diff options
Diffstat (limited to 'sound')
117 files changed, 4643 insertions, 1211 deletions
diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 3e598e756e54..4562c898a7ef 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -4,6 +4,8 @@ menuconfig SND_SOC tristate "ALSA for SoC audio support" + select LZO_COMPRESS + select LZO_DECOMPRESS select SND_PCM select AC97_BUS if SND_SOC_AC97_BUS select SND_JACK if INPUT=y || INPUT=SND diff --git a/sound/soc/atmel/playpaq_wm8510.c b/sound/soc/atmel/playpaq_wm8510.c index 5f4e59f4461c..aede7e74ec34 100644 --- a/sound/soc/atmel/playpaq_wm8510.c +++ b/sound/soc/atmel/playpaq_wm8510.c @@ -318,27 +318,28 @@ static const struct snd_soc_dapm_route intercon[] = { static int playpaq_wm8510_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; int i; /* * Add DAPM widgets */ for (i = 0; i < ARRAY_SIZE(playpaq_dapm_widgets); i++) - snd_soc_dapm_new_control(codec, &playpaq_dapm_widgets[i]); + snd_soc_dapm_new_control(dapm, &playpaq_dapm_widgets[i]); /* * Setup audio path interconnects */ - snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); /* always connected pins */ - snd_soc_dapm_enable_pin(codec, "Int Mic"); - snd_soc_dapm_enable_pin(codec, "Ext Spk"); - snd_soc_dapm_sync(codec); + snd_soc_dapm_enable_pin(dapm, "Int Mic"); + snd_soc_dapm_enable_pin(dapm, "Ext Spk"); + snd_soc_dapm_sync(dapm); diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index 293569dfd0ed..da9c3037496f 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -140,6 +140,7 @@ static int at91sam9g20ek_wm8731_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dapm_context *dapm = &codec->dapm; int ret; printk(KERN_DEBUG @@ -154,25 +155,25 @@ static int at91sam9g20ek_wm8731_init(struct snd_soc_pcm_runtime *rtd) } /* Add specific widgets */ - snd_soc_dapm_new_controls(codec, at91sam9g20ek_dapm_widgets, + snd_soc_dapm_new_controls(dapm, at91sam9g20ek_dapm_widgets, ARRAY_SIZE(at91sam9g20ek_dapm_widgets)); /* Set up specific audio path interconnects */ - snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); /* not connected */ - snd_soc_dapm_nc_pin(codec, "RLINEIN"); - snd_soc_dapm_nc_pin(codec, "LLINEIN"); + snd_soc_dapm_nc_pin(dapm, "RLINEIN"); + snd_soc_dapm_nc_pin(dapm, "LLINEIN"); #ifdef ENABLE_MIC_INPUT - snd_soc_dapm_enable_pin(codec, "Int Mic"); + snd_soc_dapm_enable_pin(dapm, "Int Mic"); #else - snd_soc_dapm_nc_pin(codec, "Int Mic"); + snd_soc_dapm_nc_pin(dapm, "Int Mic"); #endif /* always connected */ - snd_soc_dapm_enable_pin(codec, "Ext Spk"); + snd_soc_dapm_enable_pin(dapm, "Ext Spk"); - snd_soc_dapm_sync(codec); + snd_soc_dapm_sync(dapm); return 0; } diff --git a/sound/soc/atmel/snd-soc-afeb9260.c b/sound/soc/atmel/snd-soc-afeb9260.c index e3d283561c19..92c709ed0965 100644 --- a/sound/soc/atmel/snd-soc-afeb9260.c +++ b/sound/soc/atmel/snd-soc-afeb9260.c @@ -105,19 +105,20 @@ static const struct snd_soc_dapm_route audio_map[] = { static int afeb9260_tlv320aic23_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; /* Add afeb9260 specific widgets */ - snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets, + snd_soc_dapm_new_controls(dapm, tlv320aic23_dapm_widgets, ARRAY_SIZE(tlv320aic23_dapm_widgets)); /* Set up afeb9260 specific audio path audio_map */ - snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_enable_pin(codec, "Headphone Jack"); - snd_soc_dapm_enable_pin(codec, "Line In"); - snd_soc_dapm_enable_pin(codec, "Mic Jack"); + snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); + snd_soc_dapm_enable_pin(dapm, "Line In"); + snd_soc_dapm_enable_pin(dapm, "Mic Jack"); - snd_soc_dapm_sync(codec); + snd_soc_dapm_sync(dapm); return 0; } diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index 01d19e9f53f9..a15a3e974f0d 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -1172,7 +1172,7 @@ static int pm860x_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: - if (codec->bias_level == SND_SOC_BIAS_OFF) { + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { /* Enable Audio PLL & Audio section */ data = AUDIO_PLL | AUDIO_SECTION_RESET | AUDIO_SECTION_ON; @@ -1185,7 +1185,7 @@ static int pm860x_set_bias_level(struct snd_soc_codec *codec, pm860x_set_bits(codec->control_data, REG_MISC2, data, 0); break; } - codec->bias_level = level; + codec->dapm.bias_level = level; return 0; } @@ -1346,6 +1346,7 @@ EXPORT_SYMBOL_GPL(pm860x_mic_jack_detect); static int pm860x_probe(struct snd_soc_codec *codec) { struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec); + struct snd_soc_dapm_context *dapm = &codec->dapm; int i, ret; pm860x->codec = codec; @@ -1374,9 +1375,9 @@ static int pm860x_probe(struct snd_soc_codec *codec) snd_soc_add_controls(codec, pm860x_snd_controls, ARRAY_SIZE(pm860x_snd_controls)); - snd_soc_dapm_new_controls(codec, pm860x_dapm_widgets, + snd_soc_dapm_new_controls(dapm, pm860x_dapm_widgets, ARRAY_SIZE(pm860x_dapm_widgets)); - snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); return 0; out_codec: diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 3b5690d28b8b..6ebd3a685b2c 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -22,6 +22,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AK4535 if I2C select SND_SOC_AK4642 if I2C select SND_SOC_AK4671 if I2C + select SND_SOC_ALC5623 if I2C select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC select SND_SOC_CS42L51 if I2C select SND_SOC_CS4270 if I2C @@ -57,6 +58,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM8741 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8750 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8753 if SND_SOC_I2C_AND_SPI + select SND_SOC_WM8770 if SPI_MASTER select SND_SOC_WM8776 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8804 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8900 if I2C @@ -130,6 +132,9 @@ config SND_SOC_AK4642 config SND_SOC_AK4671 tristate +config SND_SOC_ALC5623 + tristate + config SND_SOC_CQ0093VC tristate @@ -240,6 +245,9 @@ config SND_SOC_WM8750 config SND_SOC_WM8753 tristate +config SND_SOC_WM8770 + tristate + config SND_SOC_WM8776 tristate @@ -318,3 +326,4 @@ config SND_SOC_WM2000 config SND_SOC_WM9090 tristate + diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index f67a2d6f7a46..42f185d233f3 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -17,6 +17,7 @@ snd-soc-da7210-objs := da7210.o 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-spdif-objs := spdif_transciever.o snd-soc-ssm2602-objs := ssm2602.o snd-soc-stac9766-objs := stac9766.o @@ -41,6 +42,7 @@ snd-soc-wm8731-objs := wm8731.o snd-soc-wm8741-objs := wm8741.o snd-soc-wm8750-objs := wm8750.o snd-soc-wm8753-objs := wm8753.o +snd-soc-wm8770-objs := wm8770.o snd-soc-wm8776-objs := wm8776.o snd-soc-wm8804-objs := wm8804.o snd-soc-wm8900-objs := wm8900.o @@ -92,6 +94,7 @@ 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-$(CONFIG_SND_SOC_ALC5623) += snd-soc-alc5623.o obj-$(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 @@ -116,6 +119,7 @@ obj-$(CONFIG_SND_SOC_WM8731) += snd-soc-wm8731.o obj-$(CONFIG_SND_SOC_WM8741) += snd-soc-wm8741.o obj-$(CONFIG_SND_SOC_WM8750) += snd-soc-wm8750.o obj-$(CONFIG_SND_SOC_WM8753) += snd-soc-wm8753.o +obj-$(CONFIG_SND_SOC_WM8770) += snd-soc-wm8770.o obj-$(CONFIG_SND_SOC_WM8776) += snd-soc-wm8776.o obj-$(CONFIG_SND_SOC_WM8804) += snd-soc-wm8804.o obj-$(CONFIG_SND_SOC_WM8900) += snd-soc-wm8900.o diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c index d272534c8f84..c71b05ddd752 100644 --- a/sound/soc/codecs/ad1836.c +++ b/sound/soc/codecs/ad1836.c @@ -220,6 +220,7 @@ static struct snd_soc_dai_driver ad1836_dai = { static int ad1836_probe(struct snd_soc_codec *codec) { struct ad1836_priv *ad1836 = snd_soc_codec_get_drvdata(codec); + struct snd_soc_dapm_context *dapm = &codec->dapm; int ret = 0; codec->control_data = ad1836->control_data; @@ -252,9 +253,9 @@ static int ad1836_probe(struct snd_soc_codec *codec) snd_soc_add_controls(codec, ad1836_snd_controls, ARRAY_SIZE(ad1836_snd_controls)); - snd_soc_dapm_new_controls(codec, ad1836_dapm_widgets, + snd_soc_dapm_new_controls(dapm, ad1836_dapm_widgets, ARRAY_SIZE(ad1836_dapm_widgets)); - snd_soc_dapm_add_routes(codec, audio_paths, ARRAY_SIZE(audio_paths)); + snd_soc_dapm_add_routes(dapm, audio_paths, ARRAY_SIZE(audio_paths)); return ret; } diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index fa2834c91b9f..dc105d8aaa0f 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -353,6 +353,7 @@ static struct snd_soc_dai_driver ad193x_dai = { static int ad193x_probe(struct snd_soc_codec *codec) { struct ad193x_priv *ad193x = snd_soc_codec_get_drvdata(codec); + struct snd_soc_dapm_context *dapm = &codec->dapm; int ret; codec->control_data = ad193x->control_data; @@ -385,9 +386,9 @@ static int ad193x_probe(struct snd_soc_codec *codec) snd_soc_add_controls(codec, ad193x_snd_controls, ARRAY_SIZE(ad193x_snd_controls)); - snd_soc_dapm_new_controls(codec, ad193x_dapm_widgets, + snd_soc_dapm_new_controls(dapm, ad193x_dapm_widgets, ARRAY_SIZE(ad193x_dapm_widgets)); - snd_soc_dapm_add_routes(codec, audio_paths, ARRAY_SIZE(audio_paths)); + snd_soc_dapm_add_routes(dapm, audio_paths, ARRAY_SIZE(audio_paths)); return ret; } diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index cd88c8f32a38..52abb93a7dce 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -290,10 +290,11 @@ static const struct snd_soc_dapm_route audio_map[] = { static int ak4535_add_widgets(struct snd_soc_codec *codec) { - snd_soc_dapm_new_controls(codec, ak4535_dapm_widgets, - ARRAY_SIZE(ak4535_dapm_widgets)); + struct snd_soc_dapm_context *dapm = &codec->dapm; - snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_new_controls(dapm, ak4535_dapm_widgets, + ARRAY_SIZE(ak4535_dapm_widgets)); + snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); return 0; } @@ -399,7 +400,7 @@ static int ak4535_set_bias_level(struct snd_soc_codec *codec, ak4535_write(codec, AK4535_PM1, i & (~0x80)); break; } - codec->bias_level = level; + codec->dapm.bias_level = level; return 0; } diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 90c90b7f4a2e..f00eba313dfd 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -26,7 +26,7 @@ #include <linux/i2c.h> #include <linux/platform_device.h> #include <linux/slab.h> -#include <sound/soc-dapm.h> +#include <sound/soc.h> #include <sound/initval.h> #include <sound/tlv.h> diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c index 24f5f49bb9d2..1d6573c38af4 100644 --- a/sound/soc/codecs/ak4671.c +++ b/sound/soc/codecs/ak4671.c @@ -437,10 +437,11 @@ static const struct snd_soc_dapm_route intercon[] = { static int ak4671_add_widgets(struct snd_soc_codec *codec) { - snd_soc_dapm_new_controls(codec, ak4671_dapm_widgets, - ARRAY_SIZE(ak4671_dapm_widgets)); + struct snd_soc_dapm_context *dapm = &codec->dapm; - snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + snd_soc_dapm_new_controls(dapm, ak4671_dapm_widgets, + ARRAY_SIZE(ak4671_dapm_widgets)); + snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); return 0; } @@ -602,7 +603,7 @@ static int ak4671_set_bias_level(struct snd_soc_codec *codec, snd_soc_write(codec, AK4671_AD_DA_POWER_MANAGEMENT, 0x00); break; } - codec->bias_level = level; + codec->dapm.bias_level = level; return 0; } diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c new file mode 100644 index 000000000000..5a45067b43ba --- /dev/null +++ b/sound/soc/codecs/alc5623.c @@ -0,0 +1,1119 @@ +/* + * alc5623.c -- alc562[123] ALSA Soc Audio driver + * + * Copyright 2008 Realtek Microelectronics + * Author: flove <flove@realtek.com> Ethan <eku@marvell.com> + * + * Copyright 2010 Arnaud Patard <arnaud.patard@rtp-net.org> + * + * + * Based on WM8753.c + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + */ + +#include <linux/module.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/slab.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/tlv.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/alc5623.h> + +#include "alc5623.h" + +static int caps_charge = 2000; +module_param(caps_charge, int, 0); +MODULE_PARM_DESC(caps_charge, "ALC5623 cap charge time (msecs)"); + +/* codec private data */ +struct alc5623_priv { + enum snd_soc_control_type control_type; + void *control_data; + struct mutex mutex; + u8 id; + unsigned int sysclk; + u16 reg_cache[ALC5623_VENDOR_ID2+2]; + unsigned int add_ctrl; + unsigned int jack_det_ctrl; +}; + +static void alc5623_fill_cache(struct snd_soc_codec *codec) +{ + int i, step = codec->driver->reg_cache_step; + u16 *cache = codec->reg_cache; + + /* not really efficient ... */ + for (i = 0 ; i < codec->driver->reg_cache_size ; i += step) + cache[i] = codec->hw_read(codec, i); +} + +static inline int alc5623_reset(struct snd_soc_codec *codec) +{ + return snd_soc_write(codec, ALC5623_RESET, 0); +} + +static int amp_mixer_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + /* to power-on/off class-d amp generators/speaker */ + /* need to write to 'index-46h' register : */ + /* so write index num (here 0x46) to reg 0x6a */ + /* and then 0xffff/0 to reg 0x6c */ + snd_soc_write(w->codec, ALC5623_HID_CTRL_INDEX, 0x46); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + snd_soc_write(w->codec, ALC5623_HID_CTRL_DATA, 0xFFFF); + break; + case SND_SOC_DAPM_POST_PMD: + snd_soc_write(w->codec, ALC5623_HID_CTRL_DATA, 0); + break; + } + + return 0; +} + +/* + * ALC5623 Controls + */ + +static const DECLARE_TLV_DB_SCALE(vol_tlv, -3450, 150, 0); +static const DECLARE_TLV_DB_SCALE(hp_tlv, -4650, 150, 0); +static const DECLARE_TLV_DB_SCALE(adc_rec_tlv, -1650, 150, 0); +static const unsigned int boost_tlv[] = { + TLV_DB_RANGE_HEAD(3), + 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), + 1, 1, TLV_DB_SCALE_ITEM(2000, 0, 0), + 2, 2, TLV_DB_SCALE_ITEM(3000, 0, 0), +}; +static const DECLARE_TLV_DB_SCALE(dig_tlv, 0, 600, 0); + +static const struct snd_kcontrol_new rt5621_vol_snd_controls[] = { + SOC_DOUBLE_TLV("Speaker Playback Volume", + ALC5623_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv), + SOC_DOUBLE("Speaker Playback Switch", + ALC5623_SPK_OUT_VOL, 15, 7, 1, 1), + SOC_DOUBLE_TLV("Headphone Playback Volume", + ALC5623_HP_OUT_VOL, 8, 0, 31, 1, hp_tlv), + SOC_DOUBLE("Headphone Playback Switch", + ALC5623_HP_OUT_VOL, 15, 7, 1, 1), +}; + +static const struct snd_kcontrol_new rt5622_vol_snd_controls[] = { + SOC_DOUBLE_TLV("Speaker Playback Volume", + ALC5623_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv), + SOC_DOUBLE("Speaker Playback Switch", + ALC5623_SPK_OUT_VOL, 15, 7, 1, 1), + SOC_DOUBLE_TLV("Line Playback Volume", + ALC5623_HP_OUT_VOL, 8, 0, 31, 1, hp_tlv), + SOC_DOUBLE("Line Playback Switch", + ALC5623_HP_OUT_VOL, 15, 7, 1, 1), +}; + +static const struct snd_kcontrol_new alc5623_vol_snd_controls[] = { + SOC_DOUBLE_TLV("Line Playback Volume", + ALC5623_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv), + SOC_DOUBLE("Line Playback Switch", + ALC5623_SPK_OUT_VOL, 15, 7, 1, 1), + SOC_DOUBLE_TLV("Headphone Playback Volume", + ALC5623_HP_OUT_VOL, 8, 0, 31, 1, hp_tlv), + SOC_DOUBLE("Headphone Playback Switch", + ALC5623_HP_OUT_VOL, 15, 7, 1, 1), +}; + +static const struct snd_kcontrol_new alc5623_snd_controls[] = { + SOC_DOUBLE_TLV("Auxout Playback Volume", + ALC5623_MONO_AUX_OUT_VOL, 8, 0, 31, 1, hp_tlv), + SOC_DOUBLE("Auxout Playback Switch", + ALC5623_MONO_AUX_OUT_VOL, 15, 7, 1, 1), + SOC_DOUBLE_TLV("PCM Playback Volume", + ALC5623_STEREO_DAC_VOL, 8, 0, 31, 1, vol_tlv), + SOC_DOUBLE_TLV("AuxI Capture Volume", + ALC5623_AUXIN_VOL, 8, 0, 31, 1, vol_tlv), + SOC_DOUBLE_TLV("LineIn Capture Volume", + ALC5623_LINE_IN_VOL, 8, 0, 31, 1, vol_tlv), + SOC_SINGLE_TLV("Mic1 Capture Volume", + ALC5623_MIC_VOL, 8, 31, 1, vol_tlv), + SOC_SINGLE_TLV("Mic2 Capture Volume", + ALC5623_MIC_VOL, 0, 31, 1, vol_tlv), + SOC_DOUBLE_TLV("Rec Capture Volume", + ALC5623_ADC_REC_GAIN, 7, 0, 31, 0, adc_rec_tlv), + SOC_SINGLE_TLV("Mic 1 Boost Volume", + ALC5623_MIC_CTRL, 10, 2, 0, boost_tlv), + SOC_SINGLE_TLV("Mic 2 Boost Volume", + ALC5623_MIC_CTRL, 8, 2, 0, boost_tlv), + SOC_SINGLE_TLV("Digital Boost Volume", + ALC5623_ADD_CTRL_REG, 4, 3, 0, dig_tlv), +}; + +/* + * DAPM Controls + */ +static const struct snd_kcontrol_new alc5623_hp_mixer_controls[] = { +SOC_DAPM_SINGLE("LI2HP Playback Switch", ALC5623_LINE_IN_VOL, 15, 1, 1), +SOC_DAPM_SINGLE("AUXI2HP Playback Switch", ALC5623_AUXIN_VOL, 15, 1, 1), +SOC_DAPM_SINGLE("MIC12HP Playback Switch", ALC5623_MIC_ROUTING_CTRL, 15, 1, 1), +SOC_DAPM_SINGLE("MIC22HP Playback Switch", ALC5623_MIC_ROUTING_CTRL, 7, 1, 1), +SOC_DAPM_SINGLE("DAC2HP Playback Switch", ALC5623_STEREO_DAC_VOL, 15, 1, 1), +}; + +static const struct snd_kcontrol_new alc5623_hpl_mixer_controls[] = { +SOC_DAPM_SINGLE("ADC2HP_L Playback Switch", ALC5623_ADC_REC_GAIN, 15, 1, 1), +}; + +static const struct snd_kcontrol_new alc5623_hpr_mixer_controls[] = { +SOC_DAPM_SINGLE("ADC2HP_R Playback Switch", ALC5623_ADC_REC_GAIN, 14, 1, 1), +}; + +static const struct snd_kcontrol_new alc5623_mono_mixer_controls[] = { +SOC_DAPM_SINGLE("ADC2MONO_L Playback Switch", ALC5623_ADC_REC_GAIN, 13, 1, 1), +SOC_DAPM_SINGLE("ADC2MONO_R Playback Switch", ALC5623_ADC_REC_GAIN, 12, 1, 1), +SOC_DAPM_SINGLE("LI2MONO Playback Switch", ALC5623_LINE_IN_VOL, 13, 1, 1), +SOC_DAPM_SINGLE("AUXI2MONO Playback Switch", ALC5623_AUXIN_VOL, 13, 1, 1), +SOC_DAPM_SINGLE("MIC12MONO Playback Switch", ALC5623_MIC_ROUTING_CTRL, 13, 1, 1), +SOC_DAPM_SINGLE("MIC22MONO Playback Switch", ALC5623_MIC_ROUTING_CTRL, 5, 1, 1), +SOC_DAPM_SINGLE("DAC2MONO Playback Switch", ALC5623_STEREO_DAC_VOL, 13, 1, 1), +}; + +static const struct snd_kcontrol_new alc5623_speaker_mixer_controls[] = { +SOC_DAPM_SINGLE("LI2SPK Playback Switch", ALC5623_LINE_IN_VOL, 14, 1, 1), +SOC_DAPM_SINGLE("AUXI2SPK Playback Switch", ALC5623_AUXIN_VOL, 14, 1, 1), +SOC_DAPM_SINGLE("MIC12SPK Playback Switch", ALC5623_MIC_ROUTING_CTRL, 14, 1, 1), +SOC_DAPM_SINGLE("MIC22SPK Playback Switch", ALC5623_MIC_ROUTING_CTRL, 6, 1, 1), +SOC_DAPM_SINGLE("DAC2SPK Playback Switch", ALC5623_STEREO_DAC_VOL, 14, 1, 1), +}; + +/* Left Record Mixer */ +static const struct snd_kcontrol_new alc5623_captureL_mixer_controls[] = { +SOC_DAPM_SINGLE("Mic1 Capture Switch", ALC5623_ADC_REC_MIXER, 14, 1, 1), +SOC_DAPM_SINGLE("Mic2 Capture Switch", ALC5623_ADC_REC_MIXER, 13, 1, 1), +SOC_DAPM_SINGLE("LineInL Capture Switch", ALC5623_ADC_REC_MIXER, 12, 1, 1), +SOC_DAPM_SINGLE("Left AuxI Capture Switch", ALC5623_ADC_REC_MIXER, 11, 1, 1), +SOC_DAPM_SINGLE("HPMixerL Capture Switch", ALC5623_ADC_REC_MIXER, 10, 1, 1), +SOC_DAPM_SINGLE("SPKMixer Capture Switch", ALC5623_ADC_REC_MIXER, 9, 1, 1), +SOC_DAPM_SINGLE("MonoMixer Capture Switch", ALC5623_ADC_REC_MIXER, 8, 1, 1), +}; + +/* Right Record Mixer */ +static const struct snd_kcontrol_new alc5623_captureR_mixer_controls[] = { +SOC_DAPM_SINGLE("Mic1 Capture Switch", ALC5623_ADC_REC_MIXER, 6, 1, 1), +SOC_DAPM_SINGLE("Mic2 Capture Switch", ALC5623_ADC_REC_MIXER, 5, 1, 1), +SOC_DAPM_SINGLE("LineInR Capture Switch", ALC5623_ADC_REC_MIXER, 4, 1, 1), +SOC_DAPM_SINGLE("Right AuxI Capture Switch", ALC5623_ADC_REC_MIXER, 3, 1, 1), +SOC_DAPM_SINGLE("HPMixerR Capture Switch", ALC5623_ADC_REC_MIXER, 2, 1, 1), +SOC_DAPM_SINGLE("SPKMixer Capture Switch", ALC5623_ADC_REC_MIXER, 1, 1, 1), +SOC_DAPM_SINGLE("MonoMixer Capture Switch", ALC5623_ADC_REC_MIXER, 0, 1, 1), +}; + +static const char *alc5623_spk_n_sour_sel[] = { + "RN/-R", "RP/+R", "LN/-R", "Vmid" }; +static const char *alc5623_hpl_out_input_sel[] = { + "Vmid", "HP Left Mix"}; +static const char *alc5623_hpr_out_input_sel[] = { + "Vmid", "HP Right Mix"}; +static const char *alc5623_spkout_input_sel[] = { + "Vmid", "HPOut Mix", "Speaker Mix", "Mono Mix"}; +static const char *alc5623_aux_out_input_sel[] = { + "Vmid", "HPOut Mix", "Speaker Mix", "Mono Mix"}; + +/* auxout output mux */ +static const struct soc_enum alc5623_aux_out_input_enum = +SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 6, 4, alc5623_aux_out_input_sel); +static const struct snd_kcontrol_new alc5623_auxout_mux_controls = +SOC_DAPM_ENUM("Route", alc5623_aux_out_input_enum); + +/* speaker output mux */ +static const struct soc_enum alc5623_spkout_input_enum = +SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 10, 4, alc5623_spkout_input_sel); +static const struct snd_kcontrol_new alc5623_spkout_mux_controls = +SOC_DAPM_ENUM("Route", alc5623_spkout_input_enum); + +/* headphone left output mux */ +static const struct soc_enum alc5623_hpl_out_input_enum = +SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 9, 2, alc5623_hpl_out_input_sel); +static const struct snd_kcontrol_new alc5623_hpl_out_mux_controls = +SOC_DAPM_ENUM("Route", alc5623_hpl_out_input_enum); + +/* headphone right output mux */ +static const struct soc_enum alc5623_hpr_out_input_enum = +SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 8, 2, alc5623_hpr_out_input_sel); +static const struct snd_kcontrol_new alc5623_hpr_out_mux_controls = +SOC_DAPM_ENUM("Route", alc5623_hpr_out_input_enum); + +/* speaker output N select */ +static const struct soc_enum alc5623_spk_n_sour_enum = +SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 14, 4, alc5623_spk_n_sour_sel); +static const struct snd_kcontrol_new alc5623_spkoutn_mux_controls = +SOC_DAPM_ENUM("Route", alc5623_spk_n_sour_enum); + +static const struct snd_soc_dapm_widget alc5623_dapm_widgets[] = { +/* Muxes */ +SND_SOC_DAPM_MUX("AuxOut Mux", SND_SOC_NOPM, 0, 0, + &alc5623_auxout_mux_controls), +SND_SOC_DAPM_MUX("SpeakerOut Mux", SND_SOC_NOPM, 0, 0, + &alc5623_spkout_mux_controls), +SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0, + &alc5623_hpl_out_mux_controls), +SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0, + &alc5623_hpr_out_mux_controls), +SND_SOC_DAPM_MUX("SpeakerOut N Mux", SND_SOC_NOPM, 0, 0, + &alc5623_spkoutn_mux_controls), + +/* output mixers */ +SND_SOC_DAPM_MIXER("HP Mix", SND_SOC_NOPM, 0, 0, + &alc5623_hp_mixer_controls[0], + ARRAY_SIZE(alc5623_hp_mixer_controls)), +SND_SOC_DAPM_MIXER("HPR Mix", ALC5623_PWR_MANAG_ADD2, 4, 0, + &alc5623_hpr_mixer_controls[0], + ARRAY_SIZE(alc5623_hpr_mixer_controls)), +SND_SOC_DAPM_MIXER("HPL Mix", ALC5623_PWR_MANAG_ADD2, 5, 0, + &alc5623_hpl_mixer_controls[0], + ARRAY_SIZE(alc5623_hpl_mixer_controls)), +SND_SOC_DAPM_MIXER("HPOut Mix", SND_SOC_NOPM, 0, 0, NULL, 0), +SND_SOC_DAPM_MIXER("Mono Mix", ALC5623_PWR_MANAG_ADD2, 2, 0, + &alc5623_mono_mixer_controls[0], + ARRAY_SIZE(alc5623_mono_mixer_controls)), +SND_SOC_DAPM_MIXER("Speaker Mix", ALC5623_PWR_MANAG_ADD2, 3, 0, + &alc5623_speaker_mixer_controls[0], + ARRAY_SIZE(alc5623_speaker_mixer_controls)), + +/* input mixers */ +SND_SOC_DAPM_MIXER("Left Capture Mix", ALC5623_PWR_MANAG_ADD2, 1, 0, + &alc5623_captureL_mixer_controls[0], + ARRAY_SIZE(alc5623_captureL_mixer_controls)), +SND_SOC_DAPM_MIXER("Right Capture Mix", ALC5623_PWR_MANAG_ADD2, 0, 0, + &alc5623_captureR_mixer_controls[0], + ARRAY_SIZE(alc5623_captureR_mixer_controls)), + +SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback", + ALC5623_PWR_MANAG_ADD2, 9, 0), +SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback", + ALC5623_PWR_MANAG_ADD2, 8, 0), +SND_SOC_DAPM_MIXER("I2S Mix", ALC5623_PWR_MANAG_ADD1, 15, 0, NULL, 0), +SND_SOC_DAPM_MIXER("AuxI Mix", SND_SOC_NOPM, 0, 0, NULL, 0), +SND_SOC_DAPM_MIXER("Line Mix", SND_SOC_NOPM, 0, 0, NULL, 0), +SND_SOC_DAPM_ADC("Left ADC", "Left HiFi Capture", + ALC5623_PWR_MANAG_ADD2, 7, 0), +SND_SOC_DAPM_ADC("Right ADC", "Right HiFi Capture", + ALC5623_PWR_MANAG_ADD2, 6, 0), +SND_SOC_DAPM_PGA("Left Headphone", ALC5623_PWR_MANAG_ADD3, 10, 0, NULL, 0), +SND_SOC_DAPM_PGA("Right Headphone", ALC5623_PWR_MANAG_ADD3, 9, 0, NULL, 0), +SND_SOC_DAPM_PGA("SpeakerOut", ALC5623_PWR_MANAG_ADD3, 12, 0, NULL, 0), +SND_SOC_DAPM_PGA("Left AuxOut", ALC5623_PWR_MANAG_ADD3, 14, 0, NULL, 0), +SND_SOC_DAPM_PGA("Right AuxOut", ALC5623_PWR_MANAG_ADD3, 13, 0, NULL, 0), +SND_SOC_DAPM_PGA("Left LineIn", ALC5623_PWR_MANAG_ADD3, 7, 0, NULL, 0), +SND_SOC_DAPM_PGA("Right LineIn", ALC5623_PWR_MANAG_ADD3, 6, 0, NULL, 0), +SND_SOC_DAPM_PGA("Left AuxI", ALC5623_PWR_MANAG_ADD3, 5, 0, NULL, 0), +SND_SOC_DAPM_PGA("Right AuxI", ALC5623_PWR_MANAG_ADD3, 4, 0, NULL, 0), +SND_SOC_DAPM_PGA("MIC1 PGA", ALC5623_PWR_MANAG_ADD3, 3, 0, NULL, 0), +SND_SOC_DAPM_PGA("MIC2 PGA", ALC5623_PWR_MANAG_ADD3, 2, 0, NULL, 0), +SND_SOC_DAPM_PGA("MIC1 Pre Amp", ALC5623_PWR_MANAG_ADD3, 1, 0, NULL, 0), +SND_SOC_DAPM_PGA("MIC2 Pre Amp", ALC5623_PWR_MANAG_ADD3, 0, 0, NULL, 0), +SND_SOC_DAPM_MICBIAS("Mic Bias1", ALC5623_PWR_MANAG_ADD1, 11, 0), + +SND_SOC_DAPM_OUTPUT("AUXOUTL"), +SND_SOC_DAPM_OUTPUT("AUXOUTR"), +SND_SOC_DAPM_OUTPUT("HPL"), +SND_SOC_DAPM_OUTPUT("HPR"), +SND_SOC_DAPM_OUTPUT("SPKOUT"), +SND_SOC_DAPM_OUTPUT("SPKOUTN"), +SND_SOC_DAPM_INPUT("LINEINL"), +SND_SOC_DAPM_INPUT("LINEINR"), +SND_SOC_DAPM_INPUT("AUXINL"), +SND_SOC_DAPM_INPUT("AUXINR"), +SND_SOC_DAPM_INPUT("MIC1"), +SND_SOC_DAPM_INPUT("MIC2"), +SND_SOC_DAPM_VMID("Vmid"), +}; + +static const char *alc5623_amp_names[] = {"AB Amp", "D Amp"}; +static const struct soc_enum alc5623_amp_enum = + SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 13, 2, alc5623_amp_names); +static const struct snd_kcontrol_new alc5623_amp_mux_controls = + SOC_DAPM_ENUM("Route", alc5623_amp_enum); + +static const struct snd_soc_dapm_widget alc5623_dapm_amp_widgets[] = { +SND_SOC_DAPM_PGA_E("D Amp", ALC5623_PWR_MANAG_ADD2, 14, 0, NULL, 0, + amp_mixer_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_PGA("AB Amp", ALC5623_PWR_MANAG_ADD2, 15, 0, NULL, 0), +SND_SOC_DAPM_MUX("AB-D Amp Mux", SND_SOC_NOPM, 0, 0, + &alc5623_amp_mux_controls), +}; + +static const struct snd_soc_dapm_route intercon[] = { + /* virtual mixer - mixes left & right channels */ + {"I2S Mix", NULL, "Left DAC"}, + {"I2S Mix", NULL, "Right DAC"}, + {"Line Mix", NULL, "Right LineIn"}, + {"Line Mix", NULL, "Left LineIn"}, + {"AuxI Mix", NULL, "Left AuxI"}, + {"AuxI Mix", NULL, "Right AuxI"}, + {"AUXOUTL", NULL, "Left AuxOut"}, + {"AUXOUTR", NULL, "Right AuxOut"}, + + /* HP mixer */ + {"HPL Mix", "ADC2HP_L Playback Switch", "Left Capture Mix"}, + {"HPL Mix", NULL, "HP Mix"}, + {"HPR Mix", "ADC2HP_R Playback Switch", "Right Capture Mix"}, + {"HPR Mix", NULL, "HP Mix"}, + {"HP Mix", "LI2HP Playback Switch", "Line Mix"}, + {"HP Mix", "AUXI2HP Playback Switch", "AuxI Mix"}, + {"HP Mix", "MIC12HP Playback Switch", "MIC1 PGA"}, + {"HP Mix", "MIC22HP Playback Switch", "MIC2 PGA"}, + {"HP Mix", "DAC2HP Playback Switch", "I2S Mix"}, + + /* speaker mixer */ + {"Speaker Mix", "LI2SPK Playback Switch", "Line Mix"}, + {"Speaker Mix", "AUXI2SPK Playback Switch", "AuxI Mix"}, + {"Speaker Mix", "MIC12SPK Playback Switch", "MIC1 PGA"}, + {"Speaker Mix", "MIC22SPK Playback Switch", "MIC2 PGA"}, + {"Speaker Mix", "DAC2SPK Playback Switch", "I2S Mix"}, + + /* mono mixer */ + {"Mono Mix", "ADC2MONO_L Playback Switch", "Left Capture Mix"}, + {"Mono Mix", "ADC2MONO_R Playback Switch", "Right Capture Mix"}, + {"Mono Mix", "LI2MONO Playback Switch", "Line Mix"}, + {"Mono Mix", "AUXI2MONO Playback Switch", "AuxI Mix"}, + {"Mono Mix", "MIC12MONO Playback Switch", "MIC1 PGA"}, + {"Mono Mix", "MIC22MONO Playback Switch", "MIC2 PGA"}, + {"Mono Mix", "DAC2MONO Playback Switch", "I2S Mix"}, + + /* Left record mixer */ + {"Left Capture Mix", "LineInL Capture Switch", "LINEINL"}, + {"Left Capture Mix", "Left AuxI Capture Switch", "AUXINL"}, + {"Left Capture Mix", "Mic1 Capture Switch", "MIC1 Pre Amp"}, + {"Left Capture Mix", "Mic2 Capture Switch", "MIC2 Pre Amp"}, + {"Left Capture Mix", "HPMixerL Capture Switch", "HPL Mix"}, + {"Left Capture Mix", "SPKMixer Capture Switch", "Speaker Mix"}, + {"Left Capture Mix", "MonoMixer Capture Switch", "Mono Mix"}, + + /*Right record mixer */ + {"Right Capture Mix", "LineInR Capture Switch", "LINEINR"}, + {"Right Capture Mix", "Right AuxI Capture Switch", "AUXINR"}, + {"Right Capture Mix", "Mic1 Capture Switch", "MIC1 Pre Amp"}, + {"Right Capture Mix", "Mic2 Capture Switch", "MIC2 Pre Amp"}, + {"Right Capture Mix", "HPMixerR Capture Switch", "HPR Mix"}, + {"Right Capture Mix", "SPKMixer Capture Switch", "Speaker Mix"}, + {"Right Capture Mix", "MonoMixer Capture Switch", "Mono Mix"}, + + /* headphone left mux */ + {"Left Headphone Mux", "HP Left Mix", "HPL Mix"}, + {"Left Headphone Mux", "Vmid", "Vmid"}, + + /* headphone right mux */ + {"Right Headphone Mux", "HP Right Mix", "HPR Mix"}, + {"Right Headphone Mux", "Vmid", "Vmid"}, + + /* speaker out mux */ + {"SpeakerOut Mux", "Vmid", "Vmid"}, + {"SpeakerOut Mux", "HPOut Mix", "HPOut Mix"}, + {"SpeakerOut Mux", "Speaker Mix", "Speaker Mix"}, + {"SpeakerOut Mux", "Mono Mix", "Mono Mix"}, + + /* Mono/Aux Out mux */ + {"AuxOut Mux", "Vmid", "Vmid"}, + {"AuxOut Mux", "HPOut Mix", "HPOut Mix"}, + {"AuxOut Mux", "Speaker Mix", "Speaker Mix"}, + {"AuxOut Mux", "Mono Mix", "Mono Mix"}, + + /* output pga */ + {"HPL", NULL, "Left Headphone"}, + {"Left Headphone", NULL, "Left Headphone Mux"}, + {"HPR", NULL, "Right Headphone"}, + {"Right Headphone", NULL, "Right Headphone Mux"}, + {"Left AuxOut", NULL, "AuxOut Mux"}, + {"Right AuxOut", NULL, "AuxOut Mux"}, + + /* input pga */ + {"Left LineIn", NULL, "LINEINL"}, + {"Right LineIn", NULL, "LINEINR"}, + {"Left AuxI", NULL, "AUXINL"}, + {"Right AuxI", NULL, "AUXINR"}, + {"MIC1 Pre Amp", NULL, "MIC1"}, + {"MIC2 Pre Amp", NULL, "MIC2"}, + {"MIC1 PGA", NULL, "MIC1 Pre Amp"}, + {"MIC2 PGA", NULL, "MIC2 Pre Amp"}, + + /* left ADC */ + {"Left ADC", NULL, "Left Capture Mix"}, + + /* right ADC */ + {"Right ADC", NULL, "Right Capture Mix"}, + + {"SpeakerOut N Mux", "RN/-R", "SpeakerOut"}, + {"SpeakerOut N Mux", "RP/+R", "SpeakerOut"}, + {"SpeakerOut N Mux", "LN/-R", "SpeakerOut"}, + {"SpeakerOut N Mux", "Vmid", "Vmid"}, + + {"SPKOUT", NULL, "SpeakerOut"}, + {"SPKOUTN", NULL, "SpeakerOut N Mux"}, +}; + +static const struct snd_soc_dapm_route intercon_spk[] = { + {"SpeakerOut", NULL, "SpeakerOut Mux"}, +}; + +static const struct snd_soc_dapm_route intercon_amp_spk[] = { + {"AB Amp", NULL, "SpeakerOut Mux"}, + {"D Amp", NULL, "SpeakerOut Mux"}, + {"AB-D Amp Mux", "AB Amp", "AB Amp"}, + {"AB-D Amp Mux", "D Amp", "D Amp"}, + {"SpeakerOut", NULL, "AB-D Amp Mux"}, +}; + +/* PLL divisors */ +struct _pll_div { + u32 pll_in; + u32 pll_out; + u16 regvalue; +}; + +/* Note : pll code from original alc5623 driver. Not sure of how good it is */ +/* usefull only for master mode */ +static const struct _pll_div codec_master_pll_div[] = { + + { 2048000, 8192000, 0x0ea0}, + { 3686400, 8192000, 0x4e27}, + { 12000000, 8192000, 0x456b}, + { 13000000, 8192000, 0x495f}, + { 13100000, 8192000, 0x0320}, + { 2048000, 11289600, 0xf637}, + { 3686400, 11289600, 0x2f22}, + { 12000000, 11289600, 0x3e2f}, + { 13000000, 11289600, 0x4d5b}, + { 13100000, 11289600, 0x363b}, + { 2048000, 16384000, 0x1ea0}, + { 3686400, 16384000, 0x9e27}, + { 12000000, 16384000, 0x452b}, + { 13000000, 16384000, 0x542f}, + { 13100000, 16384000, 0x03a0}, + { 2048000, 16934400, 0xe625}, + { 3686400, 16934400, 0x9126}, + { 12000000, 16934400, 0x4d2c}, + { 13000000, 16934400, 0x742f}, + { 13100000, 16934400, 0x3c27}, + { 2048000, 22579200, 0x2aa0}, + { 3686400, 22579200, 0x2f20}, + { 12000000, 22579200, 0x7e2f}, + { 13000000, 22579200, 0x742f}, + { 13100000, 22579200, 0x3c27}, + { 2048000, 24576000, 0x2ea0}, + { 3686400, 24576000, 0xee27}, + { 12000000, 24576000, 0x2915}, + { 13000000, 24576000, 0x772e}, + { 13100000, 24576000, 0x0d20}, +}; + +static const struct _pll_div codec_slave_pll_div[] = { + + { 1024000, 16384000, 0x3ea0}, + { 1411200, 22579200, 0x3ea0}, + { 1536000, 24576000, 0x3ea0}, + { 2048000, 16384000, 0x1ea0}, + { 2822400, 22579200, 0x1ea0}, + { 3072000, 24576000, 0x1ea0}, + +}; + +static int alc5623_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) +{ + int i; + struct snd_soc_codec *codec = codec_dai->codec; + int gbl_clk = 0, pll_div = 0; + u16 reg; + + if (pll_id < ALC5623_PLL_FR_MCLK || pll_id > ALC5623_PLL_FR_BCK) + return -ENODEV; + + /* Disable PLL power */ + snd_soc_update_bits(codec, ALC5623_PWR_MANAG_ADD2, + ALC5623_PWR_ADD2_PLL, + 0); + + /* pll is not used in slave mode */ + reg = snd_soc_read(codec, ALC5623_DAI_CONTROL); + if (reg & ALC5623_DAI_SDP_SLAVE_MODE) + return 0; + + if (!freq_in || !freq_out) + return 0; + + switch (pll_id) { + case ALC5623_PLL_FR_MCLK: + for (i = 0; i < ARRAY_SIZE(codec_master_pll_div); i++) { + if (codec_master_pll_div[i].pll_in == freq_in + && codec_master_pll_div[i].pll_out == freq_out) { + /* PLL source from MCLK */ + pll_div = codec_master_pll_div[i].regvalue; + break; + } + } + break; + case ALC5623_PLL_FR_BCK: + for (i = 0; i < ARRAY_SIZE(codec_slave_pll_div); i++) { + if (codec_slave_pll_div[i].pll_in == freq_in + && codec_slave_pll_div[i].pll_out == freq_out) { + /* PLL source from Bitclk */ + gbl_clk = ALC5623_GBL_CLK_PLL_SOUR_SEL_BITCLK; + pll_div = codec_slave_pll_div[i].regvalue; + break; + } + } + break; + default: + return -EINVAL; + } + + if (!pll_div) + return -EINVAL; + + snd_soc_write(codec, ALC5623_GLOBAL_CLK_CTRL_REG, gbl_clk); + snd_soc_write(codec, ALC5623_PLL_CTRL, pll_div); + snd_soc_update_bits(codec, ALC5623_PWR_MANAG_ADD2, + ALC5623_PWR_ADD2_PLL, + ALC5623_PWR_ADD2_PLL); + gbl_clk |= ALC5623_GBL_CLK_SYS_SOUR_SEL_PLL; + snd_soc_write(codec, ALC5623_GLOBAL_CLK_CTRL_REG, gbl_clk); + + return 0; +} + +struct _coeff_div { + u16 fs; + u16 regvalue; +}; + +/* codec hifi mclk (after PLL) clock divider coefficients */ +/* values inspired from column BCLK=32Fs of Appendix A table */ +static const struct _coeff_div coeff_div[] = { + {256*8, 0x3a69}, + {384*8, 0x3c6b}, + {256*4, 0x2a69}, + {384*4, 0x2c6b}, + {256*2, 0x1a69}, + {384*2, 0x1c6b}, + {256*1, 0x0a69}, + {384*1, 0x0c6b}, +}; + +static int get_coeff(struct snd_soc_codec *codec, int rate) +{ + struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec); + int i; + + for (i = 0; i < ARRAY_SIZE(coeff_div); i++) { + if (coeff_div[i].fs * rate == alc5623->sysclk) + return i; + } + return -EINVAL; +} + +/* + * Clock after PLL and dividers + */ +static int alc5623_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 alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec); + + switch (freq) { + case 8192000: + case 11289600: + case 12288000: + case 16384000: + case 16934400: + case 18432000: + case 22579200: + case 24576000: + alc5623->sysclk = freq; + return 0; + } + return -EINVAL; +} + +static int alc5623_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 iface = 0; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + iface = ALC5623_DAI_SDP_MASTER_MODE; + break; + case SND_SOC_DAIFMT_CBS_CFS: + iface = ALC5623_DAI_SDP_SLAVE_MODE; + break; + default: + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface |= ALC5623_DAI_I2S_DF_I2S; + break; + case SND_SOC_DAIFMT_RIGHT_J: + iface |= ALC5623_DAI_I2S_DF_RIGHT; + break; + case SND_SOC_DAIFMT_LEFT_J: + iface |= ALC5623_DAI_I2S_DF_LEFT; + break; + case SND_SOC_DAIFMT_DSP_A: + iface |= ALC5623_DAI_I2S_DF_PCM; + break; + case SND_SOC_DAIFMT_DSP_B: + iface |= ALC5623_DAI_I2S_DF_PCM | ALC5623_DAI_I2S_PCM_MODE; + break; + default: + return -EINVAL; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + iface |= ALC5623_DAI_MAIN_I2S_BCLK_POL_CTRL; + break; + case SND_SOC_DAIFMT_IB_NF: + iface |= ALC5623_DAI_MAIN_I2S_BCLK_POL_CTRL; + break; + case SND_SOC_DAIFMT_NB_IF: + break; + default: + return -EINVAL; + } + + return snd_soc_write(codec, ALC5623_DAI_CONTROL, iface); +} + +static int alc5623_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec); + int coeff, rate; + u16 iface; + + iface = snd_soc_read(codec, ALC5623_DAI_CONTROL); + iface &= ~ALC5623_DAI_I2S_DL_MASK; + + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + iface |= ALC5623_DAI_I2S_DL_16; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + iface |= ALC5623_DAI_I2S_DL_20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + iface |= ALC5623_DAI_I2S_DL_24; + break; + case SNDRV_PCM_FORMAT_S32_LE: + iface |= ALC5623_DAI_I2S_DL_32; + break; + default: + return -EINVAL; + } + + /* set iface & srate */ + snd_soc_write(codec, ALC5623_DAI_CONTROL, iface); + rate = params_rate(params); + coeff = get_coeff(codec, rate); + if (coeff < 0) + return -EINVAL; + + coeff = coeff_div[coeff].regvalue; + dev_dbg(codec->dev, "%s: sysclk=%d,rate=%d,coeff=0x%04x\n", + __func__, alc5623->sysclk, rate, coeff); + snd_soc_write(codec, ALC5623_STEREO_AD_DA_CLK_CTRL, coeff); + + return 0; +} + +static int alc5623_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u16 hp_mute = ALC5623_MISC_M_DAC_L_INPUT | ALC5623_MISC_M_DAC_R_INPUT; + u16 mute_reg = snd_soc_read(codec, ALC5623_MISC_CTRL) & ~hp_mute; + + if (mute) + mute_reg |= hp_mute; + + return snd_soc_write(codec, ALC5623_MISC_CTRL, mute_reg); +} + +#define ALC5623_ADD2_POWER_EN (ALC5623_PWR_ADD2_VREF \ + | ALC5623_PWR_ADD2_DAC_REF_CIR) + +#define ALC5623_ADD3_POWER_EN (ALC5623_PWR_ADD3_MAIN_BIAS \ + | ALC5623_PWR_ADD3_MIC1_BOOST_AD) + +#define ALC5623_ADD1_POWER_EN \ + (ALC5623_PWR_ADD1_SHORT_CURR_DET_EN | ALC5623_PWR_ADD1_SOFTGEN_EN \ + | ALC5623_PWR_ADD1_DEPOP_BUF_HP | ALC5623_PWR_ADD1_HP_OUT_AMP \ + | ALC5623_PWR_ADD1_HP_OUT_ENH_AMP) + +#define ALC5623_ADD1_POWER_EN_5622 \ + (ALC5623_PWR_ADD1_SHORT_CURR_DET_EN \ + | ALC5623_PWR_ADD1_HP_OUT_AMP) + +static void enable_power_depop(struct snd_soc_codec *codec) +{ + struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec); + + snd_soc_update_bits(codec, ALC5623_PWR_MANAG_ADD1, + ALC5623_PWR_ADD1_SOFTGEN_EN, + ALC5623_PWR_ADD1_SOFTGEN_EN); + + snd_soc_write(codec, ALC5623_PWR_MANAG_ADD3, ALC5623_ADD3_POWER_EN); + + snd_soc_update_bits(codec, ALC5623_MISC_CTRL, + ALC5623_MISC_HP_DEPOP_MODE2_EN, + ALC5623_MISC_HP_DEPOP_MODE2_EN); + + msleep(500); + + snd_soc_write(codec, ALC5623_PWR_MANAG_ADD2, ALC5623_ADD2_POWER_EN); + + /* avoid writing '1' into 5622 reserved bits */ + if (alc5623->id == 0x22) + snd_soc_write(codec, ALC5623_PWR_MANAG_ADD1, + ALC5623_ADD1_POWER_EN_5622); + else + snd_soc_write(codec, ALC5623_PWR_MANAG_ADD1, + ALC5623_ADD1_POWER_EN); + + /* disable HP Depop2 */ + snd_soc_update_bits(codec, ALC5623_MISC_CTRL, + ALC5623_MISC_HP_DEPOP_MODE2_EN, + 0); + +} + +static int alc5623_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + enable_power_depop(codec); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + /* everything off except vref/vmid, */ + snd_soc_write(codec, ALC5623_PWR_MANAG_ADD2, + ALC5623_PWR_ADD2_VREF); + snd_soc_write(codec, ALC5623_PWR_MANAG_ADD3, + ALC5623_PWR_ADD3_MAIN_BIAS); + break; + case SND_SOC_BIAS_OFF: + /* everything off, dac mute, inactive */ + snd_soc_write(codec, ALC5623_PWR_MANAG_ADD2, 0); + snd_soc_write(codec, ALC5623_PWR_MANAG_ADD3, 0); + snd_soc_write(codec, ALC5623_PWR_MANAG_ADD1, 0); + break; + } + codec->dapm.bias_level = level; + return 0; +} + +#define ALC5623_FORMATS (SNDRV_PCM_FMTBIT_S16_LE \ + | SNDRV_PCM_FMTBIT_S24_LE \ + | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops alc5623_dai_ops = { + .hw_params = alc5623_pcm_hw_params, + .digital_mute = alc5623_mute, + .set_fmt = alc5623_set_dai_fmt, + .set_sysclk = alc5623_set_dai_sysclk, + .set_pll = alc5623_set_dai_pll, +}; + +static struct snd_soc_dai_driver alc5623_dai = { + .name = "alc5623-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rate_min = 8000, + .rate_max = 48000, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = ALC5623_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rate_min = 8000, + .rate_max = 48000, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = ALC5623_FORMATS,}, + + .ops = &alc5623_dai_ops, +}; + +static int alc5623_suspend(struct snd_soc_codec *codec, pm_message_t mesg) +{ + alc5623_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int alc5623_resume(struct snd_soc_codec *codec) +{ + int i, step = codec->driver->reg_cache_step; + u16 *cache = codec->reg_cache; + + /* Sync reg_cache with the hardware */ + for (i = 2 ; i < codec->driver->reg_cache_size ; i += step) + snd_soc_write(codec, i, cache[i]); + + alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + /* charge alc5623 caps */ + if (codec->dapm.suspend_bias_level == SND_SOC_BIAS_ON) { + alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + codec->dapm.bias_level = SND_SOC_BIAS_ON; + alc5623_set_bias_level(codec, codec->dapm.bias_level); + } + + return 0; +} + +static int alc5623_probe(struct snd_soc_codec *codec) +{ + struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec); + struct snd_soc_dapm_context *dapm = &codec->dapm; + int ret; + + ret = snd_soc_codec_set_cache_io(codec, 8, 16, alc5623->control_type); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + alc5623_reset(codec); + alc5623_fill_cache(codec); + + /* power on device */ + alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + if (alc5623->add_ctrl) { + snd_soc_write(codec, ALC5623_ADD_CTRL_REG, + alc5623->add_ctrl); + } + + if (alc5623->jack_det_ctrl) { + snd_soc_write(codec, ALC5623_JACK_DET_CTRL, + alc5623->jack_det_ctrl); + } + + switch (alc5623->id) { + default: + case 0x21: + snd_soc_add_controls(codec, rt5621_vol_snd_controls, + ARRAY_SIZE(rt5621_vol_snd_controls)); + break; + case 0x22: + snd_soc_add_controls(codec, rt5622_vol_snd_controls, + ARRAY_SIZE(rt5622_vol_snd_controls)); + break; + case 0x23: + snd_soc_add_controls(codec, alc5623_vol_snd_controls, + ARRAY_SIZE(alc5623_vol_snd_controls)); + break; + } + + snd_soc_add_controls(codec, alc5623_snd_controls, + ARRAY_SIZE(alc5623_snd_controls)); + + snd_soc_dapm_new_controls(dapm, alc5623_dapm_widgets, + ARRAY_SIZE(alc5623_dapm_widgets)); + + /* set up audio path interconnects */ + snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); + + switch (alc5623->id) { + default: + case 0x21: + case 0x22: + snd_soc_dapm_new_controls(dapm, alc5623_dapm_amp_widgets, + ARRAY_SIZE(alc5623_dapm_amp_widgets)); + snd_soc_dapm_add_routes(dapm, intercon_amp_spk, + ARRAY_SIZE(intercon_amp_spk)); + break; + case 0x23: + snd_soc_dapm_add_routes(dapm, intercon_spk, + ARRAY_SIZE(intercon_spk)); + break; + } + + return ret; +} + +/* power down chip */ +static int alc5623_remove(struct snd_soc_codec *codec) +{ + alc5623_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static struct snd_soc_codec_driver soc_codec_device_alc5623 = { + .probe = alc5623_probe, + .remove = alc5623_remove, + .suspend = alc5623_suspend, + .resume = alc5623_resume, + .set_bias_level = alc5623_set_bias_level, + .reg_cache_size = ALC5623_VENDOR_ID2+2, + .reg_word_size = sizeof(u16), + .reg_cache_step = 2, +}; + +/* + * ALC5623 2 wire address is determined by A1 pin + * state during powerup. + * low = 0x1a + * high = 0x1b + */ +static int alc5623_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct alc5623_platform_data *pdata; + struct alc5623_priv *alc5623; + int ret, vid1, vid2; + + vid1 = i2c_smbus_read_word_data(client, ALC5623_VENDOR_ID1); + if (vid1 < 0) { + dev_err(&client->dev, "failed to read I2C\n"); + return -EIO; + } + vid1 = ((vid1 & 0xff) << 8) | (vid1 >> 8); + + vid2 = i2c_smbus_read_byte_data(client, ALC5623_VENDOR_ID2); + if (vid2 < 0) { + dev_err(&client->dev, "failed to read I2C\n"); + return -EIO; + } + + if ((vid1 != 0x10ec) || (vid2 != id->driver_data)) { + dev_err(&client->dev, "unknown or wrong codec\n"); + dev_err(&client->dev, "Expected %x:%lx, got %x:%x\n", + 0x10ec, id->driver_data, + vid1, vid2); + return -ENODEV; + } + + dev_dbg(&client->dev, "Found codec id : alc56%02x\n", vid2); + + alc5623 = kzalloc(sizeof(struct alc5623_priv), GFP_KERNEL); + if (alc5623 == NULL) { + ret = -ENOMEM; + goto err; + } + + pdata = client->dev.platform_data; + if (pdata) { + alc5623->add_ctrl = pdata->add_ctrl; + alc5623->jack_det_ctrl = pdata->jack_det_ctrl; + } + + alc5623->id = vid2; + switch (alc5623->id) { + case 0x21: + alc5623_dai.name = "alc5621-hifi"; + break; + case 0x22: + alc5623_dai.name = "alc5622-hifi"; + break; + default: + case 0x23: + alc5623_dai.name = "alc5623-hifi"; + break; + } + + i2c_set_clientdata(client, alc5623); + alc5623->control_data = client; + alc5623->control_type = SND_SOC_I2C; + mutex_init(&alc5623->mutex); + + ret = snd_soc_register_codec(&client->dev, + &soc_codec_device_alc5623, &alc5623_dai, 1); + if (ret != 0) { + dev_err(&client->dev, "Failed to register codec: %d\n", ret); + goto err; + } + + return 0; + +err: + return ret; +} + +static int alc5623_i2c_remove(struct i2c_client *client) +{ + struct alc5623_priv *alc5623 = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + kfree(alc5623); + return 0; +} + +static const struct i2c_device_id alc5623_i2c_table[] = { + {"alc5621", 0x21}, + {"alc5622", 0x22}, + {"alc5623", 0x23}, + {} +}; +MODULE_DEVICE_TABLE(i2c, alc5623_i2c_table); + +/* i2c codec control layer */ +static struct i2c_driver alc5623_i2c_driver = { + .driver = { + .name = "alc562x-codec", + .owner = THIS_MODULE, + }, + .probe = alc5623_i2c_probe, + .remove = __devexit_p(alc5623_i2c_remove), + .id_table = alc5623_i2c_table, +}; + +static int __init alc5623_modinit(void) +{ + int ret; + + ret = i2c_add_driver(&alc5623_i2c_driver); + if (ret != 0) { + printk(KERN_ERR "%s: can't add i2c driver", __func__); + return ret; + } + + return ret; +} +module_init(alc5623_modinit); + +static void __exit alc5623_modexit(void) +{ + i2c_del_driver(&alc5623_i2c_driver); +} +module_exit(alc5623_modexit); + +MODULE_DESCRIPTION("ASoC alc5621/2/3 driver"); +MODULE_AUTHOR("Arnaud Patard <arnaud.patard@rtp-net.org>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/alc5623.h b/sound/soc/codecs/alc5623.h new file mode 100644 index 000000000000..f3d68260d425 --- /dev/null +++ b/sound/soc/codecs/alc5623.h @@ -0,0 +1,161 @@ +/* + * alc5623.h -- alc562[123] ALSA Soc Audio driver + * + * Copyright 2008 Realtek Microelectronics + * Copyright 2010 Arnaud Patard <arnaud.patard@rtp-net.org> + * + * Author: flove <flove@realtek.com> + * Arnaud Patard <arnaud.patard@rtp-net.org> + * + * 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 _ALC5623_H +#define _ALC5623_H + +#define ALC5623_RESET 0x00 +/* 5621 5622 5623 */ +/* speaker output vol 2 2 */ +/* line output vol 4 2 */ +/* HP output vol 4 0 4 */ +#define ALC5623_SPK_OUT_VOL 0x02 +#define ALC5623_HP_OUT_VOL 0x04 +#define ALC5623_MONO_AUX_OUT_VOL 0x06 +#define ALC5623_AUXIN_VOL 0x08 +#define ALC5623_LINE_IN_VOL 0x0A +#define ALC5623_STEREO_DAC_VOL 0x0C +#define ALC5623_MIC_VOL 0x0E +#define ALC5623_MIC_ROUTING_CTRL 0x10 +#define ALC5623_ADC_REC_GAIN 0x12 +#define ALC5623_ADC_REC_MIXER 0x14 +#define ALC5623_SOFT_VOL_CTRL_TIME 0x16 +/* ALC5623_OUTPUT_MIXER_CTRL : */ +/* same remark as for reg 2 line vs speaker */ +#define ALC5623_OUTPUT_MIXER_CTRL 0x1C +#define ALC5623_MIC_CTRL 0x22 + +#define ALC5623_DAI_CONTROL 0x34 +#define ALC5623_DAI_SDP_MASTER_MODE (0 << 15) +#define ALC5623_DAI_SDP_SLAVE_MODE (1 << 15) +#define ALC5623_DAI_I2S_PCM_MODE (1 << 14) +#define ALC5623_DAI_MAIN_I2S_BCLK_POL_CTRL (1 << 7) +#define ALC5623_DAI_ADC_DATA_L_R_SWAP (1 << 5) +#define ALC5623_DAI_DAC_DATA_L_R_SWAP (1 << 4) +#define ALC5623_DAI_I2S_DL_MASK (3 << 2) +#define ALC5623_DAI_I2S_DL_32 (3 << 2) +#define ALC5623_DAI_I2S_DL_24 (2 << 2) +#define ALC5623_DAI_I2S_DL_20 (1 << 2) +#define ALC5623_DAI_I2S_DL_16 (0 << 2) +#define ALC5623_DAI_I2S_DF_PCM (3 << 0) +#define ALC5623_DAI_I2S_DF_LEFT (2 << 0) +#define ALC5623_DAI_I2S_DF_RIGHT (1 << 0) +#define ALC5623_DAI_I2S_DF_I2S (0 << 0) + +#define ALC5623_STEREO_AD_DA_CLK_CTRL 0x36 +#define ALC5623_COMPANDING_CTRL 0x38 + +#define ALC5623_PWR_MANAG_ADD1 0x3A +#define ALC5623_PWR_ADD1_MAIN_I2S_EN (1 << 15) +#define ALC5623_PWR_ADD1_ZC_DET_PD_EN (1 << 14) +#define ALC5623_PWR_ADD1_MIC1_BIAS_EN (1 << 11) +#define ALC5623_PWR_ADD1_SHORT_CURR_DET_EN (1 << 10) +#define ALC5623_PWR_ADD1_SOFTGEN_EN (1 << 8) /* rsvd on 5622 */ +#define ALC5623_PWR_ADD1_DEPOP_BUF_HP (1 << 6) /* rsvd on 5622 */ +#define ALC5623_PWR_ADD1_HP_OUT_AMP (1 << 5) +#define ALC5623_PWR_ADD1_HP_OUT_ENH_AMP (1 << 4) /* rsvd on 5622 */ +#define ALC5623_PWR_ADD1_DEPOP_BUF_AUX (1 << 2) +#define ALC5623_PWR_ADD1_AUX_OUT_AMP (1 << 1) +#define ALC5623_PWR_ADD1_AUX_OUT_ENH_AMP (1 << 0) /* rsvd on 5622 */ + +#define ALC5623_PWR_MANAG_ADD2 0x3C +#define ALC5623_PWR_ADD2_LINEOUT (1 << 15) /* rt5623 */ +#define ALC5623_PWR_ADD2_CLASS_AB (1 << 15) /* rt5621 */ +#define ALC5623_PWR_ADD2_CLASS_D (1 << 14) /* rt5621 */ +#define ALC5623_PWR_ADD2_VREF (1 << 13) +#define ALC5623_PWR_ADD2_PLL (1 << 12) +#define ALC5623_PWR_ADD2_DAC_REF_CIR (1 << 10) +#define ALC5623_PWR_ADD2_L_DAC_CLK (1 << 9) +#define ALC5623_PWR_ADD2_R_DAC_CLK (1 << 8) +#define ALC5623_PWR_ADD2_L_ADC_CLK_GAIN (1 << 7) +#define ALC5623_PWR_ADD2_R_ADC_CLK_GAIN (1 << 6) +#define ALC5623_PWR_ADD2_L_HP_MIXER (1 << 5) +#define ALC5623_PWR_ADD2_R_HP_MIXER (1 << 4) +#define ALC5623_PWR_ADD2_SPK_MIXER (1 << 3) +#define ALC5623_PWR_ADD2_MONO_MIXER (1 << 2) +#define ALC5623_PWR_ADD2_L_ADC_REC_MIXER (1 << 1) +#define ALC5623_PWR_ADD2_R_ADC_REC_MIXER (1 << 0) + +#define ALC5623_PWR_MANAG_ADD3 0x3E +#define ALC5623_PWR_ADD3_MAIN_BIAS (1 << 15) +#define ALC5623_PWR_ADD3_AUXOUT_L_VOL_AMP (1 << 14) +#define ALC5623_PWR_ADD3_AUXOUT_R_VOL_AMP (1 << 13) +#define ALC5623_PWR_ADD3_SPK_OUT (1 << 12) +#define ALC5623_PWR_ADD3_HP_L_OUT_VOL (1 << 10) +#define ALC5623_PWR_ADD3_HP_R_OUT_VOL (1 << 9) +#define ALC5623_PWR_ADD3_LINEIN_L_VOL (1 << 7) +#define ALC5623_PWR_ADD3_LINEIN_R_VOL (1 << 6) +#define ALC5623_PWR_ADD3_AUXIN_L_VOL (1 << 5) +#define ALC5623_PWR_ADD3_AUXIN_R_VOL (1 << 4) +#define ALC5623_PWR_ADD3_MIC1_FUN_CTRL (1 << 3) +#define ALC5623_PWR_ADD3_MIC2_FUN_CTRL (1 << 2) +#define ALC5623_PWR_ADD3_MIC1_BOOST_AD (1 << 1) +#define ALC5623_PWR_ADD3_MIC2_BOOST_AD (1 << 0) + +#define ALC5623_ADD_CTRL_REG 0x40 + +#define ALC5623_GLOBAL_CLK_CTRL_REG 0x42 +#define ALC5623_GBL_CLK_SYS_SOUR_SEL_PLL (1 << 15) +#define ALC5623_GBL_CLK_SYS_SOUR_SEL_MCLK (0 << 15) +#define ALC5623_GBL_CLK_PLL_SOUR_SEL_BITCLK (1 << 14) +#define ALC5623_GBL_CLK_PLL_SOUR_SEL_MCLK (0 << 14) +#define ALC5623_GBL_CLK_PLL_DIV_RATIO_DIV8 (3 << 1) +#define ALC5623_GBL_CLK_PLL_DIV_RATIO_DIV4 (2 << 1) +#define ALC5623_GBL_CLK_PLL_DIV_RATIO_DIV2 (1 << 1) +#define ALC5623_GBL_CLK_PLL_DIV_RATIO_DIV1 (0 << 1) +#define ALC5623_GBL_CLK_PLL_PRE_DIV2 (1 << 0) +#define ALC5623_GBL_CLK_PLL_PRE_DIV1 (0 << 0) + +#define ALC5623_PLL_CTRL 0x44 +#define ALC5623_PLL_CTRL_N_VAL(n) (((n)&0xff) << 8) +#define ALC5623_PLL_CTRL_K_VAL(k) (((k)&0x7) << 4) +#define ALC5623_PLL_CTRL_M_VAL(m) ((m)&0xf) + +#define ALC5623_GPIO_OUTPUT_PIN_CTRL 0x4A +#define ALC5623_GPIO_PIN_CONFIG 0x4C +#define ALC5623_GPIO_PIN_POLARITY 0x4E +#define ALC5623_GPIO_PIN_STICKY 0x50 +#define ALC5623_GPIO_PIN_WAKEUP 0x52 +#define ALC5623_GPIO_PIN_STATUS 0x54 +#define ALC5623_GPIO_PIN_SHARING 0x56 +#define ALC5623_OVER_CURR_STATUS 0x58 +#define ALC5623_JACK_DET_CTRL 0x5A + +#define ALC5623_MISC_CTRL 0x5E +#define ALC5623_MISC_DISABLE_FAST_VREG (1 << 15) +#define ALC5623_MISC_SPK_CLASS_AB_OC_PD (1 << 13) /* 5621 */ +#define ALC5623_MISC_SPK_CLASS_AB_OC_DET (1 << 12) /* 5621 */ +#define ALC5623_MISC_HP_DEPOP_MODE3_EN (1 << 10) +#define ALC5623_MISC_HP_DEPOP_MODE2_EN (1 << 9) +#define ALC5623_MISC_HP_DEPOP_MODE1_EN (1 << 8) +#define ALC5623_MISC_AUXOUT_DEPOP_MODE3_EN (1 << 6) +#define ALC5623_MISC_AUXOUT_DEPOP_MODE2_EN (1 << 5) +#define ALC5623_MISC_AUXOUT_DEPOP_MODE1_EN (1 << 4) +#define ALC5623_MISC_M_DAC_L_INPUT (1 << 3) +#define ALC5623_MISC_M_DAC_R_INPUT (1 << 2) +#define ALC5623_MISC_IRQOUT_INV_CTRL (1 << 0) + +#define ALC5623_PSEDUEO_SPATIAL_CTRL 0x60 +#define ALC5623_EQ_CTRL 0x62 +#define ALC5623_EQ_MODE_ENABLE 0x66 +#define ALC5623_AVC_CTRL 0x68 +#define ALC5623_HID_CTRL_INDEX 0x6A +#define ALC5623_HID_CTRL_DATA 0x6C +#define ALC5623_VENDOR_ID1 0x7C +#define ALC5623_VENDOR_ID2 0x7E + +#define ALC5623_PLL_FR_MCLK 0 +#define ALC5623_PLL_FR_BCK 1 +#endif diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c index 823643932dde..98b9e5294cbe 100644 --- a/sound/soc/codecs/cq93vc.c +++ b/sound/soc/codecs/cq93vc.c @@ -116,7 +116,7 @@ static int cq93vc_set_bias_level(struct snd_soc_codec *codec, DAVINCI_VC_REG12_POWER_ALL_OFF); break; } - codec->bias_level = level; + codec->dapm.bias_level = level; return 0; } diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index cb086eaf4e07..a7fdca36b490 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -519,6 +519,7 @@ static struct snd_soc_dai_driver cs42l51_dai = { static int cs42l51_probe(struct snd_soc_codec *codec) { struct cs42l51_private *cs42l51 = snd_soc_codec_get_drvdata(codec); + struct snd_soc_dapm_context *dapm = &codec->dapm; int ret, reg; codec->control_data = cs42l51->control_data; @@ -550,9 +551,9 @@ static int cs42l51_probe(struct snd_soc_codec *codec) snd_soc_add_controls(codec, cs42l51_snd_controls, ARRAY_SIZE(cs42l51_snd_controls)); - snd_soc_dapm_new_controls(codec, cs42l51_dapm_widgets, + snd_soc_dapm_new_controls(dapm, cs42l51_dapm_widgets, ARRAY_SIZE(cs42l51_dapm_widgets)); - snd_soc_dapm_add_routes(codec, cs42l51_routes, + snd_soc_dapm_add_routes(dapm, cs42l51_routes, ARRAY_SIZE(cs42l51_routes)); return 0; diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c index e8d27c8f9ba3..a9521acad99c 100644 --- a/sound/soc/codecs/cx20442.c +++ b/sound/soc/codecs/cx20442.c @@ -18,7 +18,7 @@ #include <sound/core.h> #include <sound/initval.h> -#include <sound/soc-dapm.h> +#include <sound/soc.h> #include "cx20442.h" @@ -89,10 +89,11 @@ static const struct snd_soc_dapm_route cx20442_audio_map[] = { static int cx20442_add_widgets(struct snd_soc_codec *codec) { - snd_soc_dapm_new_controls(codec, cx20442_dapm_widgets, - ARRAY_SIZE(cx20442_dapm_widgets)); + struct snd_soc_dapm_context *dapm = &codec->dapm; - snd_soc_dapm_add_routes(codec, cx20442_audio_map, + snd_soc_dapm_new_controls(dapm, cx20442_dapm_widgets, + ARRAY_SIZE(cx20442_dapm_widgets)); + snd_soc_dapm_add_routes(dapm, cx20442_audio_map, ARRAY_SIZE(cx20442_audio_map)); return 0; @@ -263,7 +264,7 @@ static void v253_close(struct tty_struct *tty) /* Prevent the codec driver from further accessing the modem */ codec->hw_write = NULL; cx20442->control_data = NULL; - codec->pop_time = 0; + codec->card->pop_time = 0; } /* Line discipline .hangup() */ @@ -291,7 +292,7 @@ static void v253_receive(struct tty_struct *tty, /* Set up codec driver access to modem controls */ cx20442->control_data = tty; codec->hw_write = (hw_write_t)tty->ops->write; - codec->pop_time = 1; + codec->card->pop_time = 1; } } @@ -348,7 +349,7 @@ static int cx20442_codec_probe(struct snd_soc_codec *codec) cx20442->control_data = NULL; codec->hw_write = NULL; - codec->pop_time = 0; + codec->card->pop_time = 0; return 0; } diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 58bb9b994811..92fd9d7a9221 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -21,7 +21,7 @@ #include <linux/slab.h> #include <sound/pcm.h> #include <sound/pcm_params.h> -#include <sound/soc-dapm.h> +#include <sound/soc.h> #include <sound/initval.h> #include <sound/tlv.h> diff --git a/sound/soc/codecs/jz4740.c b/sound/soc/codecs/jz4740.c index 16253ec9b022..8a45562a96d4 100644 --- a/sound/soc/codecs/jz4740.c +++ b/sound/soc/codecs/jz4740.c @@ -266,7 +266,7 @@ static int jz4740_codec_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: /* The only way to clear the suspend flag is to reset the codec */ - if (codec->bias_level == SND_SOC_BIAS_OFF) + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) jz4740_codec_wakeup(codec); mask = JZ4740_CODEC_1_VREF_DISABLE | @@ -288,23 +288,25 @@ static int jz4740_codec_set_bias_level(struct snd_soc_codec *codec, break; } - codec->bias_level = level; + codec->dapm.bias_level = level; return 0; } static int jz4740_codec_dev_probe(struct snd_soc_codec *codec) { + struct snd_soc_dapm_context *dapm = &codec->dapm; + snd_soc_update_bits(codec, JZ4740_REG_CODEC_1, JZ4740_CODEC_1_SW2_ENABLE, JZ4740_CODEC_1_SW2_ENABLE); snd_soc_add_controls(codec, jz4740_codec_controls, ARRAY_SIZE(jz4740_codec_controls)); - snd_soc_dapm_new_controls(codec, jz4740_codec_dapm_widgets, + snd_soc_dapm_new_controls(dapm, jz4740_codec_dapm_widgets, ARRAY_SIZE(jz4740_codec_dapm_widgets)); - snd_soc_dapm_add_routes(codec, jz4740_codec_dapm_routes, + snd_soc_dapm_add_routes(dapm, jz4740_codec_dapm_routes, ARRAY_SIZE(jz4740_codec_dapm_routes)); snd_soc_dapm_new_widgets(codec); diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index bc22ee93a75d..ef06007d8895 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -1224,15 +1224,17 @@ static const struct snd_soc_dapm_route audio_map[] = { static int max98088_add_widgets(struct snd_soc_codec *codec) { - snd_soc_dapm_new_controls(codec, max98088_dapm_widgets, + struct snd_soc_dapm_context *dapm = &codec->dapm; + + snd_soc_dapm_new_controls(dapm, max98088_dapm_widgets, ARRAY_SIZE(max98088_dapm_widgets)); - snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); snd_soc_add_controls(codec, max98088_snd_controls, ARRAY_SIZE(max98088_snd_controls)); - snd_soc_dapm_new_widgets(codec); + snd_soc_dapm_new_widgets(dapm); return 0; } @@ -1617,7 +1619,7 @@ static int max98088_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: - if (codec->bias_level == SND_SOC_BIAS_OFF) + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) max98088_sync_cache(codec); snd_soc_update_bits(codec, M98088_REG_4C_PWR_EN_IN, @@ -1630,7 +1632,7 @@ static int max98088_set_bias_level(struct snd_soc_codec *codec, codec->cache_sync = 1; break; } - codec->bias_level = level; + codec->dapm.bias_level = level; return 0; } diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 6f38d619bf8a..adbc3e8dafc8 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -207,10 +207,11 @@ static const struct snd_soc_dapm_route audio_conn[] = { static int ssm2602_add_widgets(struct snd_soc_codec *codec) { - snd_soc_dapm_new_controls(codec, ssm2602_dapm_widgets, - ARRAY_SIZE(ssm2602_dapm_widgets)); + struct snd_soc_dapm_context *dapm = &codec->dapm; - snd_soc_dapm_add_routes(codec, audio_conn, ARRAY_SIZE(audio_conn)); + snd_soc_dapm_new_controls(dapm, ssm2602_dapm_widgets, + ARRAY_SIZE(ssm2602_dapm_widgets)); + snd_soc_dapm_add_routes(dapm, audio_conn, ARRAY_SIZE(audio_conn)); return 0; } @@ -493,7 +494,7 @@ static int ssm2602_set_bias_level(struct snd_soc_codec *codec, break; } - codec->bias_level = level; + codec->dapm.bias_level = level; return 0; } diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index 00d67cc8e206..8aad3a2c4f3d 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -24,6 +24,7 @@ #include <sound/initval.h> #include <sound/pcm_params.h> #include <sound/soc.h> +#include <sound/soc-dapm.h> #include <sound/tlv.h> #include "stac9766.h" @@ -236,7 +237,7 @@ static int stac9766_set_bias_level(struct snd_soc_codec *codec, stac9766_ac97_write(codec, AC97_POWERDOWN, 0xffff); break; } - codec->bias_level = level; + codec->dapm.bias_level = level; return 0; } diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index e8652b1ae326..d9d8e844d63f 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -391,11 +391,12 @@ static int set_sample_rate_control(struct snd_soc_codec *codec, int mclk, static int tlv320aic23_add_widgets(struct snd_soc_codec *codec) { - snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets, - ARRAY_SIZE(tlv320aic23_dapm_widgets)); + struct snd_soc_dapm_context *dapm = &codec->dapm; + snd_soc_dapm_new_controls(dapm, tlv320aic23_dapm_widgets, + ARRAY_SIZE(tlv320aic23_dapm_widgets)); /* set up audio path interconnects */ - snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); return 0; } @@ -574,7 +575,7 @@ static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec, tlv320aic23_write(codec, TLV320AIC23_PWR, 0xffff); break; } - codec->bias_level = level; + codec->dapm.bias_level = level; return 0; } diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index fc687790188b..6173c2b4c364 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -183,7 +183,7 @@ static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol, if (snd_soc_test_bits(widget->codec, reg, val_mask, val)) { /* find dapm widget path assoc with kcontrol */ - list_for_each_entry(path, &widget->codec->dapm_paths, list) { + list_for_each_entry(path, &widget->dapm->paths, list) { if (path->kcontrol != kcontrol) continue; @@ -199,7 +199,7 @@ static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol, } if (found) - snd_soc_dapm_sync(widget->codec); + snd_soc_dapm_sync(widget->dapm); } ret = snd_soc_update_bits(widget->codec, reg, val_mask, val); @@ -788,17 +788,19 @@ static const struct snd_soc_dapm_route intercon_3007[] = { static int aic3x_add_widgets(struct snd_soc_codec *codec) { struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); + struct snd_soc_dapm_context *dapm = &codec->dapm; - snd_soc_dapm_new_controls(codec, aic3x_dapm_widgets, + snd_soc_dapm_new_controls(dapm, aic3x_dapm_widgets, ARRAY_SIZE(aic3x_dapm_widgets)); /* set up audio path interconnects */ - snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); if (aic3x->model == AIC3X_MODEL_3007) { - snd_soc_dapm_new_controls(codec, aic3007_dapm_widgets, + snd_soc_dapm_new_controls(dapm, aic3007_dapm_widgets, ARRAY_SIZE(aic3007_dapm_widgets)); - snd_soc_dapm_add_routes(codec, intercon_3007, ARRAY_SIZE(intercon_3007)); + snd_soc_dapm_add_routes(dapm, intercon_3007, + ARRAY_SIZE(intercon_3007)); } return 0; @@ -1135,7 +1137,7 @@ static int aic3x_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_ON: break; case SND_SOC_BIAS_PREPARE: - if (codec->bias_level == SND_SOC_BIAS_STANDBY && + if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY && aic3x->master) { /* enable pll */ reg = snd_soc_read(codec, AIC3X_PLL_PROGA_REG); @@ -1146,7 +1148,7 @@ static int aic3x_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_STANDBY: if (!aic3x->power) aic3x_set_power(codec, 1); - if (codec->bias_level == SND_SOC_BIAS_PREPARE && + if (codec->dapm.bias_level == SND_SOC_BIAS_PREPARE && aic3x->master) { /* disable pll */ reg = snd_soc_read(codec, AIC3X_PLL_PROGA_REG); @@ -1159,7 +1161,7 @@ static int aic3x_set_bias_level(struct snd_soc_codec *codec, aic3x_set_power(codec, 0); break; } - codec->bias_level = level; + codec->dapm.bias_level = level; return 0; } @@ -1351,7 +1353,7 @@ static int aic3x_probe(struct snd_soc_codec *codec) codec->control_data = aic3x->control_data; aic3x->codec = codec; - codec->idle_bias_off = 1; + codec->dapm.idle_bias_off = 1; ret = snd_soc_codec_set_cache_io(codec, 8, 8, aic3x->control_type); if (ret != 0) { diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index c5ab8c805771..7149c14b289e 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -628,11 +628,12 @@ static const struct snd_soc_dapm_route audio_map[] = { static int dac33_add_widgets(struct snd_soc_codec *codec) { - snd_soc_dapm_new_controls(codec, dac33_dapm_widgets, - ARRAY_SIZE(dac33_dapm_widgets)); + struct snd_soc_dapm_context *dapm = &codec->dapm; + snd_soc_dapm_new_controls(dapm, dac33_dapm_widgets, + ARRAY_SIZE(dac33_dapm_widgets)); /* set up audio path interconnects */ - snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); return 0; } @@ -649,7 +650,7 @@ static int dac33_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: - if (codec->bias_level == SND_SOC_BIAS_OFF) { + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { /* Coming from OFF, switch on the codec */ ret = dac33_hard_power(codec, 1); if (ret != 0) @@ -660,14 +661,14 @@ static int dac33_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_OFF: /* Do not power off, when the codec is already off */ - if (codec->bias_level == SND_SOC_BIAS_OFF) + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) return 0; ret = dac33_hard_power(codec, 0); if (ret != 0) return ret; break; } - codec->bias_level = level; + codec->dapm.bias_level = level; return 0; } @@ -1415,7 +1416,7 @@ static int dac33_soc_probe(struct snd_soc_codec *codec) codec->control_data = dac33->control_data; codec->hw_write = (hw_write_t) i2c_master_send; - codec->idle_bias_off = 1; + codec->dapm.idle_bias_off = 1; dac33->codec = codec; /* Read the tlv320dac33 ID registers */ diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index ee4fb201de60..f9a92ea6b50a 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -388,16 +388,17 @@ static const struct snd_soc_dapm_route audio_map[] = { int tpa6130a2_add_controls(struct snd_soc_codec *codec) { struct tpa6130a2_data *data; + struct snd_soc_dapm_context *dapm = &codec->dapm; if (tpa6130a2_client == NULL) return -ENODEV; data = i2c_get_clientdata(tpa6130a2_client); - snd_soc_dapm_new_controls(codec, tpa6130a2_dapm_widgets, + snd_soc_dapm_new_controls(dapm, tpa6130a2_dapm_widgets, ARRAY_SIZE(tpa6130a2_dapm_widgets)); - snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); if (data->id == TPA6140A2) return snd_soc_add_controls(codec, tpa6140a2_controls, diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index cbebec6ba1ba..f4602e8b67cc 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1621,10 +1621,11 @@ static const struct snd_soc_dapm_route intercon[] = { static int twl4030_add_widgets(struct snd_soc_codec *codec) { - snd_soc_dapm_new_controls(codec, twl4030_dapm_widgets, - ARRAY_SIZE(twl4030_dapm_widgets)); + struct snd_soc_dapm_context *dapm = &codec->dapm; - snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + snd_soc_dapm_new_controls(dapm, twl4030_dapm_widgets, + ARRAY_SIZE(twl4030_dapm_widgets)); + snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); return 0; } @@ -1638,14 +1639,14 @@ static int twl4030_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: - if (codec->bias_level == SND_SOC_BIAS_OFF) + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) twl4030_codec_enable(codec, 1); break; case SND_SOC_BIAS_OFF: twl4030_codec_enable(codec, 0); break; } - codec->bias_level = level; + codec->dapm.bias_level = level; return 0; } @@ -2245,7 +2246,7 @@ static int twl4030_soc_probe(struct snd_soc_codec *codec) snd_soc_codec_set_drvdata(codec, twl4030); /* Set the defaults, and power up the codec */ twl4030->sysclk = twl4030_codec_get_mclk() / 1000; - codec->idle_bias_off = 1; + codec->dapm.idle_bias_off = 1; twl4030_init_chip(codec); diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 10f6e5214511..0dd2d5397264 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -641,12 +641,12 @@ static const struct snd_soc_dapm_route intercon[] = { static int twl6040_add_widgets(struct snd_soc_codec *codec) { - snd_soc_dapm_new_controls(codec, twl6040_dapm_widgets, - ARRAY_SIZE(twl6040_dapm_widgets)); - - snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + struct snd_soc_dapm_context *dapm = &codec->dapm; - snd_soc_dapm_new_widgets(codec); + snd_soc_dapm_new_controls(dapm, twl6040_dapm_widgets, + ARRAY_SIZE(twl6040_dapm_widgets)); + snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); + snd_soc_dapm_new_widgets(dapm); return 0; } @@ -739,7 +739,7 @@ static int twl6040_set_bias_level(struct snd_soc_codec *codec, break; } - codec->bias_level = level; + codec->dapm.bias_level = level; return 0; } diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index 7540a509a6f5..8ea81d48124a 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -389,7 +389,7 @@ static int uda134x_set_bias_level(struct snd_soc_codec *codec, pd->power(0); break; } - codec->bias_level = level; + codec->dapm.bias_level = level; return 0; } diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index 0c6c725736c6..cd6dd19fa1aa 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -414,10 +414,11 @@ static const struct snd_soc_dapm_route audio_map[] = { static int uda1380_add_widgets(struct snd_soc_codec *codec) { - snd_soc_dapm_new_controls(codec, uda1380_dapm_widgets, - ARRAY_SIZE(uda1380_dapm_widgets)); + struct snd_soc_dapm_context *dapm = &codec->dapm; - snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_new_controls(dapm, uda1380_dapm_widgets, + ARRAY_SIZE(uda1380_dapm_widgets)); + snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); return 0; } @@ -603,7 +604,7 @@ static int uda1380_set_bias_level(struct snd_soc_codec *codec, int reg; struct uda1380_platform_data *pdata = codec->dev->platform_data; - if (codec->bias_level == level) + if (codec->dapm.bias_level == level) return 0; switch (level) { @@ -613,7 +614,7 @@ static int uda1380_set_bias_level(struct snd_soc_codec *codec, uda1380_write(codec, UDA1380_PM, R02_PON_BIAS | pm); break; case SND_SOC_BIAS_STANDBY: - if (codec->bias_level == SND_SOC_BIAS_OFF) { + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { if (gpio_is_valid(pdata->gpio_power)) { gpio_set_value(pdata->gpio_power, 1); mdelay(1); @@ -636,7 +637,7 @@ static int uda1380_set_bias_level(struct snd_soc_codec *codec, for (reg = UDA1380_MVOL; reg < UDA1380_CACHEREGNUM; reg++) set_bit(reg - 0x10, &uda1380_cache_dirty); } - codec->bias_level = level; + codec->dapm.bias_level = level; return 0; } diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index 4bcd168794e1..9277d8d7474e 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -705,6 +705,7 @@ static const struct snd_soc_dapm_route audio_map[] = { /* Called from the machine driver */ int wm2000_add_controls(struct snd_soc_codec *codec) { + struct snd_soc_dapm_context *dapm = &codec->dapm; int ret; if (!wm2000_i2c) { @@ -712,12 +713,12 @@ int wm2000_add_controls(struct snd_soc_codec *codec) return -ENODEV; } - ret = snd_soc_dapm_new_controls(codec, wm2000_dapm_widgets, + ret = snd_soc_dapm_new_controls(dapm, wm2000_dapm_widgets, ARRAY_SIZE(wm2000_dapm_widgets)); if (ret < 0) return ret; - ret = snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + ret = snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); if (ret < 0) return ret; diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 7611add7f8c3..d5e6e02ff350 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -230,8 +230,9 @@ static inline int wm8350_out2_ramp_step(struct snd_soc_codec *codec) */ static void wm8350_pga_work(struct work_struct *work) { - struct snd_soc_codec *codec = - container_of(work, struct snd_soc_codec, delayed_work.work); + struct snd_soc_dapm_context *dapm = + container_of(work, struct snd_soc_dapm_context, delayed_work.work); + struct snd_soc_codec *codec = dapm->codec; struct wm8350_data *wm8350_data = snd_soc_codec_get_drvdata(codec); struct wm8350_output *out1 = &wm8350_data->out1, *out2 = &wm8350_data->out2; @@ -302,8 +303,8 @@ static int pga_event(struct snd_soc_dapm_widget *w, out->ramp = WM8350_RAMP_UP; out->active = 1; - if (!delayed_work_pending(&codec->delayed_work)) - schedule_delayed_work(&codec->delayed_work, + if (!delayed_work_pending(&codec->dapm.delayed_work)) + schedule_delayed_work(&codec->dapm.delayed_work, msecs_to_jiffies(1)); break; @@ -311,8 +312,8 @@ static int pga_event(struct snd_soc_dapm_widget *w, out->ramp = WM8350_RAMP_DOWN; out->active = 0; - if (!delayed_work_pending(&codec->delayed_work)) - schedule_delayed_work(&codec->delayed_work, + if (!delayed_work_pending(&codec->dapm.delayed_work)) + schedule_delayed_work(&codec->dapm.delayed_work, msecs_to_jiffies(1)); break; } @@ -786,9 +787,10 @@ static const struct snd_soc_dapm_route audio_map[] = { static int wm8350_add_widgets(struct snd_soc_codec *codec) { + struct snd_soc_dapm_context *dapm = &codec->dapm; int ret; - ret = snd_soc_dapm_new_controls(codec, + ret = snd_soc_dapm_new_controls(dapm, wm8350_dapm_widgets, ARRAY_SIZE(wm8350_dapm_widgets)); if (ret != 0) { @@ -797,7 +799,7 @@ static int wm8350_add_widgets(struct snd_soc_codec *codec) } /* set up audio paths */ - ret = snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + ret = snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); if (ret != 0) { dev_err(codec->dev, "DAPM route register failed\n"); return ret; @@ -1184,7 +1186,7 @@ static int wm8350_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: - if (codec->bias_level == SND_SOC_BIAS_OFF) { + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { ret = regulator_bulk_enable(ARRAY_SIZE(priv->supplies), priv->supplies); if (ret != 0) @@ -1317,7 +1319,7 @@ static int wm8350_set_bias_level(struct snd_soc_codec *codec, priv->supplies); break; } - codec->bias_level = level; + codec->dapm.bias_level = level; return 0; } @@ -1550,7 +1552,7 @@ static int wm8350_codec_probe(struct snd_soc_codec *codec) /* Put the codec into reset if it wasn't already */ wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_5, WM8350_CODEC_ENA); - INIT_DELAYED_WORK(&codec->delayed_work, wm8350_pga_work); + INIT_DELAYED_WORK(&codec->dapm.delayed_work, wm8350_pga_work); /* Enable the codec */ wm8350_set_bits(wm8350, WM8350_POWER_MGMT_5, WM8350_CODEC_ENA); @@ -1642,12 +1644,12 @@ static int wm8350_codec_remove(struct snd_soc_codec *codec) priv->mic.jack = NULL; /* cancel any work waiting to be queued. */ - ret = cancel_delayed_work(&codec->delayed_work); + ret = cancel_delayed_work(&codec->dapm.delayed_work); /* if there was any work waiting then we run it now and * wait for its completion */ if (ret) { - schedule_delayed_work(&codec->delayed_work, 0); + schedule_delayed_work(&codec->dapm.delayed_work, 0); flush_scheduled_work(); } diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index 850299786e02..96927a457a34 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -911,10 +911,11 @@ static const struct snd_soc_dapm_route audio_map[] = { static int wm8400_add_widgets(struct snd_soc_codec *codec) { - snd_soc_dapm_new_controls(codec, wm8400_dapm_widgets, - ARRAY_SIZE(wm8400_dapm_widgets)); + struct snd_soc_dapm_context *dapm = &codec->dapm; - snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_new_controls(dapm, wm8400_dapm_widgets, + ARRAY_SIZE(wm8400_dapm_widgets)); + snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); return 0; } @@ -1219,7 +1220,7 @@ static int wm8400_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: - if (codec->bias_level == SND_SOC_BIAS_OFF) { + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { ret = regulator_bulk_enable(ARRAY_SIZE(power), &power[0]); if (ret != 0) { @@ -1306,7 +1307,7 @@ static int wm8400_set_bias_level(struct snd_soc_codec *codec, break; } - codec->bias_level = level; + codec->dapm.bias_level = level; return 0; } diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 8f107095760e..6b3833c7bdf3 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -216,10 +216,11 @@ static const struct snd_soc_dapm_route audio_map[] = { static int wm8510_add_widgets(struct snd_soc_codec *codec) { - snd_soc_dapm_new_controls(codec, wm8510_dapm_widgets, - ARRAY_SIZE(wm8510_dapm_widgets)); + struct snd_soc_dapm_context *dapm = &codec->dapm; - snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_new_controls(dapm, wm8510_dapm_widgets, + ARRAY_SIZE(wm8510_dapm_widgets)); + snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); return 0; } @@ -478,7 +479,7 @@ static int wm8510_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_STANDBY: power1 |= WM8510_POWER1_BIASEN | WM8510_POWER1_BUFIOEN; - if (codec->bias_level == SND_SOC_BIAS_OFF) { + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { /* Initial cap charge at VMID 5k */ snd_soc_write(codec, WM8510_POWER1, power1 | 0x3); mdelay(100); @@ -495,7 +496,7 @@ static int wm8510_set_bias_level(struct snd_soc_codec *codec, break; } - codec->bias_level = level; + codec->dapm.bias_level = level; return 0; } diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c index 712ef7c76f90..d3318886f43e 100644 --- a/sound/soc/codecs/wm8523.c +++ b/sound/soc/codecs/wm8523.c @@ -110,10 +110,11 @@ static const struct snd_soc_dapm_route intercon[] = { static int wm8523_add_widgets(struct snd_soc_codec *codec) { - snd_soc_dapm_new_controls(codec, wm8523_dapm_widgets, - ARRAY_SIZE(wm8523_dapm_widgets)); + struct snd_soc_dapm_context *dapm = &codec->dapm; - snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + snd_soc_dapm_new_controls(dapm, wm8523_dapm_widgets, + ARRAY_SIZE(wm8523_dapm_widgets)); + snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); return 0; } @@ -328,7 +329,7 @@ static int wm8523_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: - if (codec->bias_level == SND_SOC_BIAS_OFF) { + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { ret = regulator_bulk_enable(ARRAY_SIZE(wm8523->supplies), wm8523->supplies); if (ret != 0) { @@ -367,7 +368,7 @@ static int wm8523_set_bias_level(struct snd_soc_codec *codec, wm8523->supplies); break; } - codec->bias_level = level; + codec->dapm.bias_level = level; return 0; } diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index a2e0ed59b376..36c035bf4e42 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -302,10 +302,11 @@ static const struct snd_soc_dapm_route audio_map[] = { static int wm8580_add_widgets(struct snd_soc_codec *codec) { - snd_soc_dapm_new_controls(codec, wm8580_dapm_widgets, - ARRAY_SIZE(wm8580_dapm_widgets)); + struct snd_soc_dapm_context *dapm = &codec->dapm; - snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_new_controls(dapm, wm8580_dapm_widgets, + ARRAY_SIZE(wm8580_dapm_widgets)); + snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); return 0; } @@ -767,7 +768,7 @@ static int wm8580_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: - if (codec->bias_level == SND_SOC_BIAS_OFF) { + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { /* Power up and get individual control of the DACs */ reg = snd_soc_read(codec, WM8580_PWRDN1); reg &= ~(WM8580_PWRDN1_PWDN | WM8580_PWRDN1_ALLDACPD); @@ -785,7 +786,7 @@ static int wm8580_set_bias_level(struct snd_soc_codec *codec, snd_soc_write(codec, WM8580_PWRDN1, reg | WM8580_PWRDN1_PWDN); break; } - codec->bias_level = level; + codec->dapm.bias_level = level; return 0; } @@ -905,7 +906,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8580 = { .set_bias_level = wm8580_set_bias_level, .reg_cache_size = ARRAY_SIZE(wm8580_reg), .reg_word_size = sizeof(u16), - .reg_cache_default = &wm8580_reg, + .reg_cache_default = wm8580_reg, }; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c index 54fbd76c8bca..ea2daf4da57c 100644 --- a/sound/soc/codecs/wm8711.c +++ b/sound/soc/codecs/wm8711.c @@ -93,10 +93,11 @@ static const struct snd_soc_dapm_route intercon[] = { static int wm8711_add_widgets(struct snd_soc_codec *codec) { - snd_soc_dapm_new_controls(codec, wm8711_dapm_widgets, - ARRAY_SIZE(wm8711_dapm_widgets)); + struct snd_soc_dapm_context *dapm = &codec->dapm; - snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + snd_soc_dapm_new_controls(dapm, wm8711_dapm_widgets, + ARRAY_SIZE(wm8711_dapm_widgets)); + snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); return 0; } @@ -318,7 +319,7 @@ static int wm8711_set_bias_level(struct snd_soc_codec *codec, snd_soc_write(codec, WM8711_PWR, 0xffff); break; } - codec->bias_level = level; + codec->dapm.bias_level = level; return 0; } diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c index 075f35e4f4cb..23939976c3cc 100644 --- a/sound/soc/codecs/wm8728.c +++ b/sound/soc/codecs/wm8728.c @@ -73,10 +73,11 @@ static const struct snd_soc_dapm_route intercon[] = { static int wm8728_add_widgets(struct snd_soc_codec *codec) { - snd_soc_dapm_new_controls(codec, wm8728_dapm_widgets, - ARRAY_SIZE(wm8728_dapm_widgets)); + struct snd_soc_dapm_context *dapm = &codec->dapm; - snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + snd_soc_dapm_new_controls(dapm, wm8728_dapm_widgets, + ARRAY_SIZE(wm8728_dapm_widgets)); + snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); return 0; } @@ -180,7 +181,7 @@ static int wm8728_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_ON: case SND_SOC_BIAS_PREPARE: case SND_SOC_BIAS_STANDBY: - if (codec->bias_level == SND_SOC_BIAS_OFF) { + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { /* Power everything up... */ reg = snd_soc_read(codec, WM8728_DACCTL); snd_soc_write(codec, WM8728_DACCTL, reg & ~0x4); @@ -197,7 +198,7 @@ static int wm8728_set_bias_level(struct snd_soc_codec *codec, snd_soc_write(codec, WM8728_DACCTL, reg | 0x4); break; } - codec->bias_level = level; + codec->dapm.bias_level = level; return 0; } diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 631385802eb4..95ade3245056 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -165,10 +165,11 @@ static const struct snd_soc_dapm_route intercon[] = { static int wm8731_add_widgets(struct snd_soc_codec *codec) { - snd_soc_dapm_new_controls(codec, wm8731_dapm_widgets, - ARRAY_SIZE(wm8731_dapm_widgets)); + struct snd_soc_dapm_context *dapm = &codec->dapm; - snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + snd_soc_dapm_new_controls(dapm, wm8731_dapm_widgets, + ARRAY_SIZE(wm8731_dapm_widgets)); + snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); return 0; } @@ -319,7 +320,7 @@ static int wm8731_set_dai_sysclk(struct snd_soc_dai *codec_dai, return -EINVAL; } - snd_soc_dapm_sync(codec); + snd_soc_dapm_sync(&codec->dapm); return 0; } @@ -399,7 +400,7 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: - if (codec->bias_level == SND_SOC_BIAS_OFF) { + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { ret = regulator_bulk_enable(ARRAY_SIZE(wm8731->supplies), wm8731->supplies); if (ret != 0) @@ -428,7 +429,7 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec, wm8731->supplies); break; } - codec->bias_level = level; + codec->dapm.bias_level = level; return 0; } diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index 90e31e9aa6f7..2543a26513fb 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -95,10 +95,11 @@ static const struct snd_soc_dapm_route intercon[] = { static int wm8741_add_widgets(struct snd_soc_codec *codec) { - snd_soc_dapm_new_controls(codec, wm8741_dapm_widgets, - ARRAY_SIZE(wm8741_dapm_widgets)); + struct snd_soc_dapm_context *dapm = &codec->dapm; - snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + snd_soc_dapm_new_controls(dapm, wm8741_dapm_widgets, + ARRAY_SIZE(wm8741_dapm_widgets)); + snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); return 0; } @@ -455,7 +456,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8741 = { .resume = wm8741_resume, .reg_cache_size = ARRAY_SIZE(wm8741_reg_defaults), .reg_word_size = sizeof(u16), - .reg_cache_default = &wm8741_reg_defaults, + .reg_cache_default = wm8741_reg_defaults, }; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 6c924cd2cfd4..178b967af73f 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -399,10 +399,11 @@ static const struct snd_soc_dapm_route audio_map[] = { static int wm8750_add_widgets(struct snd_soc_codec *codec) { - snd_soc_dapm_new_controls(codec, wm8750_dapm_widgets, - ARRAY_SIZE(wm8750_dapm_widgets)); + struct snd_soc_dapm_context *dapm = &codec->dapm; - snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_new_controls(dapm, wm8750_dapm_widgets, + ARRAY_SIZE(wm8750_dapm_widgets)); + snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); return 0; } @@ -615,7 +616,7 @@ static int wm8750_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: - if (codec->bias_level == SND_SOC_BIAS_OFF) { + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { /* Set VMID to 5k */ snd_soc_write(codec, WM8750_PWR1, pwr_reg | 0x01c1); @@ -630,7 +631,7 @@ static int wm8750_set_bias_level(struct snd_soc_codec *codec, snd_soc_write(codec, WM8750_PWR1, 0x0001); break; } - codec->bias_level = level; + codec->dapm.bias_level = level; return 0; } diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 8f679a13f2bc..26096b47a493 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -670,10 +670,11 @@ static const struct snd_soc_dapm_route audio_map[] = { static int wm8753_add_widgets(struct snd_soc_codec *codec) { - snd_soc_dapm_new_controls(codec, wm8753_dapm_widgets, - ARRAY_SIZE(wm8753_dapm_widgets)); + struct snd_soc_dapm_context *dapm = &codec->dapm; - snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_new_controls(dapm, wm8753_dapm_widgets, + ARRAY_SIZE(wm8753_dapm_widgets)); + snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); return 0; } @@ -1292,7 +1293,7 @@ static int wm8753_set_bias_level(struct snd_soc_codec *codec, wm8753_write(codec, WM8753_PWR1, 0x0001); break; } - codec->bias_level = level; + codec->dapm.bias_level = level; return 0; } @@ -1482,9 +1483,11 @@ static void wm8753_set_dai_mode(struct snd_soc_codec *codec, static void wm8753_work(struct work_struct *work) { - struct snd_soc_codec *codec = - container_of(work, struct snd_soc_codec, delayed_work.work); - wm8753_set_bias_level(codec, codec->bias_level); + struct snd_soc_dapm_context *dapm = + container_of(work, struct snd_soc_dapm_context, + delayed_work.work); + struct snd_soc_codec *codec = dapm->codec; + wm8753_set_bias_level(codec, dapm->bias_level); } static int wm8753_suspend(struct snd_soc_codec *codec, pm_message_t state) @@ -1516,10 +1519,10 @@ static int wm8753_resume(struct snd_soc_codec *codec) wm8753_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* charge wm8753 caps */ - if (codec->suspend_bias_level == SND_SOC_BIAS_ON) { + if (codec->dapm.suspend_bias_level == SND_SOC_BIAS_ON) { wm8753_set_bias_level(codec, SND_SOC_BIAS_PREPARE); - codec->bias_level = SND_SOC_BIAS_ON; - schedule_delayed_work(&codec->delayed_work, + codec->dapm.bias_level = SND_SOC_BIAS_ON; + schedule_delayed_work(&codec->dapm.delayed_work, msecs_to_jiffies(caps_charge)); } @@ -1550,7 +1553,7 @@ static int wm8753_probe(struct snd_soc_codec *codec) struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec); int ret = 0, reg; - INIT_DELAYED_WORK(&codec->delayed_work, wm8753_work); + INIT_DELAYED_WORK(&codec->dapm.delayed_work, wm8753_work); ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8753->control_type); if (ret < 0) { @@ -1569,7 +1572,7 @@ static int wm8753_probe(struct snd_soc_codec *codec) /* charge output caps */ wm8753_set_bias_level(codec, SND_SOC_BIAS_PREPARE); - schedule_delayed_work(&codec->delayed_work, + schedule_delayed_work(&codec->dapm.delayed_work, msecs_to_jiffies(caps_charge)); /* set the update bits */ @@ -1604,7 +1607,7 @@ static int wm8753_probe(struct snd_soc_codec *codec) /* power down chip */ static int wm8753_remove(struct snd_soc_codec *codec) { - run_delayed_work(&codec->delayed_work); + run_delayed_work(&codec->dapm.delayed_work); wm8753_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; diff --git a/sound/soc/codecs/wm8770.c b/sound/soc/codecs/wm8770.c new file mode 100644 index 000000000000..8608b4a10b0a --- /dev/null +++ b/sound/soc/codecs/wm8770.c @@ -0,0 +1,750 @@ +/* + * wm8770.c -- WM8770 ALSA SoC Audio driver + * + * Copyright 2010 Wolfson Microelectronics plc + * + * Author: Dimitris Papastamos <dp@opensource.wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/platform_device.h> +#include <linux/spi/spi.h> +#include <linux/regulator/consumer.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "wm8770.h" + +#define WM8770_NUM_SUPPLIES 3 +static const char *wm8770_supply_names[WM8770_NUM_SUPPLIES] = { + "AVDD1", + "AVDD2", + "DVDD" +}; + +static const u16 wm8770_reg_defs[WM8770_CACHEREGNUM] = { + 0x7f, 0x7f, 0x7f, 0x7f, + 0x7f, 0x7f, 0x7f, 0x7f, + 0x7f, 0xff, 0xff, 0xff, + 0xff, 0xff, 0xff, 0xff, + 0xff, 0xff, 0, 0x90, 0, + 0, 0x22, 0x22, 0x3e, + 0xc, 0xc, 0x100, 0x189, + 0x189, 0x8770 +}; + +struct wm8770_priv { + enum snd_soc_control_type control_type; + struct regulator_bulk_data supplies[WM8770_NUM_SUPPLIES]; + struct notifier_block disable_nb[WM8770_NUM_SUPPLIES]; + struct snd_soc_codec *codec; + int sysclk; +}; + +static int vout12supply_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event); +static int vout34supply_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event); + +/* + * 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 WM8770_REGULATOR_EVENT(n) \ +static int wm8770_regulator_event_##n(struct notifier_block *nb, \ + unsigned long event, void *data) \ +{ \ + struct wm8770_priv *wm8770 = container_of(nb, struct wm8770_priv, \ + disable_nb[n]); \ + if (event & REGULATOR_EVENT_DISABLE) { \ + wm8770->codec->cache_sync = 1; \ + } \ + return 0; \ +} + +WM8770_REGULATOR_EVENT(0) +WM8770_REGULATOR_EVENT(1) +WM8770_REGULATOR_EVENT(2) + +static const DECLARE_TLV_DB_SCALE(adc_tlv, -1200, 100, 0); +static const DECLARE_TLV_DB_SCALE(dac_dig_tlv, -12750, 50, 1); +static const DECLARE_TLV_DB_SCALE(dac_alg_tlv, -12700, 100, 1); + +static const char *dac_phase_text[][2] = { + { "DAC1 Normal", "DAC1 Inverted" }, + { "DAC2 Normal", "DAC2 Inverted" }, + { "DAC3 Normal", "DAC3 Inverted" }, + { "DAC4 Normal", "DAC4 Inverted" }, +}; + +static const struct soc_enum dac_phase[] = { + SOC_ENUM_DOUBLE(WM8770_DACPHASE, 0, 1, 2, dac_phase_text[0]), + SOC_ENUM_DOUBLE(WM8770_DACPHASE, 2, 3, 2, dac_phase_text[1]), + SOC_ENUM_DOUBLE(WM8770_DACPHASE, 4, 5, 2, dac_phase_text[2]), + SOC_ENUM_DOUBLE(WM8770_DACPHASE, 6, 7, 2, dac_phase_text[3]), +}; + +static const struct snd_kcontrol_new wm8770_snd_controls[] = { + /* global DAC playback controls */ + SOC_SINGLE_TLV("DAC Playback Volume", WM8770_MSDIGVOL, 0, 255, 0, + dac_dig_tlv), + SOC_SINGLE("DAC Playback Switch", WM8770_DACMUTE, 4, 1, 1), + SOC_SINGLE("DAC Playback ZC Switch", WM8770_DACCTRL1, 0, 1, 0), + + /* global VOUT playback controls */ + SOC_SINGLE_TLV("VOUT Playback Volume", WM8770_MSALGVOL, 0, 127, 0, + dac_alg_tlv), + SOC_SINGLE("VOUT Playback ZC Switch", WM8770_MSALGVOL, 7, 1, 0), + + /* VOUT1/2/3/4 specific controls */ + SOC_DOUBLE_R_TLV("VOUT1 Playback Volume", WM8770_VOUT1LVOL, + WM8770_VOUT1RVOL, 0, 127, 0, dac_alg_tlv), + SOC_DOUBLE_R("VOUT1 Playback ZC Switch", WM8770_VOUT1LVOL, + WM8770_VOUT1RVOL, 7, 1, 0), + SOC_DOUBLE_R_TLV("VOUT2 Playback Volume", WM8770_VOUT2LVOL, + WM8770_VOUT2RVOL, 0, 127, 0, dac_alg_tlv), + SOC_DOUBLE_R("VOUT2 Playback ZC Switch", WM8770_VOUT2LVOL, + WM8770_VOUT2RVOL, 7, 1, 0), + SOC_DOUBLE_R_TLV("VOUT3 Playback Volume", WM8770_VOUT3LVOL, + WM8770_VOUT3RVOL, 0, 127, 0, dac_alg_tlv), + SOC_DOUBLE_R("VOUT3 Playback ZC Switch", WM8770_VOUT3LVOL, + WM8770_VOUT3RVOL, 7, 1, 0), + SOC_DOUBLE_R_TLV("VOUT4 Playback Volume", WM8770_VOUT4LVOL, + WM8770_VOUT4RVOL, 0, 127, 0, dac_alg_tlv), + SOC_DOUBLE_R("VOUT4 Playback ZC Switch", WM8770_VOUT4LVOL, + WM8770_VOUT4RVOL, 7, 1, 0), + + /* DAC1/2/3/4 specific controls */ + SOC_DOUBLE_R_TLV("DAC1 Playback Volume", WM8770_DAC1LVOL, + WM8770_DAC1RVOL, 0, 255, 0, dac_dig_tlv), + SOC_SINGLE("DAC1 Deemphasis Switch", WM8770_DACCTRL2, 0, 1, 0), + SOC_ENUM("DAC1 Phase", dac_phase[0]), + SOC_DOUBLE_R_TLV("DAC2 Playback Volume", WM8770_DAC2LVOL, + WM8770_DAC2RVOL, 0, 255, 0, dac_dig_tlv), + SOC_SINGLE("DAC2 Deemphasis Switch", WM8770_DACCTRL2, 1, 1, 0), + SOC_ENUM("DAC2 Phase", dac_phase[1]), + SOC_DOUBLE_R_TLV("DAC3 Playback Volume", WM8770_DAC3LVOL, + WM8770_DAC3RVOL, 0, 255, 0, dac_dig_tlv), + SOC_SINGLE("DAC3 Deemphasis Switch", WM8770_DACCTRL2, 2, 1, 0), + SOC_ENUM("DAC3 Phase", dac_phase[2]), + SOC_DOUBLE_R_TLV("DAC4 Playback Volume", WM8770_DAC4LVOL, + WM8770_DAC4RVOL, 0, 255, 0, dac_dig_tlv), + SOC_SINGLE("DAC4 Deemphasis Switch", WM8770_DACCTRL2, 3, 1, 0), + SOC_ENUM("DAC4 Phase", dac_phase[3]), + + /* ADC specific controls */ + SOC_DOUBLE_R_TLV("Capture Volume", WM8770_ADCLCTRL, WM8770_ADCRCTRL, + 0, 31, 0, adc_tlv), + SOC_DOUBLE_R("Capture Switch", WM8770_ADCLCTRL, WM8770_ADCRCTRL, + 5, 1, 1), + + /* other controls */ + SOC_SINGLE("ADC 128x Oversampling Switch", WM8770_MSTRCTRL, 3, 1, 0), + SOC_SINGLE("ADC Highpass Filter Switch", WM8770_IFACECTRL, 8, 1, 1) +}; + +static const char *ain_text[] = { + "AIN1", "AIN2", "AIN3", "AIN4", + "AIN5", "AIN6", "AIN7", "AIN8" +}; + +static const struct soc_enum ain_enum = + SOC_ENUM_DOUBLE(WM8770_ADCMUX, 0, 4, 8, ain_text); + +static const struct snd_kcontrol_new ain_mux = + SOC_DAPM_ENUM("Capture Mux", ain_enum); + +static const struct snd_kcontrol_new vout1_mix_controls[] = { + SOC_DAPM_SINGLE("DAC1 Switch", WM8770_OUTMUX1, 0, 1, 0), + SOC_DAPM_SINGLE("AUX1 Switch", WM8770_OUTMUX1, 1, 1, 0), + SOC_DAPM_SINGLE("Bypass Switch", WM8770_OUTMUX1, 2, 1, 0) +}; + +static const struct snd_kcontrol_new vout2_mix_controls[] = { + SOC_DAPM_SINGLE("DAC2 Switch", WM8770_OUTMUX1, 3, 1, 0), + SOC_DAPM_SINGLE("AUX2 Switch", WM8770_OUTMUX1, 4, 1, 0), + SOC_DAPM_SINGLE("Bypass Switch", WM8770_OUTMUX1, 5, 1, 0) +}; + +static const struct snd_kcontrol_new vout3_mix_controls[] = { + SOC_DAPM_SINGLE("DAC3 Switch", WM8770_OUTMUX2, 0, 1, 0), + SOC_DAPM_SINGLE("AUX3 Switch", WM8770_OUTMUX2, 1, 1, 0), + SOC_DAPM_SINGLE("Bypass Switch", WM8770_OUTMUX2, 2, 1, 0) +}; + +static const struct snd_kcontrol_new vout4_mix_controls[] = { + SOC_DAPM_SINGLE("DAC4 Switch", WM8770_OUTMUX2, 3, 1, 0), + SOC_DAPM_SINGLE("Bypass Switch", WM8770_OUTMUX2, 4, 1, 0) +}; + +static const struct snd_soc_dapm_widget wm8770_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("AUX1"), + SND_SOC_DAPM_INPUT("AUX2"), + SND_SOC_DAPM_INPUT("AUX3"), + + SND_SOC_DAPM_INPUT("AIN1"), + SND_SOC_DAPM_INPUT("AIN2"), + SND_SOC_DAPM_INPUT("AIN3"), + SND_SOC_DAPM_INPUT("AIN4"), + SND_SOC_DAPM_INPUT("AIN5"), + SND_SOC_DAPM_INPUT("AIN6"), + SND_SOC_DAPM_INPUT("AIN7"), + SND_SOC_DAPM_INPUT("AIN8"), + + SND_SOC_DAPM_MUX("Capture Mux", WM8770_ADCMUX, 8, 1, &ain_mux), + + SND_SOC_DAPM_ADC("ADC", "Capture", WM8770_PWDNCTRL, 1, 1), + + SND_SOC_DAPM_DAC("DAC1", "Playback", WM8770_PWDNCTRL, 2, 1), + SND_SOC_DAPM_DAC("DAC2", "Playback", WM8770_PWDNCTRL, 3, 1), + SND_SOC_DAPM_DAC("DAC3", "Playback", WM8770_PWDNCTRL, 4, 1), + SND_SOC_DAPM_DAC("DAC4", "Playback", WM8770_PWDNCTRL, 5, 1), + + SND_SOC_DAPM_SUPPLY("VOUT12 Supply", SND_SOC_NOPM, 0, 0, + vout12supply_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY("VOUT34 Supply", SND_SOC_NOPM, 0, 0, + vout34supply_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_MIXER("VOUT1 Mixer", SND_SOC_NOPM, 0, 0, + vout1_mix_controls, ARRAY_SIZE(vout1_mix_controls)), + SND_SOC_DAPM_MIXER("VOUT2 Mixer", SND_SOC_NOPM, 0, 0, + vout2_mix_controls, ARRAY_SIZE(vout2_mix_controls)), + SND_SOC_DAPM_MIXER("VOUT3 Mixer", SND_SOC_NOPM, 0, 0, + vout3_mix_controls, ARRAY_SIZE(vout3_mix_controls)), + SND_SOC_DAPM_MIXER("VOUT4 Mixer", SND_SOC_NOPM, 0, 0, + vout4_mix_controls, ARRAY_SIZE(vout4_mix_controls)), + + SND_SOC_DAPM_OUTPUT("VOUT1"), + SND_SOC_DAPM_OUTPUT("VOUT2"), + SND_SOC_DAPM_OUTPUT("VOUT3"), + SND_SOC_DAPM_OUTPUT("VOUT4") +}; + +static const struct snd_soc_dapm_route wm8770_intercon[] = { + { "Capture Mux", "AIN1", "AIN1" }, + { "Capture Mux", "AIN2", "AIN2" }, + { "Capture Mux", "AIN3", "AIN3" }, + { "Capture Mux", "AIN4", "AIN4" }, + { "Capture Mux", "AIN5", "AIN5" }, + { "Capture Mux", "AIN6", "AIN6" }, + { "Capture Mux", "AIN7", "AIN7" }, + { "Capture Mux", "AIN8", "AIN8" }, + + { "ADC", NULL, "Capture Mux" }, + + { "VOUT1 Mixer", NULL, "VOUT12 Supply" }, + { "VOUT1 Mixer", "DAC1 Switch", "DAC1" }, + { "VOUT1 Mixer", "AUX1 Switch", "AUX1" }, + { "VOUT1 Mixer", "Bypass Switch", "Capture Mux" }, + + { "VOUT2 Mixer", NULL, "VOUT12 Supply" }, + { "VOUT2 Mixer", "DAC2 Switch", "DAC2" }, + { "VOUT2 Mixer", "AUX2 Switch", "AUX2" }, + { "VOUT2 Mixer", "Bypass Switch", "Capture Mux" }, + + { "VOUT3 Mixer", NULL, "VOUT34 Supply" }, + { "VOUT3 Mixer", "DAC3 Switch", "DAC3" }, + { "VOUT3 Mixer", "AUX3 Switch", "AUX3" }, + { "VOUT3 Mixer", "Bypass Switch", "Capture Mux" }, + + { "VOUT4 Mixer", NULL, "VOUT34 Supply" }, + { "VOUT4 Mixer", "DAC4 Switch", "DAC4" }, + { "VOUT4 Mixer", "Bypass Switch", "Capture Mux" }, + + { "VOUT1", NULL, "VOUT1 Mixer" }, + { "VOUT2", NULL, "VOUT2 Mixer" }, + { "VOUT3", NULL, "VOUT3 Mixer" }, + { "VOUT4", NULL, "VOUT4 Mixer" } +}; + +static int vout12supply_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec; + + codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + snd_soc_update_bits(codec, WM8770_OUTMUX1, 0x180, 0); + break; + case SND_SOC_DAPM_POST_PMD: + snd_soc_update_bits(codec, WM8770_OUTMUX1, 0x180, 0x180); + break; + } + + return 0; +} + +static int vout34supply_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec; + + codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + snd_soc_update_bits(codec, WM8770_OUTMUX2, 0x180, 0); + break; + case SND_SOC_DAPM_POST_PMD: + snd_soc_update_bits(codec, WM8770_OUTMUX2, 0x180, 0x180); + break; + } + + return 0; +} + +static int wm8770_reset(struct snd_soc_codec *codec) +{ + return snd_soc_write(codec, WM8770_RESET, 0); +} + +static int wm8770_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec; + int iface, master; + + codec = dai->codec; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + master = 0x100; + break; + case SND_SOC_DAIFMT_CBS_CFS: + master = 0; + break; + default: + return -EINVAL; + } + + iface = 0; + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface |= 0x2; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + iface |= 0x1; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + iface |= 0xc; + break; + case SND_SOC_DAIFMT_IB_NF: + iface |= 0x8; + break; + case SND_SOC_DAIFMT_NB_IF: + iface |= 0x4; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, WM8770_IFACECTRL, 0xf, iface); + snd_soc_update_bits(codec, WM8770_MSTRCTRL, 0x100, master); + + return 0; +} + +static const int mclk_ratios[] = { + 128, + 192, + 256, + 384, + 512, + 768 +}; + +static int wm8770_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec; + struct wm8770_priv *wm8770; + int i; + int iface; + int shift; + int ratio; + + codec = dai->codec; + wm8770 = snd_soc_codec_get_drvdata(codec); + + iface = 0; + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + iface |= 0x10; + break; + case SNDRV_PCM_FORMAT_S24_LE: + iface |= 0x20; + break; + case SNDRV_PCM_FORMAT_S32_LE: + iface |= 0x30; + break; + } + + switch (substream->stream) { + case SNDRV_PCM_STREAM_PLAYBACK: + i = 0; + shift = 4; + break; + case SNDRV_PCM_STREAM_CAPTURE: + i = 2; + shift = 0; + break; + default: + return -EINVAL; + } + + /* Only need to set MCLK/LRCLK ratio if we're master */ + if (snd_soc_read(codec, WM8770_MSTRCTRL) & 0x100) { + for (; i < ARRAY_SIZE(mclk_ratios); ++i) { + ratio = wm8770->sysclk / params_rate(params); + if (ratio == mclk_ratios[i]) + break; + } + + if (i == ARRAY_SIZE(mclk_ratios)) { + dev_err(codec->dev, + "Unable to configure MCLK ratio %d/%d\n", + wm8770->sysclk, params_rate(params)); + return -EINVAL; + } + + dev_dbg(codec->dev, "MCLK is %dfs\n", mclk_ratios[i]); + + snd_soc_update_bits(codec, WM8770_MSTRCTRL, 0x7 << shift, + i << shift); + } + + snd_soc_update_bits(codec, WM8770_IFACECTRL, 0x30, iface); + + return 0; +} + +static int wm8770_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec; + + codec = dai->codec; + return snd_soc_update_bits(codec, WM8770_DACMUTE, 0x10, + !!mute << 4); +} + +static int wm8770_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec; + struct wm8770_priv *wm8770; + + codec = dai->codec; + wm8770 = snd_soc_codec_get_drvdata(codec); + wm8770->sysclk = freq; + return 0; +} + +static void wm8770_sync_cache(struct snd_soc_codec *codec) +{ + int i; + u16 *cache; + + if (!codec->cache_sync) + return; + + codec->cache_only = 0; + cache = codec->reg_cache; + for (i = 0; i < codec->driver->reg_cache_size; i++) { + if (i == WM8770_RESET || cache[i] == wm8770_reg_defs[i]) + continue; + snd_soc_write(codec, i, cache[i]); + } + codec->cache_sync = 0; +} + +static int wm8770_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + int ret; + struct wm8770_priv *wm8770; + + wm8770 = snd_soc_codec_get_drvdata(codec); + + 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 = regulator_bulk_enable(ARRAY_SIZE(wm8770->supplies), + wm8770->supplies); + if (ret) { + dev_err(codec->dev, + "Failed to enable supplies: %d\n", + ret); + return ret; + } + wm8770_sync_cache(codec); + /* global powerup */ + snd_soc_write(codec, WM8770_PWDNCTRL, 0); + } + break; + case SND_SOC_BIAS_OFF: + /* global powerdown */ + snd_soc_write(codec, WM8770_PWDNCTRL, 1); + regulator_bulk_disable(ARRAY_SIZE(wm8770->supplies), + wm8770->supplies); + break; + } + + codec->dapm.bias_level = level; + return 0; +} + +#define WM8770_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops wm8770_dai_ops = { + .digital_mute = wm8770_mute, + .hw_params = wm8770_hw_params, + .set_fmt = wm8770_set_fmt, + .set_sysclk = wm8770_set_sysclk, +}; + +static struct snd_soc_dai_driver wm8770_dai = { + .name = "wm8770-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = WM8770_FORMATS + }, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = WM8770_FORMATS + }, + .ops = &wm8770_dai_ops, + .symmetric_rates = 1 +}; + +#ifdef CONFIG_PM +static int wm8770_suspend(struct snd_soc_codec *codec, pm_message_t state) +{ + wm8770_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int wm8770_resume(struct snd_soc_codec *codec) +{ + wm8770_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + return 0; +} +#else +#define wm8770_suspend NULL +#define wm8770_resume NULL +#endif + +static int wm8770_probe(struct snd_soc_codec *codec) +{ + struct wm8770_priv *wm8770; + int ret; + int i; + + wm8770 = snd_soc_codec_get_drvdata(codec); + wm8770->codec = codec; + + codec->dapm.idle_bias_off = 1; + + ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8770->control_type); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + for (i = 0; i < ARRAY_SIZE(wm8770->supplies); i++) + wm8770->supplies[i].supply = wm8770_supply_names[i]; + + ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(wm8770->supplies), + wm8770->supplies); + if (ret) { + dev_err(codec->dev, "Failed to request supplies: %d\n", ret); + return ret; + } + + wm8770->disable_nb[0].notifier_call = wm8770_regulator_event_0; + wm8770->disable_nb[1].notifier_call = wm8770_regulator_event_1; + wm8770->disable_nb[2].notifier_call = wm8770_regulator_event_2; + + /* This should really be moved into the regulator core */ + for (i = 0; i < ARRAY_SIZE(wm8770->supplies); i++) { + ret = regulator_register_notifier(wm8770->supplies[i].consumer, + &wm8770->disable_nb[i]); + if (ret) { + dev_err(codec->dev, + "Failed to register regulator notifier: %d\n", + ret); + } + } + + ret = regulator_bulk_enable(ARRAY_SIZE(wm8770->supplies), + wm8770->supplies); + if (ret) { + dev_err(codec->dev, "Failed to enable supplies: %d\n", ret); + goto err_reg_get; + } + + ret = wm8770_reset(codec); + if (ret < 0) { + dev_err(codec->dev, "Failed to issue reset: %d\n", ret); + goto err_reg_enable; + } + + wm8770_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + /* latch the volume update bits */ + snd_soc_update_bits(codec, WM8770_MSDIGVOL, 0x100, 0x100); + snd_soc_update_bits(codec, WM8770_MSALGVOL, 0x100, 0x100); + snd_soc_update_bits(codec, WM8770_VOUT1RVOL, 0x100, 0x100); + snd_soc_update_bits(codec, WM8770_VOUT2RVOL, 0x100, 0x100); + snd_soc_update_bits(codec, WM8770_VOUT3RVOL, 0x100, 0x100); + snd_soc_update_bits(codec, WM8770_VOUT4RVOL, 0x100, 0x100); + snd_soc_update_bits(codec, WM8770_DAC1RVOL, 0x100, 0x100); + snd_soc_update_bits(codec, WM8770_DAC2RVOL, 0x100, 0x100); + snd_soc_update_bits(codec, WM8770_DAC3RVOL, 0x100, 0x100); + snd_soc_update_bits(codec, WM8770_DAC4RVOL, 0x100, 0x100); + + /* mute all DACs */ + snd_soc_update_bits(codec, WM8770_DACMUTE, 0x10, 0x10); + + snd_soc_add_controls(codec, wm8770_snd_controls, + ARRAY_SIZE(wm8770_snd_controls)); + snd_soc_dapm_new_controls(&codec->dapm, wm8770_dapm_widgets, + ARRAY_SIZE(wm8770_dapm_widgets)); + snd_soc_dapm_add_routes(&codec->dapm, wm8770_intercon, + ARRAY_SIZE(wm8770_intercon)); + return 0; + +err_reg_enable: + regulator_bulk_disable(ARRAY_SIZE(wm8770->supplies), wm8770->supplies); +err_reg_get: + regulator_bulk_free(ARRAY_SIZE(wm8770->supplies), wm8770->supplies); + return ret; +} + +static int wm8770_remove(struct snd_soc_codec *codec) +{ + struct wm8770_priv *wm8770; + int i; + + wm8770 = snd_soc_codec_get_drvdata(codec); + wm8770_set_bias_level(codec, SND_SOC_BIAS_OFF); + + for (i = 0; i < ARRAY_SIZE(wm8770->supplies); ++i) + regulator_unregister_notifier(wm8770->supplies[i].consumer, + &wm8770->disable_nb[i]); + regulator_bulk_free(ARRAY_SIZE(wm8770->supplies), wm8770->supplies); + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_wm8770 = { + .probe = wm8770_probe, + .remove = wm8770_remove, + .suspend = wm8770_suspend, + .resume = wm8770_resume, + .set_bias_level = wm8770_set_bias_level, + .reg_cache_size = ARRAY_SIZE(wm8770_reg_defs), + .reg_word_size = sizeof (u16), + .reg_cache_default = wm8770_reg_defs +}; + +#if defined(CONFIG_SPI_MASTER) +static int __devinit wm8770_spi_probe(struct spi_device *spi) +{ + struct wm8770_priv *wm8770; + int ret; + + wm8770 = kzalloc(sizeof(struct wm8770_priv), GFP_KERNEL); + if (!wm8770) + return -ENOMEM; + + wm8770->control_type = SND_SOC_SPI; + spi_set_drvdata(spi, wm8770); + + ret = snd_soc_register_codec(&spi->dev, + &soc_codec_dev_wm8770, &wm8770_dai, 1); + if (ret < 0) + kfree(wm8770); + return ret; +} + +static int __devexit wm8770_spi_remove(struct spi_device *spi) +{ + snd_soc_unregister_codec(&spi->dev); + kfree(spi_get_drvdata(spi)); + return 0; +} + +static struct spi_driver wm8770_spi_driver = { + .driver = { + .name = "wm8770", + .owner = THIS_MODULE, + }, + .probe = wm8770_spi_probe, + .remove = __devexit_p(wm8770_spi_remove) +}; +#endif + +static int __init wm8770_modinit(void) +{ + int ret = 0; + +#if defined(CONFIG_SPI_MASTER) + ret = spi_register_driver(&wm8770_spi_driver); + if (ret) { + printk(KERN_ERR "Failed to register wm8770 SPI driver: %d\n", + ret); + } +#endif + return ret; +} +module_init(wm8770_modinit); + +static void __exit wm8770_exit(void) +{ +#if defined(CONFIG_SPI_MASTER) + spi_unregister_driver(&wm8770_spi_driver); +#endif +} +module_exit(wm8770_exit); + +MODULE_DESCRIPTION("ASoC WM8770 driver"); +MODULE_AUTHOR("Dimitris Papastamos <dp@opensource.wolfsonmicro.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8770.h b/sound/soc/codecs/wm8770.h new file mode 100644 index 000000000000..5f1b3bda6cc8 --- /dev/null +++ b/sound/soc/codecs/wm8770.h @@ -0,0 +1,51 @@ +/* + * wm8770.h -- WM8770 ASoC driver + * + * Copyright 2010 Wolfson Microelectronics plc + * + * Author: Dimitris Papastamos <dp@opensource.wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _WM8770_H +#define _WM8770_H + +/* Registers */ +#define WM8770_VOUT1LVOL 0 +#define WM8770_VOUT1RVOL 0x1 +#define WM8770_VOUT2LVOL 0x2 +#define WM8770_VOUT2RVOL 0x3 +#define WM8770_VOUT3LVOL 0x4 +#define WM8770_VOUT3RVOL 0x5 +#define WM8770_VOUT4LVOL 0x6 +#define WM8770_VOUT4RVOL 0x7 +#define WM8770_MSALGVOL 0x8 +#define WM8770_DAC1LVOL 0x9 +#define WM8770_DAC1RVOL 0xa +#define WM8770_DAC2LVOL 0xb +#define WM8770_DAC2RVOL 0xc +#define WM8770_DAC3LVOL 0xd +#define WM8770_DAC3RVOL 0xe +#define WM8770_DAC4LVOL 0xf +#define WM8770_DAC4RVOL 0x10 +#define WM8770_MSDIGVOL 0x11 +#define WM8770_DACPHASE 0x12 +#define WM8770_DACCTRL1 0x13 +#define WM8770_DACMUTE 0x14 +#define WM8770_DACCTRL2 0x15 +#define WM8770_IFACECTRL 0x16 +#define WM8770_MSTRCTRL 0x17 +#define WM8770_PWDNCTRL 0x18 +#define WM8770_ADCLCTRL 0x19 +#define WM8770_ADCRCTRL 0x1a +#define WM8770_ADCMUX 0x1b +#define WM8770_OUTMUX1 0x1c +#define WM8770_OUTMUX2 0x1d +#define WM8770_RESET 0x31 + +#define WM8770_CACHEREGNUM 0x20 + +#endif diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c index 0132a27140ae..e09ed65e0868 100644 --- a/sound/soc/codecs/wm8776.c +++ b/sound/soc/codecs/wm8776.c @@ -306,7 +306,7 @@ static int wm8776_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: - if (codec->bias_level == SND_SOC_BIAS_OFF) { + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { /* Disable the global powerdown; DAPM does the rest */ snd_soc_update_bits(codec, WM8776_PWRDOWN, 1, 0); } @@ -317,7 +317,7 @@ static int wm8776_set_bias_level(struct snd_soc_codec *codec, break; } - codec->bias_level = level; + codec->dapm.bias_level = level; return 0; } @@ -404,6 +404,7 @@ static int wm8776_resume(struct snd_soc_codec *codec) static int wm8776_probe(struct snd_soc_codec *codec) { struct wm8776_priv *wm8776 = snd_soc_codec_get_drvdata(codec); + struct snd_soc_dapm_context *dapm = &codec->dapm; int ret = 0; ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8776->control_type); @@ -427,9 +428,9 @@ static int wm8776_probe(struct snd_soc_codec *codec) snd_soc_add_controls(codec, wm8776_snd_controls, ARRAY_SIZE(wm8776_snd_controls)); - snd_soc_dapm_new_controls(codec, wm8776_dapm_widgets, + snd_soc_dapm_new_controls(dapm, wm8776_dapm_widgets, ARRAY_SIZE(wm8776_dapm_widgets)); - snd_soc_dapm_add_routes(codec, routes, ARRAY_SIZE(routes)); + snd_soc_dapm_add_routes(dapm, routes, ARRAY_SIZE(routes)); return ret; } diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index 4599e8e95aa2..031a0d421108 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -515,7 +515,7 @@ static int wm8804_set_bias_level(struct snd_soc_codec *codec, snd_soc_update_bits(codec, WM8804_PWRDN, 0x9, 0); break; case SND_SOC_BIAS_STANDBY: - if (codec->bias_level == SND_SOC_BIAS_OFF) { + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { ret = regulator_bulk_enable(ARRAY_SIZE(wm8804->supplies), wm8804->supplies); if (ret) { @@ -537,7 +537,7 @@ static int wm8804_set_bias_level(struct snd_soc_codec *codec, break; } - codec->bias_level = level; + codec->dapm.bias_level = level; return 0; } @@ -581,7 +581,7 @@ static int wm8804_probe(struct snd_soc_codec *codec) wm8804 = snd_soc_codec_get_drvdata(codec); wm8804->codec = codec; - codec->idle_bias_off = 1; + codec->dapm.idle_bias_off = 1; ret = snd_soc_codec_set_cache_io(codec, 8, 8, wm8804->control_type); if (ret < 0) { diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index aca4b1ea10bb..06ea9c0f863b 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -611,10 +611,11 @@ static const struct snd_soc_dapm_route audio_map[] = { static int wm8900_add_widgets(struct snd_soc_codec *codec) { - snd_soc_dapm_new_controls(codec, wm8900_dapm_widgets, - ARRAY_SIZE(wm8900_dapm_widgets)); + struct snd_soc_dapm_context *dapm = &codec->dapm; - snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_new_controls(dapm, wm8900_dapm_widgets, + ARRAY_SIZE(wm8900_dapm_widgets)); + snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); return 0; } @@ -1051,7 +1052,7 @@ static int wm8900_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_STANDBY: /* Charge capacitors if initial power up */ - if (codec->bias_level == SND_SOC_BIAS_OFF) { + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { /* STARTUP_BIAS_ENA on */ snd_soc_write(codec, WM8900_REG_POWER1, WM8900_REG_POWER1_STARTUP_BIAS_ENA); @@ -1119,7 +1120,7 @@ static int wm8900_set_bias_level(struct snd_soc_codec *codec, WM8900_REG_POWER2_SYSCLK_ENA); break; } - codec->bias_level = level; + codec->dapm.bias_level = level; return 0; } diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 622b60238a82..4a6df4b69a04 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -923,10 +923,11 @@ static const struct snd_soc_dapm_route intercon[] = { static int wm8903_add_widgets(struct snd_soc_codec *codec) { - snd_soc_dapm_new_controls(codec, wm8903_dapm_widgets, - ARRAY_SIZE(wm8903_dapm_widgets)); + struct snd_soc_dapm_context *dapm = &codec->dapm; - snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + snd_soc_dapm_new_controls(dapm, wm8903_dapm_widgets, + ARRAY_SIZE(wm8903_dapm_widgets)); + snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); return 0; } @@ -946,7 +947,7 @@ static int wm8903_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: - if (codec->bias_level == SND_SOC_BIAS_OFF) { + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { snd_soc_write(codec, WM8903_CLOCK_RATES_2, WM8903_CLK_SYS_ENA); @@ -991,7 +992,7 @@ static int wm8903_set_bias_level(struct snd_soc_codec *codec, break; } - codec->bias_level = level; + codec->dapm.bias_level = level; return 0; } diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 33be84e506ea..be90399c1cb4 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -1428,10 +1428,11 @@ static const struct snd_soc_dapm_route wm8912_intercon[] = { static int wm8904_add_widgets(struct snd_soc_codec *codec) { struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec); + struct snd_soc_dapm_context *dapm = &codec->dapm; - snd_soc_dapm_new_controls(codec, wm8904_core_dapm_widgets, + snd_soc_dapm_new_controls(dapm, wm8904_core_dapm_widgets, ARRAY_SIZE(wm8904_core_dapm_widgets)); - snd_soc_dapm_add_routes(codec, core_intercon, + snd_soc_dapm_add_routes(dapm, core_intercon, ARRAY_SIZE(core_intercon)); switch (wm8904->devtype) { @@ -1443,20 +1444,20 @@ static int wm8904_add_widgets(struct snd_soc_codec *codec) snd_soc_add_controls(codec, wm8904_snd_controls, ARRAY_SIZE(wm8904_snd_controls)); - snd_soc_dapm_new_controls(codec, wm8904_adc_dapm_widgets, + snd_soc_dapm_new_controls(dapm, wm8904_adc_dapm_widgets, ARRAY_SIZE(wm8904_adc_dapm_widgets)); - snd_soc_dapm_new_controls(codec, wm8904_dac_dapm_widgets, + snd_soc_dapm_new_controls(dapm, wm8904_dac_dapm_widgets, ARRAY_SIZE(wm8904_dac_dapm_widgets)); - snd_soc_dapm_new_controls(codec, wm8904_dapm_widgets, + snd_soc_dapm_new_controls(dapm, wm8904_dapm_widgets, ARRAY_SIZE(wm8904_dapm_widgets)); - snd_soc_dapm_add_routes(codec, core_intercon, + snd_soc_dapm_add_routes(dapm, core_intercon, ARRAY_SIZE(core_intercon)); - snd_soc_dapm_add_routes(codec, adc_intercon, + snd_soc_dapm_add_routes(dapm, adc_intercon, ARRAY_SIZE(adc_intercon)); - snd_soc_dapm_add_routes(codec, dac_intercon, + snd_soc_dapm_add_routes(dapm, dac_intercon, ARRAY_SIZE(dac_intercon)); - snd_soc_dapm_add_routes(codec, wm8904_intercon, + snd_soc_dapm_add_routes(dapm, wm8904_intercon, ARRAY_SIZE(wm8904_intercon)); break; @@ -1464,17 +1465,17 @@ static int wm8904_add_widgets(struct snd_soc_codec *codec) snd_soc_add_controls(codec, wm8904_dac_snd_controls, ARRAY_SIZE(wm8904_dac_snd_controls)); - snd_soc_dapm_new_controls(codec, wm8904_dac_dapm_widgets, + snd_soc_dapm_new_controls(dapm, wm8904_dac_dapm_widgets, ARRAY_SIZE(wm8904_dac_dapm_widgets)); - snd_soc_dapm_add_routes(codec, dac_intercon, + snd_soc_dapm_add_routes(dapm, dac_intercon, ARRAY_SIZE(dac_intercon)); - snd_soc_dapm_add_routes(codec, wm8912_intercon, + snd_soc_dapm_add_routes(dapm, wm8912_intercon, ARRAY_SIZE(wm8912_intercon)); break; } - snd_soc_dapm_new_widgets(codec); + snd_soc_dapm_new_widgets(dapm); return 0; } @@ -2139,7 +2140,7 @@ static int wm8904_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: - if (codec->bias_level == SND_SOC_BIAS_OFF) { + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { ret = regulator_bulk_enable(ARRAY_SIZE(wm8904->supplies), wm8904->supplies); if (ret != 0) { @@ -2198,7 +2199,7 @@ static int wm8904_set_bias_level(struct snd_soc_codec *codec, wm8904->supplies); break; } - codec->bias_level = level; + codec->dapm.bias_level = level; return 0; } @@ -2373,7 +2374,7 @@ static int wm8904_probe(struct snd_soc_codec *codec) int ret, i; codec->cache_sync = 1; - codec->idle_bias_off = 1; + codec->dapm.idle_bias_off = 1; switch (wm8904->devtype) { case WM8904: diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index 2cb16f895c46..c2def1b01ae0 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -291,13 +291,14 @@ static const struct snd_soc_dapm_route audio_map[] = { static int wm8940_add_widgets(struct snd_soc_codec *codec) { + struct snd_soc_dapm_context *dapm = &codec->dapm; int ret; - ret = snd_soc_dapm_new_controls(codec, wm8940_dapm_widgets, + ret = snd_soc_dapm_new_controls(dapm, wm8940_dapm_widgets, ARRAY_SIZE(wm8940_dapm_widgets)); if (ret) goto error_ret; - ret = snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + ret = snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); if (ret) goto error_ret; diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c index f89ad6c9a80b..df1940fdbf69 100644 --- a/sound/soc/codecs/wm8955.c +++ b/sound/soc/codecs/wm8955.c @@ -577,13 +577,14 @@ static const struct snd_soc_dapm_route wm8955_intercon[] = { static int wm8955_add_widgets(struct snd_soc_codec *codec) { + struct snd_soc_dapm_context *dapm = &codec->dapm; + snd_soc_add_controls(codec, wm8955_snd_controls, ARRAY_SIZE(wm8955_snd_controls)); - snd_soc_dapm_new_controls(codec, wm8955_dapm_widgets, + snd_soc_dapm_new_controls(dapm, wm8955_dapm_widgets, ARRAY_SIZE(wm8955_dapm_widgets)); - - snd_soc_dapm_add_routes(codec, wm8955_intercon, + snd_soc_dapm_add_routes(dapm, wm8955_intercon, ARRAY_SIZE(wm8955_intercon)); return 0; @@ -786,7 +787,7 @@ static int wm8955_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: - if (codec->bias_level == SND_SOC_BIAS_OFF) { + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { ret = regulator_bulk_enable(ARRAY_SIZE(wm8955->supplies), wm8955->supplies); if (ret != 0) { @@ -850,7 +851,7 @@ static int wm8955_set_bias_level(struct snd_soc_codec *codec, wm8955->supplies); break; } - codec->bias_level = level; + codec->dapm.bias_level = level; return 0; } diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 8d5efb333c33..0ea578815003 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -388,27 +388,28 @@ static int wm8960_add_widgets(struct snd_soc_codec *codec) { struct wm8960_data *pdata = codec->dev->platform_data; struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + struct snd_soc_dapm_context *dapm = &codec->dapm; struct snd_soc_dapm_widget *w; - snd_soc_dapm_new_controls(codec, wm8960_dapm_widgets, + snd_soc_dapm_new_controls(dapm, wm8960_dapm_widgets, ARRAY_SIZE(wm8960_dapm_widgets)); - snd_soc_dapm_add_routes(codec, audio_paths, ARRAY_SIZE(audio_paths)); + snd_soc_dapm_add_routes(dapm, audio_paths, ARRAY_SIZE(audio_paths)); /* In capless mode OUT3 is used to provide VMID for the * headphone outputs, otherwise it is used as a mono mixer. */ if (pdata && pdata->capless) { - snd_soc_dapm_new_controls(codec, wm8960_dapm_widgets_capless, + snd_soc_dapm_new_controls(dapm, wm8960_dapm_widgets_capless, ARRAY_SIZE(wm8960_dapm_widgets_capless)); - snd_soc_dapm_add_routes(codec, audio_paths_capless, + snd_soc_dapm_add_routes(dapm, audio_paths_capless, ARRAY_SIZE(audio_paths_capless)); } else { - snd_soc_dapm_new_controls(codec, wm8960_dapm_widgets_out3, + snd_soc_dapm_new_controls(dapm, wm8960_dapm_widgets_out3, ARRAY_SIZE(wm8960_dapm_widgets_out3)); - snd_soc_dapm_add_routes(codec, audio_paths_out3, + snd_soc_dapm_add_routes(dapm, audio_paths_out3, ARRAY_SIZE(audio_paths_out3)); } @@ -417,7 +418,7 @@ static int wm8960_add_widgets(struct snd_soc_codec *codec) * list each time to find the desired power state do so now * and save the result. */ - list_for_each_entry(w, &codec->dapm_widgets, list) { + list_for_each_entry(w, &codec->dapm.widgets, list) { if (strcmp(w->name, "LOUT1 PGA") == 0) wm8960->lout1 = w; if (strcmp(w->name, "ROUT1 PGA") == 0) @@ -572,7 +573,7 @@ static int wm8960_set_bias_level_out3(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: - if (codec->bias_level == SND_SOC_BIAS_OFF) { + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { /* Enable anti-pop features */ snd_soc_write(codec, WM8960_APOP1, WM8960_POBCTRL | WM8960_SOFT_ST | @@ -610,7 +611,7 @@ static int wm8960_set_bias_level_out3(struct snd_soc_codec *codec, break; } - codec->bias_level = level; + codec->dapm.bias_level = level; return 0; } @@ -626,7 +627,7 @@ static int wm8960_set_bias_level_capless(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_PREPARE: - switch (codec->bias_level) { + switch (codec->dapm.bias_level) { case SND_SOC_BIAS_STANDBY: /* Enable anti pop mode */ snd_soc_update_bits(codec, WM8960_APOP1, @@ -681,7 +682,7 @@ static int wm8960_set_bias_level_capless(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: - switch (codec->bias_level) { + switch (codec->dapm.bias_level) { case SND_SOC_BIAS_PREPARE: /* Disable HP discharge */ snd_soc_update_bits(codec, WM8960_APOP2, @@ -705,7 +706,7 @@ static int wm8960_set_bias_level_capless(struct snd_soc_codec *codec, break; } - codec->bias_level = level; + codec->dapm.bias_level = level; return 0; } diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index 4f326f604104..79b650945bb2 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -882,7 +882,7 @@ static int wm8961_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_PREPARE: - if (codec->bias_level == SND_SOC_BIAS_STANDBY) { + if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY) { /* Enable bias generation */ reg = snd_soc_read(codec, WM8961_ANTI_POP); reg |= WM8961_BUFIOEN | WM8961_BUFDCOPEN; @@ -897,7 +897,7 @@ static int wm8961_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: - if (codec->bias_level == SND_SOC_BIAS_PREPARE) { + if (codec->dapm.bias_level == SND_SOC_BIAS_PREPARE) { /* VREF off */ reg = snd_soc_read(codec, WM8961_PWR_MGMT_1); reg &= ~WM8961_VREF; @@ -919,7 +919,7 @@ static int wm8961_set_bias_level(struct snd_soc_codec *codec, break; } - codec->bias_level = level; + codec->dapm.bias_level = level; return 0; } @@ -959,6 +959,7 @@ static struct snd_soc_dai_driver wm8961_dai = { static int wm8961_probe(struct snd_soc_codec *codec) { + struct snd_soc_dapm_context *dapm = &codec->dapm; int ret = 0; u16 reg; @@ -1024,9 +1025,9 @@ static int wm8961_probe(struct snd_soc_codec *codec) snd_soc_add_controls(codec, wm8961_snd_controls, ARRAY_SIZE(wm8961_snd_controls)); - snd_soc_dapm_new_controls(codec, wm8961_dapm_widgets, + snd_soc_dapm_new_controls(dapm, wm8961_dapm_widgets, ARRAY_SIZE(wm8961_dapm_widgets)); - snd_soc_dapm_add_routes(codec, audio_paths, ARRAY_SIZE(audio_paths)); + snd_soc_dapm_add_routes(dapm, audio_paths, ARRAY_SIZE(audio_paths)); return 0; } diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index e8092745a207..80986105f52e 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2682,6 +2682,7 @@ static const struct snd_soc_dapm_route wm8962_spk_stereo_intercon[] = { static int wm8962_add_widgets(struct snd_soc_codec *codec) { struct wm8962_pdata *pdata = dev_get_platdata(codec->dev); + struct snd_soc_dapm_context *dapm = &codec->dapm; snd_soc_add_controls(codec, wm8962_snd_controls, ARRAY_SIZE(wm8962_snd_controls)); @@ -2693,26 +2694,26 @@ static int wm8962_add_widgets(struct snd_soc_codec *codec) ARRAY_SIZE(wm8962_spk_stereo_controls)); - snd_soc_dapm_new_controls(codec, wm8962_dapm_widgets, + snd_soc_dapm_new_controls(dapm, wm8962_dapm_widgets, ARRAY_SIZE(wm8962_dapm_widgets)); if (pdata && pdata->spk_mono) - snd_soc_dapm_new_controls(codec, wm8962_dapm_spk_mono_widgets, + snd_soc_dapm_new_controls(dapm, wm8962_dapm_spk_mono_widgets, ARRAY_SIZE(wm8962_dapm_spk_mono_widgets)); else - snd_soc_dapm_new_controls(codec, wm8962_dapm_spk_stereo_widgets, + snd_soc_dapm_new_controls(dapm, wm8962_dapm_spk_stereo_widgets, ARRAY_SIZE(wm8962_dapm_spk_stereo_widgets)); - snd_soc_dapm_add_routes(codec, wm8962_intercon, + snd_soc_dapm_add_routes(dapm, wm8962_intercon, ARRAY_SIZE(wm8962_intercon)); if (pdata && pdata->spk_mono) - snd_soc_dapm_add_routes(codec, wm8962_spk_mono_intercon, + snd_soc_dapm_add_routes(dapm, wm8962_spk_mono_intercon, ARRAY_SIZE(wm8962_spk_mono_intercon)); else - snd_soc_dapm_add_routes(codec, wm8962_spk_stereo_intercon, + snd_soc_dapm_add_routes(dapm, wm8962_spk_stereo_intercon, ARRAY_SIZE(wm8962_spk_stereo_intercon)); - snd_soc_dapm_disable_pin(codec, "Beep"); + snd_soc_dapm_disable_pin(dapm, "Beep"); return 0; } @@ -2819,7 +2820,7 @@ static int wm8962_set_bias_level(struct snd_soc_codec *codec, struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); int ret; - if (level == codec->bias_level) + if (level == codec->dapm.bias_level) return 0; switch (level) { @@ -2833,7 +2834,7 @@ static int wm8962_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: - if (codec->bias_level == SND_SOC_BIAS_OFF) { + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { ret = regulator_bulk_enable(ARRAY_SIZE(wm8962->supplies), wm8962->supplies); if (ret != 0) { @@ -2883,7 +2884,7 @@ static int wm8962_set_bias_level(struct snd_soc_codec *codec, wm8962->supplies); break; } - codec->bias_level = level; + codec->dapm.bias_level = level; return 0; } @@ -3353,6 +3354,8 @@ static irqreturn_t wm8962_irq(int irq, void *data) if (active & (WM8962_MICSCD_EINT | WM8962_MICD_EINT)) { dev_dbg(codec->dev, "Microphone event detected\n"); + pm_wakeup_event(codec->dev, 300); + schedule_delayed_work(&wm8962->mic_work, msecs_to_jiffies(250)); } @@ -3439,6 +3442,7 @@ static void wm8962_beep_work(struct work_struct *work) struct wm8962_priv *wm8962 = container_of(work, struct wm8962_priv, beep_work); struct snd_soc_codec *codec = wm8962->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; int i; int reg = 0; int best = 0; @@ -3455,16 +3459,16 @@ static void wm8962_beep_work(struct work_struct *work) reg = WM8962_BEEP_ENA | (best << WM8962_BEEP_RATE_SHIFT); - snd_soc_dapm_enable_pin(codec, "Beep"); + snd_soc_dapm_enable_pin(dapm, "Beep"); } else { dev_dbg(codec->dev, "Disabling beep\n"); - snd_soc_dapm_disable_pin(codec, "Beep"); + snd_soc_dapm_disable_pin(dapm, "Beep"); } snd_soc_update_bits(codec, WM8962_BEEP_GENERATOR_1, WM8962_BEEP_ENA | WM8962_BEEP_RATE_MASK, reg); - snd_soc_dapm_sync(codec); + snd_soc_dapm_sync(dapm); } /* For usability define a way of injecting beep events for the device - @@ -3711,7 +3715,7 @@ static int wm8962_probe(struct snd_soc_codec *codec) INIT_DELAYED_WORK(&wm8962->mic_work, wm8962_mic_work); codec->cache_sync = 1; - codec->idle_bias_off = 1; + codec->dapm.idle_bias_off = 1; ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_I2C); if (ret != 0) { diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index 63f6dbf5d070..84b2dcb18aea 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -333,10 +333,11 @@ static const struct snd_soc_dapm_route audio_map[] = { static int wm8971_add_widgets(struct snd_soc_codec *codec) { - snd_soc_dapm_new_controls(codec, wm8971_dapm_widgets, - ARRAY_SIZE(wm8971_dapm_widgets)); + struct snd_soc_dapm_context *dapm = &codec->dapm; - snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_new_controls(dapm, wm8971_dapm_widgets, + ARRAY_SIZE(wm8971_dapm_widgets)); + snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); return 0; } @@ -553,7 +554,7 @@ static int wm8971_set_bias_level(struct snd_soc_codec *codec, snd_soc_write(codec, WM8971_PWR1, 0x0001); break; } - codec->bias_level = level; + codec->dapm.bias_level = level; return 0; } @@ -590,9 +591,11 @@ static struct snd_soc_dai_driver wm8971_dai = { static void wm8971_work(struct work_struct *work) { - struct snd_soc_codec *codec = - container_of(work, struct snd_soc_codec, delayed_work.work); - wm8971_set_bias_level(codec, codec->bias_level); + struct snd_soc_dapm_context *dapm = + container_of(work, struct snd_soc_dapm_context, + delayed_work.work); + struct snd_soc_codec *codec = dapm->codec; + wm8971_set_bias_level(codec, codec->dapm.bias_level); } static int wm8971_suspend(struct snd_soc_codec *codec, pm_message_t state) @@ -620,11 +623,11 @@ static int wm8971_resume(struct snd_soc_codec *codec) wm8971_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* charge wm8971 caps */ - if (codec->suspend_bias_level == SND_SOC_BIAS_ON) { + if (codec->dapm.suspend_bias_level == SND_SOC_BIAS_ON) { reg = snd_soc_read(codec, WM8971_PWR1) & 0xfe3e; snd_soc_write(codec, WM8971_PWR1, reg | 0x01c0); - codec->bias_level = SND_SOC_BIAS_ON; - queue_delayed_work(wm8971_workq, &codec->delayed_work, + codec->dapm.bias_level = SND_SOC_BIAS_ON; + queue_delayed_work(wm8971_workq, &codec->dapm.delayed_work, msecs_to_jiffies(1000)); } @@ -643,7 +646,7 @@ static int wm8971_probe(struct snd_soc_codec *codec) return ret; } - INIT_DELAYED_WORK(&codec->delayed_work, wm8971_work); + INIT_DELAYED_WORK(&codec->dapm.delayed_work, wm8971_work); wm8971_workq = create_workqueue("wm8971"); if (wm8971_workq == NULL) return -ENOMEM; @@ -653,8 +656,8 @@ static int wm8971_probe(struct snd_soc_codec *codec) /* charge output caps - set vmid to 5k for quick power up */ reg = snd_soc_read(codec, WM8971_PWR1) & 0xfe3e; snd_soc_write(codec, WM8971_PWR1, reg | 0x01c0); - codec->bias_level = SND_SOC_BIAS_STANDBY; - queue_delayed_work(wm8971_workq, &codec->delayed_work, + codec->dapm.bias_level = SND_SOC_BIAS_STANDBY; + queue_delayed_work(wm8971_workq, &codec->dapm.delayed_work, msecs_to_jiffies(1000)); /* set the update bits */ diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index b4363f6d19b3..d19bb14842d4 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -274,10 +274,11 @@ static const struct snd_soc_dapm_route audio_map[] = { static int wm8974_add_widgets(struct snd_soc_codec *codec) { - snd_soc_dapm_new_controls(codec, wm8974_dapm_widgets, - ARRAY_SIZE(wm8974_dapm_widgets)); + struct snd_soc_dapm_context *dapm = &codec->dapm; - snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_new_controls(dapm, wm8974_dapm_widgets, + ARRAY_SIZE(wm8974_dapm_widgets)); + snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); return 0; } @@ -530,7 +531,7 @@ static int wm8974_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_STANDBY: power1 |= WM8974_POWER1_BIASEN | WM8974_POWER1_BUFIOEN; - if (codec->bias_level == SND_SOC_BIAS_OFF) { + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { /* Initial cap charge at VMID 5k */ snd_soc_write(codec, WM8974_POWER1, power1 | 0x3); mdelay(100); @@ -547,7 +548,7 @@ static int wm8974_set_bias_level(struct snd_soc_codec *codec, break; } - codec->bias_level = level; + codec->dapm.bias_level = level; return 0; } diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index 13b979a71a7c..ac43b6088e2e 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -355,11 +355,12 @@ static const struct snd_soc_dapm_route audio_map[] = { static int wm8978_add_widgets(struct snd_soc_codec *codec) { - snd_soc_dapm_new_controls(codec, wm8978_dapm_widgets, - ARRAY_SIZE(wm8978_dapm_widgets)); + struct snd_soc_dapm_context *dapm = &codec->dapm; + snd_soc_dapm_new_controls(dapm, wm8978_dapm_widgets, + ARRAY_SIZE(wm8978_dapm_widgets)); /* set up the WM8978 audio map */ - snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); return 0; } @@ -837,7 +838,7 @@ static int wm8978_set_bias_level(struct snd_soc_codec *codec, /* bit 3: enable bias, bit 2: enable I/O tie off buffer */ power1 |= 0xc; - if (codec->bias_level == SND_SOC_BIAS_OFF) { + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { /* Initial cap charge at VMID 5k */ snd_soc_write(codec, WM8978_POWER_MANAGEMENT_1, power1 | 0x3); @@ -857,7 +858,7 @@ static int wm8978_set_bias_level(struct snd_soc_codec *codec, dev_dbg(codec->dev, "%s: %d, %x\n", __func__, level, power1); - codec->bias_level = level; + codec->dapm.bias_level = level; return 0; } diff --git a/sound/soc/codecs/wm8985.c b/sound/soc/codecs/wm8985.c index fd2e7cca1228..c3c8fd23d503 100644 --- a/sound/soc/codecs/wm8985.c +++ b/sound/soc/codecs/wm8985.c @@ -533,10 +533,11 @@ static int eqmode_put(struct snd_kcontrol *kcontrol, static int wm8985_add_widgets(struct snd_soc_codec *codec) { - snd_soc_dapm_new_controls(codec, wm8985_dapm_widgets, - ARRAY_SIZE(wm8985_dapm_widgets)); + struct snd_soc_dapm_context *dapm = &codec->dapm; - snd_soc_dapm_add_routes(codec, audio_map, + snd_soc_dapm_new_controls(dapm, wm8985_dapm_widgets, + ARRAY_SIZE(wm8985_dapm_widgets)); + snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); return 0; } @@ -879,7 +880,7 @@ static int wm8985_set_bias_level(struct snd_soc_codec *codec, 1 << WM8985_VMIDSEL_SHIFT); break; case SND_SOC_BIAS_STANDBY: - if (codec->bias_level == SND_SOC_BIAS_OFF) { + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { ret = regulator_bulk_enable(ARRAY_SIZE(wm8985->supplies), wm8985->supplies); if (ret) { @@ -939,7 +940,7 @@ static int wm8985_set_bias_level(struct snd_soc_codec *codec, break; } - codec->bias_level = level; + codec->dapm.bias_level = level; return 0; } diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c index d7f259711970..0bc2eb530c7a 100644 --- a/sound/soc/codecs/wm8988.c +++ b/sound/soc/codecs/wm8988.c @@ -677,7 +677,7 @@ static int wm8988_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: - if (codec->bias_level == SND_SOC_BIAS_OFF) { + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { /* VREF, VMID=2x5k */ snd_soc_write(codec, WM8988_PWR1, pwr_reg | 0x1c1); @@ -693,7 +693,7 @@ static int wm8988_set_bias_level(struct snd_soc_codec *codec, snd_soc_write(codec, WM8988_PWR1, 0x0000); break; } - codec->bias_level = level; + codec->dapm.bias_level = level; return 0; } @@ -759,6 +759,7 @@ static int wm8988_resume(struct snd_soc_codec *codec) static int wm8988_probe(struct snd_soc_codec *codec) { struct wm8988_priv *wm8988 = snd_soc_codec_get_drvdata(codec); + struct snd_soc_dapm_context *dapm = &codec->dapm; int ret = 0; u16 reg; @@ -790,9 +791,9 @@ static int wm8988_probe(struct snd_soc_codec *codec) snd_soc_add_controls(codec, wm8988_snd_controls, ARRAY_SIZE(wm8988_snd_controls)); - snd_soc_dapm_new_controls(codec, wm8988_dapm_widgets, + snd_soc_dapm_new_controls(dapm, wm8988_dapm_widgets, ARRAY_SIZE(wm8988_dapm_widgets)); - snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); return 0; } diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 264828e4e67c..309664ea7dc3 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -914,11 +914,12 @@ static const struct snd_soc_dapm_route audio_map[] = { static int wm8990_add_widgets(struct snd_soc_codec *codec) { - snd_soc_dapm_new_controls(codec, wm8990_dapm_widgets, - ARRAY_SIZE(wm8990_dapm_widgets)); + struct snd_soc_dapm_context *dapm = &codec->dapm; + snd_soc_dapm_new_controls(dapm, wm8990_dapm_widgets, + ARRAY_SIZE(wm8990_dapm_widgets)); /* set up the WM8990 audio map */ - snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); return 0; } @@ -1170,7 +1171,7 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: - if (codec->bias_level == SND_SOC_BIAS_OFF) { + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { /* Enable all output discharge bits */ snd_soc_write(codec, WM8990_ANTIPOP1, WM8990_DIS_LLINE | WM8990_DIS_RLINE | WM8990_DIS_OUT3 | @@ -1266,7 +1267,7 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec, break; } - codec->bias_level = level; + codec->dapm.bias_level = level; return 0; } diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 589e3fa24734..bcc54be572ce 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -735,6 +735,7 @@ static int class_w_put(struct snd_kcontrol *kcontrol, 0); } wm8993->class_w_users++; + wm8993->hubs_data.class_w = true; } /* Implement the change */ @@ -751,6 +752,7 @@ static int class_w_put(struct snd_kcontrol *kcontrol, WM8993_CP_DYN_V); } wm8993->class_w_users--; + wm8993->hubs_data.class_w = false; } dev_dbg(codec->dev, "Indirect DAC use count now %d\n", @@ -968,7 +970,7 @@ static int wm8993_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: - if (codec->bias_level == SND_SOC_BIAS_OFF) { + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { ret = regulator_bulk_enable(ARRAY_SIZE(wm8993->supplies), wm8993->supplies); if (ret != 0) @@ -1043,7 +1045,7 @@ static int wm8993_set_bias_level(struct snd_soc_codec *codec, break; } - codec->bias_level = level; + codec->dapm.bias_level = level; return 0; } @@ -1422,6 +1424,7 @@ static struct snd_soc_dai_driver wm8993_dai = { static int wm8993_probe(struct snd_soc_codec *codec) { struct wm8993_priv *wm8993 = snd_soc_codec_get_drvdata(codec); + struct snd_soc_dapm_context *dapm = &codec->dapm; int ret, i, val; wm8993->hubs_data.hp_startup_mode = 1; @@ -1503,11 +1506,11 @@ static int wm8993_probe(struct snd_soc_codec *codec) ARRAY_SIZE(wm8993_eq_controls)); } - snd_soc_dapm_new_controls(codec, wm8993_dapm_widgets, + snd_soc_dapm_new_controls(dapm, wm8993_dapm_widgets, ARRAY_SIZE(wm8993_dapm_widgets)); wm_hubs_add_analogue_controls(codec); - snd_soc_dapm_add_routes(codec, routes, ARRAY_SIZE(routes)); + snd_soc_dapm_add_routes(dapm, routes, ARRAY_SIZE(routes)); wm_hubs_add_analogue_routes(codec, wm8993->pdata.lineout1_diff, wm8993->pdata.lineout2_diff); diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 0db59c3aa5d4..f7dea3d34a3e 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1730,8 +1730,6 @@ static int wm8994_write(struct snd_soc_codec *codec, unsigned int reg, if (!wm8994_volatile(reg)) wm8994->reg_cache[reg] = value; - dev_dbg(codec->dev, "0x%x = 0x%x\n", reg, value); - return wm8994_reg_write(codec->control_data, reg, value); } @@ -1837,7 +1835,7 @@ static int configure_clock(struct snd_soc_codec *codec) snd_soc_update_bits(codec, WM8994_CLOCKING_1, WM8994_SYSCLK_SRC, new); - snd_soc_dapm_sync(codec); + snd_soc_dapm_sync(&codec->dapm); return 0; } @@ -2228,6 +2226,7 @@ static int clk_sys_event(struct snd_soc_dapm_widget *w, static void wm8994_update_class_w(struct snd_soc_codec *codec) { + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); int enable = 1; int source = 0; /* GCC flow analysis can't track enable */ int reg, reg_r; @@ -2278,11 +2277,13 @@ static void wm8994_update_class_w(struct snd_soc_codec *codec) WM8994_CP_DYN_PWR | WM8994_CP_DYN_SRC_SEL_MASK, source | WM8994_CP_DYN_PWR); + wm8994->hubs.class_w = true; } else { dev_dbg(codec->dev, "Class W disabled\n"); snd_soc_update_bits(codec, WM8994_CLASS_W_1, WM8994_CP_DYN_PWR, 0); + wm8994->hubs.class_w = false; } } @@ -3107,7 +3108,7 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: - if (codec->bias_level == SND_SOC_BIAS_OFF) { + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { /* Tweak DC servo and DSP configuration for * improved performance. */ if (wm8994->revision < 4) { @@ -3151,7 +3152,7 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_OFF: - if (codec->bias_level == SND_SOC_BIAS_STANDBY) { + if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY) { /* Switch over to startup biases */ snd_soc_update_bits(codec, WM8994_ANTIPOP_2, WM8994_BIAS_SRC | @@ -3186,7 +3187,7 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec, } break; } - codec->bias_level = level; + codec->dapm.bias_level = level; return 0; } @@ -3894,6 +3895,7 @@ static irqreturn_t wm8994_mic_irq(int irq, void *data) static int wm8994_codec_probe(struct snd_soc_codec *codec) { struct wm8994_priv *wm8994; + struct snd_soc_dapm_context *dapm = &codec->dapm; int ret, i; codec->control_data = dev_get_drvdata(codec->dev->parent); @@ -4032,10 +4034,10 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) wm_hubs_add_analogue_controls(codec); snd_soc_add_controls(codec, wm8994_snd_controls, ARRAY_SIZE(wm8994_snd_controls)); - snd_soc_dapm_new_controls(codec, wm8994_dapm_widgets, + snd_soc_dapm_new_controls(dapm, wm8994_dapm_widgets, ARRAY_SIZE(wm8994_dapm_widgets)); wm_hubs_add_analogue_routes(codec, 0, 0); - snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); return 0; diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index ecc7c37180c7..c03e2c3e24e1 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -805,7 +805,7 @@ static int wm9081_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_STANDBY: /* Initial cold start */ - if (codec->bias_level == SND_SOC_BIAS_OFF) { + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { /* Disable LINEOUT discharge */ reg = snd_soc_read(codec, WM9081_ANTI_POP_CONTROL); reg &= ~WM9081_LINEOUT_DISCH; @@ -865,7 +865,7 @@ static int wm9081_set_bias_level(struct snd_soc_codec *codec, break; } - codec->bias_level = level; + codec->dapm.bias_level = level; return 0; } @@ -1228,6 +1228,7 @@ 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; @@ -1269,9 +1270,9 @@ static int wm9081_probe(struct snd_soc_codec *codec) ARRAY_SIZE(wm9081_eq_controls)); } - snd_soc_dapm_new_controls(codec, wm9081_dapm_widgets, + snd_soc_dapm_new_controls(dapm, wm9081_dapm_widgets, ARRAY_SIZE(wm9081_dapm_widgets)); - snd_soc_dapm_add_routes(codec, audio_paths, ARRAY_SIZE(audio_paths)); + snd_soc_dapm_add_routes(dapm, audio_paths, ARRAY_SIZE(audio_paths)); return ret; } diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c index 99c046ba46bb..b5afa01aa383 100644 --- a/sound/soc/codecs/wm9090.c +++ b/sound/soc/codecs/wm9090.c @@ -443,31 +443,32 @@ static const struct snd_soc_dapm_route audio_map_in2_diff[] = { static int wm9090_add_controls(struct snd_soc_codec *codec) { struct wm9090_priv *wm9090 = snd_soc_codec_get_drvdata(codec); + struct snd_soc_dapm_context *dapm = &codec->dapm; int i; - snd_soc_dapm_new_controls(codec, wm9090_dapm_widgets, + snd_soc_dapm_new_controls(dapm, wm9090_dapm_widgets, ARRAY_SIZE(wm9090_dapm_widgets)); - snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); snd_soc_add_controls(codec, wm9090_controls, ARRAY_SIZE(wm9090_controls)); if (wm9090->pdata.lin1_diff) { - snd_soc_dapm_add_routes(codec, audio_map_in1_diff, + snd_soc_dapm_add_routes(dapm, audio_map_in1_diff, ARRAY_SIZE(audio_map_in1_diff)); } else { - snd_soc_dapm_add_routes(codec, audio_map_in1_se, + snd_soc_dapm_add_routes(dapm, audio_map_in1_se, ARRAY_SIZE(audio_map_in1_se)); snd_soc_add_controls(codec, wm9090_in1_se_controls, ARRAY_SIZE(wm9090_in1_se_controls)); } if (wm9090->pdata.lin2_diff) { - snd_soc_dapm_add_routes(codec, audio_map_in2_diff, + snd_soc_dapm_add_routes(dapm, audio_map_in2_diff, ARRAY_SIZE(audio_map_in2_diff)); } else { - snd_soc_dapm_add_routes(codec, audio_map_in2_se, + snd_soc_dapm_add_routes(dapm, audio_map_in2_se, ARRAY_SIZE(audio_map_in2_se)); snd_soc_add_controls(codec, wm9090_in2_se_controls, ARRAY_SIZE(wm9090_in2_se_controls)); @@ -514,7 +515,7 @@ static int wm9090_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: - if (codec->bias_level == SND_SOC_BIAS_OFF) { + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { /* Restore the register cache */ for (i = 1; i < codec->driver->reg_cache_size; i++) { if (reg_cache[i] == wm9090_reg_defaults[i]) @@ -544,7 +545,7 @@ static int wm9090_set_bias_level(struct snd_soc_codec *codec, break; } - codec->bias_level = level; + codec->dapm.bias_level = level; return 0; } diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index a144acda751c..58d120824498 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -203,9 +203,11 @@ static const struct snd_soc_dapm_route audio_map[] = { static int wm9705_add_widgets(struct snd_soc_codec *codec) { - snd_soc_dapm_new_controls(codec, wm9705_dapm_widgets, + struct snd_soc_dapm_context *dapm = &codec->dapm; + + snd_soc_dapm_new_controls(dapm, wm9705_dapm_widgets, ARRAY_SIZE(wm9705_dapm_widgets)); - snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); return 0; } diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index d2f224d62744..3ca42a35e03a 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -432,10 +432,11 @@ static const struct snd_soc_dapm_route audio_map[] = { static int wm9712_add_widgets(struct snd_soc_codec *codec) { - snd_soc_dapm_new_controls(codec, wm9712_dapm_widgets, - ARRAY_SIZE(wm9712_dapm_widgets)); + struct snd_soc_dapm_context *dapm = &codec->dapm; - snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_new_controls(dapm, wm9712_dapm_widgets, + ARRAY_SIZE(wm9712_dapm_widgets)); + snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); return 0; } @@ -570,7 +571,7 @@ static int wm9712_set_bias_level(struct snd_soc_codec *codec, ac97_write(codec, AC97_POWERDOWN, 0xffff); break; } - codec->bias_level = level; + codec->dapm.bias_level = level; return 0; } diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 7da13b07a53d..87b236b16016 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -647,10 +647,12 @@ static const struct snd_soc_dapm_route audio_map[] = { static int wm9713_add_widgets(struct snd_soc_codec *codec) { - snd_soc_dapm_new_controls(codec, wm9713_dapm_widgets, + struct snd_soc_dapm_context *dapm = &codec->dapm; + + snd_soc_dapm_new_controls(dapm, wm9713_dapm_widgets, ARRAY_SIZE(wm9713_dapm_widgets)); - snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); return 0; } @@ -1147,7 +1149,7 @@ static int wm9713_set_bias_level(struct snd_soc_codec *codec, ac97_write(codec, AC97_POWERDOWN, 0xffff); break; } - codec->bias_level = level; + codec->dapm.bias_level = level; return 0; } diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 19ca782ac970..8aff0efe72f5 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -94,6 +94,18 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec) struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); u16 reg, reg_l, reg_r, dcs_cfg; + /* If we're using a digital only path and have a previously + * callibrated DC servo offset stored then use that. */ + if (hubs->class_w && hubs->class_w_dcs) { + dev_dbg(codec->dev, "Using cached DC servo offset %x\n", + hubs->class_w_dcs); + snd_soc_write(codec, WM8993_DC_SERVO_3, hubs->class_w_dcs); + wait_for_dc_servo(codec, + WM8993_DCS_TRIG_DAC_WR_0 | + WM8993_DCS_TRIG_DAC_WR_1); + return; + } + /* Set for 32 series updates */ snd_soc_update_bits(codec, WM8993_DC_SERVO_1, WM8993_DCS_SERIES_NO_01_MASK, @@ -101,34 +113,34 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec) wait_for_dc_servo(codec, WM8993_DCS_TRIG_SERIES_0 | WM8993_DCS_TRIG_SERIES_1); + /* Different chips in the family support different readback + * methods. + */ + switch (hubs->dcs_readback_mode) { + case 0: + reg_l = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_1) + & WM8993_DCS_INTEG_CHAN_0_MASK;; + reg_r = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_2) + & WM8993_DCS_INTEG_CHAN_1_MASK; + break; + case 1: + reg = snd_soc_read(codec, WM8993_DC_SERVO_3); + reg_l = (reg & WM8993_DCS_DAC_WR_VAL_1_MASK) + >> WM8993_DCS_DAC_WR_VAL_1_SHIFT; + reg_r = reg & WM8993_DCS_DAC_WR_VAL_0_MASK; + break; + default: + WARN(1, "Unknown DCS readback method\n"); + break; + } + + dev_dbg(codec->dev, "DCS input: %x %x\n", reg_l, reg_r); + /* Apply correction to DC servo result */ if (hubs->dcs_codes) { dev_dbg(codec->dev, "Applying %d code DC servo correction\n", hubs->dcs_codes); - /* Different chips in the family support different - * readback methods. - */ - switch (hubs->dcs_readback_mode) { - case 0: - reg_l = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_1) - & WM8993_DCS_INTEG_CHAN_0_MASK;; - reg_r = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_2) - & WM8993_DCS_INTEG_CHAN_1_MASK; - break; - case 1: - reg = snd_soc_read(codec, WM8993_DC_SERVO_3); - reg_l = (reg & WM8993_DCS_DAC_WR_VAL_1_MASK) - >> WM8993_DCS_DAC_WR_VAL_1_SHIFT; - reg_r = reg & WM8993_DCS_DAC_WR_VAL_0_MASK; - break; - default: - WARN(1, "Unknown DCS readback method\n"); - break; - } - - dev_dbg(codec->dev, "DCS input: %x %x\n", reg_l, reg_r); - /* HPOUT1L */ if (reg_l + hubs->dcs_codes > 0 && reg_l + hubs->dcs_codes < 0xff) @@ -148,7 +160,15 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec) wait_for_dc_servo(codec, WM8993_DCS_TRIG_DAC_WR_0 | WM8993_DCS_TRIG_DAC_WR_1); + } else { + dcs_cfg = reg_l << WM8993_DCS_DAC_WR_VAL_1_SHIFT; + dcs_cfg |= reg_r; } + + /* Save the callibrated offset if we're in class W mode and + * therefore don't have any analogue signal mixed in. */ + if (hubs->class_w) + hubs->class_w_dcs = dcs_cfg; } /* @@ -163,6 +183,9 @@ static int wm8993_put_dc_servo(struct snd_kcontrol *kcontrol, ret = snd_soc_put_volsw_2r(kcontrol, ucontrol); + /* Updating the analogue gains invalidates the DC servo cache */ + hubs->class_w_dcs = 0; + /* If we're applying an offset correction then updating the * callibration would be likely to introduce further offsets. */ if (hubs->dcs_codes) @@ -791,6 +814,8 @@ static const struct snd_soc_dapm_route lineout2_se_routes[] = { int wm_hubs_add_analogue_controls(struct snd_soc_codec *codec) { + struct snd_soc_dapm_context *dapm = &codec->dapm; + /* Latch volume update bits & default ZC on */ snd_soc_update_bits(codec, WM8993_LEFT_LINE_INPUT_1_2_VOLUME, WM8993_IN1_VU, WM8993_IN1_VU); @@ -819,7 +844,7 @@ int wm_hubs_add_analogue_controls(struct snd_soc_codec *codec) snd_soc_add_controls(codec, analogue_snd_controls, ARRAY_SIZE(analogue_snd_controls)); - snd_soc_dapm_new_controls(codec, analogue_dapm_widgets, + snd_soc_dapm_new_controls(dapm, analogue_dapm_widgets, ARRAY_SIZE(analogue_dapm_widgets)); return 0; } @@ -828,24 +853,26 @@ EXPORT_SYMBOL_GPL(wm_hubs_add_analogue_controls); int wm_hubs_add_analogue_routes(struct snd_soc_codec *codec, int lineout1_diff, int lineout2_diff) { - snd_soc_dapm_add_routes(codec, analogue_routes, + struct snd_soc_dapm_context *dapm = &codec->dapm; + + snd_soc_dapm_add_routes(dapm, analogue_routes, ARRAY_SIZE(analogue_routes)); if (lineout1_diff) - snd_soc_dapm_add_routes(codec, + snd_soc_dapm_add_routes(dapm, lineout1_diff_routes, ARRAY_SIZE(lineout1_diff_routes)); else - snd_soc_dapm_add_routes(codec, + snd_soc_dapm_add_routes(dapm, lineout1_se_routes, ARRAY_SIZE(lineout1_se_routes)); if (lineout2_diff) - snd_soc_dapm_add_routes(codec, + snd_soc_dapm_add_routes(dapm, lineout2_diff_routes, ARRAY_SIZE(lineout2_diff_routes)); else - snd_soc_dapm_add_routes(codec, + snd_soc_dapm_add_routes(dapm, lineout2_se_routes, ARRAY_SIZE(lineout2_se_routes)); @@ -872,7 +899,7 @@ int wm_hubs_handle_analogue_pdata(struct snd_soc_codec *codec, * VMID as an output and can disable it. */ if (lineout1_diff && lineout2_diff) - codec->idle_bias_off = 1; + codec->dapm.idle_bias_off = 1; if (lineout1fb) snd_soc_update_bits(codec, WM8993_ADDITIONAL_CONTROL, diff --git a/sound/soc/codecs/wm_hubs.h b/sound/soc/codecs/wm_hubs.h index e51c16683589..f8a5e976b5e6 100644 --- a/sound/soc/codecs/wm_hubs.h +++ b/sound/soc/codecs/wm_hubs.h @@ -23,6 +23,9 @@ struct wm_hubs_data { int dcs_codes; int dcs_readback_mode; int hp_startup_mode; + + bool class_w; + u16 class_w_dcs; }; extern int wm_hubs_add_analogue_controls(struct snd_soc_codec *); diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 2b07b17a6b2d..a2cf64b221e5 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -132,26 +132,27 @@ static const struct snd_soc_dapm_route audio_map[] = { static int evm_aic3x_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; /* Add davinci-evm specific widgets */ - snd_soc_dapm_new_controls(codec, aic3x_dapm_widgets, + snd_soc_dapm_new_controls(dapm, aic3x_dapm_widgets, ARRAY_SIZE(aic3x_dapm_widgets)); /* Set up davinci-evm specific audio path audio_map */ - snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); /* not connected */ - snd_soc_dapm_disable_pin(codec, "MONO_LOUT"); - snd_soc_dapm_disable_pin(codec, "HPLCOM"); - snd_soc_dapm_disable_pin(codec, "HPRCOM"); + snd_soc_dapm_disable_pin(dapm, "MONO_LOUT"); + snd_soc_dapm_disable_pin(dapm, "HPLCOM"); + snd_soc_dapm_disable_pin(dapm, "HPRCOM"); /* always connected */ - snd_soc_dapm_enable_pin(codec, "Headphone Jack"); - snd_soc_dapm_enable_pin(codec, "Line Out"); - snd_soc_dapm_enable_pin(codec, "Mic Jack"); - snd_soc_dapm_enable_pin(codec, "Line In"); + snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); + snd_soc_dapm_enable_pin(dapm, "Line Out"); + snd_soc_dapm_enable_pin(dapm, "Mic Jack"); + snd_soc_dapm_enable_pin(dapm, "Line In"); - snd_soc_dapm_sync(codec); + snd_soc_dapm_sync(dapm); return 0; } diff --git a/sound/soc/ep93xx/snappercl15.c b/sound/soc/ep93xx/snappercl15.c index 28ab5ff772ac..f1c78516ccac 100644 --- a/sound/soc/ep93xx/snappercl15.c +++ b/sound/soc/ep93xx/snappercl15.c @@ -79,11 +79,12 @@ static const struct snd_soc_dapm_route audio_map[] = { static int snappercl15_tlv320aic23_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; - snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets, + snd_soc_dapm_new_controls(dapm, tlv320aic23_dapm_widgets, ARRAY_SIZE(tlv320aic23_dapm_widgets)); - snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); return 0; } diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c index d2d98c75ee8a..ad21f814e6a1 100644 --- a/sound/soc/imx/imx-ssi.c +++ b/sound/soc/imx/imx-ssi.c @@ -456,13 +456,13 @@ static int imx_ssi_dai_probe(struct snd_soc_dai *dai) static struct snd_soc_dai_driver imx_ssi_dai = { .probe = imx_ssi_dai_probe, .playback = { - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_96000, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .capture = { - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_96000, .formats = SNDRV_PCM_FMTBIT_S16_LE, diff --git a/sound/soc/imx/wm1133-ev1.c b/sound/soc/imx/wm1133-ev1.c index 30fdb15065be..46fadf497243 100644 --- a/sound/soc/imx/wm1133-ev1.c +++ b/sound/soc/imx/wm1133-ev1.c @@ -213,11 +213,12 @@ static struct snd_soc_jack_pin mic_jack_pins[] = { static int wm1133_ev1_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; - snd_soc_dapm_new_controls(codec, wm1133_ev1_widgets, + snd_soc_dapm_new_controls(dapm, wm1133_ev1_widgets, ARRAY_SIZE(wm1133_ev1_widgets)); - snd_soc_dapm_add_routes(codec, wm1133_ev1_map, + snd_soc_dapm_add_routes(dapm, wm1133_ev1_map, ARRAY_SIZE(wm1133_ev1_map)); /* Headphone jack detection */ @@ -234,7 +235,7 @@ static int wm1133_ev1_init(struct snd_soc_pcm_runtime *rtd) wm8350_mic_jack_detect(codec, &mic_jack, SND_JACK_MICROPHONE, SND_JACK_BTN_0); - snd_soc_dapm_force_enable_pin(codec, "Mic Bias"); + snd_soc_dapm_force_enable_pin(dapm, "Mic Bias"); return 0; } diff --git a/sound/soc/jz4740/qi_lb60.c b/sound/soc/jz4740/qi_lb60.c index ef1a99e6a3bd..70afbfada9fd 100644 --- a/sound/soc/jz4740/qi_lb60.c +++ b/sound/soc/jz4740/qi_lb60.c @@ -59,10 +59,11 @@ static int qi_lb60_codec_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dapm_context *dapm = &codec->dapm; int ret; - snd_soc_dapm_nc_pin(codec, "LIN"); - snd_soc_dapm_nc_pin(codec, "RIN"); + snd_soc_dapm_nc_pin(dapm, "LIN"); + snd_soc_dapm_nc_pin(dapm, "RIN"); ret = snd_soc_dai_set_fmt(cpu_dai, QI_LB60_DAIFMT); if (ret < 0) { @@ -70,9 +71,11 @@ static int qi_lb60_codec_init(struct snd_soc_pcm_runtime *rtd) return ret; } - snd_soc_dapm_new_controls(codec, qi_lb60_widgets, ARRAY_SIZE(qi_lb60_widgets)); - snd_soc_dapm_add_routes(codec, qi_lb60_routes, ARRAY_SIZE(qi_lb60_routes)); - snd_soc_dapm_sync(codec); + snd_soc_dapm_new_controls(dapm, qi_lb60_widgets, + ARRAY_SIZE(qi_lb60_widgets)); + snd_soc_dapm_add_routes(dapm, qi_lb60_routes, + ARRAY_SIZE(qi_lb60_routes)); + snd_soc_dapm_sync(dapm); return 