bias_level, + .reg_cache_size = ARRAY_SIZE(wm9081_reg_defaults), .reg_word_size = sizeof(u16), .reg_cache_default = wm9081_reg_defaults, .volatile_register = wm9081_volatile_register, + .dapm_widgets = wm9081_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(wm9081_dapm_widgets), .dapm_routes = wm9081_audio_paths, -- cgit v1.2.3-70-g09d2 From 62f75aafdf180554b4fad29ff1f3827b151d39db Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Tue, 8 Mar 2011 14:39:24 +0300 Subject: ASoC: sgtl5000: use after free in ldo_regulator_register() The "ldo" variable was dereferenced after free on the error path. Signed-off-by: Dan Carpenter Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index b7e97c02689..1f7217f703e 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -875,11 +875,13 @@ static int ldo_regulator_register(struct snd_soc_codec *codec, ldo->dev = regulator_register(&ldo->desc, codec->dev, init_data, ldo); if (IS_ERR(ldo->dev)) { + int ret = PTR_ERR(ldo->dev); + dev_err(codec->dev, "failed to register regulator\n"); kfree(ldo->desc.name); kfree(ldo); - return PTR_ERR(ldo->dev); + return ret; } return 0; -- cgit v1.2.3-70-g09d2 From 378a90f4540cc113e6ef36861ae914b0c63700a0 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 8 Mar 2011 18:52:08 +0000 Subject: ASoC: Simplify WM9081 speaker startup by using widgets for spaker output Now we have a register write minimisation code in DAPM we don't need to worry about the ordering of the enable and disable of the PGA and the output stage. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm9081.c | 32 ++++++-------------------------- 1 file changed, 6 insertions(+), 26 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index effaf75cacc..55cdf298202 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -389,27 +389,6 @@ SOC_DAPM_SINGLE("IN2 Switch", WM9081_ANALOGUE_MIXER, 2, 1, 0), SOC_DAPM_SINGLE("Playback Switch", WM9081_ANALOGUE_MIXER, 4, 1, 0), }; -static int speaker_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) -{ - struct snd_soc_codec *codec = w->codec; - unsigned int reg = snd_soc_read(codec, WM9081_POWER_MANAGEMENT); - - switch (event) { - case SND_SOC_DAPM_POST_PMU: - reg |= WM9081_SPK_ENA; - break; - - case SND_SOC_DAPM_PRE_PMD: - reg &= ~WM9081_SPK_ENA; - break; - } - - snd_soc_write(codec, WM9081_POWER_MANAGEMENT, reg); - - return 0; -} - struct _fll_div { u16 fll_fratio; u16 fll_outdiv; @@ -747,9 +726,8 @@ SND_SOC_DAPM_MIXER_NAMED_CTL("Mixer", SND_SOC_NOPM, 0, 0, SND_SOC_DAPM_PGA("LINEOUT PGA", WM9081_POWER_MANAGEMENT, 4, 0, NULL, 0), -SND_SOC_DAPM_PGA_E("Speaker PGA", WM9081_POWER_MANAGEMENT, 2, 0, NULL, 0, - speaker_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), +SND_SOC_DAPM_PGA("Speaker PGA", WM9081_POWER_MANAGEMENT, 2, 0, NULL, 0), +SND_SOC_DAPM_PGA("Speaker", WM9081_POWER_MANAGEMENT, 1, 0, NULL, 0), SND_SOC_DAPM_OUTPUT("LINEOUT"), SND_SOC_DAPM_OUTPUT("SPKN"), @@ -780,8 +758,10 @@ static const struct snd_soc_dapm_route wm9081_audio_paths[] = { { "Speaker PGA", NULL, "TOCLK" }, { "Speaker PGA", NULL, "CLK_SYS" }, - { "SPKN", NULL, "Speaker PGA" }, - { "SPKP", NULL, "Speaker PGA" }, + { "Speaker", NULL, "Speaker PGA" }, + + { "SPKN", NULL, "Speaker" }, + { "SPKP", NULL, "Speaker" }, }; static int wm9081_set_bias_level(struct snd_soc_codec *codec, -- cgit v1.2.3-70-g09d2 From 5c3a12e96c9f3158602b2feac8583ae35c10b80e Mon Sep 17 00:00:00 2001 From: Alexander Sverdlin Date: Mon, 7 Mar 2011 20:29:45 +0300 Subject: ASoC: Manage mode and rate bits correctly for CS4271 CODEC. Manage mode and rate bits correctly, according to datasheet in CS4271 CODEC. This is done to make capture work properly. Signed-off-by: Alexander Sverdlin Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/cs4271.c | 83 ++++++++++++++++++++++++++++++----------------- 1 file changed, 53 insertions(+), 30 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 538e814136d..083aab96ca8 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -168,27 +168,6 @@ struct cs4271_private { int gpio_disable; }; -struct cs4271_clk_cfg { - unsigned int ratio; /* MCLK / sample rate */ - u8 speed_mode; /* codec speed mode: 1x, 2x, 4x */ - u8 mclk_master; /* ratio bit mask for Master mode */ - u8 mclk_slave; /* ratio bit mask for Slave mode */ -}; - -static struct cs4271_clk_cfg cs4271_clk_tab[] = { - {64, CS4271_MODE1_MODE_4X, CS4271_MODE1_DIV_1, CS4271_MODE1_DIV_1}, - {96, CS4271_MODE1_MODE_4X, CS4271_MODE1_DIV_15, CS4271_MODE1_DIV_1}, - {128, CS4271_MODE1_MODE_2X, CS4271_MODE1_DIV_1, CS4271_MODE1_DIV_1}, - {192, CS4271_MODE1_MODE_2X, CS4271_MODE1_DIV_15, CS4271_MODE1_DIV_1}, - {256, CS4271_MODE1_MODE_1X, CS4271_MODE1_DIV_1, CS4271_MODE1_DIV_1}, - {384, CS4271_MODE1_MODE_1X, CS4271_MODE1_DIV_15, CS4271_MODE1_DIV_1}, - {512, CS4271_MODE1_MODE_1X, CS4271_MODE1_DIV_2, CS4271_MODE1_DIV_1}, - {768, CS4271_MODE1_MODE_1X, CS4271_MODE1_DIV_3, CS4271_MODE1_DIV_3}, - {1024, CS4271_MODE1_MODE_1X, CS4271_MODE1_DIV_3, CS4271_MODE1_DIV_3} -}; - -#define CS4171_NR_RATIOS ARRAY_SIZE(cs4271_clk_tab) - /* * @freq is the desired MCLK rate * MCLK rate should (c) be the sample rate, multiplied by one of the @@ -297,6 +276,45 @@ static int cs4271_put_deemph(struct snd_kcontrol *kcontrol, return cs4271_set_deemph(codec); } +struct cs4271_clk_cfg { + bool master; /* codec mode */ + u8 speed_mode; /* codec speed mode: 1x, 2x, 4x */ + unsigned short ratio; /* MCLK / sample rate */ + u8 ratio_mask; /* ratio bit mask for Master mode */ +}; + +static struct cs4271_clk_cfg cs4271_clk_tab[] = { + {1, CS4271_MODE1_MODE_1X, 256, CS4271_MODE1_DIV_1}, + {1, CS4271_MODE1_MODE_1X, 384, CS4271_MODE1_DIV_15}, + {1, CS4271_MODE1_MODE_1X, 512, CS4271_MODE1_DIV_2}, + {1, CS4271_MODE1_MODE_1X, 768, CS4271_MODE1_DIV_3}, + {1, CS4271_MODE1_MODE_2X, 128, CS4271_MODE1_DIV_1}, + {1, CS4271_MODE1_MODE_2X, 192, CS4271_MODE1_DIV_15}, + {1, CS4271_MODE1_MODE_2X, 256, CS4271_MODE1_DIV_2}, + {1, CS4271_MODE1_MODE_2X, 384, CS4271_MODE1_DIV_3}, + {1, CS4271_MODE1_MODE_4X, 64, CS4271_MODE1_DIV_1}, + {1, CS4271_MODE1_MODE_4X, 96, CS4271_MODE1_DIV_15}, + {1, CS4271_MODE1_MODE_4X, 128, CS4271_MODE1_DIV_2}, + {1, CS4271_MODE1_MODE_4X, 192, CS4271_MODE1_DIV_3}, + {0, CS4271_MODE1_MODE_1X, 256, CS4271_MODE1_DIV_1}, + {0, CS4271_MODE1_MODE_1X, 384, CS4271_MODE1_DIV_1}, + {0, CS4271_MODE1_MODE_1X, 512, CS4271_MODE1_DIV_1}, + {0, CS4271_MODE1_MODE_1X, 768, CS4271_MODE1_DIV_2}, + {0, CS4271_MODE1_MODE_1X, 1024, CS4271_MODE1_DIV_2}, + {0, CS4271_MODE1_MODE_2X, 128, CS4271_MODE1_DIV_1}, + {0, CS4271_MODE1_MODE_2X, 192, CS4271_MODE1_DIV_1}, + {0, CS4271_MODE1_MODE_2X, 256, CS4271_MODE1_DIV_1}, + {0, CS4271_MODE1_MODE_2X, 384, CS4271_MODE1_DIV_2}, + {0, CS4271_MODE1_MODE_2X, 512, CS4271_MODE1_DIV_2}, + {0, CS4271_MODE1_MODE_4X, 64, CS4271_MODE1_DIV_1}, + {0, CS4271_MODE1_MODE_4X, 96, CS4271_MODE1_DIV_1}, + {0, CS4271_MODE1_MODE_4X, 