0; } diff --git a/sound/soc/kirkwood/Kconfig b/sound/soc/kirkwood/Kconfig index 16ec2a2dba4d..54258fd9797f 100644 --- a/sound/soc/kirkwood/Kconfig +++ b/sound/soc/kirkwood/Kconfig @@ -18,3 +18,12 @@ config SND_KIRKWOOD_SOC_OPENRD Say Y if you want to add support for SoC audio on Openrd Client. +config SND_KIRKWOOD_SOC_T5325 + tristate "SoC Audio support for HP t5325" + depends on SND_KIRKWOOD_SOC && MACH_T5325 + select SND_KIRKWOOD_SOC_I2S + select SND_SOC_ALC5623 + help + Say Y if you want to add support for SoC audio on + the HP t5325 thin client. + diff --git a/sound/soc/kirkwood/Makefile b/sound/soc/kirkwood/Makefile index 33a16dcab5b5..3e62ae9e7bbe 100644 --- a/sound/soc/kirkwood/Makefile +++ b/sound/soc/kirkwood/Makefile @@ -5,5 +5,7 @@ obj-$(CONFIG_SND_KIRKWOOD_SOC) += snd-soc-kirkwood.o obj-$(CONFIG_SND_KIRKWOOD_SOC_I2S) += snd-soc-kirkwood-i2s.o snd-soc-openrd-objs := kirkwood-openrd.o +snd-soc-t5325-objs := kirkwood-t5325.o obj-$(CONFIG_SND_KIRKWOOD_SOC_OPENRD) += snd-soc-openrd.o +obj-$(CONFIG_SND_KIRKWOOD_SOC_T5325) += snd-soc-t5325.o diff --git a/sound/soc/kirkwood/kirkwood-t5325.c b/sound/soc/kirkwood/kirkwood-t5325.c new file mode 100644 index 000000000000..07b6ecaed2f2 --- /dev/null +++ b/sound/soc/kirkwood/kirkwood-t5325.c @@ -0,0 +1,142 @@ +/* + * kirkwood-t5325.c + * + * (c) 2010 Arnaud Patard <arnaud.patard@rtp-net.org> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <mach/kirkwood.h> +#include <plat/audio.h> +#include <asm/mach-types.h> +#include "../codecs/alc5623.h" + +static int t5325_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + int ret; + unsigned int freq, fmt; + + fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS; + ret = snd_soc_dai_set_fmt(cpu_dai, fmt); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(codec_dai, fmt); + if (ret < 0) + return ret; + + freq = params_rate(params) * 256; + + return snd_soc_dai_set_sysclk(codec_dai, 0, freq, SND_SOC_CLOCK_IN); + +} + +static struct snd_soc_ops t5325_ops = { + .hw_params = t5325_hw_params, +}; + +static const struct snd_soc_dapm_widget t5325_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), +}; + +static const struct snd_soc_dapm_route t5325_route[] = { + { "Headphone Jack", NULL, "HPL" }, + { "Headphone Jack", NULL, "HPR" }, + + {"Speaker", NULL, "SPKOUT"}, + {"Speaker", NULL, "SPKOUTN"}, + + { "MIC1", NULL, "Mic Jack" }, + { "MIC2", NULL, "Mic Jack" }, +}; + +static int t5325_dai_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + + snd_soc_dapm_new_controls(dapm, t5325_dapm_widgets, + ARRAY_SIZE(t5325_dapm_widgets)); + + snd_soc_dapm_add_routes(dapm, t5325_route, ARRAY_SIZE(t5325_route)); + + snd_soc_dapm_enable_pin(dapm, "Mic Jack"); + snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); + snd_soc_dapm_enable_pin(dapm, "Speaker"); + + snd_soc_dapm_sync(dapm); + + return 0; +} + +static struct snd_soc_dai_link t5325_dai[] = { +{ + .name = "ALC5621", + .stream_name = "ALC5621 HiFi", + .cpu_dai_name = "kirkwood-i2s", + .platform_name = "kirkwood-pcm-audio", + .codec_dai_name = "alc5621-hifi", + .codec_name = "alc562x-codec.0-001a", + .ops = &t5325_ops, + .init = t5325_dai_init, +}, +}; + + +static struct snd_soc_card t5325 = { + .name = "t5325", + .dai_link = t5325_dai, + .num_links = ARRAY_SIZE(t5325_dai), +}; + +static struct platform_device *t5325_snd_device; + +static int __init t5325_init(void) +{ + int ret; + + if (!machine_is_t5325()) + return 0; + + t5325_snd_device = platform_device_alloc("soc-audio", -1); + if (!t5325_snd_device) + return -ENOMEM; + + platform_set_drvdata(t5325_snd_device, + &t5325); + + ret = platform_device_add(t5325_snd_device); + if (ret) { + printk(KERN_ERR "%s: platform_device_add failed\n", __func__); + platform_device_put(t5325_snd_device); + } + + return ret; +} +module_init(t5325_init); + +static void __exit t5325_exit(void) +{ + platform_device_unregister(t5325_snd_device); +} +module_exit(t5325_exit); + +MODULE_AUTHOR("Arnaud Patard <arnaud.patard@rtp-net.org>"); +MODULE_DESCRIPTION("ALSA SoC t5325 audio client"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/omap/am3517evm.c b/sound/soc/omap/am3517evm.c index 979dd508305f..668773def0dc 100644 --- a/sound/soc/omap/am3517evm.c +++ b/sound/soc/omap/am3517evm.c @@ -114,20 +114,21 @@ static const struct snd_soc_dapm_route audio_map[] = { static int am3517evm_aic23_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; /* Add am3517-evm specific widgets */ - snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets, + snd_soc_dapm_new_controls(dapm, tlv320aic23_dapm_widgets, ARRAY_SIZE(tlv320aic23_dapm_widgets)); /* Set up davinci-evm specific audio path audio_map */ - snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); /* always connected */ - snd_soc_dapm_enable_pin(codec, "Line Out"); - snd_soc_dapm_enable_pin(codec, "Line In"); - snd_soc_dapm_enable_pin(codec, "Mic In"); + snd_soc_dapm_enable_pin(dapm, "Line Out"); + snd_soc_dapm_enable_pin(dapm, "Line In"); + snd_soc_dapm_enable_pin(dapm, "Mic In"); - snd_soc_dapm_sync(codec); + snd_soc_dapm_sync(dapm); return 0; } diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c index 438146addbb8..2101bdcee21f 100644 --- a/sound/soc/omap/ams-delta.c +++ b/sound/soc/omap/ams-delta.c @@ -26,7 +26,7 @@ #include <linux/spinlock.h> #include <linux/tty.h> -#include <sound/soc-dapm.h> +#include <sound/soc.h> #include <sound/jack.h> #include <asm/mach-types.h> @@ -94,6 +94,7 @@ static int ams_delta_set_audio_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_context *dapm = &codec->dapm; struct soc_enum *control = (struct soc_enum *)kcontrol->private_value; unsigned short pins; int pin, changed = 0; @@ -112,48 +113,48 @@ static int ams_delta_set_audio_mode(struct snd_kcontrol *kcontrol, /* Setup pins after corresponding bits if changed */ pin = !!(pins & (1 << AMS_DELTA_MOUTHPIECE)); - if (pin != snd_soc_dapm_get_pin_status(codec, "Mouthpiece")) { + if (pin != snd_soc_dapm_get_pin_status(dapm, "Mouthpiece")) { changed = 1; if (pin) - snd_soc_dapm_enable_pin(codec, "Mouthpiece"); + snd_soc_dapm_enable_pin(dapm, "Mouthpiece"); else - snd_soc_dapm_disable_pin(codec, "Mouthpiece"); + snd_soc_dapm_disable_pin(dapm, "Mouthpiece"); } pin = !!(pins & (1 << AMS_DELTA_EARPIECE)); - if (pin != snd_soc_dapm_get_pin_status(codec, "Earpiece")) { + if (pin != snd_soc_dapm_get_pin_status(dapm, "Earpiece")) { changed = 1; if (pin) - snd_soc_dapm_enable_pin(codec, "Earpiece"); + snd_soc_dapm_enable_pin(dapm, "Earpiece"); else - snd_soc_dapm_disable_pin(codec, "Earpiece"); + snd_soc_dapm_disable_pin(dapm, "Earpiece"); } pin = !!(pins & (1 << AMS_DELTA_MICROPHONE)); - if (pin != snd_soc_dapm_get_pin_status(codec, "Microphone")) { + if (pin != snd_soc_dapm_get_pin_status(dapm, "Microphone")) { changed = 1; if (pin) - snd_soc_dapm_enable_pin(codec, "Microphone"); + snd_soc_dapm_enable_pin(dapm, "Microphone"); else - snd_soc_dapm_disable_pin(codec, "Microphone"); + snd_soc_dapm_disable_pin(dapm, "Microphone"); } pin = !!(pins & (1 << AMS_DELTA_SPEAKER)); - if (pin != snd_soc_dapm_get_pin_status(codec, "Speaker")) { + if (pin != snd_soc_dapm_get_pin_status(dapm, "Speaker")) { changed = 1; if (pin) - snd_soc_dapm_enable_pin(codec, "Speaker"); + snd_soc_dapm_enable_pin(dapm, "Speaker"); else - snd_soc_dapm_disable_pin(codec, "Speaker"); + snd_soc_dapm_disable_pin(dapm, "Speaker"); } pin = !!(pins & (1 << AMS_DELTA_AGC)); if (pin != ams_delta_audio_agc) { ams_delta_audio_agc = pin; changed = 1; if (pin) - snd_soc_dapm_enable_pin(codec, "AGCIN"); + snd_soc_dapm_enable_pin(dapm, "AGCIN"); else - snd_soc_dapm_disable_pin(codec, "AGCIN"); + snd_soc_dapm_disable_pin(dapm, "AGCIN"); } if (changed) - snd_soc_dapm_sync(codec); + snd_soc_dapm_sync(dapm); mutex_unlock(&codec->mutex); @@ -164,19 +165,20 @@ static int ams_delta_get_audio_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_context *dapm = &codec->dapm; unsigned short pins, mode; - pins = ((snd_soc_dapm_get_pin_status(codec, "Mouthpiece") << + pins = ((snd_soc_dapm_get_pin_status(dapm, "Mouthpiece") << AMS_DELTA_MOUTHPIECE) | - (snd_soc_dapm_get_pin_status(codec, "Earpiece") << + (snd_soc_dapm_get_pin_status(dapm, "Earpiece") << AMS_DELTA_EARPIECE)); if (pins) - pins |= (snd_soc_dapm_get_pin_status(codec, "Microphone") << + pins |= (snd_soc_dapm_get_pin_status(dapm, "Microphone") << AMS_DELTA_MICROPHONE); else - pins = ((snd_soc_dapm_get_pin_status(codec, "Microphone") << + pins = ((snd_soc_dapm_get_pin_status(dapm, "Microphone") << AMS_DELTA_MICROPHONE) | - (snd_soc_dapm_get_pin_status(codec, "Speaker") << + (snd_soc_dapm_get_pin_status(dapm, "Speaker") << AMS_DELTA_SPEAKER) | (ams_delta_audio_agc << AMS_DELTA_AGC)); @@ -300,6 +302,7 @@ static int cx81801_open(struct tty_struct *tty) static void cx81801_close(struct tty_struct *tty) { struct snd_soc_codec *codec = tty->disc_data; + struct snd_soc_dapm_context *dapm = &codec->dapm; del_timer_sync(&cx81801_timer); @@ -312,12 +315,12 @@ static void cx81801_close(struct tty_struct *tty) v253_ops.close(tty); /* Revert back to default audio input/output constellation */ - snd_soc_dapm_disable_pin(codec, "Mouthpiece"); - snd_soc_dapm_enable_pin(codec, "Earpiece"); - snd_soc_dapm_enable_pin(codec, "Microphone"); - snd_soc_dapm_disable_pin(codec, "Speaker"); - snd_soc_dapm_disable_pin(codec, "AGCIN"); - snd_soc_dapm_sync(codec); + snd_soc_dapm_disable_pin(dapm, "Mouthpiece"); + snd_soc_dapm_enable_pin(dapm, "Earpiece"); + snd_soc_dapm_enable_pin(dapm, "Microphone"); + snd_soc_dapm_disable_pin(dapm, "Speaker"); + snd_soc_dapm_disable_pin(dapm, "AGCIN"); + snd_soc_dapm_sync(dapm); } /* Line discipline .hangup() */ @@ -432,16 +435,16 @@ static int ams_delta_set_bias_level(struct snd_soc_card *card, case SND_SOC_BIAS_ON: case SND_SOC_BIAS_PREPARE: case SND_SOC_BIAS_STANDBY: - if (codec->bias_level == SND_SOC_BIAS_OFF) + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_NRESET, AMS_DELTA_LATCH2_MODEM_NRESET); break; case SND_SOC_BIAS_OFF: - if (codec->bias_level != SND_SOC_BIAS_OFF) + if (codec->dapm.bias_level != SND_SOC_BIAS_OFF) ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_NRESET, 0); } - codec->bias_level = level; + codec->dapm.bias_level = level; return 0; } @@ -492,6 +495,7 @@ static void ams_delta_shutdown(struct snd_pcm_substream *substream) static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_card *card = rtd->card; int ret; @@ -541,7 +545,7 @@ static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd) } /* Add board specific DAPM widgets and routes */ - ret = snd_soc_dapm_new_controls(codec, ams_delta_dapm_widgets, + ret = snd_soc_dapm_new_controls(dapm, ams_delta_dapm_widgets, ARRAY_SIZE(ams_delta_dapm_widgets)); if (ret) { dev_warn(card->dev, @@ -550,7 +554,7 @@ static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd) return 0; } - ret = snd_soc_dapm_add_routes(codec, ams_delta_audio_map, + ret = snd_soc_dapm_add_routes(dapm, ams_delta_audio_map, ARRAY_SIZE(ams_delta_audio_map)); if (ret) { dev_warn(card->dev, @@ -560,13 +564,13 @@ static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd) } /* Set up initial pin constellation */ - snd_soc_dapm_disable_pin(codec, "Mouthpiece"); - snd_soc_dapm_enable_pin(codec, "Earpiece"); - snd_soc_dapm_enable_pin(codec, "Microphone"); - snd_soc_dapm_disable_pin(codec, "Speaker"); - snd_soc_dapm_disable_pin(codec, "AGCIN"); - snd_soc_dapm_disable_pin(codec, "AGCOUT"); - snd_soc_dapm_sync(codec); + snd_soc_dapm_disable_pin(dapm, "Mouthpiece"); + snd_soc_dapm_enable_pin(dapm, "Earpiece"); + snd_soc_dapm_enable_pin(dapm, "Microphone"); + snd_soc_dapm_disable_pin(dapm, "Speaker"); + snd_soc_dapm_disable_pin(dapm, "AGCIN"); + snd_soc_dapm_disable_pin(dapm, "AGCOUT"); + snd_soc_dapm_sync(dapm); /* Add virtual switch */ ret = snd_soc_add_controls(codec, ams_delta_audio_controls, diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index a3b6d897ad84..296cd9b7eecb 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -58,6 +58,7 @@ static int n810_dmic_func; static void n810_ext_control(struct snd_soc_codec *codec) { + struct snd_soc_dapm_context *dapm = &codec->dapm; int hp = 0, line1l = 0; switch (n810_jack_func) { @@ -72,25 +73,25 @@ static void n810_ext_control(struct snd_soc_codec *codec) } if (n810_spk_func) - snd_soc_dapm_enable_pin(codec, "Ext Spk"); + snd_soc_dapm_enable_pin(dapm, "Ext Spk"); else - snd_soc_dapm_disable_pin(codec, "Ext Spk"); + snd_soc_dapm_disable_pin(dapm, "Ext Spk"); if (hp) - snd_soc_dapm_enable_pin(codec, "Headphone Jack"); + snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); else - snd_soc_dapm_disable_pin(codec, "Headphone Jack"); + snd_soc_dapm_disable_pin(dapm, "Headphone Jack"); if (line1l) - snd_soc_dapm_enable_pin(codec, "LINE1L"); + snd_soc_dapm_enable_pin(dapm, "LINE1L"); else - snd_soc_dapm_disable_pin(codec, "LINE1L"); + snd_soc_dapm_disable_pin(dapm, "LINE1L"); if (n810_dmic_func) - snd_soc_dapm_enable_pin(codec, "DMic"); + snd_soc_dapm_enable_pin(dapm, "DMic"); else - snd_soc_dapm_disable_pin(codec, "DMic"); + snd_soc_dapm_disable_pin(dapm, "DMic"); - snd_soc_dapm_sync(codec); + snd_soc_dapm_sync(dapm); } static int n810_startup(struct snd_pcm_substream *substream) @@ -274,17 +275,18 @@ static const struct snd_kcontrol_new aic33_n810_controls[] = { static int n810_aic33_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; int err; /* Not connected */ - snd_soc_dapm_nc_pin(codec, "MONO_LOUT"); - snd_soc_dapm_nc_pin(codec, "HPLCOM"); - snd_soc_dapm_nc_pin(codec, "HPRCOM"); - snd_soc_dapm_nc_pin(codec, "MIC3L"); - snd_soc_dapm_nc_pin(codec, "MIC3R"); - snd_soc_dapm_nc_pin(codec, "LINE1R"); - snd_soc_dapm_nc_pin(codec, "LINE2L"); - snd_soc_dapm_nc_pin(codec, "LINE2R"); + snd_soc_dapm_nc_pin(dapm, "MONO_LOUT"); + snd_soc_dapm_nc_pin(dapm, "HPLCOM"); + snd_soc_dapm_nc_pin(dapm, "HPRCOM"); + snd_soc_dapm_nc_pin(dapm, "MIC3L"); + snd_soc_dapm_nc_pin(dapm, "MIC3R"); + snd_soc_dapm_nc_pin(dapm, "LINE1R"); + snd_soc_dapm_nc_pin(dapm, "LINE2L"); + snd_soc_dapm_nc_pin(dapm, "LINE2R"); /* Add N810 specific controls */ err = snd_soc_add_controls(codec, aic33_n810_controls, @@ -293,13 +295,13 @@ static int n810_aic33_init(struct snd_soc_pcm_runtime *rtd) return err; /* Add N810 specific widgets */ - snd_soc_dapm_new_controls(codec, aic33_dapm_widgets, + snd_soc_dapm_new_controls(dapm, aic33_dapm_widgets, ARRAY_SIZE(aic33_dapm_widgets)); /* Set up N810 specific audio path audio_map */ - snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_sync(codec); + snd_soc_dapm_sync(dapm); return 0; } diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c index dbd9d96b5f92..93e83c0f6660 100644 --- a/sound/soc/omap/omap3pandora.c +++ b/sound/soc/omap/omap3pandora.c @@ -170,51 +170,53 @@ static const struct snd_soc_dapm_route omap3pandora_in_map[] = { static int omap3pandora_out_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; int ret; /* All TWL4030 output pins are floating */ - snd_soc_dapm_nc_pin(codec, "EARPIECE"); - snd_soc_dapm_nc_pin(codec, "PREDRIVEL"); - snd_soc_dapm_nc_pin(codec, "PREDRIVER"); - snd_soc_dapm_nc_pin(codec, "HSOL"); - snd_soc_dapm_nc_pin(codec, "HSOR"); - snd_soc_dapm_nc_pin(codec, "CARKITL"); - snd_soc_dapm_nc_pin(codec, "CARKITR"); - snd_soc_dapm_nc_pin(codec, "HFL"); - snd_soc_dapm_nc_pin(codec, "HFR"); - snd_soc_dapm_nc_pin(codec, "VIBRA"); - - ret = snd_soc_dapm_new_controls(codec, omap3pandora_out_dapm_widgets, + snd_soc_dapm_nc_pin(dapm, "EARPIECE"); + snd_soc_dapm_nc_pin(dapm, "PREDRIVEL"); + snd_soc_dapm_nc_pin(dapm, "PREDRIVER"); + snd_soc_dapm_nc_pin(dapm, "HSOL"); + snd_soc_dapm_nc_pin(dapm, "HSOR"); + snd_soc_dapm_nc_pin(dapm, "CARKITL"); + snd_soc_dapm_nc_pin(dapm, "CARKITR"); + snd_soc_dapm_nc_pin(dapm, "HFL"); + snd_soc_dapm_nc_pin(dapm, "HFR"); + snd_soc_dapm_nc_pin(dapm, "VIBRA"); + + ret = snd_soc_dapm_new_controls(dapm, omap3pandora_out_dapm_widgets, ARRAY_SIZE(omap3pandora_out_dapm_widgets)); if (ret < 0) return ret; - snd_soc_dapm_add_routes(codec, omap3pandora_out_map, + snd_soc_dapm_add_routes(dapm, omap3pandora_out_map, ARRAY_SIZE(omap3pandora_out_map)); - return snd_soc_dapm_sync(codec); + return snd_soc_dapm_sync(dapm); } static int omap3pandora_in_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; int ret; /* Not comnnected */ - snd_soc_dapm_nc_pin(codec, "HSMIC"); - snd_soc_dapm_nc_pin(codec, "CARKITMIC"); - snd_soc_dapm_nc_pin(codec, "DIGIMIC0"); - snd_soc_dapm_nc_pin(codec, "DIGIMIC1"); + snd_soc_dapm_nc_pin(dapm, "HSMIC"); + snd_soc_dapm_nc_pin(dapm, "CARKITMIC"); + snd_soc_dapm_nc_pin(dapm, "DIGIMIC0"); + snd_soc_dapm_nc_pin(dapm, "DIGIMIC1"); - ret = snd_soc_dapm_new_controls(codec, omap3pandora_in_dapm_widgets, + ret = snd_soc_dapm_new_controls(dapm, omap3pandora_in_dapm_widgets, ARRAY_SIZE(omap3pandora_in_dapm_widgets)); if (ret < 0) return ret; - snd_soc_dapm_add_routes(codec, omap3pandora_in_map, + snd_soc_dapm_add_routes(dapm, omap3pandora_in_map, ARRAY_SIZE(omap3pandora_in_map)); - return snd_soc_dapm_sync(codec); + return snd_soc_dapm_sync(dapm); } static struct snd_soc_ops omap3pandora_ops = { diff --git a/sound/soc/omap/osk5912.c b/sound/soc/omap/osk5912.c index f0e662556428..c2a54204559d 100644 --- a/sound/soc/omap/osk5912.c +++ b/sound/soc/omap/osk5912.c @@ -116,19 +116,20 @@ static const struct snd_soc_dapm_route audio_map[] = { static int osk_tlv320aic23_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; /* Add osk5912 specific widgets */ - snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets, + snd_soc_dapm_new_controls(dapm, tlv320aic23_dapm_widgets, ARRAY_SIZE(tlv320aic23_dapm_widgets)); /* Set up osk5912 specific audio path audio_map */ - snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_enable_pin(codec, "Headphone Jack"); - snd_soc_dapm_enable_pin(codec, "Line In"); - snd_soc_dapm_enable_pin(codec, "Mic Jack"); + snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); + snd_soc_dapm_enable_pin(dapm, "Line In"); + snd_soc_dapm_enable_pin(dapm, "Mic Jack"); - snd_soc_dapm_sync(codec); + snd_soc_dapm_sync(dapm); return 0; } diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index 04b5723bf89b..62fc7a4f306b 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -58,19 +58,21 @@ static int rx51_jack_func; static void rx51_ext_control(struct snd_soc_codec *codec) { + struct snd_soc_dapm_context *dapm = &codec->dapm; + if (rx51_spk_func) - snd_soc_dapm_enable_pin(codec, "Ext Spk"); + snd_soc_dapm_enable_pin(dapm, "Ext Spk"); else - snd_soc_dapm_disable_pin(codec, "Ext Spk"); + snd_soc_dapm_disable_pin(dapm, "Ext Spk"); if (rx51_dmic_func) - snd_soc_dapm_enable_pin(codec, "DMic"); + snd_soc_dapm_enable_pin(dapm, "DMic"); else - snd_soc_dapm_disable_pin(codec, "DMic"); + snd_soc_dapm_disable_pin(dapm, "DMic"); gpio_set_value(RX51_TVOUT_SEL_GPIO, rx51_jack_func == RX51_JACK_TVOUT); - snd_soc_dapm_sync(codec); + snd_soc_dapm_sync(dapm); } static int rx51_startup(struct snd_pcm_substream *substream) @@ -244,12 +246,13 @@ static const struct snd_kcontrol_new aic34_rx51_controls[] = { static int rx51_aic34_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; int err; /* Set up NC codec pins */ - snd_soc_dapm_nc_pin(codec, "MIC3L"); - snd_soc_dapm_nc_pin(codec, "MIC3R"); - snd_soc_dapm_nc_pin(codec, "LINE1R"); + snd_soc_dapm_nc_pin(dapm, "MIC3L"); + snd_soc_dapm_nc_pin(dapm, "MIC3R"); + snd_soc_dapm_nc_pin(dapm, "LINE1R"); /* Add RX-51 specific controls */ err = snd_soc_add_controls(codec, aic34_rx51_controls, @@ -258,13 +261,13 @@ static int rx51_aic34_init(struct snd_soc_pcm_runtime *rtd) return err; /* Add RX-51 specific widgets */ - snd_soc_dapm_new_controls(codec, aic34_dapm_widgets, + snd_soc_dapm_new_controls(dapm, aic34_dapm_widgets, ARRAY_SIZE(aic34_dapm_widgets)); /* Set up RX-51 specific audio path audio_map */ - snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_sync(codec); + snd_soc_dapm_sync(dapm); /* AV jack detection */ err = snd_soc_jack_new(codec, "AV Jack", diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c index 07fbcf7d2411..a3dd07a39fec 100644 --- a/sound/soc/omap/sdp3430.c +++ b/sound/soc/omap/sdp3430.c @@ -191,39 +191,40 @@ static const struct snd_soc_dapm_route audio_map[] = { static int sdp3430_twl4030_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; int ret; /* Add SDP3430 specific widgets */ - ret = snd_soc_dapm_new_controls(codec, sdp3430_twl4030_dapm_widgets, + ret = snd_soc_dapm_new_controls(dapm, sdp3430_twl4030_dapm_widgets, ARRAY_SIZE(sdp3430_twl4030_dapm_widgets)); if (ret) return ret; /* Set up SDP3430 specific audio path audio_map */ - snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); /* SDP3430 connected pins */ - snd_soc_dapm_enable_pin(codec, "Ext Mic"); - snd_soc_dapm_enable_pin(codec, "Ext Spk"); - snd_soc_dapm_disable_pin(codec, "Headset Mic"); - snd_soc_dapm_disable_pin(codec, "Headset Stereophone"); + snd_soc_dapm_enable_pin(dapm, "Ext Mic"); + snd_soc_dapm_enable_pin(dapm, "Ext Spk"); + snd_soc_dapm_disable_pin(dapm, "Headset Mic"); + snd_soc_dapm_disable_pin(dapm, "Headset Stereophone"); /* TWL4030 not connected pins */ - snd_soc_dapm_nc_pin(codec, "AUXL"); - snd_soc_dapm_nc_pin(codec, "AUXR"); - snd_soc_dapm_nc_pin(codec, "CARKITMIC"); - snd_soc_dapm_nc_pin(codec, "DIGIMIC0"); - snd_soc_dapm_nc_pin(codec, "DIGIMIC1"); - - snd_soc_dapm_nc_pin(codec, "OUTL"); - snd_soc_dapm_nc_pin(codec, "OUTR"); - snd_soc_dapm_nc_pin(codec, "EARPIECE"); - snd_soc_dapm_nc_pin(codec, "PREDRIVEL"); - snd_soc_dapm_nc_pin(codec, "PREDRIVER"); - snd_soc_dapm_nc_pin(codec, "CARKITL"); - snd_soc_dapm_nc_pin(codec, "CARKITR"); - - ret = snd_soc_dapm_sync(codec); + snd_soc_dapm_nc_pin(dapm, "AUXL"); + snd_soc_dapm_nc_pin(dapm, "AUXR"); + snd_soc_dapm_nc_pin(dapm, "CARKITMIC"); + snd_soc_dapm_nc_pin(dapm, "DIGIMIC0"); + snd_soc_dapm_nc_pin(dapm, "DIGIMIC1"); + + snd_soc_dapm_nc_pin(dapm, "OUTL"); + snd_soc_dapm_nc_pin(dapm, "OUTR"); + snd_soc_dapm_nc_pin(dapm, "EARPIECE"); + snd_soc_dapm_nc_pin(dapm, "PREDRIVEL"); + snd_soc_dapm_nc_pin(dapm, "PREDRIVER"); + snd_soc_dapm_nc_pin(dapm, "CARKITL"); + snd_soc_dapm_nc_pin(dapm, "CARKITR"); + + ret = snd_soc_dapm_sync(dapm); if (ret) return ret; diff --git a/sound/soc/omap/sdp4430.c b/sound/soc/omap/sdp4430.c index 4b4463db6ba0..3ce17318a291 100644 --- a/sound/soc/omap/sdp4430.c +++ b/sound/soc/omap/sdp4430.c @@ -129,6 +129,7 @@ static const struct snd_soc_dapm_route audio_map[] = { static int sdp4430_twl6040_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; int ret; /* Add SDP4430 specific controls */ @@ -138,25 +139,25 @@ static int sdp4430_twl6040_init(struct snd_soc_pcm_runtime *rtd) return ret; /* Add SDP4430 specific widgets */ - ret = snd_soc_dapm_new_controls(codec, sdp4430_twl6040_dapm_widgets, + ret = snd_soc_dapm_new_controls(dapm, sdp4430_twl6040_dapm_widgets, ARRAY_SIZE(sdp4430_twl6040_dapm_widgets)); if (ret) return ret; /* Set up SDP4430 specific audio path audio_map */ - snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); /* SDP4430 connected pins */ - snd_soc_dapm_enable_pin(codec, "Ext Mic"); - snd_soc_dapm_enable_pin(codec, "Ext Spk"); - snd_soc_dapm_enable_pin(codec, "Headset Mic"); - snd_soc_dapm_enable_pin(codec, "Headset Stereophone"); + snd_soc_dapm_enable_pin(dapm, "Ext Mic"); + snd_soc_dapm_enable_pin(dapm, "Ext Spk"); + snd_soc_dapm_enable_pin(dapm, "Headset Mic"); + snd_soc_dapm_enable_pin(dapm, "Headset Stereophone"); /* TWL6040 not connected pins */ - snd_soc_dapm_nc_pin(codec, "AFML"); - snd_soc_dapm_nc_pin(codec, "AFMR"); + snd_soc_dapm_nc_pin(dapm, "AFML"); + snd_soc_dapm_nc_pin(dapm, "AFMR"); - ret = snd_soc_dapm_sync(codec); + ret = snd_soc_dapm_sync(dapm); return ret; } diff --git a/sound/soc/omap/zoom2.c b/sound/soc/omap/zoom2.c index 718031eeac34..cc5bc523b302 100644 --- a/sound/soc/omap/zoom2.c +++ b/sound/soc/omap/zoom2.c @@ -162,35 +162,36 @@ static const struct snd_soc_dapm_route audio_map[] = { static int zoom2_twl4030_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; int ret; /* Add Zoom2 specific widgets */ - ret = snd_soc_dapm_new_controls(codec, zoom2_twl4030_dapm_widgets, + ret = snd_soc_dapm_new_controls(dapm, zoom2_twl4030_dapm_widgets, ARRAY_SIZE(zoom2_twl4030_dapm_widgets)); if (ret) return ret; /* Set up Zoom2 specific audio path audio_map */ - snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); /* Zoom2 connected pins */ - snd_soc_dapm_enable_pin(codec, "Ext Mic"); - snd_soc_dapm_enable_pin(codec, "Ext Spk"); - snd_soc_dapm_enable_pin(codec, "Headset Mic"); - snd_soc_dapm_enable_pin(codec, "Headset Stereophone"); - snd_soc_dapm_enable_pin(codec, "Aux In"); + snd_soc_dapm_enable_pin(dapm, "Ext Mic"); + snd_soc_dapm_enable_pin(dapm, "Ext Spk"); + snd_soc_dapm_enable_pin(dapm, "Headset Mic"); + snd_soc_dapm_enable_pin(dapm, "Headset Stereophone"); + snd_soc_dapm_enable_pin(dapm, "Aux In"); /* TWL4030 not connected pins */ - snd_soc_dapm_nc_pin(codec, "CARKITMIC"); - snd_soc_dapm_nc_pin(codec, "DIGIMIC0"); - snd_soc_dapm_nc_pin(codec, "DIGIMIC1"); - snd_soc_dapm_nc_pin(codec, "EARPIECE"); - snd_soc_dapm_nc_pin(codec, "PREDRIVEL"); - snd_soc_dapm_nc_pin(codec, "PREDRIVER"); - snd_soc_dapm_nc_pin(codec, "CARKITL"); - snd_soc_dapm_nc_pin(codec, "CARKITR"); - - ret = snd_soc_dapm_sync(codec); + snd_soc_dapm_nc_pin(dapm, "CARKITMIC"); + snd_soc_dapm_nc_pin(dapm, "DIGIMIC0"); + snd_soc_dapm_nc_pin(dapm, "DIGIMIC1"); + snd_soc_dapm_nc_pin(dapm, "EARPIECE"); + snd_soc_dapm_nc_pin(dapm, "PREDRIVEL"); + snd_soc_dapm_nc_pin(dapm, "PREDRIVER"); + snd_soc_dapm_nc_pin(dapm, "CARKITL"); + snd_soc_dapm_nc_pin(dapm, "CARKITR"); + + ret = snd_soc_dapm_sync(dapm); return ret; } diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index f451acd4935b..85956ff8783e 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -48,51 +48,53 @@ static int corgi_spk_func; static void corgi_ext_control(struct snd_soc_codec *codec) { + struct snd_soc_dapm_context *dapm = &codec->dapm; + /* set up jack connection */ switch (corgi_jack_func) { case CORGI_HP: /* set = unmute headphone */ gpio_set_value(CORGI_GPIO_MUTE_L, 1); gpio_set_value(CORGI_GPIO_MUTE_R, 1); - snd_soc_dapm_disable_pin(codec, "Mic Jack"); - snd_soc_dapm_disable_pin(codec, "Line Jack"); - snd_soc_dapm_enable_pin(codec, "Headphone Jack"); - snd_soc_dapm_disable_pin(codec, "Headset Jack"); + snd_soc_dapm_disable_pin(dapm, "Mic Jack"); + snd_soc_dapm_disable_pin(dapm, "Line Jack"); + snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); + snd_soc_dapm_disable_pin(dapm, "Headset Jack"); break; case CORGI_MIC: /* reset = mute headphone */ gpio_set_value(CORGI_GPIO_MUTE_L, 0); gpio_set_value(CORGI_GPIO_MUTE_R, 0); - snd_soc_dapm_enable_pin(codec, "Mic Jack"); - snd_soc_dapm_disable_pin(codec, "Line Jack"); - snd_soc_dapm_disable_pin(codec, "Headphone Jack"); - snd_soc_dapm_disable_pin(codec, "Headset Jack"); + snd_soc_dapm_enable_pin(dapm, "Mic Jack"); + snd_soc_dapm_disable_pin(dapm, "Line Jack"); + snd_soc_dapm_disable_pin(dapm, "Headphone Jack"); + snd_soc_dapm_disable_pin(dapm, "Headset Jack"); break; case CORGI_LINE: gpio_set_value(CORGI_GPIO_MUTE_L, 0); gpio_set_value(CORGI_GPIO_MUTE_R, 0); - snd_soc_dapm_disable_pin(codec, "Mic Jack"); - snd_soc_dapm_enable_pin(codec, "Line Jack"); - snd_soc_dapm_disable_pin(codec, "Headphone Jack"); - snd_soc_dapm_disable_pin(codec, "Headset Jack"); + snd_soc_dapm_disable_pin(dapm, "Mic Jack"); + snd_soc_dapm_enable_pin(dapm, "Line Jack"); + snd_soc_dapm_disable_pin(dapm, "Headphone Jack"); + snd_soc_dapm_disable_pin(dapm, "Headset Jack"); break; case CORGI_HEADSET: gpio_set_value(CORGI_GPIO_MUTE_L, 0); gpio_set_value(CORGI_GPIO_MUTE_R, 1); - snd_soc_dapm_enable_pin(codec, "Mic Jack"); - snd_soc_dapm_disable_pin(codec, "Line Jack"); - snd_soc_dapm_disable_pin(codec, "Headphone Jack"); - snd_soc_dapm_enable_pin(codec, "Headset Jack"); + snd_soc_dapm_enable_pin(dapm, "Mic Jack"); + snd_soc_dapm_disable_pin(dapm, "Line Jack"); + snd_soc_dapm_disable_pin(dapm, "Headphone Jack"); + snd_soc_dapm_enable_pin(dapm, "Headset Jack"); break; } if (corgi_spk_func == CORGI_SPK_ON) - snd_soc_dapm_enable_pin(codec, "Ext Spk"); + snd_soc_dapm_enable_pin(dapm, "Ext Spk"); else - snd_soc_dapm_disable_pin(codec, "Ext Spk"); + snd_soc_dapm_disable_pin(dapm, "Ext Spk"); /* signal a DAPM event */ - snd_soc_dapm_sync(codec); + snd_soc_dapm_sync(dapm); } static int corgi_startup(struct snd_pcm_substream *substream) @@ -279,10 +281,11 @@ static const struct snd_kcontrol_new wm8731_corgi_controls[] = { static int corgi_wm8731_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; int err; - snd_soc_dapm_nc_pin(codec, "LLINEIN"); - snd_soc_dapm_nc_pin(codec, "RLINEIN"); + snd_soc_dapm_nc_pin(dapm, "LLINEIN"); + snd_soc_dapm_nc_pin(dapm, "RLINEIN"); /* Add corgi specific controls */ err = snd_soc_add_controls(codec, wm8731_corgi_controls, @@ -291,13 +294,13 @@ static int corgi_wm8731_init(struct snd_soc_pcm_runtime *rtd) return err; /* Add corgi specific widgets */ - snd_soc_dapm_new_controls(codec, wm8731_dapm_widgets, + snd_soc_dapm_new_controls(dapm, wm8731_dapm_widgets, ARRAY_SIZE(wm8731_dapm_widgets)); /* Set up corgi specific audio path audio_map */ - snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_sync(codec); + snd_soc_dapm_sync(dapm); return 0; } diff --git a/sound/soc/pxa/e740_wm9705.c b/sound/soc/pxa/e740_wm9705.c index c82cedb602fd..38a84b821ff4 100644 --- a/sound/soc/pxa/e740_wm9705.c +++ b/sound/soc/pxa/e740_wm9705.c @@ -92,23 +92,24 @@ static const struct snd_soc_dapm_route audio_map[] = { static int e740_ac97_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; - - snd_soc_dapm_nc_pin(codec, "HPOUTL"); - snd_soc_dapm_nc_pin(codec, "HPOUTR"); - snd_soc_dapm_nc_pin(codec, "PHONE"); - snd_soc_dapm_nc_pin(codec, "LINEINL"); - snd_soc_dapm_nc_pin(codec, "LINEINR"); - snd_soc_dapm_nc_pin(codec, "CDINL"); - snd_soc_dapm_nc_pin(codec, "CDINR"); - snd_soc_dapm_nc_pin(codec, "PCBEEP"); - snd_soc_dapm_nc_pin(codec, "MIC2"); - - snd_soc_dapm_new_controls(codec, e740_dapm_widgets, + struct snd_soc_dapm_context *dapm = &codec->dapm; + + snd_soc_dapm_nc_pin(dapm, "HPOUTL"); + snd_soc_dapm_nc_pin(dapm, "HPOUTR"); + snd_soc_dapm_nc_pin(dapm, "PHONE"); + snd_soc_dapm_nc_pin(dapm, "LINEINL"); + snd_soc_dapm_nc_pin(dapm, "LINEINR"); + snd_soc_dapm_nc_pin(dapm, "CDINL"); + snd_soc_dapm_nc_pin(dapm, "CDINR"); + snd_soc_dapm_nc_pin(dapm, "PCBEEP"); + snd_soc_dapm_nc_pin(dapm, "MIC2"); + + snd_soc_dapm_new_controls(dapm, e740_dapm_widgets, ARRAY_SIZE(e740_dapm_widgets)); - snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_sync(codec); + snd_soc_dapm_sync(dapm); return 0; } diff --git a/sound/soc/pxa/e750_wm9705.c b/sound/soc/pxa/e750_wm9705.c index 4c143803a75e..2bc97e92446b 100644 --- a/sound/soc/pxa/e750_wm9705.c +++ b/sound/soc/pxa/e750_wm9705.c @@ -74,23 +74,24 @@ static const struct snd_soc_dapm_route audio_map[] = { static int e750_ac97_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; - - snd_soc_dapm_nc_pin(codec, "LOUT"); - snd_soc_dapm_nc_pin(codec, "ROUT"); - snd_soc_dapm_nc_pin(codec, "PHONE"); - snd_soc_dapm_nc_pin(codec, "LINEINL"); - snd_soc_dapm_nc_pin(codec, "LINEINR"); - snd_soc_dapm_nc_pin(codec, "CDINL"); - snd_soc_dapm_nc_pin(codec, "CDINR"); - snd_soc_dapm_nc_pin(codec, "PCBEEP"); - snd_soc_dapm_nc_pin(codec, "MIC2"); - - snd_soc_dapm_new_controls(codec, e750_dapm_widgets, + struct snd_soc_dapm_context *dapm = &codec->dapm; + + snd_soc_dapm_nc_pin(dapm, "LOUT"); + snd_soc_dapm_nc_pin(dapm, "ROUT"); + snd_soc_dapm_nc_pin(dapm, "PHONE"); + snd_soc_dapm_nc_pin(dapm, "LINEINL"); + snd_soc_dapm_nc_pin(dapm, "LINEINR"); + snd_soc_dapm_nc_pin(dapm, "CDINL"); + snd_soc_dapm_nc_pin(dapm, "CDINR"); + snd_soc_dapm_nc_pin(dapm, "PCBEEP"); + snd_soc_dapm_nc_pin(dapm, "MIC2"); + + snd_soc_dapm_new_controls(dapm, e750_dapm_widgets, ARRAY_SIZE(e750_dapm_widgets)); - snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_sync(codec); + snd_soc_dapm_sync(dapm); return 0; } diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c index d42e5fe832c5..eac846c7bd9c 100644 --- a/sound/soc/pxa/e800_wm9712.c +++ b/sound/soc/pxa/e800_wm9712.c @@ -75,12 +75,13 @@ static const struct snd_soc_dapm_route audio_map[] = { static int e800_ac97_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; - snd_soc_dapm_new_controls(codec, e800_dapm_widgets, + snd_soc_dapm_new_controls(dapm, e800_dapm_widgets, ARRAY_SIZE(e800_dapm_widgets)); - snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_sync(codec); + snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_sync(dapm); return 0; } diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c index 5ef0526924b9..98cb9906e795 100644 --- a/sound/soc/pxa/magician.c +++ b/sound/soc/pxa/magician.c @@ -44,27 +44,29 @@ static int magician_in_sel = MAGICIAN_MIC; static void magician_ext_control(struct snd_soc_codec *codec) { + struct snd_soc_dapm_context *dapm = &codec->dapm; + if (magician_spk_switch) - snd_soc_dapm_enable_pin(codec, "Speaker"); + snd_soc_dapm_enable_pin(dapm, "Speaker"); else - snd_soc_dapm_disable_pin(codec, "Speaker"); + snd_soc_dapm_disable_pin(dapm, "Speaker"); if (magician_hp_switch) - snd_soc_dapm_enable_pin(codec, "Headphone Jack"); + snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); else - snd_soc_dapm_disable_pin(codec, "Headphone Jack"); + snd_soc_dapm_disable_pin(dapm, "Headphone Jack"); switch (magician_in_sel) { case MAGICIAN_MIC: - snd_soc_dapm_disable_pin(codec, "Headset Mic"); - snd_soc_dapm_enable_pin(codec, "Call Mic"); + snd_soc_dapm_disable_pin(dapm, "Headset Mic"); + snd_soc_dapm_enable_pin(dapm, "Call Mic"); break; case MAGICIAN_MIC_EXT: - snd_soc_dapm_disable_pin(codec, "Call Mic"); - snd_soc_dapm_enable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(dapm, "Call Mic"); + snd_soc_dapm_enable_pin(dapm, "Headset Mic"); break; } - snd_soc_dapm_sync(codec); + snd_soc_dapm_sync(dapm); } static int magician_startup(struct snd_pcm_substream *substream) @@ -399,15 +401,16 @@ static const struct snd_kcontrol_new uda1380_magician_controls[] = { static int magician_uda1380_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; int err; /* NC codec pins */ - snd_soc_dapm_nc_pin(codec, "VOUTLHP"); - snd_soc_dapm_nc_pin(codec, "VOUTRHP"); + snd_soc_dapm_nc_pin(dapm, "VOUTLHP"); + snd_soc_dapm_nc_pin(dapm, "VOUTRHP"); /* FIXME: is anything connected here? */ - snd_soc_dapm_nc_pin(codec, "VINL"); - snd_soc_dapm_nc_pin(codec, "VINR"); + snd_soc_dapm_nc_pin(dapm, "VINL"); + snd_soc_dapm_nc_pin(dapm, "VINR"); /* Add magician specific controls */ err = snd_soc_add_controls(codec, uda1380_magician_controls, @@ -416,13 +419,13 @@ static int magician_uda1380_init(struct snd_soc_pcm_runtime *rtd) return err; /* Add magician specific widgets */ - snd_soc_dapm_new_controls(codec, uda1380_dapm_widgets, + snd_soc_dapm_new_controls(dapm, uda1380_dapm_widgets, ARRAY_SIZE(uda1380_dapm_widgets)); /* Set up magician specific audio path interconnects */ - snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_sync(codec); + snd_soc_dapm_sync(dapm); return 0; } diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c index f284cc54bc80..f7a1e8f09f9a 100644 --- a/sound/soc/pxa/mioa701_wm9713.c +++ b/sound/soc/pxa/mioa701_wm9713.c @@ -130,13 +130,14 @@ static const struct snd_soc_dapm_route audio_map[] = { static int mioa701_wm9713_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; unsigned short reg; /* Add mioa701 specific widgets */ - snd_soc_dapm_new_controls(codec, ARRAY_AND_SIZE(mioa701_dapm_widgets)); + snd_soc_dapm_new_controls(dapm, ARRAY_AND_SIZE(mioa701_dapm_widgets)); /* Set up mioa701 specific audio path audio_mapnects */ - snd_soc_dapm_add_routes(codec, ARRAY_AND_SIZE(audio_map)); + snd_soc_dapm_add_routes(dapm, ARRAY_AND_SIZE(audio_map)); /* Prepare GPIO8 for rear speaker amplifier */ reg = codec->driver->read(codec, AC97_GPIO_CFG); @@ -146,12 +147,12 @@ static int mioa701_wm9713_init(struct snd_soc_pcm_runtime *rtd) reg = codec->driver->read(codec, AC97_3D_CONTROL); codec->driver->write(codec, AC97_3D_CONTROL, reg | 0xc000); - snd_soc_dapm_enable_pin(codec, "Front Speaker"); - snd_soc_dapm_enable_pin(codec, "Rear Speaker"); - snd_soc_dapm_enable_pin(codec, "Front Mic"); - snd_soc_dapm_enable_pin(codec, "GSM Line In"); - snd_soc_dapm_enable_pin(codec, "GSM Line Out"); - snd_soc_dapm_sync(codec); + snd_soc_dapm_enable_pin(dapm, "Front Speaker"); + snd_soc_dapm_enable_pin(dapm, "Rear Speaker"); + snd_soc_dapm_enable_pin(dapm, "Front Mic"); + snd_soc_dapm_enable_pin(dapm, "GSM Line In"); + snd_soc_dapm_enable_pin(dapm, "GSM Line Out"); + snd_soc_dapm_sync(dapm); return 0; } diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c index 13f6d485d571..530064dd06a9 100644 --- a/sound/soc/pxa/palm27x.c +++ b/sound/soc/pxa/palm27x.c @@ -77,37 +77,38 @@ static struct snd_soc_card palm27x_asoc; static int palm27x_ac97_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; int err; /* add palm27x specific widgets */ - err = snd_soc_dapm_new_controls(codec, palm27x_dapm_widgets, + err = snd_soc_dapm_new_controls(dapm, palm27x_dapm_widgets, ARRAY_SIZE(palm27x_dapm_widgets)); if (err) return err; /* set up palm27x specific audio path audio_map */ - err = snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + err = snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); if (err) return err; /* connected pins */ if (machine_is_palmld()) - snd_soc_dapm_enable_pin(codec, "MIC1"); - snd_soc_dapm_enable_pin(codec, "HPOUTL"); - snd_soc_dapm_enable_pin(codec, "HPOUTR"); - snd_soc_dapm_enable_pin(codec, "LOUT2"); - snd_soc_dapm_enable_pin(codec, "ROUT2"); + snd_soc_dapm_enable_pin(dapm, "MIC1"); + snd_soc_dapm_enable_pin(dapm, "HPOUTL"); + snd_soc_dapm_enable_pin(dapm, "HPOUTR"); + snd_soc_dapm_enable_pin(dapm, "LOUT2"); + snd_soc_dapm_enable_pin(dapm, "ROUT2"); /* not connected pins */ - snd_soc_dapm_nc_pin(codec, "OUT3"); - snd_soc_dapm_nc_pin(codec, "MONOOUT"); - snd_soc_dapm_nc_pin(codec, "LINEINL"); - snd_soc_dapm_nc_pin(codec, "LINEINR"); - snd_soc_dapm_nc_pin(codec, "PCBEEP"); - snd_soc_dapm_nc_pin(codec, "PHONE"); - snd_soc_dapm_nc_pin(codec, "MIC2"); - - err = snd_soc_dapm_sync(codec); + snd_soc_dapm_nc_pin(dapm, "OUT3"); + snd_soc_dapm_nc_pin(dapm, "MONOOUT"); + snd_soc_dapm_nc_pin(dapm, "LINEINL"); + snd_soc_dapm_nc_pin(dapm, "LINEINR"); + snd_soc_dapm_nc_pin(dapm, "PCBEEP"); + snd_soc_dapm_nc_pin(dapm, "PHONE"); + snd_soc_dapm_nc_pin(dapm, "MIC2"); + + err = snd_soc_dapm_sync(dapm); if (err) return err; diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index 84edd0385a21..f45ea408852b 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -46,6 +46,8 @@ static int poodle_spk_func; static void poodle_ext_control(struct snd_soc_codec *codec) { + struct snd_soc_dapm_context *dapm = &codec->dapm; + /* set up jack connection */ if (poodle_jack_func == POODLE_HP) { /* set = unmute headphone */ @@ -53,23 +55,23 @@ static void poodle_ext_control(struct snd_soc_codec *codec) POODLE_LOCOMO_GPIO_MUTE_L, 1); locomo_gpio_write(&poodle_locomo_device.dev, POODLE_LOCOMO_GPIO_MUTE_R, 1); - snd_soc_dapm_enable_pin(codec, "Headphone Jack"); + snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); } else { locomo_gpio_write(&poodle_locomo_device.dev, POODLE_LOCOMO_GPIO_MUTE_L, 0); locomo_gpio_write(&poodle_locomo_device.dev, POODLE_LOCOMO_GPIO_MUTE_R, 0); - snd_soc_dapm_disable_pin(codec, "Headphone Jack"); + snd_soc_dapm_disable_pin(dapm, "Headphone Jack"); } /* set the enpoints to their new connetion states */ if (poodle_spk_func == POODLE_SPK_ON) - snd_soc_dapm_enable_pin(codec, "Ext Spk"); + snd_soc_dapm_enable_pin(dapm, "Ext Spk"); else - snd_soc_dapm_disable_pin(codec, "Ext Spk"); + snd_soc_dapm_disable_pin(dapm, "Ext Spk"); /* signal a DAPM event */ - snd_soc_dapm_sync(codec); + snd_soc_dapm_sync(dapm); } static int poodle_startup(struct snd_pcm_substream *substream) @@ -244,11 +246,12 @@ static const struct snd_kcontrol_new wm8731_poodle_controls[] = { static int poodle_wm8731_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; int err; - snd_soc_dapm_nc_pin(codec, "LLINEIN"); - snd_soc_dapm_nc_pin(codec, "RLINEIN"); - snd_soc_dapm_enable_pin(codec, "MICIN"); + snd_soc_dapm_nc_pin(dapm, "LLINEIN"); + snd_soc_dapm_nc_pin(dapm, "RLINEIN"); + snd_soc_dapm_enable_pin(dapm, "MICIN"); /* Add poodle specific controls */ err = snd_soc_add_controls(codec, wm8731_poodle_controls, @@ -257,13 +260,13 @@ static int poodle_wm8731_init(struct snd_soc_pcm_runtime *rtd) return err; /* Add poodle specific widgets */ - snd_soc_dapm_new_controls(codec, wm8731_dapm_widgets, + snd_soc_dapm_new_controls(dapm, wm8731_dapm_widgets, ARRAY_SIZE(wm8731_dapm_widgets)); /* Set up poodle specific audio path audio_map */ - snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_sync(codec); + snd_soc_dapm_sync(dapm); return 0; } diff --git a/sound/soc/pxa/saarb.c b/sound/soc/pxa/saarb.c index d63cb474b4e1..ee06f9982c09 100644 --- a/sound/soc/pxa/saarb.c +++ b/sound/soc/pxa/saarb.c @@ -133,20 +133,21 @@ static struct snd_soc_card snd_soc_card_saarb = { static int saarb_pm860x_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; int ret; - snd_soc_dapm_new_controls(codec, saarb_dapm_widgets, + snd_soc_dapm_new_controls(dapm, saarb_dapm_widgets, ARRAY_SIZE(saarb_dapm_widgets)); - snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); /* connected pins */ - snd_soc_dapm_enable_pin(codec, "Ext Speaker"); - snd_soc_dapm_enable_pin(codec, "Ext Mic 1"); - snd_soc_dapm_enable_pin(codec, "Ext Mic 3"); - snd_soc_dapm_disable_pin(codec, "Headset Mic 2"); - snd_soc_dapm_disable_pin(codec, "Headset Stereophone"); + snd_soc_dapm_enable_pin(dapm, "Ext Speaker"); + snd_soc_dapm_enable_pin(dapm, "Ext Mic 1"); + snd_soc_dapm_enable_pin(dapm, "Ext Mic 3"); + snd_soc_dapm_disable_pin(dapm, "Headset Mic 2"); + snd_soc_dapm_disable_pin(dapm, "Headset Stereophone"); - ret = snd_soc_dapm_sync(codec); + ret = snd_soc_dapm_sync(dapm); if (ret) return ret; diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index 0b30d7de24ec..7e13440cca41 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -46,61 +46,63 @@ static int spitz_spk_func; static void spitz_ext_control(struct snd_soc_codec *codec) { + struct snd_soc_dapm_context *dapm = &codec->dapm; + if (spitz_spk_func == SPITZ_SPK_ON) - snd_soc_dapm_enable_pin(codec, "Ext Spk"); + snd_soc_dapm_enable_pin(dapm, "Ext Spk"); else - snd_soc_dapm_disable_pin(codec, "Ext Spk"); + snd_soc_dapm_disable_pin(dapm, "Ext Spk"); /* set up jack connection */ switch (spitz_jack_func) { case SPITZ_HP: /* enable and unmute hp jack, disable mic bias */ - snd_soc_dapm_disable_pin(codec, "Headset Jack"); - snd_soc_dapm_disable_pin(codec, "Mic Jack"); - snd_soc_dapm_disable_pin(codec, "Line Jack"); - snd_soc_dapm_enable_pin(codec, "Headphone Jack"); + snd_soc_dapm_disable_pin(dapm, "Headset Jack"); + snd_soc_dapm_disable_pin(dapm, "Mic Jack"); + snd_soc_dapm_disable_pin(dapm, "Line Jack"); + snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); gpio_set_value(SPITZ_GPIO_MUTE_L, 1); gpio_set_value(SPITZ_GPIO_MUTE_R, 1); break; case SPITZ_MIC: /* enable mic jack and bias, mute hp */ - snd_soc_dapm_disable_pin(codec, "Headphone Jack"); - snd_soc_dapm_disable_pin(codec, "Headset Jack"); - snd_soc_dapm_disable_pin(codec, "Line Jack"); - snd_soc_dapm_enable_pin(codec, "Mic Jack"); + snd_soc_dapm_disable_pin(dapm, "Headphone Jack"); + snd_soc_dapm_disable_pin(dapm, "Headset Jack"); + snd_soc_dapm_disable_pin(dapm, "Line Jack"); + snd_soc_dapm_enable_pin(dapm, "Mic Jack"); gpio_set_value(SPITZ_GPIO_MUTE_L, 0); gpio_set_value(SPITZ_GPIO_MUTE_R, 0); break; case SPITZ_LINE: /* enable line jack, disable mic bias and mute hp */ - snd_soc_dapm_disable_pin(codec, "Headphone Jack"); - snd_soc_dapm_disable_pin(codec, "Headset Jack"); - snd_soc_dapm_disable_pin(codec, "Mic Jack"); - snd_soc_dapm_enable_pin(codec, "Line Jack"); + snd_soc_dapm_disable_pin(dapm, "Headphone Jack"); + snd_soc_dapm_disable_pin(dapm, "Headset Jack"); + snd_soc_dapm_disable_pin(dapm, "Mic Jack"); + snd_soc_dapm_enable_pin(dapm, "Line Jack"); gpio_set_value(SPITZ_GPIO_MUTE_L, 0); gpio_set_value(SPITZ_GPIO_MUTE_R, 0); break; case SPITZ_HEADSET: /* enable and unmute headset jack enable mic bias, mute L hp */ - snd_soc_dapm_disable_pin(codec, "Headphone Jack"); - snd_soc_dapm_enable_pin(codec, "Mic Jack"); - snd_soc_dapm_disable_pin(codec, "Line Jack"); - snd_soc_dapm_enable_pin(codec, "Headset Jack"); + snd_soc_dapm_disable_pin(dapm, "Headphone Jack"); + snd_soc_dapm_enable_pin(dapm, "Mic Jack"); + snd_soc_dapm_disable_pin(dapm, "Line Jack"); + snd_soc_dapm_enable_pin(dapm, "Headset Jack"); gpio_set_value(SPITZ_GPIO_MUTE_L, 0); gpio_set_value(SPITZ_GPIO_MUTE_R, 1); break; case SPITZ_HP_OFF: /* jack removed, everything off */ - snd_soc_dapm_disable_pin(codec, "Headphone Jack"); - snd_soc_dapm_disable_pin(codec, "Headset Jack"); - snd_soc_dapm_disable_pin(codec, "Mic Jack"); - snd_soc_dapm_disable_pin(codec, "Line Jack"); + snd_soc_dapm_disable_pin(dapm, "Headphone Jack"); + snd_soc_dapm_disable_pin(dapm, "Headset Jack"); + snd_soc_dapm_disable_pin(dapm, "Mic Jack"); + snd_soc_dapm_disable_pin(dapm, "Line Jack"); gpio_set_value(SPITZ_GPIO_MUTE_L, 0); gpio_set_value(SPITZ_GPIO_MUTE_R, 0); break; } - snd_soc_dapm_sync(codec); + snd_soc_dapm_sync(dapm); } static int spitz_startup(struct snd_pcm_substream *substream) @@ -281,16 +283,17 @@ static const struct snd_kcontrol_new wm8750_spitz_controls[] = { static int spitz_wm8750_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; int err; /* NC codec pins */ - snd_soc_dapm_nc_pin(codec, "RINPUT1"); - snd_soc_dapm_nc_pin(codec, "LINPUT2"); - snd_soc_dapm_nc_pin(codec, "RINPUT2"); - snd_soc_dapm_nc_pin(codec, "LINPUT3"); - snd_soc_dapm_nc_pin(codec, "RINPUT3"); - snd_soc_dapm_nc_pin(codec, "OUT3"); - snd_soc_dapm_nc_pin(codec, "MONO1"); + snd_soc_dapm_nc_pin(dapm, "RINPUT1"); + snd_soc_dapm_nc_pin(dapm, "LINPUT2"); + snd_soc_dapm_nc_pin(dapm, "RINPUT2"); + snd_soc_dapm_nc_pin(dapm, "LINPUT3"); + snd_soc_dapm_nc_pin(dapm, "RINPUT3"); + snd_soc_dapm_nc_pin(dapm, "OUT3"); + snd_soc_dapm_nc_pin(dapm, "MONO1"); /* Add spitz specific controls */ err = snd_soc_add_controls(codec, wm8750_spitz_controls, @@ -299,13 +302,13 @@ static int spitz_wm8750_init(struct snd_soc_pcm_runtime *rtd) return err; /* Add spitz specific widgets */ - snd_soc_dapm_new_controls(codec, wm8750_dapm_widgets, + snd_soc_dapm_new_controls(dapm, wm8750_dapm_widgets, ARRAY_SIZE(wm8750_dapm_widgets)); /* Set up spitz specific audio paths */ - snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_sync(codec); + snd_soc_dapm_sync(dapm); return 0; } diff --git a/sound/soc/pxa/tavorevb3.c b/sound/soc/pxa/tavorevb3.c index 248c283fc4df..18cbe0e7c223 100644 --- a/sound/soc/pxa/tavorevb3.c +++ b/sound/soc/pxa/tavorevb3.c @@ -133,20 +133,21 @@ static struct snd_soc_card snd_soc_card_evb3 = { static int evb3_pm860x_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; int ret; - snd_soc_dapm_new_controls(codec, evb3_dapm_widgets, + snd_soc_dapm_new_controls(dapm, evb3_dapm_widgets, ARRAY_SIZE(evb3_dapm_widgets)); - snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); /* connected pins */ - snd_soc_dapm_enable_pin(codec, "Ext Speaker"); - snd_soc_dapm_enable_pin(codec, "Ext Mic 1"); - snd_soc_dapm_enable_pin(codec, "Ext Mic 3"); - snd_soc_dapm_disable_pin(codec, "Headset Mic 2"); - snd_soc_dapm_disable_pin(codec, "Headset Stereophone"); + snd_soc_dapm_enable_pin(dapm, "Ext Speaker"); + snd_soc_dapm_enable_pin(dapm, "Ext Mic 1"); + snd_soc_dapm_enable_pin(dapm, "Ext Mic 3"); + snd_soc_dapm_disable_pin(dapm, "Headset Mic 2"); + snd_soc_dapm_disable_pin(dapm, "Headset Stereophone"); - ret = snd_soc_dapm_sync(codec); + ret = snd_soc_dapm_sync(dapm); if (ret) return ret; diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index 7b983f935454..7f81f82ba788 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -49,31 +49,33 @@ static int tosa_spk_func; static void tosa_ext_control(struct snd_soc_codec *codec) { + struct snd_soc_dapm_context *dapm = &codec->dapm; + /* set up jack connection */ switch (tosa_jack_func) { case TOSA_HP: - snd_soc_dapm_disable_pin(codec, "Mic (Internal)"); - snd_soc_dapm_enable_pin(codec, "Headphone Jack"); - snd_soc_dapm_disable_pin(codec, "Headset Jack"); + snd_soc_dapm_disable_pin(dapm, "Mic (Internal)"); + snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); + snd_soc_dapm_disable_pin(dapm, "Headset Jack"); break; case TOSA_MIC_INT: - snd_soc_dapm_enable_pin(codec, "Mic (Internal)"); - snd_soc_dapm_disable_pin(codec, "Headphone Jack"); - snd_soc_dapm_disable_pin(codec, "Headset Jack"); + snd_soc_dapm_enable_pin(dapm, "Mic (Internal)"); + snd_soc_dapm_disable_pin(dapm, "Headphone Jack"); + snd_soc_dapm_disable_pin(dapm, "Headset Jack"); break; case TOSA_HEADSET: - snd_soc_dapm_disable_pin(codec, "Mic (Internal)"); - snd_soc_dapm_disable_pin(codec, "Headphone Jack"); - snd_soc_dapm_enable_pin(codec, "Headset Jack"); + snd_soc_dapm_disable_pin(dapm, "Mic (Internal)"); + snd_soc_dapm_disable_pin(dapm, "Headphone Jack"); + snd_soc_dapm_enable_pin(dapm, "Headset Jack"); break; } if (tosa_spk_func == TOSA_SPK_ON) - snd_soc_dapm_enable_pin(codec, "Speaker"); + snd_soc_dapm_enable_pin(dapm, "Speaker"); else - snd_soc_dapm_disable_pin(codec, "Speaker"); + snd_soc_dapm_disable_pin(dapm, "Speaker"); - snd_soc_dapm_sync(codec); + snd_soc_dapm_sync(dapm); } static int tosa_startup(struct snd_pcm_substream *substream) @@ -191,10 +193,11 @@ static const struct snd_kcontrol_new tosa_controls[] = { static int tosa_ac97_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; int err; - snd_soc_dapm_nc_pin(codec, "OUT3"); - snd_soc_dapm_nc_pin(codec, "MONOOUT"); + snd_soc_dapm_nc_pin(dapm, "OUT3"); + snd_soc_dapm_nc_pin(dapm, "MONOOUT"); /* add tosa specific controls */ err = snd_soc_add_controls(codec, tosa_controls, @@ -203,13 +206,13 @@ static int tosa_ac97_init(struct snd_soc_pcm_runtime *rtd) return err; /* add tosa specific widgets */ - snd_soc_dapm_new_controls(codec, tosa_dapm_widgets, + snd_soc_dapm_new_controls(dapm, tosa_dapm_widgets, ARRAY_SIZE(tosa_dapm_widgets)); /* set up tosa specific audio path audio_map */ - snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_sync(codec); + snd_soc_dapm_sync(dapm); return 0; } diff --git a/sound/soc/pxa/z2.c b/sound/soc/pxa/z2.c index 4cc841b44182..cacbcd4a55eb 100644 --- a/sound/soc/pxa/z2.c +++ b/sound/soc/pxa/z2.c @@ -140,22 +140,23 @@ static const struct snd_soc_dapm_route audio_map[] = { static int z2_wm8750_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; int ret; /* NC codec pins */ - snd_soc_dapm_disable_pin(codec, "LINPUT3"); - snd_soc_dapm_disable_pin(codec, "RINPUT3"); - snd_soc_dapm_disable_pin(codec, "OUT3"); - snd_soc_dapm_disable_pin(codec, "MONO"); + snd_soc_dapm_disable_pin(dapm, "LINPUT3"); + snd_soc_dapm_disable_pin(dapm, "RINPUT3"); + snd_soc_dapm_disable_pin(dapm, "OUT3"); + snd_soc_dapm_disable_pin(dapm, "MONO"); /* Add z2 specific widgets */ - snd_soc_dapm_new_controls(codec, wm8750_dapm_widgets, + snd_soc_dapm_new_controls(dapm, wm8750_dapm_widgets, ARRAY_SIZE(wm8750_dapm_widgets)); /* Set up z2 specific audio paths */ - snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - ret = snd_soc_dapm_sync(codec); + ret = snd_soc_dapm_sync(dapm); if (ret) goto err; diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c index d27e05af7759..c74eac30ebff 100644 --- a/sound/soc/pxa/zylonite.c +++ b/sound/soc/pxa/zylonite.c @@ -73,21 +73,22 @@ static const struct snd_soc_dapm_route audio_map[] = { static int zylonite_wm9713_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; if (clk_pout) snd_soc_dai_set_pll(rtd->codec_dai, 0, 0, clk_get_rate(pout), 0); - snd_soc_dapm_new_controls(codec, zylonite_dapm_widgets, + snd_soc_dapm_new_controls(dapm, zylonite_dapm_widgets, ARRAY_SIZE(zylonite_dapm_widgets)); - snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); /* Static setup for now */ - snd_soc_dapm_enable_pin(codec, "Headphone"); - snd_soc_dapm_enable_pin(codec, "Headset Earpiece"); + snd_soc_dapm_enable_pin(dapm, "Headphone"); + snd_soc_dapm_enable_pin(dapm, "Headset Earpiece"); - snd_soc_dapm_sync(codec); + snd_soc_dapm_sync(dapm); return 0; } diff --git a/sound/soc/s3c24xx/aquila_wm8994.c b/sound/soc/s3c24xx/aquila_wm8994.c index 235d1973f7d0..33bebdae08a7 100644 --- a/sound/soc/s3c24xx/aquila_wm8994.c +++ b/sound/soc/s3c24xx/aquila_wm8994.c @@ -93,27 +93,28 @@ static const struct snd_soc_dapm_route aquila_dapm_routes[] = { static int aquila_wm8994_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; int ret; /* add aquila specific widgets */ - snd_soc_dapm_new_controls(codec, aquila_dapm_widgets, + snd_soc_dapm_new_controls(dapm, aquila_dapm_widgets, ARRAY_SIZE(aquila_dapm_widgets)); /* set up aquila specific audio routes */ - snd_soc_dapm_add_routes(codec, aquila_dapm_routes, + snd_soc_dapm_add_routes(dapm, aquila_dapm_routes, ARRAY_SIZE(aquila_dapm_routes)); /* set endpoints to not connected */ - snd_soc_dapm_nc_pin(codec, "IN2LP:VXRN"); - snd_soc_dapm_nc_pin(codec, "IN2RP:VXRP"); - snd_soc_dapm_nc_pin(codec, "LINEOUT1N"); - snd_soc_dapm_nc_pin(codec, "LINEOUT1P"); - snd_soc_dapm_nc_pin(codec, "LINEOUT2N"); - snd_soc_dapm_nc_pin(codec, "LINEOUT2P"); - snd_soc_dapm_nc_pin(codec, "SPKOUTRN"); - snd_soc_dapm_nc_pin(codec, "SPKOUTRP"); - - snd_soc_dapm_sync(codec); + snd_soc_dapm_nc_pin(dapm, "IN2LP:VXRN"); + snd_soc_dapm_nc_pin(dapm, "IN2RP:VXRP"); + snd_soc_dapm_nc_pin(dapm, "LINEOUT1N"); + snd_soc_dapm_nc_pin(dapm, "LINEOUT1P"); + snd_soc_dapm_nc_pin(dapm, "LINEOUT2N"); + snd_soc_dapm_nc_pin(dapm, "LINEOUT2P"); + snd_soc_dapm_nc_pin(dapm, "SPKOUTRN"); + snd_soc_dapm_nc_pin(dapm, "SPKOUTRP"); + + snd_soc_dapm_sync(dapm); /* Headset jack detection */ ret = snd_soc_jack_new(&aquila, "Headset Jack", diff --git a/sound/soc/s3c24xx/goni_wm8994.c b/sound/soc/s3c24xx/goni_wm8994.c index 694f702cc8e2..052729c6540d 100644 --- a/sound/soc/s3c24xx/goni_wm8994.c +++ b/sound/soc/s3c24xx/goni_wm8994.c @@ -97,25 +97,26 @@ static const struct snd_soc_dapm_route goni_dapm_routes[] = { static int goni_wm8994_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; int ret; /* add goni specific widgets */ - snd_soc_dapm_new_controls(codec, goni_dapm_widgets, + snd_soc_dapm_new_controls(dapm, goni_dapm_widgets, ARRAY_SIZE(goni_dapm_widgets)); /* set up goni specific audio routes */ - snd_soc_dapm_add_routes(codec, goni_dapm_routes, + snd_soc_dapm_add_routes(dapm, goni_dapm_routes, ARRAY_SIZE(goni_dapm_routes)); /* set endpoints to not connected */ - snd_soc_dapm_nc_pin(codec, "IN2LP:VXRN"); - snd_soc_dapm_nc_pin(codec, "IN2RP:VXRP"); - snd_soc_dapm_nc_pin(codec, "LINEOUT1N"); - snd_soc_dapm_nc_pin(codec, "LINEOUT1P"); - snd_soc_dapm_nc_pin(codec, "LINEOUT2N"); - snd_soc_dapm_nc_pin(codec, "LINEOUT2P"); - - snd_soc_dapm_sync(codec); + snd_soc_dapm_nc_pin(dapm, "IN2LP:VXRN"); + snd_soc_dapm_nc_pin(dapm, "IN2RP:VXRP"); + snd_soc_dapm_nc_pin(dapm, "LINEOUT1N"); + snd_soc_dapm_nc_pin(dapm, "LINEOUT1P"); + snd_soc_dapm_nc_pin(dapm, "LINEOUT2N"); + snd_soc_dapm_nc_pin(dapm, "LINEOUT2P"); + + snd_soc_dapm_sync(dapm); /* Headset jack detection */ ret = snd_soc_jack_new(&goni, "Headset Jack", diff --git a/sound/soc/s3c24xx/jive_wm8750.c b/sound/soc/s3c24xx/jive_wm8750.c index 49605cd83947..e3599e283568 100644 --- a/sound/soc/s3c24xx/jive_wm8750.c +++ b/sound/soc/s3c24xx/jive_wm8750.c @@ -111,18 +111,19 @@ static struct snd_soc_ops jive_ops = { static int jive_wm8750_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; int err; /* These endpoints are not being used. */ - snd_soc_dapm_nc_pin(codec, "LINPUT2"); - snd_soc_dapm_nc_pin(codec, "RINPUT2"); - snd_soc_dapm_nc_pin(codec, "LINPUT3"); - snd_soc_dapm_nc_pin(codec, "RINPUT3"); - snd_soc_dapm_nc_pin(codec, "OUT3"); - snd_soc_dapm_nc_pin(codec, "MONO"); + snd_soc_dapm_nc_pin(dapm, "LINPUT2"); + snd_soc_dapm_nc_pin(dapm, "RINPUT2"); + snd_soc_dapm_nc_pin(dapm, "LINPUT3"); + snd_soc_dapm_nc_pin(dapm, "RINPUT3"); + snd_soc_dapm_nc_pin(dapm, "OUT3"); + snd_soc_dapm_nc_pin(dapm, "MONO"); /* Add jive specific widgets */ - err = snd_soc_dapm_new_controls(codec, wm8750_dapm_widgets, + err = snd_soc_dapm_new_controls(dapm, wm8750_dapm_widgets, ARRAY_SIZE(wm8750_dapm_widgets)); if (err) { printk(KERN_ERR "%s: failed to add widgets (%d)\n", @@ -130,8 +131,8 @@ static int jive_wm8750_init(struct snd_soc_pcm_runtime *rtd) return err; } - snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_sync(codec); + snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_sync(dapm); return 0; } diff --git a/sound/soc/s3c24xx/neo1973_gta02_wm8753.c b/sound/soc/s3c24xx/neo1973_gta02_wm8753.c index e97bdf150a03..c3f63ef8ab12 100644 --- a/sound/soc/s3c24xx/neo1973_gta02_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_gta02_wm8753.c @@ -333,16 +333,17 @@ static const struct snd_kcontrol_new wm8753_neo1973_gta02_controls[] = { static int neo1973_gta02_wm8753_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; int err; /* set up NC codec pins */ - snd_soc_dapm_nc_pin(codec, "OUT3"); - snd_soc_dapm_nc_pin(codec, "OUT4"); - snd_soc_dapm_nc_pin(codec, "LINE1"); - snd_soc_dapm_nc_pin(codec, "LINE2"); + snd_soc_dapm_nc_pin(dapm, "OUT3"); + snd_soc_dapm_nc_pin(dapm, "OUT4"); + snd_soc_dapm_nc_pin(dapm, "LINE1"); + snd_soc_dapm_nc_pin(dapm, "LINE2"); /* Add neo1973 gta02 specific widgets */ - snd_soc_dapm_new_controls(codec, wm8753_dapm_widgets, + snd_soc_dapm_new_controls(dapm, wm8753_dapm_widgets, ARRAY_SIZE(wm8753_dapm_widgets)); /* add neo1973 gta02 specific controls */ @@ -353,25 +354,25 @@ static int neo1973_gta02_wm8753_init(struct snd_soc_pcm_runtime *rtd) return err; /* set up neo1973 gta02 specific audio path audio_map */ - snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); /* set endpoints to default off mode */ - snd_soc_dapm_disable_pin(codec, "Stereo Out"); - snd_soc_dapm_disable_pin(codec, "GSM Line Out"); - snd_soc_dapm_disable_pin(codec, "GSM Line In"); - snd_soc_dapm_disable_pin(codec, "Headset Mic"); - snd_soc_dapm_disable_pin(codec, "Handset Mic"); - snd_soc_dapm_disable_pin(codec, "Handset Spk"); + snd_soc_dapm_disable_pin(dapm, "Stereo 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, "Handset Mic"); + snd_soc_dapm_disable_pin(dapm, "Handset Spk"); /* allow audio paths from the GSM modem to run during suspend */ - snd_soc_dapm_ignore_suspend(codec, "Stereo Out"); - snd_soc_dapm_ignore_suspend(codec, "GSM Line Out"); - snd_soc_dapm_ignore_suspend(codec, "GSM Line In"); - snd_soc_dapm_ignore_suspend(codec, "Headset Mic"); - snd_soc_dapm_ignore_suspend(codec, "Handset Mic"); - snd_soc_dapm_ignore_suspend(codec, "Handset Spk"); - - snd_soc_dapm_sync(codec); + snd_soc_dapm_ignore_suspend(dapm, "Stereo Out"); + snd_soc_dapm_ignore_suspend(dapm, "GSM Line Out"); + snd_soc_dapm_ignore_suspend(dapm, "GSM Line In"); + snd_soc_dapm_ignore_suspend(dapm, "Headset Mic"); + snd_soc_dapm_ignore_suspend(dapm, "Handset Mic"); + snd_soc_dapm_ignore_suspend(dapm, "Handset Spk"); + + snd_soc_dapm_sync(dapm); return 0; } diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c index f4f2ee731f01..e94ffe01a4a5 100644 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_wm8753.c @@ -237,81 +237,83 @@ static int neo1973_get_scenario(struct snd_kcontrol *kcontrol, static int set_scenario_endpoints(struct snd_soc_codec *codec, int scenario) { + struct snd_soc_dapm_context *dapm = &codec->dapm; + pr_debug("Entered %s\n", __func__); switch (neo1973_scenario) { case NEO_AUDIO_OFF: - snd_soc_dapm_disable_pin(codec, "Audio Out"); - snd_soc_dapm_disable_pin(codec, "GSM Line Out"); - snd_soc_dapm_disable_pin(codec, "GSM Line In"); - snd_soc_dapm_disable_pin(codec, "Headset Mic"); - snd_soc_dapm_disable_pin(codec, "Call Mic"); + 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"); break; case NEO_GSM_CALL_AUDIO_HANDSET: - snd_soc_dapm_enable_pin(codec, "Audio Out"); - snd_soc_dapm_enable_pin(codec, "GSM Line Out"); - snd_soc_dapm_enable_pin(codec, "GSM Line In"); - snd_soc_dapm_disable_pin(codec, "Headset Mic"); - snd_soc_dapm_enable_pin(codec, "Call Mic"); + snd_soc_dapm_enable_pin(dapm, "Audio Out"); + snd_soc_dapm_enable_pin(dapm, "GSM Line Out"); + snd_soc_dapm_enable_pin(dapm, "GSM Line In"); + snd_soc_dapm_disable_pin(dapm, "Headset Mic"); + snd_soc_dapm_enable_pin(dapm, "Call Mic"); break; case NEO_GSM_CALL_AUDIO_HEADSET: - snd_soc_dapm_enable_pin(codec, "Audio Out"); - snd_soc_dapm_enable_pin(codec, "GSM Line Out"); - snd_soc_dapm_enable_pin(codec, "GSM Line In"); - snd_soc_dapm_enable_pin(codec, "Headset Mic"); - snd_soc_dapm_disable_pin(codec, "Call Mic"); + snd_soc_dapm_enable_pin(dapm, "Audio Out"); + snd_soc_dapm_enable_pin(dapm, "GSM Line Out"); + snd_soc_dapm_enable_pin(dapm, "GSM Line In"); + snd_soc_dapm_enable_pin(dapm, "Headset Mic"); + snd_soc_dapm_disable_pin(dapm, "Call Mic"); break; case NEO_GSM_CALL_AUDIO_BLUETOOTH: - snd_soc_dapm_disable_pin(codec, "Audio Out"); - snd_soc_dapm_enable_pin(codec, "GSM Line Out"); - snd_soc_dapm_enable_pin(codec, "GSM Line In"); - snd_soc_dapm_disable_pin(codec, "Headset Mic"); - snd_soc_dapm_disable_pin(codec, "Call Mic"); + snd_soc_dapm_disable_pin(dapm, "Audio Out"); + snd_soc_dapm_enable_pin(dapm, "GSM Line Out"); + snd_soc_dapm_enable_pin(dapm, "GSM Line In"); + snd_soc_dapm_disable_pin(dapm, "Headset Mic"); + snd_soc_dapm_disable_pin(dapm, "Call Mic"); break; case NEO_STEREO_TO_SPEAKERS: - snd_soc_dapm_enable_pin(codec, "Audio Out"); - snd_soc_dapm_disable_pin(codec, "GSM Line Out"); - snd_soc_dapm_disable_pin(codec, "GSM Line In"); - snd_soc_dapm_disable_pin(codec, "Headset Mic"); - snd_soc_dapm_disable_pin(codec, "Call Mic"); + snd_soc_dapm_enable_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"); break; case NEO_STEREO_TO_HEADPHONES: - snd_soc_dapm_enable_pin(codec, "Audio Out"); - snd_soc_dapm_disable_pin(codec, "GSM Line Out"); - snd_soc_dapm_disable_pin(codec, "GSM Line In"); - snd_soc_dapm_disable_pin(codec, "Headset Mic"); - snd_soc_dapm_disable_pin(codec, "Call Mic"); + snd_soc_dapm_enable_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"); break; case NEO_CAPTURE_HANDSET: - snd_soc_dapm_disable_pin(codec, "Audio Out"); - snd_soc_dapm_disable_pin(codec, "GSM Line Out"); - snd_soc_dapm_disable_pin(codec, "GSM Line In"); - snd_soc_dapm_disable_pin(codec, "Headset Mic"); - snd_soc_dapm_enable_pin(codec, "Call Mic"); + 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_enable_pin(dapm, "Call Mic"); break; case NEO_CAPTURE_HEADSET: - snd_soc_dapm_disable_pin(codec, "Audio Out"); - snd_soc_dapm_disable_pin(codec, "GSM Line Out"); - snd_soc_dapm_disable_pin(codec, "GSM Line In"); - snd_soc_dapm_enable_pin(codec, "Headset Mic"); - snd_soc_dapm_disable_pin(codec, "Call Mic"); + 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_enable_pin(dapm, "Headset Mic"); + snd_soc_dapm_disable_pin(dapm, "Call Mic"); break; case NEO_CAPTURE_BLUETOOTH: - snd_soc_dapm_disable_pin(codec, "Audio Out"); - snd_soc_dapm_disable_pin(codec, "GSM Line Out"); - snd_soc_dapm_disable_pin(codec, "GSM Line In"); - snd_soc_dapm_disable_pin(codec, "Headset Mic"); - snd_soc_dapm_disable_pin(codec, "Call Mic"); + 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"); break; default: - snd_soc_dapm_disable_pin(codec, "Audio Out"); - snd_soc_dapm_disable_pin(codec, "GSM Line Out"); - snd_soc_dapm_disable_pin(codec, "GSM Line In"); - snd_soc_dapm_disable_pin(codec, "Headset Mic"); - snd_soc_dapm_disable_pin(codec, "Call Mic"); + 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"); } - snd_soc_dapm_sync(codec); + snd_soc_dapm_sync(dapm); return 0; } @@ -502,20 +504,21 @@ static const struct snd_kcontrol_new wm8753_neo1973_controls[] = { static int neo1973_wm8753_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; int err; pr_debug("Entered %s\n", __func__); /* set up NC codec pins */ - snd_soc_dapm_nc_pin(codec, "LOUT2"); - snd_soc_dapm_nc_pin(codec, "ROUT2"); - snd_soc_dapm_nc_pin(codec, "OUT3"); - snd_soc_dapm_nc_pin(codec, "OUT4"); - snd_soc_dapm_nc_pin(codec, "LINE1"); - snd_soc_dapm_nc_pin(codec, "LINE2"); + snd_soc_dapm_nc_pin(dapm, "LOUT2"); + snd_soc_dapm_nc_pin(dapm, "ROUT2"); + snd_soc_dapm_nc_pin(dapm, "OUT3"); + snd_soc_dapm_nc_pin(dapm, "OUT4"); + snd_soc_dapm_nc_pin(dapm, "LINE1"); + snd_soc_dapm_nc_pin(dapm, "LINE2"); /* Add neo1973 specific widgets */ - snd_soc_dapm_new_controls(codec, wm8753_dapm_widgets, + snd_soc_dapm_new_controls(dapm, wm8753_dapm_widgets, ARRAY_SIZE(wm8753_dapm_widgets)); /* set endpoints to default mode */ @@ -528,10 +531,10 @@ static int neo1973_wm8753_init(struct snd_soc_pcm_runtime *rtd) return err; /* set up neo1973 specific audio routes */ - err = snd_soc_dapm_add_routes(codec, dapm_routes, + err = snd_soc_dapm_add_routes(dapm, dapm_routes, ARRAY_SIZE(dapm_routes)); - snd_soc_dapm_sync(codec); + snd_soc_dapm_sync(dapm); return 0; } diff --git a/sound/soc/s3c24xx/rx1950_uda1380.c b/sound/soc/s3c24xx/rx1950_uda1380.c index ffd5cf2fb0a9..105d177fa427 100644 --- a/sound/soc/s3c24xx/rx1950_uda1380.c +++ b/sound/soc/s3c24xx/rx1950_uda1380.c @@ -232,26 +232,27 @@ static int rx1950_hw_params(struct snd_pcm_substream *substream, static int rx1950_uda1380_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; int err; /* Add rx1950 specific widgets */ - err = snd_soc_dapm_new_controls(codec, uda1380_dapm_widgets, + err = snd_soc_dapm_new_controls(dapm, uda1380_dapm_widgets, ARRAY_SIZE(uda1380_dapm_widgets)); if (err) return err; /* Set up rx1950 specific audio path audio_mapnects */ - err = snd_soc_dapm_add_routes(codec, audio_map, + err = snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); if (err) return err; - snd_soc_dapm_enable_pin(codec, "Headphone Jack"); - snd_soc_dapm_enable_pin(codec, "Speaker"); + snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); + snd_soc_dapm_enable_pin(dapm, "Speaker"); - snd_soc_dapm_sync(codec); + snd_soc_dapm_sync(dapm); snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE, &hp_jack); diff --git a/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c b/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c index f88453735ae2..05c793705d90 100644 --- a/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c +++ b/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c @@ -76,19 +76,20 @@ static const struct snd_soc_dapm_route base_map[] = { static int simtec_hermes_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; - snd_soc_dapm_new_controls(codec, dapm_widgets, + snd_soc_dapm_new_controls(dapm, dapm_widgets, ARRAY_SIZE(dapm_widgets)); - snd_soc_dapm_add_routes(codec, base_map, ARRAY_SIZE(base_map)); + snd_soc_dapm_add_routes(dapm, base_map, ARRAY_SIZE(base_map)); - snd_soc_dapm_enable_pin(codec, "Headphone Jack"); - snd_soc_dapm_enable_pin(codec, "Line In"); - snd_soc_dapm_enable_pin(codec, "Line Out"); - snd_soc_dapm_enable_pin(codec, "Mic Jack"); + snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); + snd_soc_dapm_enable_pin(dapm, "Line In"); + snd_soc_dapm_enable_pin(dapm, "Line Out"); + snd_soc_dapm_enable_pin(dapm, "Mic Jack"); simtec_audio_init(rtd); - snd_soc_dapm_sync(codec); + snd_soc_dapm_sync(dapm); return 0; } diff --git a/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c b/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c index c0967593510d..653dc7592e81 100644 --- a/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c +++ b/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c @@ -65,19 +65,20 @@ static const struct snd_soc_dapm_route base_map[] = { static int simtec_tlv320aic23_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; - snd_soc_dapm_new_controls(codec, dapm_widgets, + snd_soc_dapm_new_controls(dapm, dapm_widgets, ARRAY_SIZE(dapm_widgets)); - snd_soc_dapm_add_routes(codec, base_map, ARRAY_SIZE(base_map)); + snd_soc_dapm_add_routes(dapm, base_map, ARRAY_SIZE(base_map)); - snd_soc_dapm_enable_pin(codec, "Headphone Jack"); - snd_soc_dapm_enable_pin(codec, "Line In"); - snd_soc_dapm_enable_pin(codec, "Line Out"); - snd_soc_dapm_enable_pin(codec, "Mic Jack"); + snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); + snd_soc_dapm_enable_pin(dapm, "Line In"); + snd_soc_dapm_enable_pin(dapm, "Line Out"); + snd_soc_dapm_enable_pin(dapm, "Mic Jack"); simtec_audio_init(rtd); - snd_soc_dapm_sync(codec); + snd_soc_dapm_sync(dapm); return 0; } diff --git a/sound/soc/s3c24xx/smartq_wm8987.c b/sound/soc/s3c24xx/smartq_wm8987.c index dd20ca7f4681..1f6da1e27b1e 100644 --- a/sound/soc/s3c24xx/smartq_wm8987.c +++ b/sound/soc/s3c24xx/smartq_wm8987.c @@ -158,10 +158,11 @@ static const struct snd_soc_dapm_route audio_map[] = { static int smartq_wm8987_init(struct snd_soc_codec *codec) { + struct snd_soc_dapm_context *dapm = &codec->dapm; int err = 0; /* Add SmartQ specific widgets */ - snd_soc_dapm_new_controls(codec, wm8987_dapm_widgets, + snd_soc_dapm_new_controls(dapm, wm8987_dapm_widgets, ARRAY_SIZE(wm8987_dapm_widgets)); /* add SmartQ specific controls */ @@ -172,20 +173,20 @@ static int smartq_wm8987_init(struct snd_soc_codec *codec) return err; /* setup SmartQ specific audio path */ - snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); /* set endpoints to not connected */ - snd_soc_dapm_nc_pin(codec, "LINPUT1"); - snd_soc_dapm_nc_pin(codec, "RINPUT1"); - snd_soc_dapm_nc_pin(codec, "OUT3"); - snd_soc_dapm_nc_pin(codec, "ROUT1"); + snd_soc_dapm_nc_pin(dapm, "LINPUT1"); + snd_soc_dapm_nc_pin(dapm, "RINPUT1"); + snd_soc_dapm_nc_pin(dapm, "OUT3"); + snd_soc_dapm_nc_pin(dapm, "ROUT1"); /* set endpoints to default off mode */ - snd_soc_dapm_enable_pin(codec, "Internal Speaker"); - snd_soc_dapm_enable_pin(codec, "Internal Mic"); - snd_soc_dapm_disable_pin(codec, "Headphone Jack"); + snd_soc_dapm_enable_pin(dapm, "Internal Speaker"); + snd_soc_dapm_enable_pin(dapm, "Internal Mic"); + snd_soc_dapm_disable_pin(dapm, "Headphone Jack"); - err = snd_soc_dapm_sync(codec); + err = snd_soc_dapm_sync(dapm); if (err) return err; diff --git a/sound/soc/s3c24xx/smdk64xx_wm8580.c b/sound/soc/s3c24xx/smdk64xx_wm8580.c index 052e499b68d1..291939cf8483 100644 --- a/sound/soc/s3c24xx/smdk64xx_wm8580.c +++ b/sound/soc/s3c24xx/smdk64xx_wm8580.c @@ -182,21 +182,22 @@ static const struct snd_soc_dapm_route audio_map_rx[] = { static int smdk64xx_wm8580_init_paiftx(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; /* Add smdk64xx specific Capture widgets */ - snd_soc_dapm_new_controls(codec, wm8580_dapm_widgets_cpt, + snd_soc_dapm_new_controls(dapm, wm8580_dapm_widgets_cpt, ARRAY_SIZE(wm8580_dapm_widgets_cpt)); /* Set up PAIFTX audio path */ - snd_soc_dapm_add_routes(codec, audio_map_tx, ARRAY_SIZE(audio_map_tx)); + snd_soc_dapm_add_routes(dapm, audio_map_tx, ARRAY_SIZE(audio_map_tx)); /* Enabling the microphone requires the fitting of a 0R * resistor to connect the line from the microphone jack. */ - snd_soc_dapm_disable_pin(codec, "MicIn"); + snd_soc_dapm_disable_pin(dapm, "MicIn"); /* signal a DAPM event */ - snd_soc_dapm_sync(codec); + snd_soc_dapm_sync(dapm); return 0; } @@ -204,16 +205,17 @@ static int smdk64xx_wm8580_init_paiftx(struct snd_soc_pcm_runtime *rtd) static int smdk64xx_wm8580_init_paifrx(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; /* Add smdk64xx specific Playback widgets */ - snd_soc_dapm_new_controls(codec, wm8580_dapm_widgets_pbk, + snd_soc_dapm_new_controls(dapm, wm8580_dapm_widgets_pbk, ARRAY_SIZE(wm8580_dapm_widgets_pbk)); /* Set up PAIFRX audio path */ - snd_soc_dapm_add_routes(codec, audio_map_rx, ARRAY_SIZE(audio_map_rx)); + snd_soc_dapm_add_routes(dapm, audio_map_rx, ARRAY_SIZE(audio_map_rx)); /* signal a DAPM event */ - snd_soc_dapm_sync(codec); + snd_soc_dapm_sync(dapm); return 0; } diff --git a/sound/soc/s6000/s6105-ipcam.c b/sound/soc/s6000/s6105-ipcam.c index 96c05e137538..db1803d9665a 100644 --- a/sound/soc/s6000/s6105-ipcam.c +++ b/sound/soc/s6000/s6105-ipcam.c @@ -107,6 +107,7 @@ static int output_type_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = kcontrol->private_data; + struct snd_soc_dapm_context *dapm = &codec->dapm; unsigned int val = (ucontrol->value.enumerated.item[0] != 0); char *differential = "Audio Out Differential"; char *stereo = "Audio Out Stereo"; @@ -114,10 +115,10 @@ static int output_type_put(struct snd_kcontrol *kcontrol, if (kcontrol->private_value == val) return 0; kcontrol->private_value = val; - snd_soc_dapm_disable_pin(codec, val ? differential : stereo); - snd_soc_dapm_sync(codec); - snd_soc_dapm_enable_pin(codec, val ? stereo : differential); - snd_soc_dapm_sync(codec); + snd_soc_dapm_disable_pin(dapm, val ? differential : stereo); + snd_soc_dapm_sync(dapm); + snd_soc_dapm_enable_pin(dapm, val ? stereo : differential); + snd_soc_dapm_sync(dapm); return 1; } @@ -137,35 +138,36 @@ static const struct snd_kcontrol_new audio_out_mux = { static int s6105_aic3x_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; /* Add s6105 specific widgets */ - snd_soc_dapm_new_controls(codec, aic3x_dapm_widgets, + snd_soc_dapm_new_controls(dapm, aic3x_dapm_widgets, ARRAY_SIZE(aic3x_dapm_widgets)); /* Set up s6105 specific audio path audio_map */ - snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); /* not present */ - snd_soc_dapm_nc_pin(codec, "MONO_LOUT"); - snd_soc_dapm_nc_pin(codec, "LINE2L"); - snd_soc_dapm_nc_pin(codec, "LINE2R"); + snd_soc_dapm_nc_pin(dapm, "MONO_LOUT"); + snd_soc_dapm_nc_pin(dapm, "LINE2L"); + snd_soc_dapm_nc_pin(dapm, "LINE2R"); /* not connected */ - snd_soc_dapm_nc_pin(codec, "MIC3L"); /* LINE2L on this chip */ - snd_soc_dapm_nc_pin(codec, "MIC3R"); /* LINE2R on this chip */ - snd_soc_dapm_nc_pin(codec, "LLOUT"); - snd_soc_dapm_nc_pin(codec, "RLOUT"); - snd_soc_dapm_nc_pin(codec, "HPRCOM"); + snd_soc_dapm_nc_pin(dapm, "MIC3L"); /* LINE2L on this chip */ + snd_soc_dapm_nc_pin(dapm, "MIC3R"); /* LINE2R on this chip */ + snd_soc_dapm_nc_pin(dapm, "LLOUT"); + snd_soc_dapm_nc_pin(dapm, "RLOUT"); + snd_soc_dapm_nc_pin(dapm, "HPRCOM"); /* always connected */ - snd_soc_dapm_enable_pin(codec, "Audio In"); + snd_soc_dapm_enable_pin(dapm, "Audio In"); /* must correspond to audio_out_mux.private_value initializer */ - snd_soc_dapm_disable_pin(codec, "Audio Out Differential"); - snd_soc_dapm_sync(codec); - snd_soc_dapm_enable_pin(codec, "Audio Out Stereo"); + snd_soc_dapm_disable_pin(dapm, "Audio Out Differential"); + snd_soc_dapm_sync(dapm); + snd_soc_dapm_enable_pin(dapm, "Audio Out Stereo"); - snd_soc_dapm_sync(codec); + snd_soc_dapm_sync(dapm); snd_ctl_add(codec->snd_card, snd_ctl_new1(&audio_out_mux, codec)); diff --git a/sound/soc/sh/migor.c b/sound/soc/sh/migor.c index ac6c49ce6fdf..c61fc188394d 100644 --- a/sound/soc/sh/migor.c +++ b/sound/soc/sh/migor.c @@ -140,11 +140,12 @@ static const struct snd_soc_dapm_route audio_map[] = { static int migor_dai_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; - snd_soc_dapm_new_controls(codec, migor_dapm_widgets, + snd_soc_dapm_new_controls(dapm, migor_dapm_widgets, ARRAY_SIZE(migor_dapm_widgets)); - snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); return 0; } diff --git a/sound/soc/sh/sh7760-ac97.c b/sound/soc/sh/sh7760-ac97.c index f8e0ab82ef59..105d4112e3ba 100644 --- a/sound/soc/sh/sh7760-ac97.c +++ b/sound/soc/sh/sh7760-ac97.c @@ -23,7 +23,7 @@ extern struct snd_soc_platform_driver sh7760_soc_platform; static int machine_init(struct snd_soc_pcm_runtime *rtd) { - snd_soc_dapm_sync(rtd->codec); + snd_soc_dapm_sync(&rtd->codec->dapm); return 0; } diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index d214f02cbb65..6c0589e3fefb 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -14,27 +14,34 @@ #include <linux/i2c.h> #include <linux/spi/spi.h> #include <sound/soc.h> +#include <linux/lzo.h> +#include <linux/bitmap.h> +#include <linux/rbtree.h> static unsigned int snd_soc_4_12_read(struct snd_soc_codec *codec, unsigned int reg) { - u16 *cache = codec->reg_cache; + int ret; + unsigned int val; if (reg >= codec->driver->reg_cache_size || snd_soc_codec_volatile_register(codec, reg)) { if (codec->cache_only) return -1; + BUG_ON(!codec->hw_read); return codec->hw_read(codec, reg); } - return cache[reg]; + ret = snd_soc_cache_read(codec, reg, &val); + if (ret < 0) + return -1; + return val; } static int snd_soc_4_12_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { - u16 *cache = codec->reg_cache; u8 data[2]; int ret; @@ -42,16 +49,17 @@ static int snd_soc_4_12_write(struct snd_soc_codec *codec, unsigned int reg, data[1] = value & 0x00ff; if (!snd_soc_codec_volatile_register(codec, reg) && - reg < codec->driver->reg_cache_size) - cache[reg] = value; + reg < codec->driver->reg_cache_size) { + ret = snd_soc_cache_write(codec, reg, value); + if (ret < 0) + return -1; + } if (codec->cache_only) { codec->cache_sync = 1; return 0; } - dev_dbg(codec->dev, "0x%x = 0x%x\n", reg, value); - ret = codec->hw_write(codec->control_data, data, 2); if (ret == 2) return 0; @@ -94,23 +102,27 @@ static int snd_soc_4_12_spi_write(void *control_data, const char *data, static unsigned int snd_soc_7_9_read(struct snd_soc_codec *codec, unsigned int reg) { - u16 *cache = codec->reg_cache; + int ret; + unsigned int val; if (reg >= codec->driver->reg_cache_size || snd_soc_codec_volatile_register(codec, reg)) { if (codec->cache_only) return -1; + BUG_ON(!codec->hw_read); return codec->hw_read(codec, reg); } - return cache[reg]; + ret = snd_soc_cache_read(codec, reg, &val); + if (ret < 0) + return -1; + return val; } static int snd_soc_7_9_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { - u16 *cache = codec->reg_cache; u8 data[2]; int ret; @@ -118,16 +130,17 @@ static int snd_soc_7_9_write(struct snd_soc_codec *codec, unsigned int reg, data[1] = value & 0x00ff; if (!snd_soc_codec_volatile_register(codec, reg) && - reg < codec->driver->reg_cache_size) - cache[reg] = value; + reg < codec->driver->reg_cache_size) { + ret = snd_soc_cache_write(codec, reg, value); + if (ret < 0) + return -1; + } if (codec->cache_only) { codec->cache_sync = 1; return 0; } - dev_dbg(codec->dev, "0x%x = 0x%x\n", reg, value); - ret = codec->hw_write(codec->control_data, data, 2); if (ret == 2) return 0; @@ -170,24 +183,25 @@ static int snd_soc_7_9_spi_write(void *control_data, const char *data, static int snd_soc_8_8_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { - u8 *cache = codec->reg_cache; u8 data[2]; + int ret; reg &= 0xff; data[0] = reg; data[1] = value & 0xff; if (!snd_soc_codec_volatile_register(codec, reg) && - reg < codec->driver->reg_cache_size) - cache[reg] = value; + reg < codec->driver->reg_cache_size) { + ret = snd_soc_cache_write(codec, reg, value); + if (ret < 0) + return -1; + } if (codec->cache_only) { codec->cache_sync = 1; return 0; } - dev_dbg(codec->dev, "0x%x = 0x%x\n", reg, value); - if (codec->hw_write(codec->control_data, data, 2) == 2) return 0; else @@ -197,7 +211,8 @@ static int snd_soc_8_8_write(struct snd_soc_codec *codec, unsigned int reg, static unsigned int snd_soc_8_8_read(struct snd_soc_codec *codec, unsigned int reg) { - u8 *cache = codec->reg_cache; + int ret; + unsigned int val; reg &= 0xff; if (reg >= codec->driver->reg_cache_size || @@ -205,10 +220,14 @@ static unsigned int snd_soc_8_8_read(struct snd_soc_codec *codec, if (codec->cache_only) return -1; + BUG_ON(!codec->hw_read); return codec->hw_read(codec, reg); } - return cache[reg]; + ret = snd_soc_cache_read(codec, reg, &val); + if (ret < 0) + return -1; + return val; } #if defined(CONFIG_SPI_MASTER) @@ -244,24 +263,25 @@ static int snd_soc_8_8_spi_write(void *control_data, const char *data, static int snd_soc_8_16_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { - u16 *reg_cache = codec->reg_cache; u8 data[3]; + int ret; data[0] = reg; data[1] = (value >> 8) & 0xff; data[2] = value & 0xff; if (!snd_soc_codec_volatile_register(codec, reg) && - reg < codec->driver->reg_cache_size) - reg_cache[reg] = value; + reg < codec->driver->reg_cache_size) { + ret = snd_soc_cache_write(codec, reg, value); + if (ret < 0) + return -1; + } if (codec->cache_only) { codec->cache_sync = 1; return 0; } - dev_dbg(codec->dev, "0x%x = 0x%x\n", reg, value); - if (codec->hw_write(codec->control_data, data, 3) == 3) return 0; else @@ -271,17 +291,22 @@ static int snd_soc_8_16_write(struct snd_soc_codec *codec, unsigned int reg, static unsigned int snd_soc_8_16_read(struct snd_soc_codec *codec, unsigned int reg) { - u16 *cache = codec->reg_cache; + int ret; + unsigned int val; if (reg >= codec->driver->reg_cache_size || snd_soc_codec_volatile_register(codec, reg)) { if (codec->cache_only) return -1; + BUG_ON(!codec->hw_read); return codec->hw_read(codec, reg); - } else { - return cache[reg]; } + + ret = snd_soc_cache_read(codec, reg, &val); + if (ret < 0) + return -1; + return val; } #if defined(CONFIG_SPI_MASTER) @@ -420,7 +445,8 @@ static unsigned int snd_soc_16_8_read_i2c(struct snd_soc_codec *codec, static unsigned int snd_soc_16_8_read(struct snd_soc_codec *codec, unsigned int reg) { - u8 *cache = codec->reg_cache; + int ret; + unsigned int val; reg &= 0xff; if (reg >= codec->driver->reg_cache_size || @@ -428,16 +454,19 @@ static unsigned int snd_soc_16_8_read(struct snd_soc_codec *codec, if (codec->cache_only) return -1; + BUG_ON(!codec->hw_read); return codec->hw_read(codec, reg); } - return cache[reg]; + ret = snd_soc_cache_read(codec, reg, &val); + if (ret < 0) + return -1; + return val; } static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { - u8 *cache = codec->reg_cache; u8 data[3]; int ret; @@ -447,16 +476,17 @@ static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg, reg &= 0xff; if (!snd_soc_codec_volatile_register(codec, reg) && - reg < codec->driver->reg_cache_size) - cache[reg] = value; + reg < codec->driver->reg_cache_size) { + ret = snd_soc_cache_write(codec, reg, value); + if (ret < 0) + return -1; + } if (codec->cache_only) { codec->cache_sync = 1; return 0; } - dev_dbg(codec->dev, "0x%x = 0x%x\n", reg, value); - ret = codec->hw_write(codec->control_data, data, 3); if (ret == 3) return 0; @@ -534,23 +564,28 @@ static unsigned int snd_soc_16_16_read_i2c(struct snd_soc_codec *codec, static unsigned int snd_soc_16_16_read(struct snd_soc_codec *codec, unsigned int reg) { - u16 *cache = codec->reg_cache; + int ret; + unsigned int val; if (reg >= codec->driver->reg_cache_size || snd_soc_codec_volatile_register(codec, reg)) { if (codec->cache_only) return -1; + BUG_ON(!codec->hw_read); return codec->hw_read(codec, reg); } - return cache[reg]; + ret = snd_soc_cache_read(codec, reg, &val); + if (ret < 0) + return -1; + + return val; } static int snd_soc_16_16_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { - u16 *cache = codec->reg_cache; u8 data[4]; int ret; @@ -560,16 +595,17 @@ static int snd_soc_16_16_write(struct snd_soc_codec *codec, unsigned int reg, data[3] = value & 0xff; if (!snd_soc_codec_volatile_register(codec, reg) && - reg < codec->driver->reg_cache_size) - cache[reg] = value; + reg < codec->driver->reg_cache_size) { + ret = snd_soc_cache_write(codec, reg, value); + if (ret < 0) + return -1; + } if (codec->cache_only) { codec->cache_sync = 1; return 0; } - dev_dbg(codec->dev, "0x%x = 0x%x\n", reg, value); - ret = codec->hw_write(codec->control_data, data, 4); if (ret == 4) return 0; @@ -724,3 +760,883 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, return 0; } EXPORT_SYMBOL_GPL(snd_soc_codec_set_cache_io); + +struct snd_soc_rbtree_node { + struct rb_node node; + unsigned int reg; + unsigned int value; + unsigned int defval; +} __attribute__ ((packed)); + +struct snd_soc_rbtree_ctx { + struct rb_root root; +}; + +static struct snd_soc_rbtree_node *snd_soc_rbtree_lookup( + struct rb_root *root, unsigned int reg) +{ + struct rb_node *node; + struct snd_soc_rbtree_node *rbnode; + + node = root->rb_node; + while (node) { + rbnode = container_of(node, struct snd_soc_rbtree_node, node); + if (rbnode->reg < reg) + node = node->rb_left; + else if (rbnode->reg > reg) + node = node->rb_right; + else + return rbnode; + } + + return NULL; +} + + +static int snd_soc_rbtree_insert(struct rb_root *root, + struct snd_soc_rbtree_node *rbnode) +{ + struct rb_node **new, *parent; + struct snd_soc_rbtree_node *rbnode_tmp; + + parent = NULL; + new = &root->rb_node; + while (*new) { + rbnode_tmp = container_of(*new, struct snd_soc_rbtree_node, + node); + parent = *new; + if (rbnode_tmp->reg < rbnode->reg) + new = &((*new)->rb_left); + else if (rbnode_tmp->reg > rbnode->reg) + new = &((*new)->rb_right); + else + return 0; + } + + /* insert the node into the rbtree */ + rb_link_node(&rbnode->node, parent, new); + rb_insert_color(&rbnode->node, root); + + return 1; +} + +static int snd_soc_rbtree_cache_sync(struct snd_soc_codec *codec) +{ + struct snd_soc_rbtree_ctx *rbtree_ctx; + struct rb_node *node; + struct snd_soc_rbtree_node *rbnode; + unsigned int val; + + rbtree_ctx = codec->reg_cache; + for (node = rb_first(&rbtree_ctx->root); node; node = rb_next(node)) { + rbnode = rb_entry(node, struct snd_soc_rbtree_node, node); + if (rbnode->value == rbnode->defval) + continue; + snd_soc_cache_read(codec, rbnode->reg, &val); + snd_soc_write(codec, rbnode->reg, val); + dev_dbg(codec->dev, "Synced register %#x, value = %#x\n", + rbnode->reg, val); + } + + return 0; +} + +static int snd_soc_rbtree_cache_write(struct snd_soc_codec *codec, + unsigned int reg, unsigned int value) +{ + struct snd_soc_rbtree_ctx *rbtree_ctx; + struct snd_soc_rbtree_node *rbnode; + + rbtree_ctx = codec->reg_cache; + rbnode = snd_soc_rbtree_lookup(&rbtree_ctx->root, reg); + if (rbnode) { + if (rbnode->value == value) + return 0; + rbnode->value = value; + } else { + /* bail out early, no need to create the rbnode yet */ + if (!value) + return 0; + /* + * for uninitialized registers whose value is changed + * from the default zero, create an rbnode and insert + * it into the tree. + */ + rbnode = kzalloc(sizeof *rbnode, GFP_KERNEL); + if (!rbnode) + return -ENOMEM; + rbnode->reg = reg; + rbnode->value = value; + snd_soc_rbtree_insert(&rbtree_ctx->root, rbnode); + } + + return 0; +} + +static int snd_soc_rbtree_cache_read(struct snd_soc_codec *codec, + unsigned int reg, unsigned int *value) +{ + struct snd_soc_rbtree_ctx *rbtree_ctx; + struct snd_soc_rbtree_node *rbnode; + + rbtree_ctx = codec->reg_cache; + rbnode = snd_soc_rbtree_lookup(&rbtree_ctx->root, reg); + if (rbnode) { + *value = rbnode->value; + } else { + /* uninitialized registers default to 0 */ + *value = 0; + } + + return 0; +} + +static int snd_soc_rbtree_cache_exit(struct snd_soc_codec *codec) +{ + struct rb_node *next; + struct snd_soc_rbtree_ctx *rbtree_ctx; + struct snd_soc_rbtree_node *rbtree_node; + + /* if we've already been called then just return */ + rbtree_ctx = codec->reg_cache; + if (!rbtree_ctx) + return 0; + + /* free up the rbtree */ + next = rb_first(&rbtree_ctx->root); + while (next) { + rbtree_node = rb_entry(next, struct snd_soc_rbtree_node, node); + next = rb_next(&rbtree_node->node); + rb_erase(&rbtree_node->node, &rbtree_ctx->root); + kfree(rbtree_node); + } + + /* release the resources */ + kfree(codec->reg_cache); + codec->reg_cache = NULL; + + return 0; +} + +static int snd_soc_rbtree_cache_init(struct snd_soc_codec *codec) +{ + struct snd_soc_rbtree_ctx *rbtree_ctx; + + codec->reg_cache = kmalloc(sizeof *rbtree_ctx, GFP_KERNEL); + if (!codec->reg_cache) + return -ENOMEM; + + rbtree_ctx = codec->reg_cache; + rbtree_ctx->root = RB_ROOT; + + if (!codec->driver->reg_cache_default) + return 0; + +/* + * populate the rbtree with the initialized registers. All other + * registers will be inserted into the tree when they are first written. + * + * The reasoning behind this, is that we need to step through and + * dereference the cache in u8/u16 increments without sacrificing + * portability. This could also be done using memcpy() but that would + * be slightly more cryptic. + */ +#define snd_soc_rbtree_populate(cache) \ +({ \ + int ret, i; \ + struct snd_soc_rbtree_node *rbtree_node; \ + \ + ret = 0; \ + cache = codec->driver->reg_cache_default; \ + for (i = 0; i < codec->driver->reg_cache_size; ++i) { \ + if (!cache[i]) \ + continue; \ + rbtree_node = kzalloc(sizeof *rbtree_node, GFP_KERNEL); \ + if (!rbtree_node) { \ + ret = -ENOMEM; \ + snd_soc_cache_exit(codec); \ + break; \ + } \ + rbtree_node->reg = i; \ + rbtree_node->value = cache[i]; \ + rbtree_node->defval = cache[i]; \ + snd_soc_rbtree_insert(&rbtree_ctx->root, \ + rbtree_node); \ + } \ + ret; \ +}) + + switch (codec->driver->reg_word_size) { + case 1: { + const u8 *cache; + + return snd_soc_rbtree_populate(cache); + } + case 2: { + const u16 *cache; + + return snd_soc_rbtree_populate(cache); + } + default: + BUG(); + } + + return 0; +} + +struct snd_soc_lzo_ctx { + void *wmem; + void *dst; + const void *src; + size_t src_len; + size_t dst_len; + size_t decompressed_size; + unsigned long *sync_bmp; + int sync_bmp_nbits; +}; + +#define LZO_BLOCK_NUM 8 +static int snd_soc_lzo_block_count(void) +{ + return LZO_BLOCK_NUM; +} + +static int snd_soc_lzo_prepare(struct snd_soc_lzo_ctx *lzo_ctx) +{ + lzo_ctx->wmem = kmalloc(LZO1X_MEM_COMPRESS, GFP_KERNEL); + if (!lzo_ctx->wmem) + return -ENOMEM; + return 0; +} + +static int snd_soc_lzo_compress(struct snd_soc_lzo_ctx *lzo_ctx) +{ + size_t compress_size; + int ret; + + ret = lzo1x_1_compress(lzo_ctx->src, lzo_ctx->src_len, + lzo_ctx->dst, &compress_size, lzo_ctx->wmem); + if (ret != LZO_E_OK || compress_size > lzo_ctx->dst_len) + return -EINVAL; + lzo_ctx->dst_len = compress_size; + return 0; +} + +static int snd_soc_lzo_decompress(struct snd_soc_lzo_ctx *lzo_ctx) +{ + size_t dst_len; + int ret; + + dst_len = lzo_ctx->dst_len; + ret = lzo1x_decompress_safe(lzo_ctx->src, lzo_ctx->src_len, + lzo_ctx->dst, &dst_len); + if (ret != LZO_E_OK || dst_len != lzo_ctx->dst_len) + return -EINVAL; + return 0; +} + +static int snd_soc_lzo_compress_cache_block(struct snd_soc_codec *codec, + struct snd_soc_lzo_ctx *lzo_ctx) +{ + int ret; + + lzo_ctx->dst_len = lzo1x_worst_compress(PAGE_SIZE); + lzo_ctx->dst = kmalloc(lzo_ctx->dst_len, GFP_KERNEL); + if (!lzo_ctx->dst) { + lzo_ctx->dst_len = 0; + return -ENOMEM; + } + + ret = snd_soc_lzo_compress(lzo_ctx); + if (ret < 0) + return ret; + return 0; +} + +static int snd_soc_lzo_decompress_cache_block(struct snd_soc_codec *codec, + struct snd_soc_lzo_ctx *lzo_ctx) +{ + int ret; + + lzo_ctx->dst_len = lzo_ctx->decompressed_size; + lzo_ctx->dst = kmalloc(lzo_ctx->dst_len, GFP_KERNEL); + if (!lzo_ctx->dst) { + lzo_ctx->dst_len = 0; + return -ENOMEM; + } + + ret = snd_soc_lzo_decompress(lzo_ctx); + if (ret < 0) + return ret; + return 0; +} + +static inline int snd_soc_lzo_get_blkindex(struct snd_soc_codec *codec, + unsigned int reg) +{ + struct snd_soc_codec_driver *codec_drv; + size_t reg_size; + + codec_drv = codec->driver; + reg_size = codec_drv->reg_cache_size * codec_drv->reg_word_size; + return (reg * codec_drv->reg_word_size) / + DIV_ROUND_UP(reg_size, snd_soc_lzo_block_count()); +} + +static inline int snd_soc_lzo_get_blkpos(struct snd_soc_codec *codec, + unsigned int reg) +{ + struct snd_soc_codec_driver *codec_drv; + size_t reg_size; + + codec_drv = codec->driver; + reg_size = codec_drv->reg_cache_size * codec_drv->reg_word_size; + return reg % (DIV_ROUND_UP(reg_size, snd_soc_lzo_block_count()) / + codec_drv->reg_word_size); +} + +static inline int snd_soc_lzo_get_blksize(struct snd_soc_codec *codec) +{ + struct snd_soc_codec_driver *codec_drv; + size_t reg_size; + + codec_drv = codec->driver; + reg_size = codec_drv->reg_cache_size * codec_drv->reg_word_size; + return DIV_ROUND_UP(reg_size, snd_soc_lzo_block_count()); +} + +static int snd_soc_lzo_cache_sync(struct snd_soc_codec *codec) +{ + struct snd_soc_lzo_ctx **lzo_blocks; + unsigned int val; + int i; + + lzo_blocks = codec->reg_cache; + for_each_set_bit(i, lzo_blocks[0]->sync_bmp, lzo_blocks[0]->sync_bmp_nbits) { + snd_soc_cache_read(codec, i, &val); + snd_soc_write(codec, i, val); + dev_dbg(codec->dev, "Synced register %#x, value = %#x\n", + i, val); + } + + return 0; +} + +static int snd_soc_lzo_cache_write(struct snd_soc_codec *codec, + unsigned int reg, unsigned int value) +{ + struct snd_soc_lzo_ctx *lzo_block, **lzo_blocks; + int ret, blkindex, blkpos; + size_t blksize, tmp_dst_len; + void *tmp_dst; + + /* index of the compressed lzo block */ + blkindex = snd_soc_lzo_get_blkindex(codec, reg); + /* register index within the decompressed block */ + blkpos = snd_soc_lzo_get_blkpos(codec, reg); + /* size of the compressed block */ + blksize = snd_soc_lzo_get_blksize(codec); + lzo_blocks = codec->reg_cache; + lzo_block = lzo_blocks[blkindex]; + + /* save the pointer and length of the compressed block */ + tmp_dst = lzo_block->dst; + tmp_dst_len = lzo_block->dst_len; + + /* prepare the source to be the compressed block */ + lzo_block->src = lzo_block->dst; + lzo_block->src_len = lzo_block->dst_len; + + /* decompress the block */ + ret = snd_soc_lzo_decompress_cache_block(codec, lzo_block); + if (ret < 0) { + kfree(lzo_block->dst); + goto out; + } + + /* write the new value to the cache */ + switch (codec->driver->reg_word_size) { + case 1: { + u8 *cache; + cache = lzo_block->dst; + if (cache[blkpos] == value) { + kfree(lzo_block->dst); + goto out; + } + cache[blkpos] = value; + } + break; + case 2: { + u16 *cache; + cache = lzo_block->dst; + if (cache[blkpos] == value) { + kfree(lzo_block->dst); + goto out; + } + cache[blkpos] = value; + } + break; + default: + BUG(); + } + + /* prepare the source to be the decompressed block */ + lzo_block->src = lzo_block->dst; + lzo_block->src_len = lzo_block->dst_len; + + /* compress the block */ + ret = snd_soc_lzo_compress_cache_block(codec, lzo_block); + if (ret < 0) { + kfree(lzo_block->dst); + kfree(lzo_block->src); + goto out; + } + + /* set the bit so we know we have to sync this register */ + set_bit(reg, lzo_block->sync_bmp); + kfree(tmp_dst); + kfree(lzo_block->src); + return 0; +out: + lzo_block->dst = tmp_dst; + lzo_block->dst_len = tmp_dst_len; + return ret; +} + +static int snd_soc_lzo_cache_read(struct snd_soc_codec *codec, + unsigned int reg, unsigned int *value) +{ + struct snd_soc_lzo_ctx *lzo_block, **lzo_blocks; + int ret, blkindex, blkpos; + size_t blksize, tmp_dst_len; + void *tmp_dst; + + *value = 0; + /* index of the compressed lzo block */ + blkindex = snd_soc_lzo_get_blkindex(codec, reg); + /* register index within the decompressed block */ + blkpos = snd_soc_lzo_get_blkpos(codec, reg); + /* size of the compressed block */ + blksize = snd_soc_lzo_get_blksize(codec); + lzo_blocks = codec->reg_cache; + lzo_block = lzo_blocks[blkindex]; + + /* save the pointer and length of the compressed block */ + tmp_dst = lzo_block->dst; + tmp_dst_len = lzo_block->dst_len; + + /* prepare the source to be the compressed block */ + lzo_block->src = lzo_block->dst; + lzo_block->src_len = lzo_block->dst_len; + + /* decompress the block */ + ret = snd_soc_lzo_decompress_cache_block(codec, lzo_block); + if (ret >= 0) { + /* fetch the value from the cache */ + switch (codec->driver->reg_word_size) { + case 1: { + u8 *cache; + cache = lzo_block->dst; + *value = cache[blkpos]; + } + break; + case 2: { + u16 *cache; + cache = lzo_block->dst; + *value = cache[blkpos]; + } + break; + default: + BUG(); + } + } + + kfree(lzo_block->dst); + /* restore the pointer and length of the compressed block */ + lzo_block->dst = tmp_dst; + lzo_block->dst_len = tmp_dst_len; + return 0; +} + +static int snd_soc_lzo_cache_exit(struct snd_soc_codec *codec) +{ + struct snd_soc_lzo_ctx **lzo_blocks; + int i, blkcount; + + lzo_blocks = codec->reg_cache; + if (!lzo_blocks) + return 0; + + blkcount = snd_soc_lzo_block_count(); + /* + * the pointer to the bitmap used for syncing the cache + * is shared amongst all lzo_blocks. Ensure it is freed + * only once. + */ + if (lzo_blocks[0]) + kfree(lzo_blocks[0]->sync_bmp); + for (i = 0; i < blkcount; ++i) { + if (lzo_blocks[i]) { + kfree(lzo_blocks[i]->wmem); + kfree(lzo_blocks[i]->dst); + } + /* each lzo_block is a pointer returned by kmalloc or NULL */ + kfree(lzo_blocks[i]); + } + kfree(lzo_blocks); + codec->reg_cache = NULL; + return 0; +} + +static int snd_soc_lzo_cache_init(struct snd_soc_codec *codec) +{ + struct snd_soc_lzo_ctx **lzo_blocks; + size_t reg_size, bmp_size; + struct snd_soc_codec_driver *codec_drv; + int ret, tofree, i, blksize, blkcount; + const char *p, *end; + unsigned long *sync_bmp; + + ret = 0; + codec_drv = codec->driver; + reg_size = codec_drv->reg_cache_size * codec_drv->reg_word_size; + + /* + * If we have not been given a default register cache + * then allocate a dummy zero-ed out region, compress it + * and remember to free it afterwards. + */ + tofree = 0; + if (!codec_drv->reg_cache_default) + tofree = 1; + + if (!codec_drv->reg_cache_default) { + codec_drv->reg_cache_default = kzalloc(reg_size, + GFP_KERNEL); + if (!codec_drv->reg_cache_default) + return -ENOMEM; + } + + blkcount = snd_soc_lzo_block_count(); + codec->reg_cache = kzalloc(blkcount * sizeof *lzo_blocks, + GFP_KERNEL); + if (!codec->reg_cache) { + ret = -ENOMEM; + goto err_tofree; + } + lzo_blocks = codec->reg_cache; + + /* + * allocate a bitmap to be used when syncing the cache with + * the hardware. Each time a register is modified, the corresponding + * bit is set in the bitmap, so we know that we have to sync + * that register. + */ + bmp_size = codec_drv->reg_cache_size; + sync_bmp = kmalloc(BITS_TO_LONGS(bmp_size) * sizeof (long), + GFP_KERNEL); + if (!sync_bmp) { + ret = -ENOMEM; + goto err; + } + bitmap_zero(sync_bmp, reg_size); + + /* allocate the lzo blocks and initialize them */ + for (i = 0; i < blkcount; ++i) { + lzo_blocks[i] = kzalloc(sizeof **lzo_blocks, + GFP_KERNEL); + if (!lzo_blocks[i]) { + kfree(sync_bmp); + ret = -ENOMEM; + goto err; + } + lzo_blocks[i]->sync_bmp = sync_bmp; + lzo_blocks[i]->sync_bmp_nbits = reg_size; + /* alloc the working space for the compressed block */ + ret = snd_soc_lzo_prepare(lzo_blocks[i]); + if (ret < 0) + goto err; + } + + blksize = snd_soc_lzo_get_blksize(codec); + p = codec_drv->reg_cache_default; + end = codec_drv->reg_cache_default + reg_size; + /* compress the register map and fill the lzo blocks */ + for (i = 0; i < blkcount; ++i, p += blksize) { + lzo_blocks[i]->src = p; + if (p + blksize > end) + lzo_blocks[i]->src_len = end - p; + else + lzo_blocks[i]->src_len = blksize; + ret = snd_soc_lzo_compress_cache_block(codec, + lzo_blocks[i]); + if (ret < 0) + goto err; + lzo_blocks[i]->decompressed_size = + lzo_blocks[i]->src_len; + } + + if (tofree) + kfree(codec_drv->reg_cache_default); + return 0; +err: + snd_soc_cache_exit(codec); +err_tofree: + if (tofree) + kfree(codec_drv->reg_cache_default); + return ret; +} + +static int snd_soc_flat_cache_sync(struct snd_soc_codec *codec) +{ + int i; + struct snd_soc_codec_driver *codec_drv; + unsigned int val; + + codec_drv = codec->driver; + for (i = 0; i < codec_drv->reg_cache_size; ++i) { + snd_soc_cache_read(codec, i, &val); + if (codec_drv->reg_cache_default) { + switch (codec_drv->reg_word_size) { + case 1: { + const u8 *cache; + + cache = codec_drv->reg_cache_default; + if (cache[i] == val) + continue; + } + break; + case 2: { + const u16 *cache; + + cache = codec_drv->reg_cache_default; + if (cache[i] == val) + continue; + } + break; + default: + BUG(); + } + } + snd_soc_write(codec, i, val); + dev_dbg(codec->dev, "Synced register %#x, value = %#x\n", + i, val); + } + return 0; +} + +static int snd_soc_flat_cache_write(struct snd_soc_codec *codec, + unsigned int reg, unsigned int value) +{ + switch (codec->driver->reg_word_size) { + case 1: { + u8 *cache; + + cache = codec->reg_cache; + cache[reg] = value; + } + break; + case 2: { + u16 *cache; + + cache = codec->reg_cache; + cache[reg] = value; + } + break; + default: + BUG(); + } + + return 0; +} + +static int snd_soc_flat_cache_read(struct snd_soc_codec *codec, + unsigned int reg, unsigned int *value) +{ + switch (codec->driver->reg_word_size) { + case 1: { + u8 *cache; + + cache = codec->reg_cache; + *value = cache[reg]; + } + break; + case 2: { + u16 *cache; + + cache = codec->reg_cache; + *value = cache[reg]; + } + break; + default: + BUG(); + } + + return 0; +} + +static int snd_soc_flat_cache_exit(struct snd_soc_codec *codec) +{ + if (!codec->reg_cache) + return 0; + kfree(codec->reg_cache); + codec->reg_cache = NULL; + return 0; +} + +static int snd_soc_flat_cache_init(struct snd_soc_codec *codec) +{ + struct snd_soc_codec_driver *codec_drv; + size_t reg_size; + + codec_drv = codec->driver; + reg_size = codec_drv->reg_cache_size * codec_drv->reg_word_size; + + if (codec_drv->reg_cache_default) + codec->reg_cache = kmemdup(codec_drv->reg_cache_default, + reg_size, GFP_KERNEL); + else + codec->reg_cache = kzalloc(reg_size, GFP_KERNEL); + if (!codec->reg_cache) + return -ENOMEM; + + return 0; +} + +/* an array of all supported compression types */ +static const struct snd_soc_cache_ops cache_types[] = { + { + .id = SND_SOC_NO_COMPRESSION, + .init = snd_soc_flat_cache_init, + .exit = snd_soc_flat_cache_exit, + .read = snd_soc_flat_cache_read, + .write = snd_soc_flat_cache_write, + .sync = snd_soc_flat_cache_sync + }, + { + .id = SND_SOC_LZO_COMPRESSION, + .init = snd_soc_lzo_cache_init, + .exit = snd_soc_lzo_cache_exit, + .read = snd_soc_lzo_cache_read, + .write = snd_soc_lzo_cache_write, + .sync = snd_soc_lzo_cache_sync + }, + { + .id = SND_SOC_RBTREE_COMPRESSION, + .init = snd_soc_rbtree_cache_init, + .exit = snd_soc_rbtree_cache_exit, + .read = snd_soc_rbtree_cache_read, + .write = snd_soc_rbtree_cache_write, + .sync = snd_soc_rbtree_cache_sync + } +}; + +int snd_soc_cache_init(struct snd_soc_codec *codec) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(cache_types); ++i) + if (cache_types[i].id == codec->driver->compress_type) + break; + if (i == ARRAY_SIZE(cache_types)) { + dev_err(codec->dev, "Could not match compress type: %d\n", + codec->driver->compress_type); + return -EINVAL; + } + + mutex_init(&codec->cache_rw_mutex); + codec->cache_ops = &cache_types[i]; + + if (codec->cache_ops->init) + return codec->cache_ops->init(codec); + return -EINVAL; +} + +/* + * NOTE: keep in mind that this function might be called + * multiple times. + */ +int snd_soc_cache_exit(struct snd_soc_codec *codec) +{ + if (codec->cache_ops && codec->cache_ops->exit) + return codec->cache_ops->exit(codec); + return -EINVAL; +} + +/** + * snd_soc_cache_read: Fetch the value of a given register from the cache. + * + * @codec: CODEC to configure. + * @reg: The register index. + * @value: The value to be returned. + */ +int snd_soc_cache_read(struct snd_soc_codec *codec, + unsigned int reg, unsigned int *value) +{ + int ret; + + mutex_lock(&codec->cache_rw_mutex); + + if (value && codec->cache_ops && codec->cache_ops->read) { + ret = codec->cache_ops->read(codec, reg, value); + mutex_unlock(&codec->cache_rw_mutex); + return ret; + } + + mutex_unlock(&codec->cache_rw_mutex); + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_cache_read); + +/** + * snd_soc_cache_write: Set the value of a given register in the cache. + * + * @codec: CODEC to configure. + * @reg: The register index. + * @value: The new register value. + */ +int snd_soc_cache_write(struct snd_soc_codec *codec, + unsigned int reg, unsigned int value) +{ + int ret; + + mutex_lock(&codec->cache_rw_mutex); + + if (codec->cache_ops && codec->cache_ops->write) { + ret = codec->cache_ops->write(codec, reg, value); + mutex_unlock(&codec->cache_rw_mutex); + return ret; + } + + mutex_unlock(&codec->cache_rw_mutex); + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_cache_write); + +/** + * snd_soc_cache_sync: Sync the register cache with the hardware. + * + * @codec: CODEC to configure. + * + * Any registers that should not be synced should be marked as + * volatile. In general drivers can choose not to use the provided + * syncing functionality if they so require. + */ +int snd_soc_cache_sync(struct snd_soc_codec *codec) +{ + int ret; + + if (!codec->cache_sync) { + return 0; + } + + if (codec->cache_ops && codec->cache_ops->sync) { + ret = codec->cache_ops->sync(codec); + if (!ret) + codec->cache_sync = 0; + return ret; + } + + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_cache_sync); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 441285ade024..3d70ce58d03c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -39,6 +39,9 @@ #include <sound/soc-dapm.h> #include <sound/initval.h> +#define CREATE_TRACE_POINTS +#include <trace/events/asoc.h> + #define NAME_SIZE 32 static DEFINE_MUTEX(pcm_mutex); @@ -238,8 +241,10 @@ static const struct file_operations codec_reg_fops = { static void soc_init_codec_debugfs(struct snd_soc_codec *codec) { - codec->debugfs_codec_root = debugfs_create_dir(codec->name , - debugfs_root); + struct dentry *debugfs_card_root = codec->card->debugfs_card_root; + + codec->debugfs_codec_root = debugfs_create_dir(codec->name, + debugfs_card_root); if (!codec->debugfs_codec_root) { printk(KERN_WARNING "ASoC: Failed to create codec debugfs directory\n"); @@ -253,20 +258,13 @@ static void soc_init_codec_debugfs(struct snd_soc_codec *codec) printk(KERN_WARNING "ASoC: Failed to create codec register debugfs file\n"); - codec->debugfs_pop_time = debugfs_create_u32("dapm_pop_time", 0644, - codec->debugfs_codec_root, - &codec->pop_time); - if (!codec->debugfs_pop_time) - printk(KERN_WARNING - "Failed to create pop time debugfs file\n"); - - codec->debugfs_dapm = debugfs_create_dir("dapm", + codec->dapm.debugfs_dapm = debugfs_create_dir("dapm", codec->debugfs_codec_root); - if (!codec->debugfs_dapm) + if (!codec->dapm.debugfs_dapm) printk(KERN_WARNING "Failed to create DAPM debugfs directory\n"); - snd_soc_dapm_debugfs_init(codec); + snd_soc_dapm_debugfs_init(&codec->dapm); } static void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec) @@ -374,6 +372,29 @@ static const struct file_operations platform_list_fops = { .llseek = default_llseek,/* read accesses f_pos */ }; +static void soc_init_card_debugfs(struct snd_soc_card *card) +{ + card->debugfs_card_root = debugfs_create_dir(card->name, + debugfs_root); + if (!card->debugfs_card_root) { + dev_warn(card->dev, + "ASoC: Failed to create codec debugfs directory\n"); + return; + } + + card->debugfs_pop_time = debugfs_create_u32("dapm_pop_time", 0644, + card->debugfs_card_root, + &card->pop_time); + if (!card->debugfs_pop_time) + dev_warn(card->dev, + "Failed to create pop time debugfs file\n"); +} + +static void soc_cleanup_card_debugfs(struct snd_soc_card *card) +{ + debugfs_remove_recursive(card->debugfs_card_root); +} + #else static inline void soc_init_codec_debugfs(struct snd_soc_codec *codec) @@ -383,6 +404,14 @@ static inline void soc_init_codec_debugfs(struct snd_soc_codec *codec) static inline void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec) { } + +static inline void soc_init_card_debugfs(struct snd_soc_card *card) +{ +} + +static inline void soc_cleanup_card_debugfs(struct snd_soc_card *card) +{ +} #endif #ifdef CONFIG_SND_SOC_AC97_BUS @@ -1017,7 +1046,7 @@ static int soc_suspend(struct device *dev) /* close any waiting streams and save state */ for (i = 0; i < card->num_rtd; i++) { run_delayed_work(&card->rtd[i].delayed_work); - card->rtd[i].codec->suspend_bias_level = card->rtd[i].codec->bias_level; + card->rtd[i].codec->dapm.suspend_bias_level = card->rtd[i].codec->dapm.bias_level; } for (i = 0; i < card->num_rtd; i++) { @@ -1041,7 +1070,7 @@ static int soc_suspend(struct device *dev) /* If there are paths active then the CODEC will be held with * bias _ON and should not be suspended. */ if (!codec->suspended && codec->driver->suspend) { - switch (codec->bias_level) { + switch (codec->dapm.bias_level) { case SND_SOC_BIAS_STANDBY: case SND_SOC_BIAS_OFF: codec->driver->suspend(codec, PMSG_SUSPEND); @@ -1110,7 +1139,7 @@ static void soc_resume_deferred(struct work_struct *work) * resume. Otherwise the suspend was suppressed. */ if (codec->driver->resume && codec->suspended) { - switch (codec->bias_level) { + switch (codec->dapm.bias_level) { case SND_SOC_BIAS_STANDBY: case SND_SOC_BIAS_OFF: codec->driver->resume(codec); @@ -1346,7 +1375,7 @@ static void soc_remove_dai_link(struct snd_soc_card *card, int num) } /* Make sure all DAPM widgets are freed */ - snd_soc_dapm_free(codec); + snd_soc_dapm_free(&codec->dapm); soc_cleanup_codec_debugfs(codec); device_remove_file(&rtd->dev, &dev_attr_codec_reg); @@ -1410,6 +1439,7 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num) /* probe the CODEC */ if (!codec->probed) { + codec->dapm.card = card; if (codec->driver->probe) { ret = codec->driver->probe(codec); if (ret < 0) { @@ -1470,8 +1500,8 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num) } /* Make sure all DAPM widgets are instantiated */ - snd_soc_dapm_new_widgets(codec); - snd_soc_dapm_sync(codec); + snd_soc_dapm_new_widgets(&codec->dapm); + snd_soc_dapm_sync(&codec->dapm); /* register the rtd device */ rtd->dev.release = rtd_release; @@ -1667,6 +1697,8 @@ static int soc_probe(struct platform_device *pdev) INIT_LIST_HEAD(&card->codec_dev_list); INIT_LIST_HEAD(&card->platform_dev_list); + soc_init_card_debugfs(card); + ret = snd_soc_register_card(card); if (ret != 0) { dev_err(&pdev->dev, "Failed to register card\n"); @@ -1694,6 +1726,8 @@ static int soc_remove(struct platform_device *pdev) for (i = 0; i < card->num_rtd; i++) soc_remove_dai_link(card, i); + soc_cleanup_card_debugfs(card); + /* remove the card */ if (card->remove) card->remove(pdev); @@ -1877,6 +1911,27 @@ void snd_soc_free_ac97_codec(struct snd_soc_codec *codec) } EXPORT_SYMBOL_GPL(snd_soc_free_ac97_codec); +unsigned int snd_soc_read(struct snd_soc_codec *codec, unsigned int reg) +{ + unsigned int ret; + + ret = codec->driver->read(codec, reg); + dev_dbg(codec->dev, "read %x => %x\n", reg, ret); + trace_snd_soc_reg_read(codec, reg, ret); + + return ret; +} +EXPORT_SYMBOL_GPL(snd_soc_read); + +unsigned int snd_soc_write(struct snd_soc_codec *codec, + unsigned int reg, unsigned int val) +{ + dev_dbg(codec->dev, "write %x = %x\n", reg, val); + trace_snd_soc_reg_write(codec, reg, val); + return codec->driver->write(codec, reg, val); +} +EXPORT_SYMBOL_GPL(snd_soc_write); + /** * snd_soc_update_bits - update codec register bits * @codec: audio codec @@ -3219,30 +3274,25 @@ int snd_soc_register_codec(struct device *dev, return -ENOMEM; } - /* allocate CODEC register cache */ - if (codec_drv->reg_cache_size && codec_drv->reg_word_size) { - - if (codec_drv->reg_cache_default) - codec->reg_cache = kmemdup(codec_drv->reg_cache_default, - codec_drv->reg_cache_size * codec_drv->reg_word_size, GFP_KERNEL); - else - codec->reg_cache = kzalloc(codec_drv->reg_cache_size * - codec_drv->reg_word_size, GFP_KERNEL); - - if (codec->reg_cache == NULL) { - kfree(codec->name); - kfree(codec); - return -ENOMEM; - } - } - + INIT_LIST_HEAD(&codec->dapm.widgets); + INIT_LIST_HEAD(&codec->dapm.paths); + codec->dapm.bias_level = SND_SOC_BIAS_OFF; + codec->dapm.dev = dev; + codec->dapm.codec = codec; codec->dev = dev; codec->driver = codec_drv; - codec->bias_level = SND_SOC_BIAS_OFF; codec->num_dai = num_dai; mutex_init(&codec->mutex); - INIT_LIST_HEAD(&codec->dapm_widgets); - INIT_LIST_HEAD(&codec->dapm_paths); + + /* allocate CODEC register cache */ + if (codec_drv->reg_cache_size && codec_drv->reg_word_size) { + ret = snd_soc_cache_init(codec); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache compression type: %d\n", + ret); + goto error_cache; + } + } for (i = 0; i < num_dai; i++) { fixup_codec_formats(&dai_drv[i].playback); @@ -3253,7 +3303,7 @@ int snd_soc_register_codec(struct device *dev, if (num_dai) { ret = snd_soc_register_dais(dev, dai_drv, num_dai); if (ret < 0) - goto error; + goto error_dais; } mutex_lock(&client_mutex); @@ -3264,9 +3314,9 @@ int snd_soc_register_codec(struct device *dev, pr_debug("Registered codec '%s'\n", codec->name); return 0; -error: - if (codec->reg_cache) - kfree(codec->reg_cache); +error_dais: + snd_soc_cache_exit(codec); +error_cache: kfree(codec->name); kfree(codec); return ret; @@ -3300,8 +3350,7 @@ found: pr_debug("Unregistered codec '%s'\n", codec->name); - if (codec->reg_cache) - kfree(codec->reg_cache); + snd_soc_cache_exit(codec); kfree(codec->name); kfree(codec); } diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 7d85c6496afa..8352430f4043 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -42,9 +42,12 @@ #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> +#include <sound/soc.h> #include <sound/soc-dapm.h> #include <sound/initval.h> +#include <trace/events/asoc.h> + /* dapm power sequences - make this per codec in the future */ static int dapm_up_seq[] = { [snd_soc_dapm_pre] = 0, @@ -90,17 +93,24 @@ static void pop_wait(u32 pop_time) schedule_timeout_uninterruptible(msecs_to_jiffies(pop_time)); } -static void pop_dbg(u32 pop_time, const char *fmt, ...) +static void pop_dbg(struct device *dev, u32 pop_time, const char *fmt, ...) { va_list args; + char *buf; - va_start(args, fmt); + if (!pop_time) + return; - if (pop_time) { - vprintk(fmt, args); - } + buf = kmalloc(PAGE_SIZE, GFP_KERNEL); + if (buf == NULL) + return; + va_start(args, fmt); + vsnprintf(buf, PAGE_SIZE, fmt, args); + dev_info(dev, buf); va_end(args); + + kfree(buf); } /* create a new dapm widget */ @@ -120,37 +130,42 @@ static inline struct snd_soc_dapm_widget *dapm_cnew_widget( * Returns 0 for success else error. */ static int snd_soc_dapm_set_bias_level(struct snd_soc_card *card, - struct snd_soc_codec *codec, enum snd_soc_bias_level level) + struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level) { int ret = 0; switch (level) { case SND_SOC_BIAS_ON: - dev_dbg(codec->dev, "Setting full bias\n"); + dev_dbg(dapm->dev, "Setting full bias\n"); break; case SND_SOC_BIAS_PREPARE: - dev_dbg(codec->dev, "Setting bias prepare\n"); + dev_dbg(dapm->dev, "Setting bias prepare\n"); break; case SND_SOC_BIAS_STANDBY: - dev_dbg(codec->dev, "Setting standby bias\n"); + dev_dbg(dapm->dev, "Setting standby bias\n"); break; case SND_SOC_BIAS_OFF: - dev_dbg(codec->dev, "Setting bias off\n"); + dev_dbg(dapm->dev, "Setting bias off\n"); break; default: - dev_err(codec->dev, "Setting invalid bias %d\n", level); + dev_err(dapm->dev, "Setting invalid bias %d\n", level); return -EINVAL; } + trace_snd_soc_bias_level_start(card, level); + if (card && card->set_bias_level) ret = card->set_bias_level(card, level); if (ret == 0) { - if (codec->driver->set_bias_level) - ret = codec->driver->set_bias_level(codec, level); + if (dapm->codec && dapm->codec->driver->set_bias_level) + ret = dapm->codec->driver->set_bias_level(dapm->codec, level); else - codec->bias_level = level; + dapm->bias_level = level; } + trace_snd_soc_bias_level_done(card, level); + return ret; } @@ -241,7 +256,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, } /* connect mux widget to its interconnecting audio paths */ -static int dapm_connect_mux(struct snd_soc_codec *codec, +static int dapm_connect_mux(struct snd_soc_dapm_context *dapm, struct snd_soc_dapm_widget *src, struct snd_soc_dapm_widget *dest, struct snd_soc_dapm_path *path, const char *control_name, const struct snd_kcontrol_new *kcontrol) @@ -251,7 +266,7 @@ static int dapm_connect_mux(struct snd_soc_codec *codec, for (i = 0; i < e->max; i++) { if (!