128, CS4271_MODE1_DIV_1}, + {0, CS4271_MODE1_MODE_4X, 192, CS4271_MODE1_DIV_2}, + {0, CS4271_MODE1_MODE_4X, 256, CS4271_MODE1_DIV_2}, +}; + +#define CS4171_NR_RATIOS ARRAY_SIZE(cs4271_clk_tab) + static int cs4271_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -308,23 +326,28 @@ static int cs4271_hw_params(struct snd_pcm_substream *substream, unsigned int ratio, val; cs4271->rate = params_rate(params); + + /* Configure DAC */ + if (cs4271->rate < 50000) + val = CS4271_MODE1_MODE_1X; + else if (cs4271->rate < 100000) + val = CS4271_MODE1_MODE_2X; + else + val = CS4271_MODE1_MODE_4X; + ratio = cs4271->mclk / cs4271->rate; for (i = 0; i < CS4171_NR_RATIOS; i++) - if (cs4271_clk_tab[i].ratio == ratio) + if ((cs4271_clk_tab[i].master == cs4271->master) && + (cs4271_clk_tab[i].speed_mode == val) && + (cs4271_clk_tab[i].ratio == ratio)) break; - if ((i == CS4171_NR_RATIOS) || ((ratio == 1024) && cs4271->master)) { + if (i == CS4171_NR_RATIOS) { dev_err(codec->dev, "Invalid sample rate\n"); return -EINVAL; } - /* Configure DAC */ - val = cs4271_clk_tab[i].speed_mode; - - if (cs4271->master) - val |= cs4271_clk_tab[i].mclk_master; - else - val |= cs4271_clk_tab[i].mclk_slave; + val |= cs4271_clk_tab[i].ratio_mask; ret = snd_soc_update_bits(codec, CS4271_MODE1, CS4271_MODE1_MODE_MASK | CS4271_MODE1_DIV_MASK, val); -- cgit v1.2.3-70-g09d2 From 0e45cab64449660fe83bb71208ab43b8d22a5648 Mon Sep 17 00:00:00 2001 From: Christian Glindkamp Date: Wed, 9 Mar 2011 11:20:04 +0100 Subject: ASoC: Add MAX9850 codec driver This patch adds ASoC support for the MAX9850 codec with headphone amplifier. Supported features: - Playback - 16, 20 and 24 bit audio - 8k - 48k sample rates - DAPM Signed-off-by: Christian Glindkamp Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/max9850.c | 389 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/max9850.h | 38 +++++ 4 files changed, 433 insertions(+) create mode 100644 sound/soc/codecs/max9850.c create mode 100644 sound/soc/codecs/max9850.h (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 10f11dd98c2..d63c1754e05 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -33,6 +33,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_JZ4740_CODEC if SOC_JZ4740 select SND_SOC_LM4857 if I2C select SND_SOC_MAX98088 if I2C + select SND_SOC_MAX9850 if I2C select SND_SOC_MAX9877 if I2C select SND_SOC_PCM3008 select SND_SOC_SGTL5000 if I2C @@ -186,6 +187,9 @@ config SND_SOC_DMIC config SND_SOC_MAX98088 tristate +config SND_SOC_MAX9850 + tristate + config SND_SOC_PCM3008 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index ebb059c0eef..379bc55f072 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -19,6 +19,7 @@ snd-soc-dfbmcs320-objs := dfbmcs320.o snd-soc-dmic-objs := dmic.o snd-soc-l3-objs := l3.o snd-soc-max98088-objs := max98088.o +snd-soc-max9850-objs := max9850.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-sgtl5000-objs := sgtl5000.o snd-soc-alc5623-objs := alc5623.o @@ -107,6 +108,7 @@ obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o +obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o obj-$(CONFIG_SND_SOC_SN95031) +=snd-soc-sn95031.o diff --git a/sound/soc/codecs/max9850.c b/sound/soc/codecs/max9850.c new file mode 100644 index 00000000000..6d2c5a37550 --- /dev/null +++ b/sound/soc/codecs/max9850.c @@ -0,0 +1,389 @@ +/* + * max9850.c -- codec driver for max9850 + * + * Copyright (C) 2011 taskit GmbH + * + * Author: Christian Glindkamp + * + * Initial development of this code was funded by + * MICRONIC Computer Systeme GmbH, http://www.mcsberlin.de/ + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#include +#include +#include +#include +#include +#include +#include +#include + +#include "max9850.h" + +struct max9850_priv { + unsigned int sysclk; +}; + +/* max9850 register cache */ +static const u8 max9850_reg[MAX9850_CACHEREGNUM] = { + 0x00, 0x00, 0x0c, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00 +}; + +/* these registers are not used at the moment but provided for the sake of + * completeness */ +static int max9850_volatile_register(struct snd_soc_codec *codec, + unsigned int reg) +{ + switch (reg) { + case MAX9850_STATUSA: + case MAX9850_STATUSB: + return 1; + default: + return 0; + } +} + +static const unsigned int max9850_tlv[] = { + TLV_DB_RANGE_HEAD(4), + 0x18, 0x1f, TLV_DB_SCALE_ITEM(-7450, 400, 0), + 0x20, 0x33, TLV_DB_SCALE_ITEM(-4150, 200, 0), + 0x34, 0x37, TLV_DB_SCALE_ITEM(-150, 100, 0), + 0x38, 0x3f, TLV_DB_SCALE_ITEM(250, 50, 0), +}; + +static const struct snd_kcontrol_new max9850_controls[] = { +SOC_SINGLE_TLV("Headphone Volume", MAX9850_VOLUME, 0, 0x3f, 1, max9850_tlv), +SOC_SINGLE("Headphone Switch", MAX9850_VOLUME, 7, 1, 1), +SOC_SINGLE("Mono Switch", MAX9850_GENERAL_PURPOSE, 2, 1, 0), +}; + +static const struct snd_kcontrol_new max9850_mixer_controls[] = { + SOC_DAPM_SINGLE("Line In Switch", MAX9850_ENABLE, 1, 1, 0), +}; + +static const struct snd_soc_dapm_widget max9850_dapm_widgets[] = { +SND_SOC_DAPM_SUPPLY("Charge Pump 1", MAX9850_ENABLE, 4, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("Charge Pump 2", MAX9850_ENABLE, 5, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("MCLK", MAX9850_ENABLE, 6, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("SHDN", MAX9850_ENABLE, 7, 0, NULL, 0), +SND_SOC_DAPM_MIXER_NAMED_CTL("Output Mixer", MAX9850_ENABLE, 2, 0, + &max9850_mixer_controls[0], + ARRAY_SIZE(max9850_mixer_controls)), +SND_SOC_DAPM_PGA("Headphone Output", MAX9850_ENABLE, 3, 0, NULL, 0), +SND_SOC_DAPM_DAC("DAC", "HiFi Playback", MAX9850_ENABLE, 0, 0), +SND_SOC_DAPM_OUTPUT("OUTL"), +SND_SOC_DAPM_OUTPUT("HPL"), +SND_SOC_DAPM_OUTPUT("OUTR"), +SND_SOC_DAPM_OUTPUT("HPR"), +SND_SOC_DAPM_MIXER("Line Input", SND_SOC_NOPM, 0, 0, NULL, 0), +SND_SOC_DAPM_INPUT("INL"), +SND_SOC_DAPM_INPUT("INR"), +}; + +static const struct snd_soc_dapm_route intercon[] = { + /* output mixer */ + {"Output Mixer", NULL, "DAC"}, + {"Output Mixer", "Line In Switch", "Line Input"}, + + /* outputs */ + {"Headphone Output", NULL, "Output Mixer"}, + {"HPL", NULL, "Headphone Output"}, + {"HPR", NULL, "Headphone Output"}, + {"OUTL", NULL, "Output Mixer"}, + {"OUTR", NULL, "Output Mixer"}, + + /* inputs */ + {"Line Input", NULL, "INL"}, + {"Line Input", NULL, "INR"}, + + /* supplies */ + {"Output Mixer", NULL, "Charge Pump 1"}, + {"Output Mixer", NULL, "Charge Pump 2"}, + {"Output Mixer", NULL, "SHDN"}, + {"DAC", NULL, "MCLK"}, +}; + +static int max9850_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct max9850_priv *max9850 = snd_soc_codec_get_drvdata(codec); + u64 lrclk_div; + u8 sf, da; + + if(!max9850->sysclk) + return -EINVAL; + + /* lrclk_div = 2^22 * rate / iclk with iclk = mclk / sf */ + sf = (snd_soc_read(codec, MAX9850_CLOCK) >> 2) + 1; + lrclk_div = (1 << 22); + lrclk_div *= params_rate(params); + lrclk_div *= sf; + do_div(lrclk_div, max9850->sysclk); + + snd_soc_write(codec, MAX9850_LRCLK_MSB, (lrclk_div >> 8) & 0x7f); + snd_soc_write(codec, MAX9850_LRCLK_LSB, lrclk_div & 0xff); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + da = 0; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + da = 0x2; + break; + case SNDRV_PCM_FORMAT_S24_LE: + da = 0x3; + break; + default: + return -EINVAL; + } + snd_soc_update_bits(codec, MAX9850_DIGITAL_AUDIO, 0x3, da); + + return 0; +} + +static int max9850_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 max9850_priv *max9850 = snd_soc_codec_get_drvdata(codec); + + /* calculate mclk -> iclk divider */ + if (freq <= 13000000) + snd_soc_write(codec, MAX9850_CLOCK, 0x0); + else if (freq <= 26000000) + snd_soc_write(codec, MAX9850_CLOCK, 0x4); + else if (freq <= 40000000) + snd_soc_write(codec, MAX9850_CLOCK, 0x8); + else + return -EINVAL; + + max9850->sysclk = freq; + return 0; +} + +static int max9850_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u8 da = 0; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + da |= MAX9850_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: + da |= MAX9850_DLY; + break; + case SND_SOC_DAIFMT_RIGHT_J: + da |= MAX9850_RTJ; + break; + case SND_SOC_DAIFMT_LEFT_J: + 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: + da |= MAX9850_BCINV | MAX9850_INV; + break; + case SND_SOC_DAIFMT_IB_NF: + da |= MAX9850_BCINV; + break; + case SND_SOC_DAIFMT_NB_IF: + da |= MAX9850_INV; + break; + default: + return -EINVAL; + } + + /* set da */ + snd_soc_write(codec, MAX9850_DIGITAL_AUDIO, da); + + return 0; +} + +static int max9850_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + int ret; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + ret = snd_soc_cache_sync(codec); + if (ret) { + dev_err(codec->dev, + "Failed to sync cache: %d\n", ret); + return ret; + } + } + break; + case SND_SOC_BIAS_OFF: + break; + } + codec->dapm.bias_level = level; + return 0; +} + +#define MAX9850_RATES SNDRV_PCM_RATE_8000_48000 + +#define MAX9850_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_ops max9850_dai_ops = { + .hw_params = max9850_hw_params, + .set_sysclk = max9850_set_dai_sysclk, + .set_fmt = max9850_set_dai_fmt, +}; + +static struct snd_soc_dai_driver max9850_dai = { + .name = "max9850-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = MAX9850_RATES, + .formats = MAX9850_FORMATS + }, + .ops = &max9850_dai_ops, +}; + +#ifdef CONFIG_PM +static int max9850_suspend(struct snd_soc_codec *codec, pm_message_t state) +{ + max9850_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int max9850_resume(struct snd_soc_codec *codec) +{ + max9850_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} +#else +#define max9850_suspend NULL +#define max9850_resume NULL +#endif + +static int max9850_probe(struct snd_soc_codec *codec) +{ + struct snd_soc_dapm_context *dapm = &codec->dapm; + int ret; + + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + /* enable zero-detect */ + snd_soc_update_bits(codec, MAX9850_GENERAL_PURPOSE, 1, 1); + /* enable slew-rate control */ + snd_soc_update_bits(codec, MAX9850_VOLUME, 0x40, 0x40); + /* set slew-rate 125ms */ + snd_soc_update_bits(codec, MAX9850_CHARGE_PUMP, 0xff, 0xc0); + + snd_soc_dapm_new_controls(dapm, max9850_dapm_widgets, + ARRAY_SIZE(max9850_dapm_widgets)); + snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); + + snd_soc_add_controls(codec, max9850_controls, + ARRAY_SIZE(max9850_controls)); + + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_max9850 = { + .probe = max9850_probe, + .suspend = max9850_suspend, + .resume = max9850_resume, + .set_bias_level = max9850_set_bias_level, + .reg_cache_size = ARRAY_SIZE(max9850_reg), + .reg_word_size = sizeof(u8), + .reg_cache_default = max9850_reg, + .volatile_register = max9850_volatile_register, +}; + +static int __devinit max9850_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct max9850_priv *max9850; + int ret; + + max9850 = kzalloc(sizeof(struct max9850_priv), GFP_KERNEL); + if (max9850 == NULL) + return -ENOMEM; + + i2c_set_clientdata(i2c, max9850); + + ret = snd_soc_register_codec(&i2c->dev, + &soc_codec_dev_max9850, &max9850_dai, 1); + if (ret < 0) + kfree(max9850); + return ret; +} + +static __devexit int max9850_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + kfree(i2c_get_clientdata(client)); + return 0; +} + +static const struct i2c_device_id max9850_i2c_id[] = { + { "max9850", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, max9850_i2c_id); + +static struct i2c_driver max9850_i2c_driver = { + .driver = { + .name = "max9850", + .owner = THIS_MODULE, + }, + .probe = max9850_i2c_probe, + .remove = __devexit_p(max9850_i2c_remove), + .id_table = max9850_i2c_id, +}; + +static int __init max9850_init(void) +{ + return i2c_add_driver(&max9850_i2c_driver); +} +module_init(max9850_init); + +static void __exit max9850_exit(void) +{ + i2c_del_driver(&max9850_i2c_driver); +} +module_exit(max9850_exit); + +MODULE_AUTHOR("Christian Glindkamp "); +MODULE_DESCRIPTION("ASoC MAX9850 codec driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/max9850.h b/sound/soc/codecs/max9850.h new file mode 100644 index 00000000000..72b1ddb04b0 --- /dev/null +++ b/sound/soc/codecs/max9850.h @@ -0,0 +1,38 @@ +/* + * max9850.h -- codec driver for max9850 + * + * Copyright (C) 2011 taskit GmbH + * Author: Christian Glindkamp + * + * 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. + * + */ + +#ifndef _MAX9850_H +#define _MAX9850_H + +#define MAX9850_STATUSA 0x00 +#define MAX9850_STATUSB 0x01 +#define MAX9850_VOLUME 0x02 +#define MAX9850_GENERAL_PURPOSE 0x03 +#define MAX9850_INTERRUPT 0x04 +#define MAX9850_ENABLE 0x05 +#define MAX9850_CLOCK 0x06 +#define MAX9850_CHARGE_PUMP 0x07 +#define MAX9850_LRCLK_MSB 0x08 +#define MAX9850_LRCLK_LSB 0x09 +#define MAX9850_DIGITAL_AUDIO 0x0a + +#define MAX9850_CACHEREGNUM 11 + +/* MAX9850_DIGITAL_AUDIO */ +#define MAX9850_MASTER (1<<7) +#define MAX9850_INV (1<<6) +#define MAX9850_BCINV (1<<5) +#define MAX9850_DLY (1<<3) +#define MAX9850_RTJ (1<<2) + +#endif -- cgit v1.2.3-70-g09d2 From 27380fb83079bc7bd644e1115bb001dfdcec307f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 11 Mar 2011 12:07:31 +0000 Subject: ASoC: Fix spacing in MAX8950 Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/max9850.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/max9850.c b/sound/soc/codecs/max9850.c index 6d2c5a37550..208d2ee6185 100644 --- a/sound/soc/codecs/max9850.c +++ b/sound/soc/codecs/max9850.c @@ -118,7 +118,7 @@ static int max9850_hw_params(struct snd_pcm_substream *substream, u64 lrclk_div; u8 sf, da; - if(!max9850->sysclk) + if (!max9850->sysclk) return -EINVAL; /* lrclk_div = 2^22 * rate / iclk with iclk = mclk / sf */ -- cgit v1.2.3-70-g09d2