(strcmp(control_name, e->texts[i]))) { - list_add(&path->list, &codec->dapm_paths); + list_add(&path->list, &dapm->paths); list_add(&path->list_sink, &dest->sources); list_add(&path->list_source, &src->sinks); path->name = (char*)e->texts[i]; @@ -264,7 +279,7 @@ static int dapm_connect_mux(struct snd_soc_codec *codec, } /* connect mixer widget to its interconnecting audio paths */ -static int dapm_connect_mixer(struct snd_soc_codec *codec, +static int dapm_connect_mixer(struct snd_soc_dapm_context *dapm, struct snd_soc_dapm_widget *src, struct snd_soc_dapm_widget *dest, struct snd_soc_dapm_path *path, const char *control_name) { @@ -273,7 +288,7 @@ static int dapm_connect_mixer(struct snd_soc_codec *codec, /* search for mixer kcontrol */ for (i = 0; i < dest->num_kcontrols; i++) { if (!strcmp(control_name, dest->kcontrols[i].name)) { - list_add(&path->list, &codec->dapm_paths); + list_add(&path->list, &dapm->paths); list_add(&path->list_sink, &dest->sources); list_add(&path->list_source, &src->sinks); path->name = dest->kcontrols[i].name; @@ -290,6 +305,8 @@ static int dapm_update_bits(struct snd_soc_dapm_widget *widget) int change, power; unsigned int old, new; struct snd_soc_codec *codec = widget->codec; + struct snd_soc_dapm_context *dapm = widget->dapm; + struct snd_soc_card *card = dapm->card; /* check for valid widgets */ if (widget->reg < 0 || widget->id == snd_soc_dapm_input || @@ -309,24 +326,26 @@ static int dapm_update_bits(struct snd_soc_dapm_widget *widget) change = old != new; if (change) { - pop_dbg(codec->pop_time, "pop test %s : %s in %d ms\n", + pop_dbg(dapm->dev, card->pop_time, + "pop test %s : %s in %d ms\n", widget->name, widget->power ? "on" : "off", - codec->pop_time); - pop_wait(codec->pop_time); + card->pop_time); + pop_wait(card->pop_time); snd_soc_write(codec, widget->reg, new); } - pr_debug("reg %x old %x new %x change %d\n", widget->reg, - old, new, change); + dev_dbg(dapm->dev, "reg %x old %x new %x change %d\n", widget->reg, + old, new, change); return change; } /* create new dapm mixer control */ -static int dapm_new_mixer(struct snd_soc_codec *codec, +static int dapm_new_mixer(struct snd_soc_dapm_context *dapm, struct snd_soc_dapm_widget *w) { int i, ret = 0; size_t name_len; struct snd_soc_dapm_path *path; + struct snd_card *card = dapm->codec->card->snd_card; /* add kcontrol */ for (i = 0; i < w->num_kcontrols; i++) { @@ -368,11 +387,11 @@ static int dapm_new_mixer(struct snd_soc_codec *codec, path->kcontrol = snd_soc_cnew(&w->kcontrols[i], w, path->long_name); - ret = snd_ctl_add(codec->card->snd_card, path->kcontrol); + ret = snd_ctl_add(card, path->kcontrol); if (ret < 0) { - printk(KERN_ERR "asoc: failed to add dapm kcontrol %s: %d\n", - path->long_name, - ret); + dev_err(dapm->dev, + "asoc: failed to add dapm kcontrol %s: %d\n", + path->long_name, ret); kfree(path->long_name); path->long_name = NULL; return ret; @@ -383,20 +402,22 @@ static int dapm_new_mixer(struct snd_soc_codec *codec, } /* create new dapm mux control */ -static int dapm_new_mux(struct snd_soc_codec *codec, +static int dapm_new_mux(struct snd_soc_dapm_context *dapm, struct snd_soc_dapm_widget *w) { struct snd_soc_dapm_path *path = NULL; struct snd_kcontrol *kcontrol; + struct snd_card *card = dapm->codec->card->snd_card; int ret = 0; if (!w->num_kcontrols) { - printk(KERN_ERR "asoc: mux %s has no controls\n", w->name); + dev_err(dapm->dev, "asoc: mux %s has no controls\n", w->name); return -EINVAL; } kcontrol = snd_soc_cnew(&w->kcontrols[0], w, w->name); - ret = snd_ctl_add(codec->card->snd_card, kcontrol); + ret = snd_ctl_add(card, kcontrol); + if (ret < 0) goto err; @@ -406,26 +427,27 @@ static int dapm_new_mux(struct snd_soc_codec *codec, return ret; err: - printk(KERN_ERR "asoc: failed to add kcontrol %s\n", w->name); + dev_err(dapm->dev, "asoc: failed to add kcontrol %s\n", w->name); return ret; } /* create new dapm volume control */ -static int dapm_new_pga(struct snd_soc_codec *codec, +static int dapm_new_pga(struct snd_soc_dapm_context *dapm, struct snd_soc_dapm_widget *w) { if (w->num_kcontrols) - pr_err("asoc: PGA controls not supported: '%s'\n", w->name); + dev_err(w->dapm->dev, + "asoc: PGA controls not supported: '%s'\n", w->name); return 0; } /* reset 'walked' bit for each dapm path */ -static inline void dapm_clear_walk(struct snd_soc_codec *codec) +static inline void dapm_clear_walk(struct snd_soc_dapm_context *dapm) { struct snd_soc_dapm_path *p; - list_for_each_entry(p, &codec->dapm_paths, list) + list_for_each_entry(p, &dapm->paths, list) p->walked = 0; } @@ -435,13 +457,14 @@ static inline void dapm_clear_walk(struct snd_soc_codec *codec) */ static int snd_soc_dapm_suspend_check(struct snd_soc_dapm_widget *widget) { - int level = snd_power_get_state(widget->codec->card->snd_card); + int level = snd_power_get_state(widget->dapm->codec->card->snd_card); switch (level) { case SNDRV_CTL_POWER_D3hot: case SNDRV_CTL_POWER_D3cold: if (widget->ignore_suspend) - pr_debug("%s ignoring suspend\n", widget->name); + dev_dbg(widget->dapm->dev, "%s ignoring suspend\n", + widget->name); return widget->ignore_suspend; default: return 1; @@ -572,7 +595,7 @@ static int dapm_generic_apply_power(struct snd_soc_dapm_widget *w) /* call any power change event handlers */ if (w->event) - pr_debug("power %s event for %s flags %x\n", + dev_dbg(w->dapm->dev, "power %s event for %s flags %x\n", w->power ? "on" : "off", w->name, w->event_flags); @@ -621,9 +644,9 @@ static int dapm_generic_check_power(struct snd_soc_dapm_widget *w) int in, out; in = is_connected_input_ep(w); - dapm_clear_walk(w->codec); + dapm_clear_walk(w->dapm); out = is_connected_output_ep(w); - dapm_clear_walk(w->codec); + dapm_clear_walk(w->dapm); return out != 0 && in != 0; } @@ -634,7 +657,7 @@ static int dapm_adc_check_power(struct snd_soc_dapm_widget *w) if (w->active) { in = is_connected_input_ep(w); - dapm_clear_walk(w->codec); + dapm_clear_walk(w->dapm); return in != 0; } else { return dapm_generic_check_power(w); @@ -648,7 +671,7 @@ static int dapm_dac_check_power(struct snd_soc_dapm_widget *w) if (w->active) { out = is_connected_output_ep(w); - dapm_clear_walk(w->codec); + dapm_clear_walk(w->dapm); return out != 0; } else { return dapm_generic_check_power(w); @@ -674,7 +697,7 @@ static int dapm_supply_check_power(struct snd_soc_dapm_widget *w) } } - dapm_clear_walk(w->codec); + dapm_clear_walk(w->dapm); return power; } @@ -709,12 +732,57 @@ static void dapm_seq_insert(struct snd_soc_dapm_widget *new_widget, list_add_tail(&new_widget->power_list, list); } +static void dapm_seq_check_event(struct snd_soc_dapm_context *dapm, + struct snd_soc_dapm_widget *w, int event) +{ + struct snd_soc_card *card = dapm->card; + const char *ev_name; + int power, ret; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + ev_name = "PRE_PMU"; + power = 1; + break; + case SND_SOC_DAPM_POST_PMU: + ev_name = "POST_PMU"; + power = 1; + break; + case SND_SOC_DAPM_PRE_PMD: + ev_name = "PRE_PMD"; + power = 0; + break; + case SND_SOC_DAPM_POST_PMD: + ev_name = "POST_PMD"; + power = 0; + break; + default: + BUG(); + return; + } + + if (w->power != power) + return; + + if (w->event && (w->event_flags & event)) { + pop_dbg(dapm->dev, card->pop_time, "pop test : %s %s\n", + w->name, ev_name); + trace_snd_soc_dapm_widget_event_start(w, event); + ret = w->event(w, NULL, event); + trace_snd_soc_dapm_widget_event_done(w, event); + if (ret < 0) + pr_err("%s: %s event failed: %d\n", + ev_name, w->name, ret); + } +} + /* Apply the coalesced changes from a DAPM sequence */ -static void dapm_seq_run_coalesced(struct snd_soc_codec *codec, +static void dapm_seq_run_coalesced(struct snd_soc_dapm_context *dapm, struct list_head *pending) { + struct snd_soc_card *card = dapm->card; struct snd_soc_dapm_widget *w; - int reg, power, ret; + int reg, power; unsigned int value = 0; unsigned int mask = 0; unsigned int cur_mask; @@ -735,64 +803,26 @@ static void dapm_seq_run_coalesced(struct snd_soc_codec *codec, if (power) value |= cur_mask; - pop_dbg(codec->pop_time, + pop_dbg(dapm->dev, card->pop_time, "pop test : Queue %s: reg=0x%x, 0x%x/0x%x\n", w->name, reg, value, mask); - /* power up pre event */ - if (w->power && w->event && - (w->event_flags & SND_SOC_DAPM_PRE_PMU)) { - pop_dbg(codec->pop_time, "pop test : %s PRE_PMU\n", - w->name); - ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMU); - if (ret < 0) - pr_err("%s: pre event failed: %d\n", - w->name, ret); - } - - /* power down pre event */ - if (!w->power && w->event && - (w->event_flags & SND_SOC_DAPM_PRE_PMD)) { - pop_dbg(codec->pop_time, "pop test : %s PRE_PMD\n", - w->name); - ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMD); - if (ret < 0) - pr_err("%s: pre event failed: %d\n", - w->name, ret); - } + /* Check for events */ + dapm_seq_check_event(dapm, w, SND_SOC_DAPM_PRE_PMU); + dapm_seq_check_event(dapm, w, SND_SOC_DAPM_PRE_PMD); } if (reg >= 0) { - pop_dbg(codec->pop_time, + pop_dbg(dapm->dev, card->pop_time, "pop test : Applying 0x%x/0x%x to %x in %dms\n", - value, mask, reg, codec->pop_time); - pop_wait(codec->pop_time); - snd_soc_update_bits(codec, reg, mask, value); + value, mask, reg, card->pop_time); + pop_wait(card->pop_time); + snd_soc_update_bits(dapm->codec, reg, mask, value); } list_for_each_entry(w, pending, power_list) { - /* power up post event */ - if (w->power && w->event && - (w->event_flags & SND_SOC_DAPM_POST_PMU)) { - pop_dbg(codec->pop_time, "pop test : %s POST_PMU\n", - w->name); - ret = w->event(w, - NULL, SND_SOC_DAPM_POST_PMU); - if (ret < 0) - pr_err("%s: post event failed: %d\n", - w->name, ret); - } - - /* power down post event */ - if (!w->power && w->event && - (w->event_flags & SND_SOC_DAPM_POST_PMD)) { - pop_dbg(codec->pop_time, "pop test : %s POST_PMD\n", - w->name); - ret = w->event(w, NULL, SND_SOC_DAPM_POST_PMD); - if (ret < 0) - pr_err("%s: post event failed: %d\n", - w->name, ret); - } + dapm_seq_check_event(dapm, w, SND_SOC_DAPM_POST_PMU); + dapm_seq_check_event(dapm, w, SND_SOC_DAPM_POST_PMD); } } @@ -804,8 +834,8 @@ static void dapm_seq_run_coalesced(struct snd_soc_codec *codec, * Currently anything that requires more than a single write is not * handled. */ -static void dapm_seq_run(struct snd_soc_codec *codec, struct list_head *list, - int event, int sort[]) +static void dapm_seq_run(struct snd_soc_dapm_context *dapm, + struct list_head *list, int event, int sort[]) { struct snd_soc_dapm_widget *w, *n; LIST_HEAD(pending); @@ -819,7 +849,7 @@ static void dapm_seq_run(struct snd_soc_codec *codec, struct list_head *list, /* Do we need to apply any queued changes? */ if (sort[w->id] != cur_sort || w->reg != cur_reg) { if (!list_empty(&pending)) - dapm_seq_run_coalesced(codec, &pending); + dapm_seq_run_coalesced(dapm, &pending); INIT_LIST_HEAD(&pending); cur_sort = -1; @@ -872,12 +902,12 @@ static void dapm_seq_run(struct snd_soc_codec *codec, struct list_head *list, } if (ret < 0) - pr_err("Failed to apply widget power: %d\n", - ret); + dev_err(w->dapm->dev, + "Failed to apply widget power: %d\n", ret); } if (!list_empty(&pending)) - dapm_seq_run_coalesced(codec, &pending); + dapm_seq_run_coalesced(dapm, &pending); } /* @@ -889,9 +919,9 @@ static void dapm_seq_run(struct snd_soc_codec *codec, struct list_head *list, * o Input pin to Output pin (bypass, sidetone) * o DAC to ADC (loopback). */ -static int dapm_power_widgets(struct snd_soc_codec *codec, int event) +static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) { - struct snd_soc_card *card = codec->card; + struct snd_soc_card *card = dapm->codec->card; struct snd_soc_dapm_widget *w; LIST_HEAD(up_list); LIST_HEAD(down_list); @@ -899,10 +929,12 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) int power; int sys_power = 0; + trace_snd_soc_dapm_start(card); + /* Check which widgets we need to power and store them in * lists indicating if they should be powered up or down. */ - list_for_each_entry(w, &codec->dapm_widgets, list) { + list_for_each_entry(w, &dapm->widgets, list) { switch (w->id) { case snd_soc_dapm_pre: dapm_seq_insert(w, &down_list, dapm_down_seq); @@ -925,6 +957,8 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) if (w->power == power) continue; + trace_snd_soc_dapm_widget_power(w, power); + if (power) dapm_seq_insert(w, &up_list, dapm_up_seq); else @@ -938,7 +972,7 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) /* If there are no DAPM widgets then try to figure out power from the * event type. */ - if (list_empty(&codec->dapm_widgets)) { + if (list_empty(&dapm->widgets)) { switch (event) { case SND_SOC_DAPM_STREAM_START: case SND_SOC_DAPM_STREAM_RESUME: @@ -948,7 +982,7 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) sys_power = 0; break; case SND_SOC_DAPM_STREAM_NOP: - switch (codec->bias_level) { + switch (dapm->bias_level) { case SND_SOC_BIAS_STANDBY: case SND_SOC_BIAS_OFF: sys_power = 0; @@ -963,52 +997,59 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) } } - if (sys_power && codec->bias_level == SND_SOC_BIAS_OFF) { - ret = snd_soc_dapm_set_bias_level(card, codec, + if (sys_power && dapm->bias_level == SND_SOC_BIAS_OFF) { + ret = snd_soc_dapm_set_bias_level(card, dapm, SND_SOC_BIAS_STANDBY); if (ret != 0) - pr_err("Failed to turn on bias: %d\n", ret); + dev_err(dapm->dev, + "Failed to turn on bias: %d\n", ret); } /* If we're changing to all on or all off then prepare */ - if ((sys_power && codec->bias_level == SND_SOC_BIAS_STANDBY) || - (!sys_power && codec->bias_level == SND_SOC_BIAS_ON)) { - ret = snd_soc_dapm_set_bias_level(card, codec, SND_SOC_BIAS_PREPARE); + if ((sys_power && dapm->bias_level == SND_SOC_BIAS_STANDBY) || + (!sys_power && dapm->bias_level == SND_SOC_BIAS_ON)) { + ret = snd_soc_dapm_set_bias_level(card, dapm, SND_SOC_BIAS_PREPARE); if (ret != 0) - pr_err("Failed to prepare bias: %d\n", ret); + dev_err(dapm->dev, + "Failed to prepare bias: %d\n", ret); } /* Power down widgets first; try to avoid amplifying pops. */ - dapm_seq_run(codec, &down_list, event, dapm_down_seq); + dapm_seq_run(dapm, &down_list, event, dapm_down_seq); /* Now power up. */ - dapm_seq_run(codec, &up_list, event, dapm_up_seq); + dapm_seq_run(dapm, &up_list, event, dapm_up_seq); /* If we just powered the last thing off drop to standby bias */ - if (codec->bias_level == SND_SOC_BIAS_PREPARE && !sys_power) { - ret = snd_soc_dapm_set_bias_level(card, codec, SND_SOC_BIAS_STANDBY); + if (dapm->bias_level == SND_SOC_BIAS_PREPARE && !sys_power) { + ret = snd_soc_dapm_set_bias_level(card, dapm, SND_SOC_BIAS_STANDBY); if (ret != 0) - pr_err("Failed to apply standby bias: %d\n", ret); + dev_err(dapm->dev, + "Failed to apply standby bias: %d\n", ret); } /* If we're in standby and can support bias off then do that */ - if (codec->bias_level == SND_SOC_BIAS_STANDBY && - codec->idle_bias_off) { - ret = snd_soc_dapm_set_bias_level(card, codec, SND_SOC_BIAS_OFF); + if (dapm->bias_level == SND_SOC_BIAS_STANDBY && + dapm->idle_bias_off) { + ret = snd_soc_dapm_set_bias_level(card, dapm, SND_SOC_BIAS_OFF); if (ret != 0) - pr_err("Failed to turn off bias: %d\n", ret); + dev_err(dapm->dev, + "Failed to turn off bias: %d\n", ret); } /* If we just powered up then move to active bias */ - if (codec->bias_level == SND_SOC_BIAS_PREPARE && sys_power) { - ret = snd_soc_dapm_set_bias_level(card, codec, SND_SOC_BIAS_ON); + if (dapm->bias_level == SND_SOC_BIAS_PREPARE && sys_power) { + ret = snd_soc_dapm_set_bias_level(card, dapm, SND_SOC_BIAS_ON); if (ret != 0) - pr_err("Failed to apply active bias: %d\n", ret); + dev_err(dapm->dev, + "Failed to apply active bias: %d\n", ret); } - pop_dbg(codec->pop_time, "DAPM sequencing finished, waiting %dms\n", - codec->pop_time); - pop_wait(codec->pop_time); + pop_dbg(dapm->dev, card->pop_time, + "DAPM sequencing finished, waiting %dms\n", card->pop_time); + pop_wait(card->pop_time); + + trace_snd_soc_dapm_done(card); return 0; } @@ -1035,9 +1076,9 @@ static ssize_t dapm_widget_power_read_file(struct file *file, return -ENOMEM; in = is_connected_input_ep(w); - dapm_clear_walk(w->codec); + dapm_clear_walk(w->dapm); out = is_connected_output_ep(w); - dapm_clear_walk(w->codec); + dapm_clear_walk(w->dapm); ret = snprintf(buf, PAGE_SIZE, "%s: %s in %d out %d", w->name, w->power ? "On" : "Off", in, out); @@ -1087,29 +1128,29 @@ static const struct file_operations dapm_widget_power_fops = { .llseek = default_llseek, }; -void snd_soc_dapm_debugfs_init(struct snd_soc_codec *codec) +void snd_soc_dapm_debugfs_init(struct snd_soc_dapm_context *dapm) { struct snd_soc_dapm_widget *w; struct dentry *d; - if (!codec->debugfs_dapm) + if (!dapm->debugfs_dapm) return; - list_for_each_entry(w, &codec->dapm_widgets, list) { + list_for_each_entry(w, &dapm->widgets, list) { if (!w->name) continue; d = debugfs_create_file(w->name, 0444, - codec->debugfs_dapm, w, + dapm->debugfs_dapm, w, &dapm_widget_power_fops); if (!d) - printk(KERN_WARNING - "ASoC: Failed to create %s debugfs file\n", - w->name); + dev_warn(w->dapm->dev, + "ASoC: Failed to create %s debugfs file\n", + w->name); } } #else -void snd_soc_dapm_debugfs_init(struct snd_soc_codec *codec) +void snd_soc_dapm_debugfs_init(struct snd_soc_dapm_context *dapm) { } #endif @@ -1130,7 +1171,7 @@ static int dapm_mux_update_power(struct snd_soc_dapm_widget *widget, return 0; /* find dapm widget path assoc with kcontrol */ - list_for_each_entry(path, &widget->codec->dapm_paths, list) { + list_for_each_entry(path, &widget->dapm->paths, list) { if (path->kcontrol != kcontrol) continue; @@ -1146,7 +1187,7 @@ static int dapm_mux_update_power(struct snd_soc_dapm_widget *widget, } if (found) - dapm_power_widgets(widget->codec, SND_SOC_DAPM_STREAM_NOP); + dapm_power_widgets(widget->dapm, SND_SOC_DAPM_STREAM_NOP); return 0; } @@ -1164,7 +1205,7 @@ static int dapm_mixer_update_power(struct snd_soc_dapm_widget *widget, return -ENODEV; /* find dapm widget path assoc with kcontrol */ - list_for_each_entry(path, &widget->codec->dapm_paths, list) { + list_for_each_entry(path, &widget->dapm->paths, list) { if (path->kcontrol != kcontrol) continue; @@ -1175,7 +1216,7 @@ static int dapm_mixer_update_power(struct snd_soc_dapm_widget *widget, } if (found) - dapm_power_widgets(widget->codec, SND_SOC_DAPM_STREAM_NOP); + dapm_power_widgets(widget->dapm, SND_SOC_DAPM_STREAM_NOP); return 0; } @@ -1191,7 +1232,7 @@ static ssize_t dapm_widget_show(struct device *dev, int count = 0; char *state = "not set"; - list_for_each_entry(w, &codec->dapm_widgets, list) { + list_for_each_entry(w, &codec->dapm.widgets, list) { /* only display widgets that burnm power */ switch (w->id) { @@ -1215,7 +1256,7 @@ static ssize_t dapm_widget_show(struct device *dev, } } - switch (codec->bias_level) { + switch (codec->dapm.bias_level) { case SND_SOC_BIAS_ON: state = "On"; break; @@ -1247,31 +1288,32 @@ static void snd_soc_dapm_sys_remove(struct device *dev) } /* free all dapm widgets and resources */ -static void dapm_free_widgets(struct snd_soc_codec *codec) +static void dapm_free_widgets(struct snd_soc_dapm_context *dapm) { struct snd_soc_dapm_widget *w, *next_w; struct snd_soc_dapm_path *p, *next_p; - list_for_each_entry_safe(w, next_w, &codec->dapm_widgets, list) { + list_for_each_entry_safe(w, next_w, &dapm->widgets, list) { list_del(&w->list); kfree(w); } - list_for_each_entry_safe(p, next_p, &codec->dapm_paths, list) { + list_for_each_entry_safe(p, next_p, &dapm->paths, list) { list_del(&p->list); kfree(p->long_name); kfree(p); } } -static int snd_soc_dapm_set_pin(struct snd_soc_codec *codec, +static int snd_soc_dapm_set_pin(struct snd_soc_dapm_context *dapm, const char *pin, int status) { struct snd_soc_dapm_widget *w; - list_for_each_entry(w, &codec->dapm_widgets, list) { + list_for_each_entry(w, &dapm->widgets, list) { if (!strcmp(w->name, pin)) { - pr_debug("dapm: %s: pin %s\n", codec->name, pin); + dev_dbg(w->dapm->dev, "dapm: pin %s = %d\n", + pin, status); w->connected = status; /* Allow disabling of forced pins */ if (status == 0) @@ -1280,26 +1322,26 @@ static int snd_soc_dapm_set_pin(struct snd_soc_codec *codec, } } - pr_err("dapm: %s: configuring unknown pin %s\n", codec->name, pin); + dev_err(dapm->dev, "dapm: unknown pin %s\n", pin); return -EINVAL; } /** * snd_soc_dapm_sync - scan and power dapm paths - * @codec: audio codec + * @dapm: DAPM context * * Walks all dapm audio paths and powers widgets according to their * stream or path usage. * * Returns 0 for success. */ -int snd_soc_dapm_sync(struct snd_soc_codec *codec) +int snd_soc_dapm_sync(struct snd_soc_dapm_context *dapm) { - return dapm_power_widgets(codec, SND_SOC_DAPM_STREAM_NOP); + return dapm_power_widgets(dapm, SND_SOC_DAPM_STREAM_NOP); } EXPORT_SYMBOL_GPL(snd_soc_dapm_sync); -static int snd_soc_dapm_add_route(struct snd_soc_codec *codec, +static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm, const struct snd_soc_dapm_route *route) { struct snd_soc_dapm_path *path; @@ -1310,7 +1352,7 @@ static int snd_soc_dapm_add_route(struct snd_soc_codec *codec, int ret = 0; /* find src and dest widgets */ - list_for_each_entry(w, &codec->dapm_widgets, list) { + list_for_each_entry(w, &dapm->widgets, list) { if (!wsink && !(strcmp(w->name, sink))) { wsink = w; @@ -1353,7 +1395,7 @@ static int snd_soc_dapm_add_route(struct snd_soc_codec *codec, /* connect static paths */ if (control == NULL) { - list_add(&path->list, &codec->dapm_paths); + list_add(&path->list, &dapm->paths); list_add(&path->list_sink, &wsink->sources); list_add(&path->list_source, &wsource->sinks); path->connect = 1; @@ -1374,14 +1416,14 @@ static int snd_soc_dapm_add_route(struct snd_soc_codec *codec, case snd_soc_dapm_supply: case snd_soc_dapm_aif_in: case snd_soc_dapm_aif_out: - list_add(&path->list, &codec->dapm_paths); + list_add(&path->list, &dapm->paths); list_add(&path->list_sink, &wsink->sources); list_add(&path->list_source, &wsource->sinks); path->connect = 1; return 0; case snd_soc_dapm_mux: case snd_soc_dapm_value_mux: - ret = dapm_connect_mux(codec, wsource, wsink, path, control, + ret = dapm_connect_mux(dapm, wsource, wsink, path, control, &wsink->kcontrols[0]); if (ret != 0) goto err; @@ -1389,7 +1431,7 @@ static int snd_soc_dapm_add_route(struct snd_soc_codec *codec, case snd_soc_dapm_switch: case snd_soc_dapm_mixer: case snd_soc_dapm_mixer_named_ctl: - ret = dapm_connect_mixer(codec, wsource, wsink, path, control); + ret = dapm_connect_mixer(dapm, wsource, wsink, path, control); if (ret != 0) goto err; break; @@ -1397,7 +1439,7 @@ static int snd_soc_dapm_add_route(struct snd_soc_codec *codec, case snd_soc_dapm_mic: case snd_soc_dapm_line: case snd_soc_dapm_spk: - list_add(&path->list, &codec->dapm_paths); + list_add(&path->list, &dapm->paths); list_add(&path->list_sink, &wsink->sources); list_add(&path->list_source, &wsource->sinks); path->connect = 0; @@ -1406,15 +1448,15 @@ static int snd_soc_dapm_add_route(struct snd_soc_codec *codec, return 0; err: - printk(KERN_WARNING "asoc: no dapm match for %s --> %s --> %s\n", source, - control, sink); + dev_warn(dapm->dev, "asoc: no dapm match for %s --> %s --> %s\n", + source, control, sink); kfree(path); return ret; } /** * snd_soc_dapm_add_routes - Add routes between DAPM widgets - * @codec: codec + * @dapm: DAPM context * @route: audio routes * @num: number of routes * @@ -1425,17 +1467,16 @@ err: * Returns 0 for success else error. On error all resources can be freed * with a call to snd_soc_card_free(). */ -int snd_soc_dapm_add_routes(struct snd_soc_codec *codec, +int snd_soc_dapm_add_routes(struct snd_soc_dapm_context *dapm, const struct snd_soc_dapm_route *route, int num) { int i, ret; for (i = 0; i < num; i++) { - ret = snd_soc_dapm_add_route(codec, route); + ret = snd_soc_dapm_add_route(dapm, route); if (ret < 0) { - printk(KERN_ERR "Failed to add route %s->%s\n", - route->source, - route->sink); + dev_err(dapm->dev, "Failed to add route %s->%s\n", + route->source, route->sink); return ret; } route++; @@ -1447,17 +1488,17 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_add_routes); /** * snd_soc_dapm_new_widgets - add new dapm widgets - * @codec: audio codec + * @dapm: DAPM context * * Checks the codec for any new dapm widgets and creates them if found. * * Returns 0 for success. */ -int snd_soc_dapm_new_widgets(struct snd_soc_codec *codec) +int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm) { struct snd_soc_dapm_widget *w; - list_for_each_entry(w, &codec->dapm_widgets, list) + list_for_each_entry(w, &dapm->widgets, list) { if (w->new) continue; @@ -1467,12 +1508,12 @@ int snd_soc_dapm_new_widgets(struct snd_soc_codec *codec) case snd_soc_dapm_mixer: case snd_soc_dapm_mixer_named_ctl: w->power_check = dapm_generic_check_power; - dapm_new_mixer(codec, w); + dapm_new_mixer(dapm, w); break; case snd_soc_dapm_mux: case snd_soc_dapm_value_mux: w->power_check = dapm_generic_check_power; - dapm_new_mux(codec, w); + dapm_new_mux(dapm, w); break; case snd_soc_dapm_adc: case snd_soc_dapm_aif_out: @@ -1484,7 +1525,7 @@ int snd_soc_dapm_new_widgets(struct snd_soc_codec *codec) break; case snd_soc_dapm_pga: w->power_check = dapm_generic_check_power; - dapm_new_pga(codec, w); + dapm_new_pga(dapm, w); break; case snd_soc_dapm_input: case snd_soc_dapm_output: @@ -1505,7 +1546,7 @@ int snd_soc_dapm_new_widgets(struct snd_soc_codec *codec) w->new = 1; } - dapm_power_widgets(codec, SND_SOC_DAPM_STREAM_NOP); + dapm_power_widgets(dapm, SND_SOC_DAPM_STREAM_NOP); return 0; } EXPORT_SYMBOL_GPL(snd_soc_dapm_new_widgets); @@ -1889,7 +1930,7 @@ int snd_soc_dapm_get_pin_switch(struct snd_kcontrol *kcontrol, mutex_lock(&codec->mutex); ucontrol->value.integer.value[0] = - snd_soc_dapm_get_pin_status(codec, pin); + snd_soc_dapm_get_pin_status(&codec->dapm, pin); mutex_unlock(&codec->mutex); @@ -1912,11 +1953,11 @@ int snd_soc_dapm_put_pin_switch(struct snd_kcontrol *kcontrol, mutex_lock(&codec->mutex); if (ucontrol->value.integer.value[0]) - snd_soc_dapm_enable_pin(codec, pin); + snd_soc_dapm_enable_pin(&codec->dapm, pin); else - snd_soc_dapm_disable_pin(codec, pin); + snd_soc_dapm_disable_pin(&codec->dapm, pin); - snd_soc_dapm_sync(codec); + snd_soc_dapm_sync(&codec->dapm); mutex_unlock(&codec->mutex); @@ -1926,14 +1967,14 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_put_pin_switch); /** * snd_soc_dapm_new_control - create new dapm control - * @codec: audio codec + * @dapm: DAPM context * @widget: widget template * * Creates a new dapm control based upon the template. * * Returns 0 for success else error. */ -int snd_soc_dapm_new_control(struct snd_soc_codec *codec, +int snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, const struct snd_soc_dapm_widget *widget) { struct snd_soc_dapm_widget *w; @@ -1941,11 +1982,12 @@ int snd_soc_dapm_new_control(struct snd_soc_codec *codec, if ((w = dapm_cnew_widget(widget)) == NULL) return -ENOMEM; - w->codec = codec; + w->dapm = dapm; + w->codec = dapm->codec; INIT_LIST_HEAD(&w->sources); INIT_LIST_HEAD(&w->sinks); INIT_LIST_HEAD(&w->list); - list_add(&w->list, &codec->dapm_widgets); + list_add(&w->list, &dapm->widgets); /* machine layer set ups unconnected pins and insertions */ w->connected = 1; @@ -1955,7 +1997,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_new_control); /** * snd_soc_dapm_new_controls - create new dapm controls - * @codec: audio codec + * @dapm: DAPM context * @widget: widget array * @num: number of widgets * @@ -1963,18 +2005,18 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_new_control); * * Returns 0 for success else error. */ -int snd_soc_dapm_new_controls(struct snd_soc_codec *codec, +int snd_soc_dapm_new_controls(struct snd_soc_dapm_context *dapm, const struct snd_soc_dapm_widget *widget, int num) { int i, ret; for (i = 0; i < num; i++) { - ret = snd_soc_dapm_new_control(codec, widget); + ret = snd_soc_dapm_new_control(dapm, widget); if (ret < 0) { - printk(KERN_ERR - "ASoC: Failed to create DAPM control %s: %d\n", - widget->name, ret); + dev_err(dapm->dev, + "ASoC: Failed to create DAPM control %s: %d\n", + widget->name, ret); return ret; } widget++; @@ -1983,34 +2025,17 @@ int snd_soc_dapm_new_controls(struct snd_soc_codec *codec, } EXPORT_SYMBOL_GPL(snd_soc_dapm_new_controls); - -/** - * snd_soc_dapm_stream_event - send a stream event to the dapm core - * @codec: audio codec - * @stream: stream name - * @event: stream event - * - * Sends a stream event to the dapm core. The core then makes any - * necessary widget power changes. - * - * Returns 0 for success else error. - */ -int snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, +static void soc_dapm_stream_event(struct snd_soc_dapm_context *dapm, const char *stream, int event) { - struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dapm_widget *w; - if (stream == NULL) - return 0; - - mutex_lock(&codec->mutex); - list_for_each_entry(w, &codec->dapm_widgets, list) + list_for_each_entry(w, &dapm->widgets, list) { if (!w->sname) continue; - pr_debug("widget %s\n %s stream %s event %d\n", - w->name, w->sname, stream, event); + dev_dbg(w->dapm->dev, "widget %s\n %s stream %s event %d\n", + w->name, w->sname, stream, event); if (strstr(w->sname, stream)) { switch(event) { case SND_SOC_DAPM_STREAM_START: @@ -2028,7 +2053,30 @@ int snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, } } - dapm_power_widgets(codec, event); + dapm_power_widgets(dapm, event); +} + +/** + * snd_soc_dapm_stream_event - send a stream event to the dapm core + * @rtd: PCM runtime data + * @stream: stream name + * @event: stream event + * + * Sends a stream event to the dapm core. The core then makes any + * necessary widget power changes. + * + * Returns 0 for success else error. + */ +int snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, + const char *stream, int event) +{ + struct snd_soc_codec *codec = rtd->codec; + + if (stream == NULL) + return 0; + + mutex_lock(&codec->mutex); + soc_dapm_stream_event(&codec->dapm, stream, event); mutex_unlock(&codec->mutex); return 0; } @@ -2036,7 +2084,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_stream_event); /** * snd_soc_dapm_enable_pin - enable pin. - * @codec: SoC codec + * @dapm: DAPM context * @pin: pin name * * Enables input/output pin and its parents or children widgets iff there is @@ -2044,15 +2092,15 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_stream_event); * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to * do any widget power switching. */ -int snd_soc_dapm_enable_pin(struct snd_soc_codec *codec, const char *pin) +int snd_soc_dapm_enable_pin(struct snd_soc_dapm_context *dapm, const char *pin) { - return snd_soc_dapm_set_pin(codec, pin, 1); + return snd_soc_dapm_set_pin(dapm, pin, 1); } EXPORT_SYMBOL_GPL(snd_soc_dapm_enable_pin); /** * snd_soc_dapm_force_enable_pin - force a pin to be enabled - * @codec: SoC codec + * @dapm: DAPM context * @pin: pin name * * Enables input/output pin regardless of any other state. This is @@ -2062,42 +2110,45 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_enable_pin); * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to * do any widget power switching. */ -int snd_soc_dapm_force_enable_pin(struct snd_soc_codec *codec, const char *pin) +int snd_soc_dapm_force_enable_pin(struct snd_soc_dapm_context *dapm, + const char *pin) { struct snd_soc_dapm_widget *w; - list_for_each_entry(w, &codec->dapm_widgets, list) { + list_for_each_entry(w, &dapm->widgets, list) { if (!strcmp(w->name, pin)) { - pr_debug("dapm: %s: pin %s\n", codec->name, pin); + dev_dbg(w->dapm->dev, + "dapm: force enable pin %s\n", pin); w->connected = 1; w->force = 1; return 0; } } - pr_err("dapm: %s: configuring unknown pin %s\n", codec->name, pin); + dev_err(dapm->dev, "dapm: unknown pin %s\n", pin); return -EINVAL; } EXPORT_SYMBOL_GPL(snd_soc_dapm_force_enable_pin); /** * snd_soc_dapm_disable_pin - disable pin. - * @codec: SoC codec + * @dapm: DAPM context * @pin: pin name * * Disables input/output pin and its parents or children widgets. * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to * do any widget power switching. */ -int snd_soc_dapm_disable_pin(struct snd_soc_codec *codec, const char *pin) +int snd_soc_dapm_disable_pin(struct snd_soc_dapm_context *dapm, + const char *pin) { - return snd_soc_dapm_set_pin(codec, pin, 0); + return snd_soc_dapm_set_pin(dapm, pin, 0); } EXPORT_SYMBOL_GPL(snd_soc_dapm_disable_pin); /** * snd_soc_dapm_nc_pin - permanently disable pin. - * @codec: SoC codec + * @dapm: DAPM context * @pin: pin name * * Marks the specified pin as being not connected, disabling it along @@ -2109,26 +2160,27 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_disable_pin); * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to * do any widget power switching. */ -int snd_soc_dapm_nc_pin(struct snd_soc_codec *codec, const char *pin) +int snd_soc_dapm_nc_pin(struct snd_soc_dapm_context *dapm, const char *pin) { - return snd_soc_dapm_set_pin(codec, pin, 0); + return snd_soc_dapm_set_pin(dapm, pin, 0); } EXPORT_SYMBOL_GPL(snd_soc_dapm_nc_pin); /** * snd_soc_dapm_get_pin_status - get audio pin status - * @codec: audio codec + * @dapm: DAPM context * @pin: audio signal pin endpoint (or start point) * * Get audio pin status - connected or disconnected. * * Returns 1 for connected otherwise 0. */ -int snd_soc_dapm_get_pin_status(struct snd_soc_codec *codec, const char *pin) +int snd_soc_dapm_get_pin_status(struct snd_soc_dapm_context *dapm, + const char *pin) { struct snd_soc_dapm_widget *w; - list_for_each_entry(w, &codec->dapm_widgets, list) { + list_for_each_entry(w, &dapm->widgets, list) { if (!strcmp(w->name, pin)) return w->connected; } @@ -2139,7 +2191,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_get_pin_status); /** * snd_soc_dapm_ignore_suspend - ignore suspend status for DAPM endpoint - * @codec: audio codec + * @dapm: DAPM context * @pin: audio signal pin endpoint (or start point) * * Mark the given endpoint or pin as ignoring suspend. When the @@ -2148,18 +2200,19 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_get_pin_status); * normal means at suspend time, it will not be turned on if it was not * already enabled. */ -int snd_soc_dapm_ignore_suspend(struct snd_soc_codec *codec, const char *pin) +int snd_soc_dapm_ignore_suspend(struct snd_soc_dapm_context *dapm, + const char *pin) { struct snd_soc_dapm_widget *w; - list_for_each_entry(w, &codec->dapm_widgets, list) { + list_for_each_entry(w, &dapm->widgets, list) { if (!strcmp(w->name, pin)) { w->ignore_suspend = 1; return 0; } } - pr_err("Unknown DAPM pin: %s\n", pin); + dev_err(dapm->dev, "dapm: unknown pin %s\n", pin); return -EINVAL; } EXPORT_SYMBOL_GPL(snd_soc_dapm_ignore_suspend); @@ -2170,20 +2223,20 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_ignore_suspend); * * Free all dapm widgets and resources. */ -void snd_soc_dapm_free(struct snd_soc_codec *codec) +void snd_soc_dapm_free(struct snd_soc_dapm_context *dapm) { - snd_soc_dapm_sys_remove(codec->dev); - dapm_free_widgets(codec); + snd_soc_dapm_sys_remove(dapm->dev); + dapm_free_widgets(dapm); } EXPORT_SYMBOL_GPL(snd_soc_dapm_free); -static void soc_dapm_shutdown_codec(struct snd_soc_codec *codec) +static void soc_dapm_shutdown_codec(struct snd_soc_dapm_context *dapm) { struct snd_soc_dapm_widget *w; LIST_HEAD(down_list); int powerdown = 0; - list_for_each_entry(w, &codec->dapm_widgets, list) { + list_for_each_entry(w, &dapm->widgets, list) { if (w->power) { dapm_seq_insert(w, &down_list, dapm_down_seq); w->power = 0; @@ -2195,9 +2248,9 @@ static void soc_dapm_shutdown_codec(struct snd_soc_codec *codec) * standby. */ if (powerdown) { - snd_soc_dapm_set_bias_level(NULL, codec, SND_SOC_BIAS_PREPARE); - dapm_seq_run(codec, &down_list, 0, dapm_down_seq); - snd_soc_dapm_set_bias_level(NULL, codec, SND_SOC_BIAS_STANDBY); + snd_soc_dapm_set_bias_level(NULL, dapm, SND_SOC_BIAS_PREPARE); + dapm_seq_run(dapm, &down_list, 0, dapm_down_seq); + snd_soc_dapm_set_bias_level(NULL, dapm, SND_SOC_BIAS_STANDBY); } } @@ -2208,10 +2261,10 @@ void snd_soc_dapm_shutdown(struct snd_soc_card *card) { struct snd_soc_codec *codec; - list_for_each_entry(codec, &card->codec_dev_list, list) - soc_dapm_shutdown_codec(codec); - - snd_soc_dapm_set_bias_level(card, codec, SND_SOC_BIAS_OFF); + list_for_each_entry(codec, &card->codec_dev_list, list) { + soc_dapm_shutdown_codec(&codec->dapm); + snd_soc_dapm_set_bias_level(card, &codec->dapm, SND_SOC_BIAS_OFF); + } } /* Module information */ diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 8a0a9205b1e7..4d95abb40288 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -60,6 +60,7 @@ EXPORT_SYMBOL_GPL(snd_soc_jack_new); void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) { struct snd_soc_codec *codec; + struct snd_soc_dapm_context *dapm; struct snd_soc_jack_pin *pin; int enable; int oldstatus; @@ -68,6 +69,7 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) return; codec = jack->codec; + dapm = &codec->dapm; mutex_lock(&codec->mutex); @@ -88,15 +90,15 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) enable = !enable; if (enable) - snd_soc_dapm_enable_pin(codec, pin->pin); + snd_soc_dapm_enable_pin(dapm, pin->pin); else - snd_soc_dapm_disable_pin(codec, pin->pin); + snd_soc_dapm_disable_pin(dapm, pin->pin); } /* Report before the DAPM sync to help users updating micbias status */ blocking_notifier_call_chain(&jack->notifier, status, NULL); - snd_soc_dapm_sync(codec); + snd_soc_dapm_sync(dapm); snd_jack_report(jack->jack, status); @@ -263,11 +265,12 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, INIT_DELAYED_WORK(&gpios[i].work, gpio_work); gpios[i].jack = jack; - ret = request_irq(gpio_to_irq(gpios[i].gpio), - gpio_handler, - IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING, - jack->codec->dev->driver->name, - &gpios[i]); + ret = request_any_context_irq(gpio_to_irq(gpios[i].gpio), + gpio_handler, + IRQF_TRIGGER_RISING | + IRQF_TRIGGER_FALLING, + jack->codec->dev->driver->name, + &gpios[i]); if (ret) goto err; |